Munications - Voice.over - IP.and - QoS.v8.0. (Exam.642 437) .2011.RETAiL - Ebook DeBTB00k

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CVOICE 8.0
Implementing Cisco
Unified Communications
Voice over IP and QoS v8.0
Study Guide
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John Wiley & Sons, Inc.
CVOICE 8.0
Implementing Cisco
Voice over IP and QoS v8.0
Study Guide
Andrew Froehlich
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Senior Acquisitions Editor: Jeff Kellum
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Dear Reader,
Thank you for choosing CVOICE 8.0: Implementing Cisco Unied Communications
Voice over IP and QoS v8.0 Study Guide. This book is part of a family of premium-quality
Sybex books, all of which are written by outstanding authors who combine practical
experience with a gift for teaching.
Sybex was founded in 1976. More than 30 years later, were still committed to producing
consistently exceptional books. With each of our titles, were working hard to set a new
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I hope you see all that reected in these pages. Id be very interested to hear your comments
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Acknowledgments
I would like to take this opportunity to thank the many people who collaborated with me
on the completion of this book as well as those who provided much-needed support along
the long path to completion. Many thanks to my acquisitions editor, Jeff Kellum. Jeff has
given me the opportunity to write my second book for Sybex, and I very much appreciate
his condence in me. Additionally, Id like to thank my development editor, Jim Compton;
technical editors, Scott Morris and Tyler Owen; production editor, Dassi Zeidel; and
copyeditor, Linda Recktenwald. Ive worked with all of these great people on both of my
Sybex publications, and knowing their working styles and habits has greatly helped in the
development of this book and making it technically sound, well structured, and well written.
Id also like to thank my family and friends for all of their support and encouragement.
2010 and 2011 have been a fruitful period for me both personally and professionally.
Moving overseas to Thailand and getting married, as well as consulting, freelance writing,
and publishing this book created an environment that was highly rewarding. Yet there was
no way I could have done it without the support of my friends and family. Specically, I
would like to thank my mother and father, Ron and Elaine Froehlich, and my friends Angie
Barbini, Adriana Castro, Matt and Fabiana Liska, Kevin and Ruth Ann McQuire, and
Sean and Heather Uhles.
Finally, I want to thank my wife, Manta Froehlich. She is the one person who saw the
daily effort put into writing this book and gave me the freedom to do what I needed to get
the job done. That includes many late nights and weekend work. I have learned a great deal
from her patience, and I know that having her by my side makes me a better person.
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About the Author
Andrew Froehlich, CCNA, CCDA, CCNA Voice, CCNP, CCSP, CCDP, F5 systems
engineer, is the president of West Gate Networks, a network and IT consulting rm based
in Chicago. Andrew has performed network design and support for large organizations
including the University of Chicago Medical Center, State Farm Insurance, and United
Airlines. In addition to having more than 14 years of network architecture experience, he
holds a degree in management information systems from Northern Iowa University and a
master of business administration degree from Northern Illinois University. He is also a
freelance writer and blogger for IT publications including Network World and Enterprise
Efciency. Andrew also authored Sybexs CCNA Voice Study Guide.
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Contents at a Glance
Introduction xxi
Assessment Test xxx
Chapter 1 An Introduction to Traditional Telephony
and Cisco Unified Communications 1
Chapter 2 Understanding Analog and Digital Voice 33
Chapter 3 VoIP Operation and Protocols 77
Chapter 4 The VoIP Path-Selection Process 103
Chapter 5 VoIP Design Options 145
Chapter 6 Configuring Voice Gateway Ports and DSPs 179
Chapter 7 Configuring Voice Gateway Signaling Protocols 223
Chapter 8 Configuring and Managing CUCM Express 281
Chapter 9 Advanced Voice Gateway Features 353
Chapter 10 Configuring and Managing CUBE
and H.323 Gateways 395
Chapter 11 Introduction to Quality of Service 439
Chapter 12 Configuring Quality of Service 473
Appendix About the Companion CD 529
Index 533
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Contents
Introduction xxi
Assessment Test xxx
Chapter 1 An Introduction to Traditional Telephony and Cisco
Unified Communications 1
Understanding Traditional Telephony Components 2
Telephony Edge Devices 3
Phone Switches 3
The Central Office 4
The Local Loop 5
Trunks 6
National and International Calling PSTN 8
Understanding Private Telephony Phone Systems 9
Key System 10
PBX 10
Understanding the Unified Communications Model 11
Endpoints 11
Applications 15
Call Processing Agents 15
Network Infrastructure 20
Unified Communications Deployment Models 20
The Centralized Services Deployment Model 20
The Distributed Services Deployment Model 21
The Inter-Networking of Services Deployment Model 22
The Geographical Diversity Deployment Model 22
Summary 23
Exam Essentials 24
Written Lab 1.1 25
Review Questions 26
Answers to Review Questions 30
Answers to Written Lab 1.1 32
Chapter 2 Understanding Analog and Digital Voice 33
Understanding Analog Voice Ports and Signaling 34
Analog Voice Port Types 34
Analog Voice Signaling 35
Basic Configuration of Analog Voice Ports 47
Understanding Digital Voice Ports and Signaling 51
An Overview of the Analog-to-Digital
Conversion Process 51
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xii Contents
Digital Voice Port Types 56
Digital Voice Multiplexing, Framing, and
Physical Transport 56
Digital Voice Signaling 60
Basic Configuration of Digital Voice Ports 63
Summary 66
Exam Essentials 66
Written Lab 2.1 67
Review Questions 69
Chapter 3 VoIP Operation and Protocols 77
Voice Media Transmission Protocols 78
Introduction to the Real-Time Transport Protocol 78
Introduction to the Real-time Transport
Control Protocol 81
Introduction to Compressed RTP 82
Introduction to Secure RTP 83
Voice Gateway Signaling Protocols 83
H.323 84
Session Initiation Protocol 85
Media Gateway Control Protocol 87
Skinny Client Control Protocol 88
Voice Gateway Signaling Protocol Comparison 88
An Introduction to Gatekeepers and
Other H.323 Components 89
Gatekeeper 89
H.323 Proxy Server 91
H.323 Multipoint Control Unit 91
A Typical H.323 Network 92
Choosing the Appropriate Voice Gateway
Signaling Protocol 93
Summary 94
Exam Essentials 94
Written Lab 3.1 95
Review Questions 96
Chapter 4 The VoIP Path-Selection Process 103
Understanding the Dial Plan Path-Selection Process 104
Understanding Voice Call Types 104
Path Selection and Call Routing 108
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Contents xiii
POTS and VoIP Dial Peers 108
Call Legs 110
Path-Selection Strategies 111
Introduction to PSTN and Private Numbering Plans 113
Using Wildcards to Simplify Dial-Peer Configurations 117
Site-Code Dialing 122
Dial-Plan Digit Manipulation 123
Digit Stripping 123
Forwarding the Last X Digits 124
Prefix Adding 125
Number Substitution 126
Translation Rules and Profiles 127
Verifying Dial-Plan Configurations 132
Summary 135
Exam Essentials 135
Written Lab 4.1 136
Review Questions 137
Chapter 5 VoIP Design Options 145
Voice Gateway DSP Functions 146
Understanding Voice and VoIP Quality Considerations 147
Audio Fidelity 148
Echo and Echo Cancellation 148
Background Noise 149
Voice over IP Quality Considerations 151
Defining Voice Codecs 153
Voice Codec Types 153
Understanding Codec Complexity 156
Quantifying Voice Codec Clarity 160
Mean Opinion Score 161
Perceptual Speech Quality Measure 162
Perceptual Evaluation of Speech Quality 163
Perceptual Objective Listening Quality Analysis 163
Choosing the Right Codec 163
Hardware Compatibility 163
Network Capacity 164
Codec Complexity 164
Endpoint Uses 164
Call Clarity 164
Calculating IP Voice Bandwidth Consumption 164
Frame and Bandwidth Calculations 165
Determining Packet and Frame Size Information 165
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xiv Contents
Additional Voice Packet and Frame Size Factors 166
Codec Bit Rate 166
Frame and Bandwidth Calculation Example 167
Summary 170
Exam Essentials 170
Written Lab 5.1 171
Chapter 6 Configuring Voice Gateway Ports and DSPs 179
Analog Port Configurations 180
Configuring an FXS and an FXO PLAR
OPX Port 180
Configuring FXS/DID Inbound and FXO
Outbound 184
Configuring E&M to Bridge Legacy PBX
with VoIP Networks 187
Configuring CAMA 188
Digital Port Configurations 191
Configuring a T1 CAS to Analog Cross-Connect 191
Configuring a T1 PRI 195
Configuring DSP Resources 198
Enabling a DSP Farm on a Voice Gateway 198
Creating DSP Profiles 199
Configuring SCCP Communications 200
Configuring the CUCM 201
Voice Port and Dial-Peer Verification Commands 203
show voice port 203
show controller 205
show voice dsp 205
test voice port 206
csim start 209
debug dialpeer 209
Summary 210
Exam Essentials 210
Written Lab 6.1 211
Hands-On Labs 212
Hands-On Lab 6.1: Configuring a T1 PRI 212
Hands-On Lab 6.2: Configuring a CAMA
Port for E911 Services 213
Hands-On Lab 6.3: Configuring an Outbound
Dial Peer to the PSTN 214
Dial Peer to the PSAP 214
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Contents xv
Chapter 7 Configuring Voice Gateway Signaling Protocols 223
Configuring H.323 224
Configuring an H.323 Gateway 227
H.323 show Commands 234
Configuring SIP 236
Determine the Endpoint Locations 237
Determine the Endpoint Capabilities 237
Determine Endpoint Availability 239
Establish a Session 239
Configure SIP between IP Voice Gateways 239
Configure Secure SIP Communications 241
Modify SIP Voice Gateway Settings 243
SIP show Commands 249
Configuring MGCP 253
Residential Gateways 254
Trunking Gateways 255
Configure an MGCP Residential Gateway 257
Configure an MGCP Trunking Gateway 259
MGCP show Commands 260
Summary 265
Exam Essentials 266
Written Lab 7.1 267
Hands-On Labs 268
Hands-On Lab 7.1: Configuring Basic SIP 269
Hands-On Lab 7.2: Modifying SIP Timers and Retries 270
Chapter 8 Configuring and Managing CUCM Express 281
Voice Network Infrastructure Considerations 282
Power Options for IP Phones 282
Configuring VLANs and Voice VLANs 286
Network Infrastructure Services for VoIP Support 290
An Overview of CUCM Express 293
Understanding CUCM Express Capabilities 294
Understanding CUCM Express Hardware
Requirements 295
Understanding CUCM Express Software Licensing 296
New Software-Activated Voice Licensing 297
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xvi Contents
Initial CUCM Express Configuration 297
Configuring CUCM Express as a TFTP Server 298
Configuring the Mandatory CUCM Express
System Settings Using SCCP Signaling 300
Configuring the Mandatory CUCM Express
System Settings Using SIP Signaling 305
Configuring SCCP and SIP Phones and Directory Numbers 307
Configuring Basic SCCP Ephone and Ephone-DNs 308
Configuring Basic SIP Voice Register Pools
and Voice Register DNs 310
SCCP Ephone-DN Line Configuration Options 311
Configuring Ephone-DN Shared Lines 312
Configuring Two Ephone-DNs with One Number 314
Configuring Ephone-DN Dual- and Octo-lines 315
Configuring SCCP Individual Lines 317
Configuring Ephone Button Options 318
Configuring CUCM Express Telephony Service Features 325
Configuring User Locale and Network Locale 325
Configuring the Date and Time Format 328
Modifying the Cisco IP Phone Keepalive Timer 329
Cisco IP Phone Restart versus Reset 329
Using CUCM Express Verification and
Troubleshooting Commands 332
Troubleshooting Cisco Phone Registrations 332
Determining the State of an Ephone 334
Summary 339
Exam Essentials 340
Written Lab 8.1 341
Hands-On Labs 342
Hands-On Lab 8.1: Configuring CUCM
Express as a TFTP Server 342
Hands-On Lab 8.2: Configuring CUCM Express
for Basic SCCP Phone Operation 343
Hands-On Lab 8.3: Verifying the Configuration
and Status of Your Ephones 344
Chapter 9 Advanced Voice Gateway Features 353
Configuring DTMF Relay Support 354
Configuring H.323 DTMF Relay 354
Configuring SIP DTMF Relay 355
Configuring MGCP DTMF Relay 356
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Contents xvii
Configuring Fax Support 357
Understanding Fax Relay 357
Configuring Cisco Fax Relay 359
Configuring T.38 Fax Relay 359
Understanding Fax Pass-through 364
Configuring Fax Pass-through 364
Understanding T.37 Store-and-Forward Fax 365
Configuring Modem Support 367
Configuring Modem Pass-Through 367
Configuring Modem Relay 368
Configuring Voice Backup Paths 368
Configuring a WAN-to-PSTN Fallback 369
Configuring MGCP-to-H.323 Fallback 370
Understanding and Configuring COR and SRST 372
Toll Bypass and TEHO 377
Configuring Call Blocking 380
Summary 382
Exam Essentials 382
Written Lab 9.1 384
Hands-On Labs 384
Hands-On Lab 9.1: Configuring Toll Bypass
and PSTN Redundancy 385
Hands-On Lab 9.2: Configuring TEHO 386
Chapter 10 Configuring and Managing CUBE
and H.323 Gateways 395
What Is an H.323 Gatekeeper? 396
H.323 Gatekeeper Mandatory Features 397
H.323 Gatekeeper Optional Features 398
Understanding Gatekeeper Signaling 399
RAS Gatekeeper Discovery Messages 399
RAS Gateway Registration Messages 400
RAS Call Admission Messages 400
The H.323 Gatekeeper Discovery, Registration,
and Admission Process 401
RAS Location Messages 402
RAS Resource Availability Messages 404
RAS Bandwidth Messages 404
Configuring an H.323 Gatekeeper 405
Configuring Local Zones 406
Configuring Remote Zones 406
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xviii Contents
Configuring Zone Prefixes 407
Configuring Technology Prefixes 408
Voice Gateway Interoperation with Gatekeepers 409
Configuring H.323 Interface Commands 409
Configuring Dial Peers for Gatekeeper Interoperation 410
Enabling the H.323 Service on a Voice Gateway 411
Configuring Call Admission Control on H.323
Gatekeepers 411
Understanding the CAC Bandwidth Control
on H.323 Gatekeepers 411
Configuring CAC Bandwidth Control
on H.323 Gatekeepers 412
Gatekeeper Verification and Troubleshooting Commands 414
Introducing the Cisco Unified Border Element 416
CUBE Features 417
CUBE Media Flow Options 417
CUBE Signaling Protocol Interoperation 419
CUBE RSVP-CAC 420
CUBE Call Flow Differences 421
Configuring the CUBE 422
Configuring Protocol Interoperation 422
Configuring Media Flow-Around 423
Configuring Codec Transparency 424
Configuring H.323 Fast-to-Slow-Start Signaling 424
Configuring SIP Delayed-to-Early-Offer Signaling 425
CUBE Verification and Troubleshooting Commands 425
Summary 427
Exam Essentials 427
Written Lab 10.1 428
Hands-On Labs 429
Hands-On Lab 10.1: Configuring
GB_Gatekeeper_1 430
Hands-On Lab 10.2: Configuring
London_gw1 and Glasgow_gw1 430
Chapter 11 Introduction to Quality of Service 439
Problems with Voice/Video on IP Networks 440
Mitigating IP Network Voice Issues 441
Providing Sufficient Bandwidth for a Newly
Converged Network 441
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Contents xix
Reduce End-to-End Delay 442
Reduce Jitter 442
Eliminate Packet Loss 443
Putting the Pieces Together 444
The Three-Step QoS Process 444
Traffic Classification 444
Traffic Marking 445
Traffic Queuing 445
QoS Policy Considerations 445
The Three-Step QoS Policy Development Process 445
Methods of Configuring QoS Policies 446
QoS Classification Models 447
The Best-Effort Model 447
The IntServ Model 447
The DiffServ Model 448
Comparing the QoS Models 449
Understanding the DiffServ ToS/DS Byte 449
DiffServ Service Quality Features 453
Layer 2 Class of Service and QoS Trust Boundaries 459
Layer 2 Classification and Marking with CoS 459
Identifying QoS Trust Boundaries 460
QoS Baseline Models 461
Comparing the Cisco QoS Baseline Model 461
Recommended Cisco Baseline Classification
Markings 462
Recommended Cisco Baseline Congestion-
Management and -Avoidance Tools 463
Summary 464
Exam Essentials 464
Chapter 12 Configuring Quality of Service 473
Configuring QoS Policies Using AutoQoS 474
Configuring AutoQoS for VoIP on a Router 475
Configuring AutoQoS for VoIP on a Switch 479
Configuring AutoQoS for the Enterprise on
a Router 483
Configuring QoS Policies Using MQC 488
Configuring Class Maps 490
Configuring Policy Maps 493
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xx Contents
Applying Policy Maps to Interfaces with a Service Policy 495
MQC QoS Configuration Show Commands 495
Configuring Congestion-Avoidance Techniques 498
Configuring Class-Based Traffic Policing and Shaping 500
Understanding Token Buckets 500
Understanding Traffic-Policing Token Buckets 501
Configuring Class-Based Traffic Policing 504
Configuring Class-Based Traffic Shaping 506
Configuring Link Efficiency Techniques 508
Configuring Link Fragmentation and
Interleaving for MLP and Frame Relay 509
Configuring Class-Based Header Compression 512
Configuring Trust Boundaries 513
Configuring CoS-to-DSCP Mappings 515
Summary 517
Exam Essentials 517
Hands-On Labs 518
Hands-On Lab 12.1: Configuring a Switchport
to Trust Cisco IP Phone QoS Markings 519
Hands-On Lab 12.2: Modifying CoS-to-DSCP
Mappings 519
Hands-On Lab 12.3: Configuring a Router for
QoS Using MQC 520
Appendix About the Companion CD 529
Index 533
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Introduction
Welcome to CVOICE 8.0: Implementing Cisco Unied Communications Voice over IP
and QoS v8.0 Study Guide, a comprehensive guide that covers everything you need for
Ciscos new exam 642-437. This particular exam will meet one requirement on the path to
achieve two different Cisco certication goals. Cisco has multiple levels of certications,
most of which build upon each other, as shown here:
Architect
Expert
Professional
Associate
Entry Specialist
This book covers one exam that is part of either the ve-exam CCNP Voice certication
or the two-exam Cisco Rich Media Communications Specialist certication, both of which
are highlighted. Currently, the ve exams to become CCNP Voice certied are:
642-437 CVOICE v8.0
642-447 CIPT1 v8.0
642-457 CIPT2 v8.0
642-427 TVOICE v8.0
642-467 CAPPS v8.0

The CCNP Voice certication track has the prerequisite that the test taker must currently
be CCNA Voice (640-461 or 640-460) certied.
Specialist certications are for network professionals with a very focused certication goal
in mind. Specically, the Cisco Rich Media Communications Specialist certication is for
IT professionals who must be procient in the design, implementation, and support of voice,
video, and web collaboration services on a converged IP network. Note that the technology
involved with these specialized certications is likely to change rapidly, and therefore most
specialist certications are valid for only two years. The two exams to become Cisco Rich
Media Communications Specialist certied are:
642-437 CVOICE v8.0
642-481 CRMC
The prerequisite for this Specialist certication is that the certication candidate must
currently be either CCNA certied or have any CCIE certication.
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xxii Introduction
The exams necessary to achieve either of these two Cisco certications can be taken
in any order you choose, but it is very common to start with the 642-437 CVOICE v8.0
exam, because it provides a solid foundation for the remainder of the exams.
A Closer Look at Ciscos Voice Certifications
Probably most readers of this study guide will be looking to achieve their CCNP Voice
certication, because it is part of Ciscos core structure for voice. Cisco offers three
distinct levels of core voice certications. The following diagram shows that the CCNA Voice
certication is a building block to the professional- and expert-level voice certications:
CCIE
Voice
CCNP Voice
CCNA Voice
As of the writing of this book, the CVOICE v8.0 (642-437) exam costs $250 USD. The
exam tests your knowledge a great deal in areas both theoretical and technically specic to
Cisco hardware and software.
Once you use this book to pass the CVOICE v8.0 exam, you can choose to continue
on the CCNP Voice path and pass the other four exams to achieve the CCNP Voice
certication. If you choose to achieve the CCNP Voice certication, you may want to
further your education and attempt to pass the CCIE Voice certication. But even if
you stop after achieving your CCNP Voice certication, you will have demonstrated to
your current or prospective employers that you have professional-level knowledge of the
interoperations of legacy PSTN and Cisco voice technologies. This assurance to employers
will make it easier for you to land that dream job youve always wanted!
What Skills Do You Need to Pass the CVOICE v8.0 Exam?
To pass the 642-437 exam, you should be procient in the following areas:
A thorough knowledge of analog, digital, and IP voice technologies including but

not limited to FXS, FXO, T1/E1, CAMA, voice trunks, voice packetization, codecs,
transcoding, PBX, key systems, multiplexing, IP-to-IP gateways, and QoS.
The ability to install, configure, and support Cisco voice gateways and gatekeepers.
This includes functions including, but not limited to, dial peers, digit manipulation,
path selection, calling privileges, signaling protocols, DSP farms, and analog and
digital ports.
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Introduction xxiii
The ability to install, configure, and support a Cisco Unified Communications Manager
Express (CUCM Express) system and endpoints. This also includes preparation of
CUCM Express support components, including DHCP, NTP, and TFTP.
The ability to install, configure, and support a Cisco Unified Border Element (CUBE)
for functionality including address hiding, protocol/media internetworking, and call
admission control.
A solid understanding of QoS fundamentals and how to implement them on Cisco

routers and Catalyst switches. This includes topics such as QoS requirements, IntServ/
DiffServ models, and link efficiency techniques.
What Does This Book Cover?
This book covers everything you need to know in order to pass the CVOICE v8.0 (642-437)
exam. In addition to studying this book, having the ability to study and practice with
Cisco router/switch hardware and software will provide you the condence to complete the
simulation questions found in the exam.
You will learn the following information in this book:
Chapter 1, An Introduction to Traditional Telephony and Cisco Unified

Communications, covers traditional telephony concepts and components that are
found in PSTN networks and legacy voice systems. Additionally, you are given
an introduction to Ciscos Unified Communications model and the best-practice
deployment models.
Chapter 2, Understanding Analog and Digital Voice, provides you with the
background covering traditional analog and digital telephony ports that are commonly
installed on voice gateways that connect to the PSTN or legacy PBX systems. Topics
such as network signaling, interface types, and the analog-to-digital conversion process
are covered in detail along with the basics of configuring many of these interfaces on
Cisco hardware.
Chapter 3, VoIP Operation and Protocols, introduces you to voice transport over
an IP network. Topics in this chapter include voice media transmission and control
protocols, voice gateway signaling protocols, and an introduction to common H.323
network components.
Chapter 4, The VoIP Path-Selection Process, provides you with the path-selection
process that a voice gateway goes through each time a call needs to be routed through
it. This includes a thorough understanding of the dial-plan selection process and
on- versus off-network calling. Additionally, we cover the differences between POTS
and VoIP dial peers and how to modify voice gateway path selections based on
dial-peer strategies and dial-peer wildcards, translations, and manipulation techniques.
Chapter 5, VoIP Design Options, exposes readers to VoIP network design

considerations. This includes voice quality topics such as fidelity, latency, delay, and
jitter, and it introduces you to some of the popular voice codecs used on VoIP networks.
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xxiv Introduction
This chapter also shows you how to calculate voice bandwidth consumption on an IP
network and how to choose the optimal codec based on network specifications.
Chapter 6, Configuring Voice Gateway Ports and DSPs, dives into more complex
voice gateway configuration techniques that show readers how to set gateway features
such as PLAR FXS/FXO DID, E&M bridge, and CAMA and explores several T1
scenarios. Additionally, you will learn how to configure a voice gateway as a DSP farm
for the offloading of services such as transcoding, conferencing, and MTP services.
Chapter 7, Configuring Voice Gateway Signaling Protocols, is an in-depth look at

voice gateway signaling protocols and how to configure them in multiple scenarios.
Those scenarios include the need to modify default settings, configuring protocols for
redundancy, using secure-mode communications, and best-practice configuration and
verification methods for production networks.
Chapter 8, Configuring and Managing CUCM Express, introduces you to the world
of the CUCM Express router and the functionality it can provide small to medium-size
businesses and remote-site offices. Those preparing for the CVOICE v8.0 exam must
know not only how to configure the CUCM Express router but also how to prepare the
IP network for voice communications with CUCM Express. This includes topics such
as PoE for Cisco IP phones, voice VLAN configuration best practices, and network
services for the support of IP phones including DHCP, TFTP, and NTP.
Chapter 9, Advanced Voice Gateway Features, shows readers several of the value-
added features and functionalities of voice gateways. Some of these features help
to facilitate the exchange of calls between IP and legacy PSTN networks, as is the
case with DTMF and fax/modem relay. Youll also see how to configure fallback
functionality on networks that operate both IP and PSTN networks between sites.
Lastly, we take a look at some cost-saving features inherent when you configure
features such as TEHO and call blocking.
Chapter 10, Configuring and Managing CUBE and H.323 Gateways, is an

introduction to how to configure and manage both an H.323 gatekeeper and the
Cisco Unified Boarder Element (CUBE). An H.323 gatekeeper is a value-added
component on large H.323 networks that helps to manage H.323 endpoints and
zones. The CUBE device is a router that acts as an IP-to-IP gateway between two
different networks. You will learn several methods of installing and maintaining these
hardware/software components.
Chapter 11, Introduction to Quality of Service, fully introduces the concepts

of QoS and shows you why QoS is necessary on IP networks. It also details best-
practice policy methodologies and models. You will be introduced to three different
QoS implementation methods. Finally, we cover some techniques for avoiding link
congestion; these techniques also aid in the efficient transport of time-sensitive traffic,
especially across low-speed WAN connections.
Chapter 12, Configuring Quality of Service, continues our coverage of QoS,

showing how to implement AutoQoS, MQC, and traffic policing/shaping and
flast.indd xxiv 9/20/11 10:56:06 AM
Introduction xxv
congestion-avoidance techniques, including LFI and compression. You will also learn
how to set QoS trust boundaries at various points within a network.
How to Use This Book
CVOICE 8.0: Implementing Cisco Unied Communications Voice over IP and QoS
Study Guide is designed to prepare a reader to pass the 642-437 exam, one of ve stages on
the way to the professional-level certication in Cisco voice technologies. To get the most
out of this book, I recommend you use the following study method:
1. Take the assessment test provided to you prior to Chapter 1 of this book. Try to
answer each question without looking at the answers and explanations found in the
back of the book. This should give you an indication of your skill level prior to reading
the book. Once you have completed the assessment test and graded yourself, take time
to carefully read over the explanations for any question you get wrong and note the
chapters in which the material is covered. This information should help you identify
sections of the book that you need to spend additional time on. Keep in mind, however,
that the book was designed for you to read each chapter in order. Much of the material
found in the chapters builds on knowledge learned from previous chapters.
2. Before reading each chapter, make sure to review the test objectives listed at the
beginning. These objectives are what the exam taker must ultimately know in order to
pass the CVOICE v8.0 (642-437) exam.
3. Complete each written lab at the end of each chapter. These labs are created to make
sure the reader fully understands key topics that are contained within that chapter.
Using a written format instead of multiple-choice format forces the reader to know the
answers off the top of their head instead of just eliminating options, as we often do
with multiple-choice questions.
4. Work through and fully understand the commands found in the hands-on labs in the
chapter. Not all chapters have hands-on labs, but the book focuses on the important
tasks necessary for aspiring CCNP Voicecertified network engineers. See the
accompanying sidebar for a recommended lab setup.
5. Answer all of the review questions related to each chapter. Once you have finished
answering the questions, review the answers and explanations to not only understand
the correct answers but also understand why the incorrect answers are actually
incorrect! Keep in mind that these review questions will not be the exact questions you
will find on the exam, but they will help you to understand the material from which
Cisco creates the actual exam questions.
6. Take time to review the bonus practice exams that are included on the companion CD.
Questions in these exams appear only on the CD.
7. Test yourself using all the flashcards on the included CD.
flast.indd xxv 9/20/11 10:56:06 AM
xxvi Introduction
8. Finally, make sure your mindset is in the right place. G. K. Chesterton said it best:
There is a great deal of difference between the eager man who wants to read a book
and the tired man who wants a book to read. So become that eager person when you
prepare for your exam. Youll see the payoff of your hard work before you know it!
Recommended Home Lab Setup
As stated earlier, it is critical to get some hands-on experience with Cisco voice routers
that can operate as voice gateways, H.323 gatekeepers, and CUCM Express. Additionally,
some time spent working with a Cisco Catalyst switch to congure QoS policies is highly
recommended. If you are in a classroom environment, the training center should provide
you with this equipment or a similar conguration to get you hands-on experience.
Otherwise, you will have to nd the hardware and software yourself. The following is
a list of equipment I suggest you try to acquire for your home lab studies. If you are
concerned about the high cost of purchasing the equipment, keep in mind that Cisco
hardware can be easily resold on used markets such as Craigslist or eBay. Combine that
fact with adding an extremely hot certication to your resume, and its an investment
well worth the initial cost.
Qty Item
2 Cisco ISR 2900 series router with two Fast Ethernet interfaces and one
T1 serial interface
1 Cisco Router IOS with Voice Gateway and H.323 Gatekeeper services
1 Cisco Router IOS that can operate as a Cisco Unied Communications
Manager Express
1 Cisco Catalyst switch
2 Cisco 7940 IP phones
2 Analog telephones
1 Windows PC loaded with terminal emulation software such as
PuTTY or SecureCRT
The router and switch equipment should give you the ability to practice conguring
all of the example congurations and practice labs in this study guide. The two IP
phones I recommend can also be supplemented with two Windows PCs running
the Cisco IP Communicator softphone. The analog phones in your lab are useful for
testing FXS congurations. You should also acquire the necessary analog, Ethernet,
and T1 crossover cabling for interconnecting hardware. Finally, it is important to use a
terminal emulator on which you are comfortable with both conguring Cisco hardware
and using copy and paste functions, so you can save any congurations and command
outputs that will help you with your studies.
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Introduction xxvii
Whats on the CD?
The CD included with this book includes many supplemental tools that you can use
to further your studies and achieve your goal of becoming a CCNP Voicecertied
administrator. The following content is provided for you to use to further your study.
The Sybex Test Engine
The Sybex test engine software lets readers practice all of the review and assessment
questions found in the book as well as two bonus practice exams that are found only on the
CD. The exams let potential test takers practice in an electronic test-taking environment
that is similar to the actual Cisco exam.
Electronic Flashcards
In addition to the Sybex test engine software, the CD includes over 200 electronic
ashcards with which to test yourself. These ashcards are designed to help you quickly
recognize and recall important CVOICE information that will be useful when taking the
642-437 exam.
Glossary of Terms in PDF
The CD contains a searchable Glossary of terms in PDF format. This includes an
exhaustive list of terms and denitions any CCNP Voice candidate should be familiar with.
Tips for Taking the CVOICE Exam
According to Ciscos website at https://learningnetwork.cisco.com/community/
certifications/ccvp/cvoicev8?tab=overview, the CVOICE exam contains anywhere
from 60 to 70 questions and must be completed in 90 minutes or less. English is currently
the only language in which the exam is available. A passing score varies according to the
types of questions found in the exam, but it is probably best to assume you need to get
approximately 85 percent of the questions correct to pass the exam.
When taking the exam, thoroughly read each question to make sure you know what
answer it is looking for. Cisco exam questions tend to have answers that look identical.
You will nd, however, that there are small differences in the answers that can determine a
correct or incorrect answer.
Also, keep in mind that you should choose the answer that Cisco believes is correct as
opposed to what you or other vendors believe. This is a Cisco exam, after all, so the right
answer is the one that Cisco recommends!
The format of the 642-437 exam questions might include any of the following:
Multiple-choice single-answer
Multiple-choice multiple-answerCisco will always tell you to choose two or three,

depending on the proper number of multiple correct responses.
Drag-and-drop
flast.indd xxvii 9/20/11 10:56:07 AM
xxviii Introduction
Fill-in-the-blank
Cisco voice router gateway/gatekeeper/CUCM Express, switch or CUBE simulations

Because of the limitations inherent in the Sybex test engine, this study
guide cannot include several of the exam types that you are likely to
experience on the real exam. But rest assured that if you fully understand
the material contained in the text and all the lab and practice questions,
you should have the knowledge to answer any question type you come
across on the actual exam.
Test-Day Tips for Certification Success
Arrive at least 30 minutes early to the exam center. That way you can check in and
mentally prepare for the exam without having to rush.
Take the Cisco exam tutorial found within the Cisco exam software on test day. This
tutorial is offered prior to the official start of each exam before the test timer starts.
In this tutorial you will be given an interactive lesson as to the format of the exam
and how to navigate through the different question types, including multiple-choice,
drag-and-drop, fill-in-the-blank, and simulation questions. Even if you have taken
many Cisco exams, I highly recommend going through the tutorial in case there is
something new to the exam format since the last time you took an exam.
Read both the questions and answers very carefully. Cisco often will intentionally lead
the hasty test taker, who simply glosses over a question, to quickly choose the incorrect
answer. Patience and careful thinking pay off greatly when taking Cisco exams!
Be aware that you cannot go back to change an answer once you have moved on to the
next question. Make sure that the answer you choose is the one you want to stick with,
because there is no way to change it later on.
Conventions Used in This Book
This book uses certain typographic styles in order to help you quickly identify important
information and to avoid confusion over the meaning of words such as on-screen prompts.
In particular, look for the following styles:
Italicized text indicates key terms that are described at length for the first time in a
chapter and are defined in the books Glossary.
A monospaced font indicates the contents of configuration files, messages displayed at

a command prompt, filenames, text-mode command names, and Internet URLs.
Italicized monospaced text indicates a variableinformation that differs from one

system or command run to another, such as the name of a client computer or a process
ID number.
flast.indd xxviii 9/20/11 10:56:08 AM
Introduction xxix
Bold monospaced text is information that youre to type into the computer, usually at a
command prompt.
In addition to these text conventions, which can apply to individual words or entire
paragraphs, a few conventions highlight segments of text:
A note indicates information thats useful or interesting but thats somewhat
peripheral to the main text. A note might be relevant to a small number of
networks, for instance, or it may refer to an outdated feature.
A tip provides information that can save you time or frustration and that
may not be entirely obvious. A tip might describe how to get around a
limitation or how to use a feature to perform an unusual task.
Warnings describe potential pitfalls or dangers. If you fail to heed a
warning, you may end up spending a lot of time recovering from a bug, or
you may even end up restoring your entire system from scratch.
Sidebars
A sidebar is like a note but longer. The information in a sidebar is useful, but it doesnt
t into the main ow of the text.
Real World Scenario
A real world scenario is a type of sidebar that describes a task or example thats
particularly grounded in the real world. This may be a situation that I or somebody I
know has encountered, or it may be advice on how to work around problems that are
common in real, working Cisco environments.
How to Contact Sybex
Sybex strives to keep you supplied with the latest tools and information you need for your
work. Please check our website at www.sybex.com/go/cvoice, where well post additional
content and updates that supplement this book should the need arise.
flast.indd xxix 9/20/11 10:56:08 AM
Assessment Test
1. After the AutoQoS for the Enterprise implementation phase has been completed, what final
step should be done?
A. Disable the discovery phase process within every interface it is running by issuing the
no auto discovery qos command.
B. Disable the discovery phase process globally by issuing the no auto discovery qos
command.
C. Schedule the autodiscovery phase process to run every week within every interface by
issuing the auto discovery qos 7 command.
D. Schedule the autodiscovery phase process to run every week globally by issuing the
auto discovery qos 7 command.
2. Which of the following DTMF relay methods transmit tones in an ASCII format? (Choose
all that apply.)
A. h245-signal
B. h245-alphanumeric
C. cisco-rtp
D. rtp-nte
3. Given the following information, what UC deployment model should you choose if your
business has six large (1,000 users or more) and geographically dispersed campuses that are
interconnected together by a 3 Mbps WAN link?
A. Centralized services model
B. Distributed services model
C. Inter-networking of services model
D. Geographical diversity model
4. Which of the following commands is the correct syntax and interface mode to configure
AutoQoS for VoIP on a Cisco router?
A. Router(config-if)#auto qos voip
B. Router(config-if)#auto qos voip cisco-phone
C. Router(config)#auto qos voip
D. Router(config)#auto qos voip cisco-phone
5. What is the correct command used to configure loop-start signaling on an FXS port?
A. Router(config-voiceport)#dial-type loopstart
B. Router(config-controller)#dial-type loopstart
C. Router(config-controller)#signal loopstart
D. Router(config-voiceport)#signal loopstart
flast.indd xxx 9/20/11 10:56:09 AM
6. Which of the following is not a feature of a Cisco Unified Border Element (CUBE)?
A. Call admission control (CAC)
B. Secure deployment
C. IP address hiding
D. Zone management
7. Dr. Nyquist discovered that analog samples taken at times the highest frequency would
produce high-quality sound when reconstructed using only the taken samples.
A. Three
B. Two
C. Five
D. Four
8. Which of the following is not a voice gateway signaling protocol?
A. MGCP
B. SCCP
C. Q.931
D. H.323
9. What type of voice trunk directly connects a private switch to a public switch?
A. CO trunk
B. Interoffice trunk
C. Tie trunk
D. Tandem trunk
10. What H.323 device maintains a database of telephone extensions to IP address mappings?
A. Proxy server
B. MCU
C. Gateway
D. Gatekeeper
11. A phone call enters a voice gateway. What happens if no incoming dial peer is matched?
A. The call is routed out the PSTN by default.
B. The call is dropped.
C. The voice gateway sends a redirect signal to the calling phone.
D. The call will match the default dial peer.
Assessment Test xxxi
flast.indd xxxi 9/20/11 10:56:10 AM
12. How are the voice and native data VLANs treated differently on the link between the Cisco
switch and the Cisco IP phone?
A. The voice VLAN is tagged using 802.1Q and the data VLAN is not tagged.
B. The voice VLAN is tagged using ISL and the data VLAN is tagged using 802.1Q.
C. The voice VLAN is not tagged and the data VLAN is tagged using ISL.
D. The voice VLAN is not tagged and the data VLAN is tagged using 802.1Q.
13. The following destination pattern is configured in a dial peer:
Router(config-dial-peer)# destination-pattern 34.?
Which of the following dial strings will be matched? (Choose all that apply.)
A. 3484
B. 34
C. 342
D. 3433
14. According to the ITU-T G.114 specification, packet delay for voice should not
exceed ms.
A. 30
B. 50
C. 150
D. 250
15. What is the correct configuration command for setting a voice gateway to use ISDN switch
type primary-5ess?
A. Router(config)#isdn switch-type primary-5ess
B. Router(config-controller)# isdn signaling switch-type primary-5ess
C. Router(config-controller)# isdn switch-type primary-5ess
D. Router(config)# isdn signaling switch-type primary-5ess
16. What voice gateway feature replaces lost packets with ones that are intelligently generated?
A. PESQ
B. DSP
C. PLC
D. iSAC
17. What codec quality tool has been developed to better test and grade next-generation codecs
that use wideband?
A. MOS
B. POLQA
C. PSQM
D. PESQ
xxxii Assessment Test
flast.indd xxxii 9/20/11 10:56:10 AM
18. Which of the following are limitations inherent in loop-start signaling? (Choose all that apply.)
A. It is unable to properly transition on-hook for inbound calls when FXO interfaces are used.
B. Glare.
C. It is unable to properly transition off-hook for inbound calls when FXO interfaces are used.
D. Gleam.
E. It is unable to properly transition on- or off-hook for inbound calls when FXO
interfaces are used.
19. What FXS config-voiceport command can be used to adjust the analog ring tone?
A. ring frequency
B. ring cadence
C. ring type
D. cptone
20. How many simultaneous calls can an E1 CAS circuit support?
A. 24
B. 31
C. 32
D. 30
21. Which of the following commands can be used to verify the line coding of a T1 interface?
A. show voice port
B. show voice port summary
C. show controller t1
D. show interface
22. When an H.323 gatekeeper receives an ARQ message from a registered H.323 device, what
two decisions does the gatekeeper make about a requested call?
A. What codec should be used
B. What type of H.323 device is attempting to make the call
C. Whether the call is permitted to go through
D. How the call should be routed
23. What voice signaling protocol is used by default when configuring dial peers on a router
with an IP voice gateway IOS?
A. SIP
B. SCCP
C. H.323
D. MGCP
E. SIPv2
Assessment Test xxxiii
flast.indd xxxiii 9/20/11 10:56:10 AM
24. Which of the following best describes an on-net to off-net call?
A. An internal user calling a telephone accessed through the PSTN
B. An internal user calling a remote site through the secondary PSTN path during a
WAN failure
C. An external user calling a remote site through the secondary PSTN path during a
WAN failure
D. An internal user calling a telephone accessed through the IP WAN
25. Which of the following is a common reason for adjusting the maximum number of SIP
retries?
A. If a high-compression codec is being used.
B. If the network is unreliable.
C. If the SIP gateway connects to an ISDN circuit.
D. If the SIP gateway accepts both TCP and UDP messages.
26. You are reviewing a routers configuration and see the following:
ephone 1
mac-address 0033.1c43.2533
type 7965
codec g729r8
button 1:1
What does the codec g729r8 command mean?
A. This ephone will operate only with the codec specified.
B. This is the preferred codec for the ephone.
C. This is the only codec that the IP phone understands.
D. DSP resources have been specifically set aside for this ephone.
27. Voice packets reach the destination IP phone with a delay variance between 15 and 50 ms.
What is the result?
A. The packets will be dropped.
B. Queuing buffers in the phone will smooth out any jitter.
C. The destination phone will reject the call by sending back a reorder signal to the
calling party.
D. The stream may sound garbled because it exceeds best-practice limits.
xxxiv Assessment Test
flast.indd xxxiv 9/20/11 10:56:11 AM
28. When viewing show ephone output like the following, what does SEIZE mean on
the extension?
ephone-2[1] Mac:0021.A02E.7D9A TCP socket:[5] activeLine:1 REGISTERED in SCCP
ver 12/9
mediaActive:0 offhook:1 ringing:0 ringRate: 0 reset:0 reset_sent:0 paging 0
debug:0 caps:9
IP:192.168.10.13 50271 7965 keepalive 16 max_line 6
button 1: dn 2 number 4002 CH1 SIEZE
Preferred Codec: g711ulaw
Active Call on DN 2 chan 1 :4002 0.0.0.0 0 to 0.0.0.0 0 via 0.0.0.0
G711Ulaw64k 160 bytes no vad
Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Jitter 0 Latency 0 callingDn -1 calledDn -1
A. The phone is currently in a call.
B. The phone is on-hook.
C. The phone is off-hook and unregistered.
D. The phone is off-hook.
E. The phone is receiving a call.
29. CAMA interfaces physically connect to what destination?
A. A PBX
B. The PSTN
C. The PSAP
D. A DID
30. Which of the following correctly configures a call-block profile (called block_976) for
incoming calls on a POTS dial peer?
A. call-block translation-profile block_976 incoming
B. call-block translation-profile incoming block_976
C. translation-profile call-block incoming block_976
D. translation-profile call-block block_976 incoming
31. What two methods are used to transmit RAS location messages?
A. Round-robin
B. Sequential
C. FIFO
D. Blast
Assessment Test xxxv
flast.indd xxxv 9/20/11 10:56:11 AM
32. Which of the following QoS variable-delay reduction techniques might use CBWFQ?
A. Prioritize time-sensitive traffic
B. Link fragmentation and interleaving
C. Compression
D. Bandwidth upgrades to eliminate bottlenecks
33. What markings can Cisco Catalyst L2 switches use to enforce QoS?
A. DSCP
B. IP Precedence
C. CoS
D. RSVP
34. When configuring MQC, what command is used to associate traffic class types with one or
more QoS operations?
A. class-map
B. policy-map
C. traffic-map
D. qos-map
35. Which telephony edge device converts voice into a binary stream?
A. PBX
B. Digital telephone
C. CO trunk
D. Tie trunk
36. What must be carefully watched when cRTP is configured between two voice gateways?
A. Packet fragmentation
B. Gateway CPU utilization
C. Packet delay
D. Packet jitter
37. What is the process of translating between two different codecs?
A. Transcoding
B. MTP
C. Translation
D. DSP
38. What is the proper name for the international numbering plan that was developed by the ITU?
A. G.711
B. NANP
C. E.164
D. E.711
xxxvi Assessment Test
flast.indd xxxvi 9/20/11 10:56:12 AM
Answers to Assessment Test xxxvii
Answers to Assessment Test
1. A. As soon as you have implemented AutoQoS for the Enterprise policies, you no longer
need to waste CPU resources by keeping the discovery phase running on an interface.
To disable the autodiscovery process, you should go into interface conguration mode of
each interface the processes is running and issue the no auto discovery qos command.
See Chapter 12.
2. A, B. Both the h245-alphanumeric and h245-signal DTMF relay methods convert tones
to ASCII for transmission on IP networks. See Chapter 9.
3. C. The best choice would be the inter-networking of services model because of the distributed
nature of the multisite network and the fact that the WAN links (3 Mbps) are likely to be too
small to transport voice trafc to a centralized call-processing agent. See Chapter 1.
4. A. The correct syntax for AutoQoS for VoIP on a router is auto qos voip. This command
is performed while in interface conguration mode. See Chapter 12.
5. D. Because an FXS port is an analog connection, you will be in config-voiceport mode.
The correct command while in this mode is signal loopstart. See Chapter 2.
6. D. Zone management is a feature of an H.323 gatekeeper and not a CUBE. See Chapter 10.
7. B. Sampling at a rate of twice the highest frequency to be represented follows the Nyquist
sampling theorem. See Chapter 2.
8. C. Q.931 is not a voice gateway signaling protocol. See Chapter 3.
9. A. A CO trunk is the name used to describe a circuit that connects a private PBX switch to
a public switch at the central ofce. See Chapter 1.
10. D. An H.323 gatekeeper is a server that maintains a database of telephone extensions to
IP address mappings. Before a call is made, the gatekeeper must be queried to identify the
location of the destination H.323 endpoint. See Chapter 3.
11. D. If a call is not matched against congured incoming dial peers, it is matched against the
default dial peer (dial peer 0) and processed accordingly. See Chapter 4.
12. A. Voice VLANs are tagged with 802.1Q and the native data VLAN is left untagged. See
Chapter 8.
13. B, C. The . means that any digit can be used. The ? means that the previous digit or group
will occur 0 or one time. That means that 34 and 342 will be the two choices that match
this destination pattern. See Chapter 4.
14. C. The ITU-T recommends that end-to-end delay should not exceed 150 milliseconds for
voice packets. See Chapter 5.
15. A. The ISDN switch type is congured globally in cong mode. The correct command is isdn
switch-type primary-5ess in order to set the ISDN switch the voice gateway will connect to.
See Chapter 2.
flast.indd xxxvii 9/20/11 10:56:12 AM
xxxviii Answers to Assessment Test
16. C. Packet loss concealment is a software process that replaces lost packets with ones
intelligently derived by the router. See Chapter 5.
17. B. The Perceptual Objective Listening Quality Analysis tool is an ITU-T standard that is
being developed to test and score high-delity codecs. See Chapter 5.
18. B, C. Glare can be a big problem for telephone loop-start users who make and receive
frequent telephone calls. Also, there is not a proper way for FXO ports to properly go
off-hook at the end of a call that came inbound on the interface. See Chapter 2.
19. B. The ring cadence command is used to adjust the ring tone. See Chapter 6.
20. D. Although it uses robbed-bit signaling, an E1 CAS circuit uses 2 of its 32 channels for
framing and synchronization. Therefore it can support up to 30 simultaneous calls. See
Chapter 2.
21. C. The show controller t1 command displays conguration information for T1 and E1
ports. See Chapter 6.
22. C, D. When an H.323 device attempts to make a call that utilizes an H.323 gatekeeper,
that call request goes to the gatekeeper. The gatekeeper rst determines if the call is
permitted and then uses the E.164 destination address to determine what IP address the call
should be routed to. See Chapter 10.
23. C. H.323 is the default voice gateway signaling protocol. If you want to use a different
signaling protocol, you must manually specify it. See Chapter 7.
24. B. On-net to off-net calls occur when a call is made to a remote site but for some reason
the call cannot be completed on the IP WAN. A secondary path is used to establish the call
instead using the PSTN network. See Chapter 4.
25. B. If your network is prone to packet drops and/or congestion, it is common to raise the
maximum number of SIP retry messages to help ensure that SIP messages are properly
received between endpoints. See Chapter 7.
26. B. The codec command species the preferred codec for an ephone when this phone is
calling another phone that is also congured on CUCM Express. The command can be
used while conguring individual ephones. See Chapter 8.
27. D. The maximum jitter is 30 ms between voice packets. Because this call exceeds those
limits, the result may be a voice stream that sounds garbled at the destination phone. See
Chapter 11.
28. D. When a user picks up the phone handset, the phone goes into an off-hook state. This is
referred to as a line seizure. See Chapter 8.
29. C. CAMA interfaces are used to connect to the PSAP for E911 calling. See Chapter 6.
30. B. The proper syntax is call-block translation-profile incoming block_976. This
command is performed while in config-dial-peer conguration mode. See Chapter 9.
flast.indd xxxviii 9/20/11 10:56:12 AM
Answers to Assessment Test xxxix
31. B, D. The sequential method sends an LRQ to remote gatekeepers one at a time and waits
for a response before sending another message. The blast method sends LRQ messages to
all remote gatekeepers at one time. See Chapter 10.
32. A. Trafc prioritization techniques can use CBWFQ as a way to segment trafc on a
network and give one class higher priority over another. See Chapter 11.
33. C. Cisco Layer 2 switches can read and enforce QoS using CoS markings found in Ethernet
frames. See Chapter 11.
34. B. The policy-map command associates trafc classes (segmented using class maps) and
applies QoS operations to them. See Chapter 12.
35. B. The two types of traditional telephony edge devices are analog and digital telephones.
Digital telephones take an analog stream and digitize it for transport. See Chapter 1.
36. B. cRTP is very CPU intensive and can cause the CPU to spike, which can end up causing
packet drops. See Chapter 3.
37. A. Transcoding is the process of translating between two codecs. DSP resources are used to
ofoad transcoding. See Chapter 5.
38. C. The ITU International numbering plan is formally known as E.164. See Chapter 4.
flast.indd xxxix 9/20/11 10:56:13 AM
flast.indd xl 9/20/11 10:56:13 AM
An Introduction to
Traditional Telephony
and Cisco Unified
Communications
THE FOLLOWING CVOICE EXAM
OBJECTIVES ARE COVERED IN THIS
CHAPTER:
Describe the components of a gateway.
Describe the function of gateways.

Describe a dial plan.
Describe a numbering plan.
Chapter
1
c01.indd 1 9/21/11 12:09:10 PM
Evolution is the process of something changing over time
into a more complex state where it can better adapt to its
environment. Evolution typically is triggered only when
outside forces require changes to be made. Technology also evolves into newer and more
useful tools over time. While the analog phone is still around, advances have been made
and telephones have evolved into fully digital devices. Even more recently, weve seen
more and more voice running over IP networks that share the same cabling and routing
functions with data networks.
But throughout this telephone evolution process, many of the traditional interfaces,
signaling protocols, and setups remain unchanged. In order to understand voice networks
of today, we must rst take a step back in time to discuss traditional telephony topics. Once
you have a solid foundation, you can see how many of these elements have either remained
the same or evolved over time to improve voice networks as they transition from circuit-
switched networks to packet-switched networks.
Chapter 1 will start off covering traditional telephony devices. This includes legacy
analog and digital phones as well as a look at components within public telephone networks.
We will then move on to the two private telephone network types in most organizations.
Lastly, this chapter will cover Ciscos take on IP telephony networks and how it breaks down
components into separate functionality categories and deployment models.
Understanding Traditional Telephony
Components
In 1875, Alexander Graham Bell invented the telephone, a device that transmits and
receives sound, most commonly human speech. The telephone houses a microphone that
callers speak into. With a standard analog telephone, the speech is then transported across
a pair of copper wires in the form of an electrical signal.
As the popularity of telephones grew, companies began providing a telephone network
that was used to interconnect multiple phones throughout a region. Today, public telephone
networks are a mixture of analog and digital circuits and trunks that interconnect and
cover the globe.
Telephone systems can be split into public and private sections. The private side consists
of equipment owned and maintained by an individual user or business. The public side
is owned and maintained by the telephone company, and this service is paid for by the
c01.indd 2 9/21/11 12:09:11 PM
individual or business owner who wants to use public phone services. The public switched
telephone network (PSTN) is the network that interconnects telephones found in homes
and businesses throughout towns, cities, countries around the world. It used to be that the
PSTN consisted solely of analog circuits. The rst analog circuit was just two wires, and it
was responsible for carrying a single telephone call. As technology improved, both the
public- and private-side equipment became more sophisticated. Private businesses could
own and maintain their own phone switches. These phone switches could then be
interconnected by trunk lines that were specically designed for the transport of voice
services between phone switches. In this rst section, we will investigate the traditional
telephony components that make up the private and public telephone network.
Telephony Edge Devices
The edge is the part of the phone system that end users interact with to make and receive
calls in their purest form. Traditional telephony edge devices can be divided into two
categories: analog and digital telephones. But even traditional telephony devices have
evolved to include more advanced features to make the calling experience a better one.
Here is a closer look at each of these phone types.
Analog Telephones
Analog edge devices are still somewhat common in homes and small business
environments. The analog telephone is commonly directly connected to the PSTN, so
all of the backend intelligence is the responsibility of the service provider, and the phone
user is simply responsible for purchasing and maintaining their analog telephone, which
is a very simple device. Some businesses still use analog PBX (private branch exchange)
systems, although they are becoming rare. Connecting an analog phone to a PBX provides
additional capabilities to the phone such as voicemail with message-waiting indicators, call
hold, and personalized ringtones. Other than that, the features of analog telephones are
very limited.
Digital Telephones
Digital telephone devices use special hardware to convert analog voice streams into a
digital data stream. Most legacy PBX systems are digital. It is also important to note that
the digital handsets of most of these digital PBX systems are proprietary. It is rare to be
able to mix and match different digital phones within a single digital PBX.
Phone Switches
On the public side of the overall telephony, there are public phone switches and private phone
switches. A PBX or key system can be installed by a private party to provide a multitude
of private telephone services to phones located within this private network. The differences
between a PBX and a key system are detailed later in this chapter. Extension-to-extension
c01.indd 3 9/21/11 12:09:12 PM
4 Chapter 1
An Introduction to Traditional Telephony and Communications

dialing, multiple lines, voicemail, call waiting, and call forwarding are just a few of the
services that private switches can provide.
A private switch does nothing when a call needs to be placed to a phone that is located
outside the private network. This is where a connection to the PSTN comes into play.
Privately owned phones and/or private switches connect to public telephone switches. These
switches handle public call routing and signaling.
The Central Office
A PSTN central ofce is the rst major stop where a public telephone line terminates.
Central ofce (CO) is a term used to describe a geographically located ofce that houses
PSTN switch equipment. Home and business lines are run back to the central ofce
and connect to the PSTN switches. The CO has large trunk lines that further multiplex
the phone lines and interconnect this central ofce with the larger national and global
telephone network.
In spite of the name, most modern central ofces are not really ofces at all but
underground bunkers of sorts. The switches and cabling are built underground for two
main reasons:

To help protect the cabling and switch equipment from lightning strikes

To limit the amount of electromagnetic radiation emitted by the lines and equipment,
which can interfere with analog radio and over-the-air television signals
To better understand where the CO ts into the PSTN, imagine that you are at your ofce
and need to call a customer of yours who is right down the street. Their telephone number
is 555-1717. If you are connected to the same CO (which is likely), then your call would be
directed out of your ofce on the PSTN line and reach the central ofce switch equipment.
That switch equipment would then look up the destination number of your customer
and discover that the destination terminates within the CO. The switch would then use
telephony signaling protocols to complete the connection and ring your customers phone,
as shown in Figure 1.1.
FI GURE 1.1 A PSTN call within the same central office
PSTN
central
ofce
5
5
5
-
1
7
1
7
5
5
5
-17
17
Called party
Calling party
c01.indd 4 9/21/11 12:09:12 PM
CO switches can also be compared to IP routers in a sense. From an IP router
perspective, packets enter a router interface, and they contain an IP address that identies
the destination device. The router uses the IP address to perform a routing table lookup to
see which router interface is the shortest path to the destination. A CO switch is similar
in that it too contains a table. But instead of IP addresses, the table consists of telephone
digits. These digits have a hierarchical structure similar to IP addresses. A hierarchical
structure helps to reduce the lookup table size and makes decision making faster and more
efcient. When calls enter a switch, the destination number is efciently matched within
the CO switch lookup table.
The Local Loop
The local loop is the physical connection that connects a customers private telephone
equipment to the PSTN central ofce. The loop is typically copper wiring and carries single
phone lines or multiple lines in the form of T1/E1 connections.
The local loop is sometimes referred to as the outside plant in very large
businesses with multiple connections to the CO.
Figure 1.2 shows an example of a business that has its private telephone equipment
connecting to the PSTN CO through the local loop wiring.
FI GURE 1. 2 A local loop
PSTN
Central
ofce
Local loop
PBX
PSTN
demarc
The customers site has a termination point called the demarcation point (demarc). This
point separates the customers house wiring from the PSTNs wiring and assigns the
physical cabling responsibilities accordingly. If a problem occurs on the PSTN lines, the
PSTN may visit the customers site and will troubleshoot up to the demarc. If the problem
is on the customers side of the demarc, it is the responsibility of the customer to x.
c01.indd 5 9/21/11 12:09:13 PM
6 Chapter 1

Trunks
Traditional telephony trunks are circuits that interconnect voice switches. There are three
distinct types of trunk lines:

Tie trunks

Central ofce trunks

Interofce trunks
The trunks themselves are similar for the most part except for the types of phone switches
(either public or private) they interconnect with. The following sections describe each
telephony trunk type in more detail.
Tie Trunks
A tie trunk (or tie line) is a dedicated voice circuit that directly connects two PBX switches.
This point-to-point connection is commonly used within private organizations to tie
multiple telephone systems together, as shown in Figure 1.3.
Calling
party
Called
party
4104
PBX-A
3XXX
PBX-B
4XXX
Tie trunk
4
1
0
4
4104
FI GURE 1. 3 A tie trunk
So why would a business ever need to have more than one PBX? There many reasons,
but these are some of the more popular ones:

Migrating from one PBX system to another

A merger of two or more businesses resulting in the need to combine PBX systems

A business or organization with multiple voice management groups that control their
own independent PBX systems
As a CVOICE candidate, you probably have an IP networking background, so these
reasons can be best compared to the migration and merging of IP networks. For example,
a merger between two separate PBX systems is similar to a merger of two separate IP
networks. The networks may not use the same routing protocols and therefore must either
be recongured so they use the same routing protocol or congured to redistribute into one
another. At a very high level, the same challenges found in migrating two IP systems are
c01.indd 6 9/21/11 12:09:14 PM
similar to merging two PBX systems. In both situations, similar planning methodologies
are required to successfully merge the two systems.
Central Office Trunks
Central ofce trunks are the circuits that connect a private business PBX to the PSTN.
When organizations have large PBX systems, having many users increases the number of
simultaneous calls, which requires multiple outside lines to the PSTN. The most efcient
and economical method is to have a trunk connection from the private PBX to the local
PSTN CO switch, as shown in Figure 1.4.
PSTN
Central
ofce
Calling party
External calls
External calls
CO trunk
PBX
FI GURE 1. 4 A CO trunk
Again, keep in mind that the physical wiring between the private PBX and the CO is
known as the local loop. In a large-business scenario, the local loop can be also referred to
as the central ofce trunk.
Interoffice Trunks
Interofce trunks are the backhauled connections that interconnect central ofces. Central
ofces that are connected with interofce trunk lines are considered to be interexchange
connections. Its easiest to understand interofce trunks in terms of local vs. long-distance
charges; a call whose routing goes no higher than an interofce trunk is considered local. For
example, imagine you are in your ofce and need to call someone on the other side of the
city with the number 555-1717. You pick up the phone and dial the 7-digit (or sometimes
10-digit) number. The dialed digits (known as DTMF, or dual-tone multi-frequency, as
discussed in Chapter 2, Understanding Analog and Digital Voice) are interpreted by your
local CO telephone switch, which determines that the destination phone does not reside
within the local CO but at a CO that is accessible through an interofce trunk connection.
The phone switch seizes one of the lines on the interofce trunk and communicates with the
neighboring CO switch to help terminate the call at the correct phone across the city.
Figure 1.5 depicts the call process ow using an interofce trunk between COs.
c01.indd 7 9/21/11 12:09:14 PM
8 Chapter 1

This type of interofce trunk would likely be considered a local call instead of a
long-distance call because the call uses the interofce trunk line to complete the call as
opposed to moving farther up the PSTN hierarchy. In North America, it used to be that
local calls were strictly dened by the area code they belong in. Only 7-digit dialing was
considered to be a local call and therefore did not incur long-distance charges. Over time,
and due to the U.S. government stepping in and breaking up the AT&T monopoly, it
became obvious that the area code method for determining local vs. long distance would
not be able to function in the future, for two reasons:

The deregulation of AT&T by the U.S. government meant that the FCC must decide
what would and would not be considered a long-distance call. The FCC came up with
the concept of Local Access and Transport Areas (LATAs). These LATAs were
supposed to be used by the newly formed Baby Bell companies to determine what
was considered long distance and what was not. LATAs were broken up mainly by
population and oftentimes overlapped state lines. LATAs oftentimes broke cities and
towns into multiple zones that would have necessitated the need for a long-distance
call that may have been right across the street.

The rapid growth of telephone number usage in large cities required multiple area
codes to overlap in a single geographical area. It became possible that a telephone
in one ofce might have a different area code than a different telephone in the same
building. No longer did area codes actually mean a different geographical area as they
were rst intended.
Because of the LATA and overlapping number confusion found in the U.S. telephone
numbering system, you will nd that certain 10-digit dialing is now considered to be local.
National and International Calling PSTN
For the bigger picture, we need to distinguish between, local, long distance, and
international long distance. This network setup varies from country to country, but at its
core, there is a three-step PSTN hierarchy, as shown in Figure 1.6.
Calling
party
Called
party
555-1717
PBX
4XXX
Interofce
trunk
5
5
5
-
1
7
1
7
555-1717
PSTN
Central
ofce
PBX
4XXX
PSTN
Central
ofce
FI GURE 1. 5 An Interoffice trunk connection
c01.indd 8 9/21/11 12:09:15 PM
Understanding Private Telephony Phone Systems 9
Telephone calls between COs that have interofce trunks are considered to be local
calling. If the telephones are on different networks that are not interconnected using
interofce trunks but fall within some type of border (such as by phone company, state,
or nation), the call is considered to be at the next level of the PSTN hierarchy, called the
interexchange network. Typically, long-distance charges begin to apply. Lastly, if the call is
placed between two international borders, it is considered to fall within the highest level of
the PSTN hierarchy, called the international network. International long-distance charges
begin to apply at this level.
Understanding Private Telephony
Phone Systems
In a business environment with multiple employees, you will quickly see that having
individual telephone lines run in from the PSTN is not the most efcient or economical
method for providing voice services. It would be extremely rare to nd a time when every
employee needed to access the PSTN at once. In fact, calculations show that the telephone-
to-PSTN line ratio is quite low. Therefore, business environments often implement some sort
of intelligence that allows multiple employees to have their own telephone handsets while
sharing PSTN lines. The two traditional telephone systems available are the key system and
the PBX. The next two sections briey explain the differences between these two systems.
Central ofce (local
calling)
Interexchange network
(long distance calling)
International network
(international long
distance)
L
o
c
a
l

t
o

I
n
t
e
r
n
a
t
i
o
n
a
l

c
a
l
l

o
w
FI GURE 1. 6 The PSTN local-to-international hierarchy
c01.indd 9 9/21/11 12:09:15 PM
10 Chapter 1

Key System
Very small businesses may choose to implement a key system, which is a simpler solution
and easier to manage than a PBX but offers fewer features. All of the telephone handsets
in a system are identical, and each phone shares the same small group of PSTN external
numbers, indicated by lights and selected by pressing buttons. Key systems do not assign
unique telephone numbers to individual phones. This ensures that anyone in the ofce can
answer an incoming call to any line. The key system is often called a shared-line system,
because of how lines are congured on the phones.
PBX
A private branch exchange (PBX) system is similar to PSTN switches owned and operated
by the PSTN. In fact, many PBX systems found in very large organizations use identical
switching equipment. Traditionally, businesses with employees of 20 or more will choose
to implement a PBX because of the more advanced features available to end users and
the scalability to grow both internally by adding additional handsets and externally by
connecting to other PBX systems using tie trunks or to the PSTN using CO trunks. While
PBX systems can either be analog or digital in nature, most legacy PBX systems in use
today use a digital transport method.
One of the major usage differences between PBX and key systems is where the calls that
originate within them are going. With key systems, because the businesses are typically
small, very few internal-to-internal calls are made. By contrast, a large percentage of calls
made on a PBX system are employee-to-employee calls. That is why it is very common for
PBX systems to use truncated numbers for internal dialing. These truncated numbers are
called extensions and are typically three to ve digits in length.
When Is It Time to Upgrade a Key System?
Kevin began his language service business back in the spring of 1998. When his business
was just starting up, his only employees were himself and four other persons, who
each took a portion of the sales and accounting duties. Kevin decided at that time to
implement a key system. To Kevin, this was a logical choice because he needed only
three phone lines for all his calls. Each employee had all three telephone numbers
congured on their phone. When a call came inbound from the PSTN (which was rare),
any one of the employees could answer the line. If the caller needed to speak to someone
directly, it was a simple process of placing the call on hold and yelling to the other side of
the small ofce for the proper person to pick up the line.
Over the years, Kevins business grew and with it the ofce space and number of
employees. The key system continued to be sufcient until an interesting phenomenon
c01.indd 10 9/21/11 12:09:16 PM
Understanding the Unified
Communications Model
The Cisco CVOICE exam requires that students have a basic understanding of the
end-to-end Cisco components involved in Internet Protocol Telephony (IPT), which is a
method used to transport voice communications over an IP network, and Voice over IP.
Cisco groups its components into four categories:

Endpoints

Applications

Call processing agents

Network infrastructure
Endpoints
Cisco has a plethora of both hardware- and software-based IP phones for nearly any
voice situation. In Ciscos Unied Communications model, an endpoint can be any end
device or software that interacts with Unied Communications hardware and/or software.
This is because Cisco lumps together multiple communications methods, including voice,
videoconferencing, and instant messaging, under one umbrella of services. The following
section is meant to give a broad overview of many of the different product features. It is not
a complete list of all of the phones available, however. These models are also continuously
being updated. While Cisco does not make analog telephones, it does offer hardware
occurred. After a while, the inbound calls that were directed to a specic employee
increased signicantly. This is because as the business grew, employees became
specialists in a specic part of the organization. No longer could any employee handle
any request. In addition, as the number of employees increased, a need arose to have
individual voicemail boxes, which a key system cannot handle, because no employee has
a dial-in number or extension.
Because of this, Kevin had to migrate to a new Cisco CUCM Express solution, which he
implemented as a PBX switching system. Now Kevin has the PBX set up so each one of
his 22 employees has their own unique telephone extension to make and receive both
internal and external calls. The migration from a key system to a PBX system let Kevins
business better adapt to growth.
c01.indd 11 9/21/11 12:09:16 PM
12 Chapter 1

solutions that convert analog into IP for use on modern networks. This section examines
the following categories of IP phone endpoints:

Wired IP phones

Wireless IP phones

Software IP phones

Videoconferencing phones

Analog-to-IP adapters
This should give you an understanding of the wide spectrum of Unied Communications
devices available for implementation.
Wired IP Phones
When most people think of IPT, the rst type of IP phone endpoints that they think of are
standard wired IP phones. This is likely because they most closely resemble analog telephones
of old. However, newer IP phones are beginning to look more like computers than telephones,
with the LCD displays, soft keys, and user-programmable features. Cisco divides its wired IP
phone systems into two major categories: small-business and enterprise-class phones.
Wired IP Phones for Small Business
Ciscos small-business IP phones include the SPA 300 and 500 series. These entry-level
phones are designed to work only with the Cisco Communications Manager Express call
agent solution. They are part of the Cisco Smart Business Communications System (SBCS)
suite of products and fully interoperate with products such as the UC500 series CUCM
Express and Unity Express voicemail products. This is because the phones specically
support the Cisco proprietary Smart Phone Control Protocol (SPCP), which only the
UC500 series platform call agents support. In addition, the 300 and 500 series phones
also support the Session Initiation Protocol (SIP) for call signaling on an IP network for
connecting to an Internet Telephony Service Provider (ITSP). An ITSP is a fairly new
service in which small businesses can pay a monthly fee to have a service provider manage
the backend IPT hardware while enjoying the added features and cost savings that IP
phones provide over PSTN solutions. All that is needed is a high-speed Internet connection
to the ITSP. Calls are routed to the ITSP across the Internet, and the ITSP then routes calls
out to the PSTN on its end.
Wired IP Phones for the Enterprise
The majority of Cisco IP phone offerings can be found in the enterprise class of hardware.
These X900 phones are further categorized by a numbering system in which X is the
number of a specic series. Within these categories, there are minor differences between
the features the individual phones offer. As of the writing of this book, the enterprise-class
IP phones include the following:

9900 series

8900 series

7900 series
c01.indd 12 9/21/11 12:09:17 PM

6900 series

3900 series
Each of these phone series is designed to meet a specic market niche. For example, the
3900 series phones offer basic functionality and do not include many of the add-on bells
and whistles that some of the high-end models tout. Cisco markets this line of phones for
use in lobbies, manufacturing oors, and retail-outlet oors where a basic-use phone may
be needed by employees and people in public-access areas.
It is not necessary to know all the wired/wireless IP phones that Cisco
offers. There simply are too many to list here, so we chose to discuss only
the most unique. If you want to investigate all of Ciscos IP phone offerings,
you can refer to Ciscos IP phone product web page at http://www.cisco
.com/en/US/products/sw/voicesw/index.html#~all-prod and begin
your research under the IP Communications section.
Wireless IP Phones
The Cisco 7900 series of phones offers the majority of different models. This includes the
two models of wireless IP phones currently:
7921G Wireless IP Phone This phone can operate on 802.11a/b/g networks.
7925G and 7925G-EX Wireless IP Phones These phones can operate on 802.11a/b/g
networks. In addition, they offer Bluetooth 2.0 support and a hermetically sealed and rug-
gedized case for heavy-use situations.
Unified Communications IP Soft Phones
Cisco has several software-based IP phones that let users make and receive voice calls
on computer hardware. The requirements are a compatible PC, a microphone, and
speakers/headphones. Once one of the following applications is installed and connected
to a compatible version of Cisco Unied Communications Manager (CUCM) server, you
can make and receive phone calls without an actual telephone handset. Here are the three
primary Cisco software IP phones available:
Cisco IP Communicator This software-based IP phone behaves just like a 7970 hardware-
based phone. That means that everything a hardware phone can do, the IP Communicator
can do as well. The application can be installed on Microsoft Windows XP, Vista, and
Windows 7 operating systems.
Cisco Unified Personal Communicator This software application integrates, voice,
voicemail, instant messaging, and other features into a single application that can be
installed on the latest Microsoft Windows and Mac OScompatible computers.
Cisco Unified Mobile Communicator With the increased mobility of business phone
users, thanks to 3G and 4G availability, Cisco created the Unied Mobile Communicator
software package to run on popular smartphones such as Apples iPhone and the RIM
Blackberry. The software lets users interact with the Cisco Unied Communications
c01.indd 13 9/21/11 12:09:17 PM
14 Chapter 1

platform remotely to accomplish tasks such as retrieve missed calls, join MeetingPlace
conferences, and even make and receive calls on a mobile phone, giving the impression to
the called/calling party that you are making the call from your desk phone extension. Not
only does the Mobile Communicator software make life easier on the mobile user, but it
can also dramatically decrease mobile phone bills by limiting roaming charges, because calls
made and received through the software run through the ofce telephony infrastructure
over the phones wireless data connection.
Video Phones and Tablets
Some hardware- and software-based phones integrate voice and video. The four primary
Cisco endpoints in this category are as follows:
7985G IP Video Phone This phone features a large, color LCD display and built-in
high-resolution camera for videoconferencing to other 7985G phones as well as all of
Ciscos other video hardware and software applications.
9951 IP Video Phone This phone features a touchscreen color display, built-in
high-resolution camera for videoconferencing and collaboration applications, and
Ethernet or Wi-Fi connectivity.
Cisco Video Advantage This products works with the Cisco VT Camera II USB hardware
to make and receive videoconference calls on a desktop PC running Microsoft Windows
software. The VT Camera II plugs into a PC through a USB port. Users make and receive
the video calls using their Cisco Unied IP desktop phones or the Cisco IP Communicator
software installed on the same PC as the Cisco Video Advantage software.
Cisco Cius Tablet The Cius is a new product from Cisco that combines Unied
Communications voice and video functionality with additional PC functionality. The tablet
can connect to a Cisco Unied Communications system, either wired or wireless, from inside
the enterprise. An optional 3G/4G wireless option is available to use the tablet as a mobile
communications tool from outside the ofce. Figure 1.7 shows the Cisco Cius tablet.
FI GURE 1. 7 The Cisco Cius tablet device
Courtesy of Cisco Systems, Inc. Unauthorized use not permitted.
c01.indd 14 9/21/11 12:09:18 PM
Analog-to-IP Adapters
Some people just cant let go of their analog endpoints for one reason or another. Much of
it has to do with the high cost of replacing all phones within a system. Another important
reason is that many businesses still rely on analog fax machines for their day-to-day
business operations. Cisco has anticipated this and has two major analog-to-IP adapters
to get these incompatible systems to interact on an IP phone network. Analog telephony
adapters (ATAs) are appliances that have an Ethernet port to connect to an IP LAN.
They then have two or more RJ-11 ports for connecting to analog telephones. The
appliances then use software to convert the analog signal into a digital IP packet for proper
transport on any IP network. The two primary solutions available from Cisco are these:
ATA 180 Series These are small point-solution appliances for connecting two analog
desk phones, conference phones, or fax machines to an IP network. These devices are good
for small businesses or anywhere only two analog phones are needed in one geographic
location.
VG200 Series The VG200 series appliances are Ciscos newest analog-to-IP devices that
offer additional integration to the Unied Communications features available. The form
factors of these stand-alone analog gateways include the ability to connect 2, 4, 24, or
even 48 analog devices to a single appliance. The two- and four-port models are scheduled
to completely replace the ATA 180 series once the 180 series goes end-of-life. The VG224
and 248 are high-density appliances that run on special IOS software that runs on ISR
(Integrated Services Router) equipment.
Applications
In addition to the Unied Communications platform calling features, Cisco provides
value-added Unied Communications applications that seamlessly integrate into the
product lineup. These applications include voicemail functionality with Ciscos Unity
lineup, Emergency Responder for 911 services, conference call applications in the form of
the Cisco Conference Connection suite, and billing applications. These add-on telephony
applications reside on dedicated hardware and software platforms and bolt into the Unied
Communications call processing agents that are described next.
Call Processing Agents
Call processing agents are the brains behind IP call-processing and call-control mechanisms
on a LAN. From a Cisco prospective, call agents are Cisco Unied Communications
Manager solutions. These were previously called Cisco Call Managers. Our discussion of
call agents will look at the three different Cisco call agent call-control responsibilities:

Call agents are responsible for the setup and teardown of telephone endpoints on the
local network.

Call control agents are used for IP telephone endpoint registration to the call agent.

Voice gateways are used to bridge voice networks.
c01.indd 15 9/21/11 12:09:18 PM
16 Chapter 1

Cisco Call Agent Solutions
Cisco offers three primary call agents to handle call processing for small, medium, and
large organizations. It is important to know the primary differences between the three
solutions. In addition, the 642437 exam requires that you be able to congure basic
settings on the Unied Communications Manager Express. This material is covered in
Chapter 8, Conguring and Managing the CUCM Express, of this study guide. Heres a
brief look at the three call agent solutions:
Cisco Unified Communications Manager Cisco Unied Communications Manager
(CUCM) is Ciscos agship call agent. It is a hardware appliance that runs on a hardened
Linux operating system. The current CUCM version is 8.0, which includes a number of
feature enhancements over versions 6 and 7. Each server appliance is capable of handling
up to 7,500 endpoints and can be clustered to support up to 30,000 endpoints.
Cisco Unified Communications Manager Business Edition Cisco Unied
Communications Manager Business Edition (CUCMBE), Ciscos medium-sized solution,
is basically the full-blown CUCM solution except for some key differences. The rst is
a limit of 500 endpoints on each appliance. It also does not offer the high-availability
redundancy features found in the CUCM. One major benet of the CUCMBE is that it
offers an integrated voicemail system, called Unity Connection, which runs on the same
hardware as the call agent software. This helps lower customer costs by allowing one piece
of hardware to be used for both purposes.
Cisco Unified Communications Manager Express The Cisco Unied Communications
Manager Express (CUCME) call agent differs greatly from the CUCM and CUCMBE in
the fact that the express software runs on Cisco routers. That means that the CUCME
runs a specialized version of the Cisco IOS. In addition, specialized cards or interfaces can
be installed into Cisco routers for voicemail access using Unity Express software.
This solution lets small businesses have a fully functional IP data, voice, and voicemail
solution contained in a single appliance. The CUCM Express is geared to small businesses
with up to 250 endpoint devices.
Cisco Call Control Agent Solutions
Call agents are responsible for handling IP phone endpoint setups so that the phones
receive extension numbers and other calling features unique to each phone. The IP phone
endpoints register to the call agent and communicate with it each time a call is placed on
the network. These are all functions of call control. When a user picks up an IP phone that
is registered to a call agent, that phone relies on the call agent for things such as dial tone
and other supervisory and informational signals (discussed in Chapter 2). When the end
user dials in a telephone number, the address-signaling information is sent from the phone
to the call agent. The call processing agent then has various settings and rules in place to
either permit the phone to call this number or deny it. For example, if the end user attempts
to call an international number on their desk phone, the call agent may deny this request so
the business does not incur expensive long-distance charges. If the call is allowed, the call
c01.indd 16 9/21/11 12:09:19 PM
agent performs signaling between the source IP phone and the voice gateway, as shown in
Figure 1.8.
M
Of-network
phone
IP phone
Call agent
PSTN
S
i
g
n
a
l
i
n
g
S
i
g
n
a
l
i
n
g
V
FI GURE 1. 8 Call setup signaling through the call agent
From a Cisco hardware perspective, call agents and Cisco IP endpoints can
communicate call setup information using either the Cisco proprietary Skinny Client
Control Protocol (SCCP) or the IETF-dened Session Initiation Protocol (SIP) method.
By default, Cisco call agents and most Cisco IP phones are congured for SCCP signaling.
Signaling between the call agent and the voice gateway (as shown in Figure 1.8) can be
H.323, SIP, MGCP (Media Gateway Control Protocol), or SCCP.
Once the call signaling is established between the source and destination phones, the
voice stream is transported directly between the phone and the voice gateway, which is the
nal hop on the IP network before it must be converted for proper transport on the PSTN.
The transport of voice on an IP network uses a separate protocol, as shown in Figure 1.9.
By using a separate and direct protocol for voice transport, voice packets are sent on the
most direct and efcient path.
M
Of-network
phone
IP phone
Call agent
IP voice packet transport
S
i
g
n
a
l
i
n
g
S
i
g
n
a
l
i
n
g
PSTN
V
FI GURE 1. 9 Voice transport
The protocol used to transport voice is the Real-Time Protocol (RTP) and is discussed
in detail in Chapter 4, The VoIP Path-Selection Process, of this book. When the phone
conversation is nished, both phones will again communicate call control information with
the call agent to end the call, just as they did with the setup signaling information shown in
Figure 1.8.
c01.indd 17 9/21/11 12:09:19 PM
18 Chapter 1

Cisco Voice Gateway Solutions
One major topic of the CVOICE exam is the functions of voice gateways on an IP network and
how to congure them. Voice gateways are a critical component of an IP network for several
reasons. Primarily, a voice gateway sits on the border between an internal IP voice network
and the public switched telephone network or a legacy PBX. Because these two networks
are incompatible with each other, it is the responsibility of the voice gateway to translate
between them. In order to accomplish this task, the voice gateway uses various hardware ports
to connect to the PSTN. It uses special hardware called digital signal processors (DSPs) to
translate from one medium to another for proper interoperation. In addition, voice gateways
are congured to speak one or more signaling protocols that are used to properly route calls to
and from the IP call agent, which may be one of the Cisco Unied Communications solutions
described earlier in this chapter. These signaling protocols are as follows:

H.323

SIP

MGCP

SCCP
Each of these protocols has benets and drawbacks. Careful consideration must be made
during the voice network design phase to choose the signaling protocol that best ts the
following:

Call agent hardware and software

Voice gateway hardware and software

Legacy hardware that needs to interact with the IP solution
DSPs are also used to facilitate other responsibilities (such as ofoading call conferencing
duties) for the proper operation of voice on an IP network as well as for value-added
features. All of these topics will be covered in depth in Chapters 5, VoIP Design Options,
and 6, Conguring Voice Gateway Ports and DSPs.
Voice Gateway Hardware Components
Cisco voice gateways are routers that have special IOS software designed to support voice
interface cards and voice signaling protocols. Specically, voice gateway IOS software
operates on some older router platforms, including these:

1700 series

2600XM series

3700 series
In addition, voice gateway IOS runs on newer Cisco Integrated Services Routers (ISRs).
These include the following router platforms:

1800 ISR series

2800 ISR series

3800 ISR series
c01.indd 18 9/21/11 12:09:20 PM
The ISR router series platform is slated to become end-of-life soon and will be replaced
by the ISR G2 (Generation 2) series platform as follows:

2900 ISR G2 series

3900 ISR G2 series
While there is a 1900 ISR G2 platform to replace the 1800 series, voice services will not
be available. Table 1.1 compares the 2900 and 3900 ISR G2 voice-capable routers and the
Unied Communications voice capabilities they offer.
TABLE 1.1 Comparison of the 2900 and 3900 series ISR G2 platforms
UC Feature 2900 Series ISR 3900 Series ISR
Conference call support Yes Yes
DSP support PVDM 2/3 PVDM 2/3
Max SRST calls 250 1500
Max SIP sessions 600 2500
Max digital voice calls 400 660
Max FXO ports 40 60
Max BRI ports 24 38
Lastly, voice services can be integrated into large enterprise and service provider
hardware, including these series:

1000 ASR series

9000 ASR series

6500 series

7200 series

7600 series

12000 series

AS5400 series

AS5800 series
The actual implementation of voice on the hardware listed here is different from the ISR
platform and is outside the scope of this study guide.
c01.indd 19 9/21/11 12:09:20 PM
20 Chapter 1

The primary focus of the CVOICE exam is on the ISR series of router equip-
ment. It is important to understand which equipment is considered to be
part of the ISR lineup.
Network Infrastructure
The nal Unied Communications model component in Ciscos design is the IP network
itself. This consists of standard IP devices, such as routers, layer 2/3 switches, and rewalls,
that transport both regular IP data and Unied Communications trafc over the same
physical network. The major point to note in a Unied network infrastructure is the
importance of Quality of Service (QoS) techniques that must be understood and properly
deployed to ensure proper transport of time-sensitive trafc such as voice and video.
Deployment Models
Cisco presents four different deployment models that it recommends for use within a
production network for the entire Unied Communications version 8.X suite. These models
have remained fairly static over the years, but the terminology has changed to reect the fact
that Unied Communications offers more than just voice services in todays networks. Although
this section discusses placement of the call processing agents, in reality other UC servers and
services can be centralized or distributed to perform the same way for the services they provide.
Each deployment has its benets and potential drawbacks. The primary differences between the
four models discussed next depend mostly on the following characteristics:

Number of users supported

Physical location of users

Amount of WAN bandwidth and QoS controls

Ability to offer alternative methods to achieve high availability (HA)
The Centralized Services Deployment Model
The centralized services deployment model is ideal when a single building or a group of
buildings in a campus is interconnected in a LAN environment. A single call processing
agent can be used, or multiple call processing agents can be clustered to provide scalability
and high availability to voice services. The benets of this model derive from having a
single administration point with which to manage the Unied Communications (UC)
services. A major drawback is the lack of scalability if your UC needs extend outside the
single location site. Figure 1.10 shows a typical centralized services deployment model.
c01.indd 20 9/21/11 12:09:21 PM
Unified Communications Deployment Models 21
Campus
M PSTN
V
FI GURE 1.10 The centralized services deployment model
The Distributed Services Deployment Model
The distributed services model is useful when you have a large campus site and a handful of
smaller remote sites that are connected to the primary site using high-speed WAN connections.
This model often represents the classic hub-and-spoke look. While this is a great model
in many instances, you should plan for high availability in case of a WAN outage. Cisco
commonly suggests using either Survivable Remote Site Telephony (SRST) or a CUCME in
SRST fallback mode. Both of these features allow the remote sites to route calls to outbound
PSTN links in the event of a WAN failure, at which point the remote site cannot access the
call processing agent. Figure 1.11 shows the distributed services deployment model.
Campus
M
V
PSTN
Branch Branch
SRST
V
IP WAN
V
V
V
FI GURE 1.11 The distributed services deployment model
c01.indd 21 9/21/11 12:09:21 PM
22 Chapter 1

The Inter-Networking of Services Deployment Model
The inter-networking of services model is used for organizations with multiple large and
geographically dispersed locations. In this situation, call processing agents are distributed
and located at the various sites and act largely as independent single-site deployments.
Local calling therefore never traverses over WAN links, which conserves bandwidth. This
model is also useful when WAN links are not reliable or do not have enough bandwidth
for handling voice trafc in addition to data trafc. Calling between sites can be sent
either across an IP WAN or through the PSTN. Figure 1.12 shows this model, also called
multisite with distributed call processing.
FI GURE 1.12 The inter-networking of services model
Campus
M
V
IP WAN PSTN
M
V
Campus
M
V
Campus
V
V
V
The Geographical Diversity Deployment Model
The nal deployment model is the geographical diversity deployment model. In this model,
the organization again has multiple geographically dispersed locations with a large number
of users. The call processing agents are distributed as in the case of the inter-networking of
c01.indd 22 9/21/11 12:09:22 PM
Summary 23
services model. The difference this time is that the geographical diversity call-processing
agents work as a single cluster across interconnecting IP WAN links. As with the
distributed services model, it is extremely important to have WAN connections with ample
bandwidth and QoS enabled for voice. One benet in this model is that local site calling is
contained within the LAN and does not traverse the WAN. In addition, WAN links in a
mesh design can provide some form of redundancy to the point where SRST is not required.
Figure 1.13 shows this model, also called clustering over IP WAN.
Campus
M
V
IP WAN PSTN
M
V
Campus
M
V
Campus
Cluster
V
V
V
FI GURE 1.13 The geographical diversity model
Summary
You should now have a solid understanding of the hardware and software components
involved in traditional telephony systems. In addition, this chapter covered the two
different types of private telephone equipment and when a business might choose to
implement a key system or a PBX based on internal vs. external calling patterns. Lastly,
we covered Ciscos IPT component and deployment models.
Well cover many of these same topics again throughout this book in much more detail.
This chapter was written to give you a 30,000-foot view of traditional telephony so that
c01.indd 23 9/21/11 12:09:22 PM
24 Chapter 1

when we cover newer and more advanced topics on voice in future chapters, youll be able
to understand how the voice network has evolved to the point where it is today. Many
things have changed while some things remain the same.
Exam Essentials
Know the two different types of traditional telephony edge devices. The two main types
of edge devices are analog and digital phones. Besides the obvious difference of handling
voice services as analog or digital formats, digital telephones often offer additional service
features and are more commonly found in legacy business PBX systems.
Know what a phone switch is. Phone switches are responsible for routing calls throughout
a voice network. They can be privately owned as is the case with a PBX, or they can be part
of the PSTN.
Know what a central office is and where it is located in relation to a business. The central
ofce is the rst telephone switch that personal and business telephones reach on the PSTN.
The physical cabling between the privately owned telephone equipment and the CO is
called the local loop.
Understand the difference between tie, central office, and interoffice trunks. Tie trunks
connect two PBX systems. Central ofce trunks connect a PBX to the CO. Interofce
trunks interconnect two COs.
Understand the three-tiered PSTN hierarchy. The PSTN routes calls based on central
ofce, interexchange, and international networks. The PSTN uses a hierarchical network
structure where local calls are routed through the central ofce, national calls through the
central ofce, and interexchange and international calls through all three tiersthe central
ofce, interexchange, and international networks.
Understand the difference between a key system and a PBX. Key systems are used in
smaller environments, have few features, and do not have unique extensions. PBX systems
are found in large businesses, have many features, and pool external numbers while having
unique internal extension numbers.
Know the four Unified Communications model tiers and which products fall within
each tier. The endpoints tier contains hardware and software the end user interacts with.
The applications tier contains various value-added applications used in the UC lineup. The
call-processing layer is the brains where call processing takes place. This is where the UC
Manager resides. Lastly, the network infrastructure is the IP network equipment that
transports voice and data as well as employs QoS.
Understand the four Unified Communications deployment models. Know the four
models and when they should be implemented. Understand that choosing one model over
another depends on several factors, including number of users, physical location of users,
amount of bandwidth, and alternative methods to achieve HA.
c01.indd 24 9/21/11 12:09:23 PM
Written Lab 1.1 25
Written Lab 1.1
Write the answers to the following questions:
1. These PSTN sites house telephone switch equipment that directly connects to personal
telephones and/or ofce PBX switches.
2. In large environments, the local loop is also called the outside .
3. Name the three types of voice trunks in a traditional telephony network.
4. What are the four categories of the Cisco Unied Communications model?
5. What is a private phone system that uses shared-line extensions?
6. The Cisco IP Communicator software resides in which UC model category?
7. What hardware interconnects two incompatible voice networks?
8. A CUCM is also called a agent.
9. What hardware device allows a user to connect analog telephone devices to an IP
network?
10. What UC deployment model is useful when you have a large campus site and a handful
of smaller remote sites that are connected to the primary site using high-speed WAN
connections?
(The answers to Written Lab 1.1 can be found following the answers to the review
questions for this chapter.)
c01.indd 25 9/21/11 12:09:23 PM
26 Chapter 1

Review Questions
1. What is the name of the first major stop that a public telephone line makes when leaving
the customer site?
A. Tie trunk
B. CO trunk
C. Local loop
D. Central office
2. There is a specific point within a customers site that defines the physical cabling
responsibilities between the private owner and the telephone company. What is this point
referred to as?
A. Demarc
B. Inside wiring
C. Outside wiring
D. Local loop
E. House wiring
3. What type of voice trunk directly connects two PBX systems?
A. Demarc
B. Tie trunk
C. CO trunk
D. Interoffice trunk
4. What type of voice trunk directly connects two central offices?
A. CO trunk
B. Interoffice trunk
C. Tie line
D. Interexchange trunk
5. Central offices maintain pools of what numbers?
A. Subscriber code
B. E.164 code
C. Area code
D. Interexchange code
c01.indd 26 9/21/11 12:09:24 PM
Review Questions 27
6. At what point of the PSTN hierarchy described in this Study Guide will a caller begin
incurring long-distance charges?
A. Central office
B. Local loop
C. International network
D. Interexchange network
7. Which Cisco IP phone does not support Ciscos proprietary SCCP signaling protocol?
A. 7925G series wireless phone
B. SPA 500 series phone
C. Cisco IP Communicator softphone
D. All Cisco phones support SCCP
8. The Cisco Emergency Responder belongs in what UC model category?
A. Network infrastructure
B. Applications
C. Call processing agents
D. Endpoints
9. What is an analog-to-IP adapter used for?
A. To translate between analog and IP signaling protocols for proper transport on
the PSTN
B. To translate between analog and IP signaling protocols for proper transport on an
IP network
C. To translate between voice and data signaling protocols for proper transport on the
PSTN
D. To translate between voice and data signaling protocols for proper transport on an
IP network
10. Which of the following does not reside in the call-processing agent UC model category?
A. Call agents
B. PBX agents
C. Voice gateways
D. Call control agents
11. Which signaling protocol that is compatible with Cisco IP phones is an IETF standard?
A. MGCP
B. H.323
C. SIP
D. SCCP
c01.indd 27 9/21/11 12:09:24 PM
28 Chapter 1

12. Which of the following is not a signaling protocol that can be configured on voice
gateways?
A. SPCP
B. MGCP
C. H.323
D. SIP
13. Which of the following Cisco routers cannot act as a Cisco voice gateway?
A. 2900 series ISR
B. 1800 series ISR
C. 1900 series ISR G2
D. 2600XM series
14. A 2900 Series ISR G2 with a voice gateway IOS and the proper modules can do all the
following functions except what?
A. Conference call offloading
B. SRST
C. Connect analog phones
D. Emergency Responder offloading
15. At which segment of the Cisco Unified Communications model is QoS handled?
A. Network infrastructure
B. Applications
C. Call processing agents
D. Endpoints
16. Which of the following is not a consideration when choosing a voice gateway signaling
protocol?
A. Call agent hardware and software
B. Legacy hardware used
C. Voice gateway hardware and software
D. Quality of Service requirements
17. What UC deployment model uses dispersed call agents that act as a single clustered voice
system?
A. Centralized services model
B. Distributed services model
C. Inter-networking of services model
c01.indd 28 9/21/11 12:09:24 PM
Review Questions 29
18. When using the distributed services UC deployment model, what additional feature is often
recommended?
A. WAN links 5 Mbps or higher
B. SRST
C. H.323 signaling
D. MGCP signaling
19. Which UC deployment models recommend QoS on WAN links? (Choose all that apply.)
A. Inter-networking of services model
B. Centralized services model
C. Distributed services model
20. When would a network administrator choose the inter-networking of services model over
the other three UC deployment models?
A. If there is a single building or campus site and only a few small remote offices.
B. If there are several large dispersed campus sites and WAN links are slow and/or unreli-
able.
C. If there are several large dispersed campus sites and WAN links are large enough to
handle voice traffic.
D. If there is a single building or campus site and WAN links are large enough to handle
voice traffic.
c01.indd 29 9/21/11 12:09:25 PM
30 Chapter 1

Answers to Review Questions
1. D. The central ofce is a geographically located ofce that houses PSTN switch equipment.
2. A. The demarc is the point where the wiring responsibilities are split between the private
owner and the phone company.
3. B. A tie trunk is the name used to describe a circuit that connects two PBX switches.
4. B. An interofce trunk is the name used to describe a circuit that connects two PSTN
switches located in separate COs.
5. A. Central ofces today have one or more area codes assigned to them and they maintain
pools of subscriber code numbers.
6. D. In the three-tiered PSTN hierarchy, CO-to-CO calling over interofce trunks would be
considered local calling. Long-distance charges would apply if a call needed to be sent to
the interexchange network.
7. B. The Cisco SPA 300 and SPA 500 series phones do not support SCCP.
8. B. The Cisco Emergency Responder falls within the applications category of the Unied
Communications model.
9. B. Analog-to-IP adapters sit on the edge of an IP network and translate analog signaling
into IP for proper transport on an IP network. This lets people continue to use analog
telephone hardware on an IP network.
10. B. PBX agents are not one of the three Cisco call-processing agents as dened by Cisco.
11. C. Most current Cisco phones can run either SCCP or SIP signaling protocols. SCCP is
Cisco proprietary while SIP is an IETF open standard.
12. A. SPCP is a modication of the Cisco proprietary SCCP signaling protocol that is used
only for communications between a CUCM Express call agent and Cisco SPA 300 and 500
series phones. Voice gateways cannot be congured with SPCP signaling.
13. C. The 1900 series ISR G2 cannot run voice IOS software and therefore cant be used as a
voice gateway.
14. D. The Cisco ISR can support numerous voice gateway functions, but it cannot ofoad
Cisco Emergency Responder duties.
15. A. QoS is congured and maintained on network infrastructure hardware.
16. D. Quality of Service requirements are not a factor when choosing between voice gateway
signaling protocols.
17. D. The geographical diversity model clusters call agents that communicate as a single unit
over WAN links.
c01.indd 30 9/21/11 12:09:25 PM
18. B. SRST is recommended when deploying the distributed services UC deployment model.
SRST is used when WAN links fail and remote sites need to make outbound calls.
19. C, D. The WAN is a critical component in the distributed services and geographical
diversity models. Because these two models have voice trafc sent across the WAN, QoS is
therefore recommended.
20. B. The inter-networking of services model is used in situations where there are several
dispersed buildings or campus sites and WAN links are not capable of handling voice
trafc.
c01.indd 31 9/21/11 12:09:25 PM
32 Chapter 1

Answers to Written Lab 1.1
1. Central ofce
2. Plant
3. Tie trunk, CO trunk, and interofce trunk
4. Endpoints, applications, call-processing agents, and network infrastructure
5. Key system
6. Endpoints
7. Voice gateway
8. Call processing
9. Analog-to-IP adapter
10. Distributed services
c01.indd 32 9/21/11 12:09:26 PM
Understanding Analog
and Digital Voice
THE FOLLOWING CVOICE EXAM OBJECTIVES
ARE COVERED IN THIS CHAPTER:
Describe the different types of voice ports and their usage.

Implement a gateway.
Configure analog voice ports.
Configure digital voice ports.
Chapter
2
c02.indd 33 9/21/11 11:47:48 AM
Unied Communications today still relies heavily on the
ability to connect to both legacy PBX systems and the public
telephone network. While it would be great to have an
end-to-end IP voice solution for every situation, that is simply not possible in many
businesses today. In reality, you will probably need to support legacy analog or digital
endpoints and circuits at some point.
This chapter provides a thorough introduction to analog and digital voice ports and
signaling protocols. We will also cover the analog-to-digital conversion process needed to
transform analog waveforms into binary code for transport on digital circuits. Once weve
covered the details of analog and digital telephony, you will learn how to congure the
basic settings on various analog and digital ports that are available on Cisco voice gateway
hardware.
Understanding Analog Voice
Ports and Signaling
Analog voice was the method used by the very rst telephones. The technology captures
sound and places it onto the wire using electrical currents. The process is fairly simple and
has worked now for 130 years or so, ever since the telephone was invented. While analog
ports are becoming extinct, there still are a number of situations where youre likely to
encounter analog devices and analog ports in your career. The following section covers
analog voice ports and their signaling techniques.
Analog Voice Port Types
From a Cisco perspective, there are three analog ports that you need to become familiar
with: FXS, FXO, and E&M. While many more analog port types are available out in the
wild, these are the three port types available as modules on Cisco voice gateway hardware.
Foreign Exchange Station Ports
Foreign Exchange Station (FXS) ports are used to connect plain old telephone service
(POTS) end devices to a voice gateway. FXS ports are also found in homes that directly
connect to the PSTN. FXS ports use two-wire cabling with RJ11 connectors.
c02.indd 34 9/21/11 11:47:49 AM
Foreign Exchange Office Ports
Foreign Exchange Ofce (FXO) ports connect the PSTN to a voice gateway. FXO ports
use the same two-wire RJ11 cabling that FXS ports utilize.
E&M Ports
E&M ports interconnect two PBX systems. The cabling uses either six or eight wires, which
are bundled into pairs of two. Unlike FXS and FXO ports, E&M ports can either use one
pair (two-wire) or two pairs (four-wire) for signaling purposes. This leaves two pairs for the
transport of voice communication. If you are not familiar with RJ48 cabling, it uses the same
eight-position, eight-contact (8P8C) modular connector that Ethernet uses. The difference
between RJ48 and RJ45 is in how the pins are wired. See E&M Signaling later in the
chapter for more about this signaling type.
Analog Voice Signaling
One of the rst technical objectives CVOICE candidates need to understand is how analog
telephones work on the PSTN. Telephones and telephone switches use signaling methods to
communicate various stages in the setup, transport, and teardown of a telephone call. Three
analog signaling categories will be covered in this section. Briey, they can be described as
the following:
Address Signaling Address signaling is the transmission of telephone digits from the
calling party phone to the called party phone. A unique sequence of digits identies each
individual phone on the network so the call reaches the correct destination.
Informational Signaling Informational signaling is feedback generated from the telephone
switch to the user in the form of tones or voice messages to inform the phone user what
state a call is in.
Supervisory Signaling Supervisory signaling detects changes in the status of the telephone
physical loop or trunk. The signaling is then used to set up and tear down calls. Loop-start
and ground-start analog signaling fall within this signaling category.
In addition to these three analog signaling categories that deal with signaling from the
customer premise equipment (CPE) to the PSTN, a separate set of signaling categories will
be detailed that cover signaling specically for E&M ports.
Address Signaling
A telephone number consists of a string of digits that uniquely identies a specic telephone
or telephone system on a voice network. When someone wishes to call another user, they pick
up a telephone handset and dial the unique digits that specify the telephone of the person they
wish to talk to. The interpretation and handling of the dialed digits are the responsibility of
address signaling. Two main methods are used to input telephone numbers using a telephone:

Pulse dialing

DTMF dialing
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36 Chapter 2
Understanding Analog and Digital Voice

Pulse dialing was the original method for dialing numbers on analog phones. Pulse dialing
is also known as rotary dialing because of the method used to input the numbers using the
phone handset. Figure 2.1 shows the design of a rotary phone pad.
Dial
stopper
1
2
ABC
3
DEF
4
GHI
5
JKL
6
MNO
7
PRS
8
TUV
9
XYZ
9
OPER
D
i
a
l
FI GURE 2.1 A rotary dial pad
As you can see, the digits are organized in a circle on a rotating disk. To dial a digit, the
user puts a nger in the numbered hole and turns the disk clockwise until the nger hits
the rotary dial stopper at the lower right of the disk. The user then releases the disk, and it
rotates on its own in a counterclockwise direction to its original starting position. During
this counterclockwise rotation the phone performs a series of on- and off-hook transitions.
The terms on-hook and off-hook refer to a mechanical switch that connects the telephone
circuit to power and disconnects it. The power is detected by the telephone company and
translated into dialed digits. Depending on how far the disk was rotated (based on the
digit that the user wanted to indicate) the number of on- and off-hook transitions will
specify a single number in the overall telephone number that the user wants to ring. Each
additional digit is entered this way until the complete number has been entered. The PSTN
phone switch then has the complete number of the phone that should be called, and the
other signaling steps are performed until the call has been established. As you can see, this
is a fully mechanical method of identifying digits of a phone number. It also can be time-
consuming to rotate the disk for so many digits. There is an old joke in the phone business
that if you never want to be bothered with people calling you, always request a phone
number with as many 9s and 0s in it as possible. That way, it takes so long to call you on a
rotary phone that people wont bother!
Analog telephone dialing evolved from the rotary disk to a DTMF or touch-tone pad.
DTMF stands for dual-tone multi-frequency. This method of entering telephone number
digits uses specic audible tones that are produced when a user presses a button on the dial
pad. A single button press emits two different tones simultaneously. The DTMF pad and
tone frequencies emitted for each button are shown in Figure 2.2.
c02.indd 36 9/21/11 11:47:50 AM
The A, B, C, and D tones in the diagram are simply additional tones that
function in the exact same way that the other tones do. The primary differ-
ence is that you dont typically have these buttons on telephone handsets.
The tones can be used for a variety of reasons depending on the voice net-
work being used.
The PSTN phone switch will recognize both of the simultaneous frequencies being
emitted and translate the combination into the digit that the user wishes to call. The switch
has timers for how fast or slow each number can be input into the dial pad before timeout
occurs and the user must hang up and redial the entire phone number. The least duration
between dialed digits is 45 milliseconds and the longest is 3000 milliseconds. As you can
imagine, touch-tone dialing is the far more efcient method of entering a telephone number
and is now used almost 100 percent of the time.
Informational Signaling
Many telephone users take informational signaling for granted until they make their rst
international call. Why is this, you ask? Since the telephone has been around for well over
100 years, most of us have grown up knowing what dial tones and ring-back tones sound
like in the part of the world we happen to reside in. Some people are surprised to nd out,
however, that the tones and cadences used to inform the calling party of the call progress
are very different around the world. When an overseas call is made, the user rst hears
their normal dial tone because they are connected to their local PBX or PSTN switch. The
caller then enters the international long-distance number and waits to hear the familiar
ring-back tone they are accustomed to. But because they are connecting to an international
telephone system, that ring-back tone will likely sound very different from what they are
used to. The remote-end switch always returns informational signaling, which may be
1
DTMF frequencies
1209 Hz 1336 Hz 1477Hz 1633 Hz
697 Hz
770 Hz
852 Hz
941 Hz
2
ABC
3
DEF
A
4
GHI
5
JKL
6
MNO
B
7
PQRS
8
TUV
9
WXYZ
C
* 0 # D
FI GURE 2. 2 DTMF pad and corresponding frequencies
c02.indd 37 9/21/11 11:47:51 AM
38 Chapter 2

located in a different country. The user may be confused by this and unsure if the call
is actually progressing as it should. It is at this time that many people realize just how
important and useful informational signals can be.
Information signaling consists of audible tones or recordings that indicate to the phone
user the status of the phone system and progress being made to place a call. Informational
signals are also commonly referred to as call progress (CP) tones. Table 2.1 lists the more
common informational signals and their functions.
Supervisory Signaling
An analog telephone has two wires that connect it to the PSTN. One is the ground wire (or
lead) and is called the tip wire. The other is the ring wire, which connects the phone to a
battery for power. This power source is a 48-volt DC battery. When a telephone handset
sits in the cradle and is not in use, the phone is considered to be on-hook. This means that the
circuit between the ring and tip is severed, and the battery (ring) cannot power the tip side
and therefore cannot signal to the PSTN that a user wants to use the phone. When the circuit
transitions to an off-hook state, the circuit is complete and the ring powers the tip, signaling that
digits will soon be entered into the handset that the PSTN switch must listen for and process.
The nal supervisory signal once the line is secured at both ends of the connection is to send an
electrical current to the receiving ends phone, which causes the telephone ringer to go off.
TABLE 2.1 Common informational signals
Informational Signal Type Signal Function
Dial tone Phone is in an off-hook state and ready to accept user input
with the keypad.
Busy Called number phone is currently in use.
Number not in service Called number is not available on the phone network.
Call waiting An incoming call is being made to line 2 on the phone;
line 1 is in use.
Ring-back The phone company is attempting to establish the
connection to the called party.
Reorder All local circuits are busy; thus the call cannot be completed.
Congestion The telephone network is unable to complete the call.
Receiver off-hook Someone has picked up the handset of a phone from
the cradle.
c02.indd 38 9/21/11 11:47:51 AM
There are three main types of analog supervisory signaling; each handles the on-hook
and off-hook process differently. Table 2.2 lists these methods, where they are commonly
used, and the analog interfaces they operate with on Cisco equipment.
The following sections cover each signaling technique to detail how on-hook and off-
hook transitions are handled.
Loop-Start Signaling
Most home analog telephones use loop-start signaling along the local loop. Signaling takes
place between the telephone handset and the PSTN switch. When the telephone handset is
in the phone cradle, the ring and tip connections are separated, as shown in Figure 2.3.
TABLE 2. 2 Supervisory signaling methods
Supervisory Signaling Method Common Usage Interface Types
Loop-start Home telephones FXS/FXO
Ground-start Business telephones FXS/FXO
E&M PBX to PBX E&M
Analog
phone
Ring
48 volt
PSTN central
ofce switch
Tip
FI GURE 2. 3 The on-hook status in loop-start signaling
With an FXS port, when the telephone user picks up the handset, a mechanical lever on
the phone lifts and completes the ring-and-tip circuit. If an FXO port is used, the interface
is responsible for completing the circuit loop. Figure 2.4 illustrates the off-hook status.
Analog
phone
Ring
48 volt
PSTN central
ofce switch
Tip
FI GURE 2. 4 The off-hook status in loop-start signaling
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40 Chapter 2

Once the circuit is complete, the 48-volt DC battery on the ring powers the tip. This
electricity is detected by the PSTN switch, and the switch monitors the line for telephone
number digits to be entered (either through pulse dialing or DTMF). From a PSTN
standpoint, this is known as a telephone line seizure.
Loop-start signaling suffers from two major limitations, which can cause problems for
users who make and receive frequent calls and are the reasons loop-start signaling is not
recommended for business phones.
The rst is the possibility that both the phone handset and the PSTN switch could
attempt to seize (use) the line at the same time. This is known as glare. For example,
suppose you have a home phone that uses loop-start signaling. On the other side of town,
your friend picks up the phone and calls you. Once your friend enters your telephone
number, the PSTN switch performs a lookup and determines that your phone line needs to
be seized so that the circuit can be completed between your friends phone and yours. The
CO switch seizes your line, but unfortunately, it takes up to four seconds from the time
the switch seizes your line to the time your phone actually rings to indicate that someone
is calling. Now suppose that during these four seconds, you decide to use your phone to
call your mother. You pick up your handset, but to your surprise, instead of hearing a dial
tone, you hear someone on the other end of the line. After a few seconds of confusion and
clarication, you and your friend realize that youve just experienced glare.
Also, when an FXO port is used with loop-start signaling, calls coming into the FXO
port may not properly disconnect. Remember that FXO ports are responsible for the on-
hook and off-hook transitions instead of the analog telephone itself. To the PSTN switch,
the FXO port looks like and is therefore treated just like an analog phone. The FXO port
can properly handle on- and off-hook transitions for outbound calls, but the same cannot
be said for inbound calls coming from the PSTN. The result is that the line sits in an off-
hook state long after the call ended and the circuit should have been severed.
Most businesses tend to make and receive more calls than home users. The increased
call frequency means there is a higher probability of experiencing glare. To avoid these two
problems with loop-start signaling, it is highly recommended that business analog phones
use ground-start signaling, discussed next.
Ground-Start Signaling
Ground-start signaling is used primarily on PBX-to-PBX or PBX-to-PSTN switch connections.
The major difference from loop-start is that ground-start signaling requires grounding end to
end before the ring and tip circuits are connected and the line is seized. This requirement that
the full path be free prevents the potential for glare found in loop-start signaling, which does
not verify the circuit path from the source to the destination.
Figure 2.5 details the process of transitioning a phone using ground-start signaling
from an idle on-hook state to an off-hook state. In each diagram, a PBX is connected to
the PSTN switch using ground-start signaling. Figure 2.5, Step 1, shows the PBX in an
on-hook state, with the tip and ring circuits severed at the PBX side. Also notice that the
local ring has a second severed connection on the ring wire, and the PSTN switch side has
a severed connection that feeds into a ring ground. This means that both the ring and tip
wires are disconnected from the ground.
c02.indd 40 9/21/11 11:47:52 AM
When an outside caller on the PSTN wants to call our local PBX, the PSTN switch will
ground the wire. This grounding is detected by the local PBX switch, which proceeds to
ground its ring wire, as shown in Figure 2.5, Step 2.
At this point the circuit is not yet completed, but the line is completely free from usage
end to end, so that when the line is seized, no glare will occur. The nal step in the process
is for the PBX to complete the circuit by connecting the tip and ring circuits together, which
transitions the circuit into an off-hook state. The PBX also removes the ring wire from the
ground. Figure 2.5, Step 3, details the completed circuit using ground-start signaling.
FI GURE 2. 5 Ground-start signaling
Step 1: Idle state
PBX
Ring
48 volt
PSTN central
ofce switch
Tip Ring
ground
Step 2: The ring grounded
PBX
Ring
48 volt
PSTN central
ofce switch
Tip Ring
ground
Step 3: The loop closed
PBX
Ring
48 volt
PSTN central
ofce switch
Tip Ring
ground
E&M Signaling
E&M signaling is a supervisory signaling type that is used to connect PBX switches. E&M
uses trunk lines for transport.
Youll nd more detail about trunk types later in this chapter. For now, note that E&M
signaling is used on trunk connections. A second important difference between E&M
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42 Chapter 2

signaling and loop-start/ground-start signaling is that E&M signaling can separate voice
from the signaling onto separate wiring.
E&M Physical Wiring Types
A great deal of discussion about E&M concerns the name itself. Many books and white
papers argue that E&M refers to earth and magnet; others argue that it means ear
and mouth as names for the wiring leads. For the purposes of this book, we will stick
with the terms earth and magnet for the leads. Regardless of the name debate, it is
important to note that there are six physical wiring methods for E&M signaling used
throughout the world. E&M species eight different wires, listed in Table 2.3.
E&M Type I Uses one pair of wires for E&M signaling. The PBX supplies battery power
for both the earth and the magnet leads. In an on-hook state, the earth lead is in an open
position and the magnet lead is connected to the ground. The PBX transitions from an
on-hook state to an off-hook state by connecting the magnet to the battery. The remote
line side then connects the earth lead to the ground, which completes the circuit. This is the
most common E&M method used in North America.
E&M Type II Uses two pair of wires for E&M signaling. It also offers the advantage of
producing a low amount of radiated interference, which can be benecial in environments
where there are many radio transmissions that are susceptible to interference. One of the
four E&M wires is used for the earth lead and a second wire is used as the magnet lead.
The other pair of wires is used as a signal ground (SG) and a signal battery (SB). Both
TABLE 2. 3 E&M wiring labels
E&M Wire Usage
E Signaling output
M Signaling input
SG Signal ground
SB 48 volt signal battery
T Audio input
R Audio output
T1 Secondary audio input (on four-wire E&M)
R1 Secondary audio output
c02.indd 42 9/21/11 11:47:54 AM
the earth and magnet leads are in an open state when on-hook. When transitioning to an
off-hook state, the PBX connects the magnet lead to the SB lead, and the remote line side
connects the earth lead to the SG lead, completing the circuit. You may come across E&M
type II signaling when working with legacy PBX systems.
E&M Type III Uses two pairs of wires for E&M signaling. Like types II and IV, it uses
one pair as the earth and magnet leads and the other pair for the SB and SG leads. The main
difference with E&M type II is that in the idle state for type III, the magnet lead is already
connected to the SB lead when on-hook. The PBX indicates an off-hook state by moving the
magnet lead from the SB lead to the SG lead. The remote line side will then ground the earth
lead, completing the circuit. E&M type III circuits are rarely used in production.
E&M Type IV Uses two pairs of wires for E&M signaling. Like types II and III, it uses
one pair as the earth and magnet leads and the other pair for the SB and SG leads. Also
like type II, E&M type IV signaling has both the earth and magnet leads in an open state
when on-hook. The PBX moves to an off-hook state by connecting the magnet lead to
the SB lead. The remote line then connects the earth lead to the SG lead, which is already
grounded, completing the circuit. E&M type IV is not widely used, and you are not likely
to see it in production.
E&M Type V Uses one pair of wires for E&M signaling. Similar to type I, this type uses one
wire lead for earth and the other lead for magnet. When the circuit is idle and on-hook, both
the earth and magnet leads are open. The PBX will go off-hook by grounding the magnet
lead. The remote line goes off-hook by grounding the earth lead. E&M type V signaling is the
most popular type outside North America.
SSDC5 Predominantly found in England, SSDC5 signaling uses one pair of wires and is
similar to E&M type V signaling except that the on-hook and off-hook states are ipped.
This is done so that if the trunk line breaks, the E&M interface defaults to an off-hook
state indicated by a busy signal.
Cisco hardware supports only E&M types I, II, III, and V.
While Cisco technically does not support E&M type IV on its hardware, it
operates identically to E&M type II except that the magnet lead operates
as a battery in type II when off-hook and as a ground when using type IV.
Cisco equipment can interface with type IV equipment if the necessary
magnet lead rewiring is done to account for this difference.
E&M Line-Seizure Signaling Types
In addition to the different physical wiring types, E&M has three on- and off-hook statuses
and line-seizure signaling methods: immediate-start, wink-start, and delay-dial. These
methods are illustrated in the following gures using the example of an E&M trunk between
two PBX systems. For the purposes of this example, the PBX on the left will always initiate
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44 Chapter 2

(the sending side) the off-hook status to the PBX on the right (the receiving side), which will
react to the change in status according to the signaling type detailed here:
Immediate-start The immediate-start signaling method is the easiest method to
comprehend. Figure 2.6 shows the process. In step 1, both PBX systems are in an idle and
on-hook state. Then, in step 2, the sending-side PBX wants to seize a line between the two
PBX systems, moves into an off-hook state, and informs the receiving-side PBX of this
state change. In step 3, after sending the off-hook notication to the receiving-side PBX,
the sending-side PBX waits 150 milliseconds before transmitting DTMF digits across the
trunk. This pause potentially gives the receiving-side PBX time to be notied so it can begin
listening for DTMF digits to be collected and processed. Finally, the receiving-side PBX
collects the DTMF digits, transitions to an off-hook state, and noties the sending-side PBX
of the transition change and line seizure.
Sending
PBX
Receiving
PBX
On-hook On-hook
Step 1: The idle state
Step 2: The sending side of-hook
Sending
PBX
Receiving
PBX
On-hook Of-hook
Going of-hook
Step 3: The pause before DTMF transmission
Sending
PBX
Receiving
PBX
On-hook Of-hook
DTMF digits
after 150 ms pause
Sending
PBX
Receiving
PBX
Of-hook Of-hook
Going of-hook
Step 4: The remote side of-hook
FI GURE 2. 6 Immediate-start E&M signaling
Wink-start This type of E&M signaling is very popular around the world. It is also the
default E&M signaling method on Cisco equipment. This type of signaling is different from
immediate-start signaling mainly because the sending PBX must receive feedback from the
c02.indd 44 9/21/11 11:47:55 AM
receiving PBX before sending any digits. This ensures that the receiving switch is ready
and able to collect and process digits. Figure 2.7 shows the process: In step 1, the two PBX
switches are in an idle and on-hook state. The sending PBX then (step 2) goes off-hook and
noties the receiving PBX. At this point, the sending-side PBX sits and waits for feedback
from the receiving-side PBX. When the receiving-side PBX sees the off-hook notication
from the sending-side PBX, it quickly transitions the trunk line off-hook and back on-hook.
This quick off-hook-to-on-hook transition is called a wink and lasts between 140 and 200
milliseconds. The remote-side wink is depicted in step 3. When the wink is sent across the
trunk to the sending-side PBX, that serves as a notication to go ahead and send the DTMF
digits across the trunk to the receiving-side PBX, as shown in step 4. The nal step is when
the remote-side PBX collects the DTMF digits and transitions to an off-hook state.
FI GURE 2. 7 Wink-start E&M signaling
Sending
PBX
Receiving
PBX
On-hook On-hook
Step 2: The serving side goes of-hook
Sending
PBX
Receiving
PBX
On-hook Of-hook
Going of-hook
Step 3: The remote side wink
Sending
PBX
Receiving
PBX
On-of-on hook Of-hook
Wink of/on
Step 4: DTMF transmission after receiving wink
Sending
PBX
Receiving
PBX
On-hook Of-hook
DTMF digits
after receiving wink
Step 5: The remote side goes of-hook
Sending
PBX
Receiving
PBX
Of-hook Of-hook
Going of-hook
c02.indd 45 9/21/11 11:47:56 AM
46 Chapter 2

Delay-dial This line-seizure signaling method is similar to wink-start in the fact that
the sending-side PBX waits to see if the receiving-side PBX is able to receive address
information in the form of DTMF digits. Figure 2.8 steps through the complete delay-dial
process. In step 1, both of the PBX switches are in an idle and on-hook state. In step 2, the
sending PBX goes off-hook and indicates this status change to the receiving-side PBX.
Step 3 differs from the other two E&M signaling methods in that the sending PBX is
responsible for checking to see if the receiving PBX is in an off-hook or on-hook state. If
the receiving-side PBX is off-hook already, the sending-side PBX will not send the DTMF
digits and will terminate the call accordingly. If the receiving-side PBX is on-hook, the
sending side will proceed to send address information across the trunk, as shown in step 4.
As with all E&M signaling, the final step in the process is the receiving-side PBX going off-
hook after receiving DTMF addressing and seizing the line.
Sending
PBX
Receiving
PBX
On-hook On-hook
Step 2: Sending side of-hook
Sending
PBX
Receiving
PBX
On-hook Of-hook
Going of-hook
Step 3: Sending side status check
Sending
PBX
Receiving
PBX
On- or of-hook Of-hook
Checks to see if
receiving PBX is on-
or of-hook
Step 4: DTMF transmission after receiving
side on-hook verication
Sending
PBX
Receiving
PBX
On-hook Of-hook
if Receiving PBX is
on-hook
DTMF digits
Step 5: The remote side goes of-hook
Sending
PBX
Receiving
PBX
Of-hook Of-hook
Going of-hook
FI GURE 2. 8 Delay-dial E&M signaling
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Basic Configuration of Analog Voice Ports
There are literally dozens of options to choose from when conguring analog voice ports.
For many situations, you can get by with just using the default settings and changing only
a few options. This section will go through the most basic FXS, FXO, and E&M port
congurations. Chapter 6, Conguring Voice Gateway Ports and DSPs, will cover
voice-port conguration options in greater detail.
Basic FXS Port Configuration
An FXS port most commonly connects to an analog telephone or fax machine. The
primary conguration option for this type of setup is whether the signaling type is loop-
start or ground-start. FXS ports connecting to analog phones typically use loop-start,
while specialized phones (such as pay phones) and FXS ports that connect a PBX to the
PSTN use ground-start. For example, these commands demonstrate how to congure an
FXS port for loop-start signaling on FXS port 0/0/0 of a Cisco router:
Router#configure terminal
Router(config)#voice-port 0/0/0
Router(config-voiceport)#signal loopstart
Router(config-voiceport)#
Additionally, depending on the phone type used and the country you reside in, you
may wish to congure settings that are specic to the PSTN standards in that area. These
include call progress tone and ring frequency. The call progress tone settings modify the
audible informational signaling tones, such as the ring-back tone or busy signal. These
tones vary widely depending on the part of the world. To modify these settings, you can
use the cp-tone command, as shown here:
Router(config-voiceport)#cptone ?
locale 2 letter ISO-3166 country code

AR Argentina IN India PE Peru
AU Australia ID Indonesia PH Philippines
AT Austria IE Ireland PL Poland
BE Belgium IL Israel PT Portugal
BR Brazil IT Italy RU Russian Federation
CA Canada JP Japan SA Saudi Arabia
CN China JO Jordan SG Singapore
CO Colombia KE Kenya SK Slovakia
C1 Custom1 KR Korea Republic SI Slovenia
C2 Custom2 KW Kuwait ZA South Africa
c02.indd 47 9/21/11 11:47:59 AM
48 Chapter 2

CY Cyprus LB Lebanon ES Spain
CZ Czech Republic LU Luxembourg SE Sweden
DK Denmark MY Malaysia CH Switzerland
EG Egypt MX Mexico TW Taiwan
FI Finland NP Nepal TH Thailand
FR France NL Netherlands TR Turkey
DE Germany NZ New Zealand AE United Arab Emirates
GH Ghana NG Nigeria GB United Kingdom
GR Greece NO Norway US United States
HK Hong Kong OM Oman VE Venezuela
HU Hungary PK Pakistan ZW Zimbabwe
IS Iceland PA Panama
Router(config-voiceport)#cptone
Another attribute international users may need to congure along with the call progress
tone is impedance, used to adjust the resistive strength that the attached analog device
is expecting in ohms. This example shows the different conguration options for setting
impedance on an FXS port:
Router(config-voiceport)#impedance ?
600c 600 Ohms complex
600r 600 Ohms real
900c 900 Ohms complex
900r 900 ohms real
complex1 220 ohms + (820 ohms || 115nF)
complex4 600r, line = 270 ohms + (750 ohms || 150nF)
complex5 320 + (1050 || 230 nF), line = 12Kft
complex6 600r, line = 350 + (1000 || 210nF)
Router(config-voiceport)# impedance
Analog input gain and output attenuation settings deal with how to adjust the volume
of the analog call in decibels (dB). (For reference, you should know that +3 dB = 2 power/
volume and 3 dB = power/volume for voice.) You should test carefully using these
settings, because you can make them either too high or low. This example shows an FXS
port being congured to add 2 dB of input gain and 1 dB output attenuation:
c02.indd 48 9/21/11 11:47:59 AM
Router(config-voiceport)#input gain 2
Router(config-voiceport)#output attenuation -1
Lastly, if you congure your FXS port with loop-start signaling, you know that you
have a risk of glare on the line. To help reduce instances of this, you can adjust the echo
cancellation coverage timer so that the voice port will wait the longer time it takes for
the router to keep a waveform in its memory. Here is an example of how to set echo
cancellation to 32 milliseconds:
Router(config-voiceport)#echo-cancel coverage 32
Basic FXO Port Configuration
Because FXO ports commonly connect to the PSTN, they should be congured with
ground-start signaling (although it is possible to have them congured with loop-start if
the device you are connecting to requires it). Here is an example conguring of ground-
start signaling on an FXO at port 0/1/0 on a Cisco router:
Router(config-voiceport)#signal groundStart
There are two other FXO conguration settings that you should be familiar with.
The rst setting deals with how the line will handle dialed digits. The two options are
pulse dialing and DTMF, and you congure them using the dial-type command, as
shown here:
Router(config-voiceport)#dial-type ?
dtmf touch-tone dialer
pulse pulse dialer
The other basic FXO command to consider determines the number of rings the local
voice router detects before answering an incoming call. When a voice gateway uses an FXO
port to connect to the PBX, it is the FXO port that actually answers the call as opposed to
an analog phone in an FXS port situation. Because of this, the FXO port can wait a certain
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50 Chapter 2

number of rings before answering the call. To change the number of rings that a router
waits to answer the call, you can use the ring number command, as shown here:
Router(config-voiceport)#ring number ?
<1-10> The number of rings detected before closing the loop
As you can see, the voice gateway can be congured to wait anywhere between 1 and 10
rings before answering the call (closing the loop).
Basic E&M Port Configuration
As you have seen in the preceding discussion, there are multiple ways to congure an E&M
port. Fortunately, as long as you know the settings your Cisco voice gateway requires to
connect two PBX systems, the basic conguration has only three primary options.
The rst option is to determine the E&M interface type that should be used. Again,
although there are multiple types, Cisco hardware supports only E&M types I, II, III,
and V. Here is an example of how to congure the E&M type on port 2/1/0 to use E&M
type II:
Router(config-voiceport)#type 2
Next, you should congure which physical wiring setup needs to be used on the
interface. Your choices are two-wire or four-wire. This is congured using the operation
conguration command as shown here:
Router(config-voiceport)#operation two-wire
Lastly, the E&M signaling type must be set according to what the PBX is expecting.
Your choices are wink-start, immediate, or delay-dial. To congure immediate
signaling, use the signal command and then choose an E&M signaling type, as
shown here:
Router(config-voiceport)#signal immediate
c02.indd 50 9/21/11 11:48:00 AM
Understanding Digital Voice
Ports and Signaling
Pure analog circuits are becoming extinct. After all, we live in a digital age. After the long
reign of analog telephone circuits, we entered a digitized circuit phase, which is now itself
being superseded by the contemporary Voice over IP stage, where we combine voice and
data on an IP network.
In this section, we will rst look at how analog signals are converted into digital signals.
This is a three-step process: sample, quantize, encode. A fourth step is often used on IP
voice networks to compress the signal, which helps voice to operate properly on links with
low latency and high bandwidth utilization. The remainder of this section will cover the
most popular digital circuits and the types of signaling they utilize.
An Overview of the Analog-to-Digital Conversion Process
As the PSTN grew both in numbers of phone lines and the geographical coverage across nations
and globally, the analog telephone suffered from this growth spurt in both of these areas.
The problem for coverage is that analog lines cannot travel long distances. Remember
that analog signals convert voice into electrical pulses. As these electrical pulses are sent
over a wire, they tend to get weaker and weaker the farther they travel. To help with this
problem, analog signal repeaters are installed on analog lines. These devices take in the
electrical pulses and add additional power to them, which helps them to travel longer
distances. Unfortunately, while the voice pulses are amplied, so is the background noise
that is inherently found on the line. Over time and multiple signal amplications, this noise
becomes so great that the voice quality suffers greatly.
Alternatively, when voice signals are sent in a digital format, instead of transmitting
electrical pulses over the wire, binary code in the form of 1s and 0s is transmitted. These
bits can much more easily and accurately be collected and retransmitted. Therefore, the
distance limitation is overcome with the use of digital circuits.
The other benet of transporting voice digitally is the ability to send more information
than you can in an analog format. Once analog signals are sampled, converted into binary,
and optionally compressed, multiple voice signals can share the same wire simultaneously,
whereas an analog signal requires the use of a dedicated pair of wires for transport. The
transmission of multiple voice calls on a single pair of wires lets digital systems scale much
more when compared to analog.
The following four sections cover the analog-to-digital conversion process. Technically,
only three steps are required:
1. Sample
2. Quantize
3. Encode
This process is often referred to as pulse-amplitude modulation (PAM).
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52 Chapter 2

But because this is a VoIP study guide, there is a fourth step that is often performed
when discussing IP voice codecs. The VoIP steps for digitizing voice are as follows:
1. Sample
2. Quantize
3. Encode
4. Compress (optional depending on codec used)
Lets look at each of these four steps so you can understand how this process works.
Sample the Voice Signal
Dr. Harry Nyquist, a Bell labs engineer working in the 1920s, is credited with determining
the optimal method for sampling the human voicethe rst step in digitizing voice on
digital telephone systems. This method is known as the Nyquist Sampling Theorem.
Dr. Nyquist knew that most of human speech falls within the range of 200 to 2800 Hz.
Telephones have whats known as a low-pass lter, which restricts the range of audio from
300 to 3300 Hz. For manufacturing cost reasons, these lters are not entirely accurate, so
Nyquist used the maximum collected frequency to be 4000 Hz. His research concluded
that if a sample is taken at two times the highest frequency, this sample could almost
perfectly replicate the full analog signal when reconstructed. This results in the following:
sample_size = 2 4000 Hz (cycles per second)
or
sample_size = 8,000 cycles (times) per second
The result of this sampling is referred to as a discrete signal. Figure 2.9 shows an analog
signal and the Nyquist sampling algorithm at work.
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Using this sampling method we can accurately sample voice and move onto the next step
of the process, which is quantization.
Quantize the Collected Samples
Now that we have our discrete signal by sampling our analog waveform, we must translate
the samples into some type of numbering sample. This process is called quantization
or pulse-code modulation (PCM). For voice, each of the collected samples is assigned a
numerical value based on a reference scale of 0 to 255. This process of quantifying each
sample is shown in Figure 2.10.
Now we have our samples and we have quantized them into 8-bit numbers; next we
need to encode the signal.
Encode the Quantized Samples
Even though we have quantized our samples into a numbering system using the range 0 to
255, computers understand only binary numbers, which are series of 1s and 0s. Encoding
is the process of taking the quantized samples and translating them into binary. It is no
coincidence that PCM uses a range for quantizing analog samples between 0 and 255. This
nicely translates into an 8-bit binary format. It is much like IP version 4 addresses, which
use four 8-bit octets to specify the address of a network device. Figure 2.11 shows the
encoding process in action.
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FI GURE 2.10 Quantizing each sample
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54 Chapter 2

As stated earlier, legacy digital systems stop at this point. The result is a 64 Kbps digital
voice stream that can easily be replicated accurately over and over, which overcomes the
analog distance limitation. We arrive at the 64 Kbps number because we are sampling
our analog signal 8,000 times per second. These 8,000 samples are then converted into an
8-bit binary number. Doing the math, we arrive at the following:
bit_rate_per_second = 8,000 samples per second 8 bits
bit_rate_per_second = 64,000 bits_per_second
64,000 bits_per_second = 64 Kbps
In addition, digital streams can be multiplexed over a single cable. Multiplexing is
the process of transporting multiple signals over the same cable medium. When multiple
streams are transported across a single cable, the cable is known as a digital trunk. Using
multiplexed digital trunks helps overcome the growth limitation of analog signals.
The nal step we will investigate briey is the optional fourth step of compression,
which is found in most VoIP codecs in use today. The only popular voice codec that does
not use compression is G.711, which sends voice packets in uncompressed 64 Kbps streams.
All other codecs described in the book use compression of some kind.
Compression of the Encoded Sample
Bandwidth on a given network is xed for the most part. Sure, upgrades can be performed
to increase bandwidth, but networks are built to be in place for multiple years. Because
networks typically grow over time, voice and data applications are constantly being
required to use less and less bandwidth, a limited commodity. The move from analog
to digital helped to allow multiple voice calls to share the copper medium. The 64 Kbps
stream is great, but eventually it also was too large. This required an additional step
called compression, which attempts to eliminate redundant 8-bit binary samples on the
FI GURE 2.11 Encoding the quantized sample
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receiving end by using a known sample or group of samples, which then is turned into
a smaller representation of the signal and sent across the wire to the remote end. When
the compressed signal is then decoded, it is approximately the same signal as prior to
compression.
Obviously, some compression methods are better than others. You probably can identify
voice codecs that use a high amount of compression. The voice sample after being decoded
on the other side turns a human voice into a much choppier and more robotic sound. This
means that compression has a denite tradeoff between bandwidth savings and voice quality.
What in the World Is Companding and Where Does It Fit In?
Oftentimes when discussing the analog voice and analog-to-digital conversion, the
topic of companding arises. It is a difcult concept to grasp because it has multiple uses
depending on the type of voice signal that the process is being used on.
The process of companding was originally used in analog systems to increase the
signal-to-noise ratio (SNR). With voice transmissions, the algorithm helped to amplify
the signal of a human voice while reducing background noises that are picked up by the
microphone.
In the migration from transporting analog signals to transporting digital, it was
discovered that the biggest benet to using companding methods was not so much to
increase SNR but instead to reduce the total number of bits that were required for the
digital circuit to be encoded and transported. The number of bits used to represent a
moment of voice over a period of time is called a sample size.
Therefore, the more modern denition of companding in a digital sense consists of rst
compounding the analog voice signal before it is input to an analog-to-digital converter
(ADC) and then, after the signal is digitized and transported, expanding it on the other
end of the call by a second ADC. The expanded signal is then sent to the receiver part of
the phone. (From compounding and expanding, we get the term companding.)
One popular companding algorithm is u-law, which is performed primarily on T1 circuits
in North America and J1 circuits in Japan. In regions such as Europe, the A-law algorithm
is used instead; it is similar to the u-law algorithm but performed on E1 circuits. It was
discovered that by using u-law and A-law, the sample size could be made as small as
8 bits per sample. In a way, it is the earliest form of compression but is technically not
considered as such. These u-law and A-law algorithms are dened in the G.711 (aka
pulse-code modulation (PCM)) standard. Companding is the only compression-like
technique used by PCM.
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56 Chapter 2

Digital Voice Port Types
Cisco offers several digital voice interface types that can be installed on voice gateways.
The major differences between these interfaces involve the number of simultaneous calls an
interface can handle and the geographic region where some interfaces are more likely to be
found than others. In addition, there are other differences such as the signaling, framing, and
line coding, which will be explained in the next section. For now, lets briey look at the major
differences between the T1, E1, and ISDN (Integrated Services Digital Network) BRI ports.
T1 Port
A T1 port consists of 24 separate channels, each operating at 64 Kbps. Depending on the
signaling type used, a T1 can provide either 24 simultaneous calls using Channel Associated
Signaling (CAS) or 23 simultaneous calls using Common Channel Signaling (CCS). A T1
using CCS is often called an ISDN PRI.
You may jump to the conclusion at this point that CAS is the better option because
it offers the ability to use one additional line. Make sure to read the next section, which
describes how CAS and CCS signaling work. In fact, CCS signaling is much more popular
in PSTNs today. T1 circuits are used in public networks in North America (United States,
Canada, and parts of the Caribbean).
E1 Port
An E1 port is closely related to the T1 in many ways. In fact, the E1 interface is used instead
of T1 almost everywhere outside North America and Japan. An E1 port has 32 channels,
which operate at 64 Kbps. Also, similar to the T1, an E1 port can use either CCS (E1 PRI) or
CAS signaling. E1 and T1 ports differ in the type of framing used to transport data across the
wire. An E1 port is capable of sending 30 simultaneous voice calls using either CAS or CCS.
ISDN BRI Port
The ISDN Basic Rate Interface (BRI) is a three-channel 64 Kbps port. Two of the 64 Kbps
channels are for the transport of voice/data, and the third 16 Kbps channel is for signaling.
An ISDN BRI can be thought of as the smaller version of the ISDN T1/E1 PRI, because it
operates identically using CCS. ISDN ports are in use by public telephone companies all
around the world.
Digital Voice Multiplexing, Framing,
and Physical Transport
As stated earlier, one of the primary benets of digital circuits over analog is their ability to
transport multiple calls on a single pair of wires. This is known as multiplexing. The most
common type of digital circuit multiplexing is known as time-division multiplexing (TDM).
All the previously mentioned circuits (T1, E1, and ISDN BRI) use TDM. TDM is a strict time-
based method for sharing a single cable to transport multiple voice signals. Lets take a T1
c02.indd 56 9/21/11 11:48:03 AM
circuit that uses 24 logical TDM channels, for example. Each channel is assigned a timeslot,
which reoccurs in a specic order. The timeslot is how TDM functions. It is the time that each
channel within a trunk has to transport voice. When voice data from channel 1 is sent, the
next portion of that voice call must wait until the timeslots for the other 23 channels have had
their opportunity to send data on the same wire. This is true even if the channels have nothing
to send. When each channels time is nished, the circuit moves to the next channel to either
send voice data or wait until the time limit expires and moves on to the next channel. Each
channel on a T1 circuit can send 8 bits of data at a time, as shown in Figure 2.12.
Recall from earlier in the chapter that using the Nyquist theorem, we need to be able to
send 8,000 samples each second for a single voice call. Each TDM cycle on a T1 is 193 bits.
To get to this number we multiply 8 bits for each sample sent per channel by the number of
channels, which is 24. This gives us 192 bits. Then we add 1 additional bit for framing and
we come to 193 bits for each TDM cycle.
We can then take the 8,000 sample size and multiply it by 193 bits to get the total
bandwidth of a T1 circuit:
T1_throughput = 193_bits 8,000 samples per second
T1_throughput = 1,544,000 bps
T1_throughput = 1.544 Mbps
So that is how multiplexing works to send 24 separate voice streams simultaneously over
a single copper cable. But the voice gateways need some method to package all these TDM
channels together for proper transport to the next voice gateway. This is where framing
comes into play. Historically there are two types of voice framing, but only one is in almost
universal use today.
The rst framing type is called Super Frame (SF). This bundles 12 TDM channel
cycles together in a single frame. Each frame is 193 bits in size. Super frames are the older
framing type and are almost never seen today. Because each TDM channel is 8 bits and we
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FI GURE 2.12 T1 TDM channels
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58 Chapter 2

bundle 12 of them together, that comes out to 192 bits. SF uses the extra bit to signal the
end of a TDM cycle. It does this by sending a special 12-bit pattern using bit 193 twelve
times, as shown in Figure 2.13.
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FI GURE 2.13 Super Frame
Extended Super Frame (ESF) has replaced SF because it requires less synchronization
between gateway endpoints and it includes a cyclic redundancy check (CRC) feature for
more reliable transport. Unlike SF, ESF bundles 24 TDM channel cycles together in a
single frame. Again, the last bit of the 193-bit cycle is used. Because ESF is bundling 24
cycles together in a frame, it has the benet of 24 bits per channel for framing and other
purposes. I say other purposes because ESF requires only 18 bits for framing 24 TDM
cycles instead of the 12 bits for each 12 TDM cycle required by SF. This frees up 6 bits that
are used for the CRC. More specically, ESF frames are used in the following manner:
Framing pattern: 4, 8, 12, 16, 20, 24
Data link control: 1, 3, 5, 7, 9, 11, 13, 15, 17, 19, 21, 23
CRC: 2, 6, 10, 14, 18, 22
Figure 2.14 shows how an Extended Super Frame is constructed.
FI GURE 2.14 Extended Super Frame
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One nal digital circuit parameter to mention is the circuits physical layer line-coding
characteristics. Choosing a line code type dictates how bits are sent across the digital wire,
that is, how 1s and 0s are represented electrically. On digital copper circuits such as a
T1, no change in voltage during a time frame represents binary 0, and either a positive or
negative voltage on the wire represents a binary 1.
There are two common standards in use today. The older standard is called alternate
mark inversion (AMI). AMI logically is transferred across electrical copper wiring, as
shown in Figure 2.15.
FI GURE 2.15 AMI Transport
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same electrical polarity.
Along those lines, 0s in AMI line coding are always represented with a 0 current. This
method works sufciently for up to seven 0s in a row. But if the eighth bit is also a 0, you
will receive an error on the line, because of the fact that bipolar switch equipment has
difculties dealing with 8 bits in a row with 0 voltage. Because of this limitation, a new
method of line coding was developed.
Bipolar 8-bit Zero Substitution (B8ZS) was created to eliminate the AMI methods
8-bit 0 problem. As the name states, this line-coding method will substitute for a series of
eight 0s on the wire (which would create an error) a specic pattern that is known by both
the sending and receiving voice gateways. There are two possible patterns that B8ZS will
use to display 8 bits of consecutive 0s, depending on the polarity of the previous bit sent.
Therefore, you will see one of the polarity patterns shown in Figure 2.16 to represent eight
0s using B8ZS.
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60 Chapter 2

Digital Voice Signaling
Just as with analog signaling, there are several types of digital voice signaling that you need
to understand; each method operates differently and affects the number of simultaneous
calls allowed on a single circuit. We briey discussed CAS and CCS earlier in this chapter.
This section will cover these two in greater detail and introduce two signaling subtypes,
Q.931 and QSIG.
Channel Associated Signaling
Channel Associated Signaling (CAS) is a PSTN signaling type that allows for up to 24
simultaneous calls. In order to squeeze 24 calls into an SF or ESF frame, CAS uses whats
known as in-band or robbed-bit signaling (RBS). RBS will take bits from SF framing
channels 6 and 12 and ESF framing channels 6, 12, 18, and 24 for sending signaling data
from one end of the digital circuit to the other.
This stealing of bits is done to maximize the number of calls a CAS T1 can handle. The
downside to stealing a bit used for voice is that the quality of the call suffers because you
are only able to use 56 Kbps as opposed to 64 Kbps. While this does indeed degrade the
quality of the digital voice signal, the difference between the two is slight to the ear.
CAS can also be used on E1 circuits, but it operates there very differently than with T1.
Of the thirty-two 64 Kbps channels an E1 circuit has, channel 1 is used for framing and
synchronization and channel 17 is used for signaling. Therefore, even though an E1 CAS
uses RBS, it still requires a dedicated channel on 17 for signaling. The difference between
a T1 CAS and an E1 CAS boils down to the fact that E1s do not use SF or ESF. Instead
they use a multiframe method (the details of which are outside the scope of this book) that
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It is important to understand that while an ISDN circuit has a bit rate of 192
Kbps, you cannot count framing and synchronization as bandwidth. There-
fore, an ISDN BRI actually has a bandwidth of 144 Kbps.
bundles channels into groups of 16. In this way, channel 1 is used to designate the start of the
rst 16-multiframe group, while channel 17 is used to designate the second grouping of 16.
Common Channel Signaling
Unlike CAS digital circuits that sacrice voice quality to maximize the number of simultaneous
calls, Common Channel Signaling (CCS) sets aside one or two dedicated channels for
signaling, a method known as out-of-band signaling. This means that these signaling
channels cannot be used for voice calls. However, the benets are that the remaining channels
use a full 64 Kbps for higher call quality, and the CCS channels have additional bandwidth
that can be used to provide additional value-added services to voice circuits.
By far the most common digital circuit transport standard that uses CCS is Integrated
Services Digital Network (ISDN). ISDN is a standard suite of protocols that operates on
layers 13 of the OSI model. ISDN utilizes PSTN circuits running CCS for the transport of
voice, data, and video. ISDN differs greatly from other analog and digital methods in the
fact that it can transport voice and data on the same circuit-switched connection. There are
two types of ISDN in use within phone companies today.
ISDN BRI An ISDN Basic Rate Interface (BRI) consists of three 64 Kbps digital
channels. Two 64 Kbps channels are called B, or bearer, channels. ISDN BRI circuits are
commonly found at remote sites or other locations where only two voice/data connections
are needed. These channels are responsible for transporting voice, data, or video on the
circuit. The other channel is called the D, or delta, channel. But this channel is further
broken down. 16 Kbps of the D channel are used for signaling, while the other 48 Kbps of
bandwidth are used for framing and synchronization. Table 2.4 outlines the segmentation
of the 192 Kbps ISDN BRI bit rate:
TABLE 2. 4 ISDN BRI channels
ISDN Segment Purpose Bandwidth Allocated
B channel 1 Voice/data transport 64 Kbps
B channel 2 Voice/data transport 64 Kbps
D channel CCS signaling 16 Kbps
D channel Framing and synchronization 48 Kbps
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62 Chapter 2

ISDN PRI An ISDN Primary Rate Interface (PRI) circuit is the big brother of the ISDN
BRI. It is found in larger environments where multiple voice and/or data connections
are needed. Depending on the interface type used, a PRI can carry 23 (T1) or 30 (E1)
simultaneous B channels for voice or data transport. Again, out-of-band signaling is used
in ISDN, and a 24-channel T1 circuit will set aside a full 64 Kbps for D channel signaling.
But unlike the ISDN BRI circuit, PRI requires the full 64 Kbps to support the transport of
23 voice/data channels. The D channel of a T1 circuit is almost always the last one, channel
24 when all channels are numbered. The larger ISDN E1 interface has 32 channels for use.
With PRI, ISDN uses 30 B channels for the transport of voice or data and 1 D channel.
Channel 17 is used for the single D channel, while channel 1 is designated for framing and
synchronization, similar to an ISDN BRI circuit.
Q.931 Signaling
The D channel of both ISDN PRI and BRI circuits uses the Q.931 signaling protocol. Q.931
is an ITU-T (International Telecommunications Union Telecommunication Standardization
Sector) standard that is responsible for the setup and teardown of B channel connections
whether they are voice or data connections. The protocol uses signaling messages between
voice gateways, including commonly used signaling messages such as these:

Call setup

Call in process

Remote end ringing

Call connected

Call disconnected

Release channel
QSIG
You may run into situations where a PBX hardware vendor uses a proprietary ISDN
signaling protocol. Unless you have the same PBX on both ends of your circuit, you need a
way so the two PBXs or voice gateways can properly communicate signaling information
back and forth. Q signaling (QSIG) is an open standard protocol designed to overcome
this exact issue. QSIG uses Q.931 as its underlying signaling protocol but tweaks it so
that other proprietary ISDN signaling protocols can also be used and understood.
These additional proprietary signals can then be used within a network that has
different PBX systems but with the appearance of one unied system with uniform services
across the board. QSIG not only helps in the administration process of operating different
PBX systems in a single organization, it also helps the end users because every phone
operates the same way and they can utilize the same voice services no matter what PBX
they are connected to. Some examples of QSIG signaling messages include these:

Caller ID

Call transfer
c02.indd 62 9/21/11 11:48:07 AM

Call redirect

Do not disturb
Basic Configuration of Digital Voice Ports
When working through the FXS, FXO, and E&M analog conguration ports, you may
have noticed that all of these ports were congured by issuing the voice-port command.
Similarly, digital voice ports are congured using the controller command. The
next section will go through how to congure basic settings for a T1 CAS and ISDN T1
PRI port.
Basic T1 CAS Configuration
T1 CAS circuits can be congured with either Super Frame or Extended Super Frame,
using the framing command. You need to set the framing type to match what your
PSTN provider has congured on its end. As was stated earlier, ESF is used almost
exclusively everywhere in the world. Here is an example of choosing ESF framing on
T1 port 2/1:
Router#config t
Router(config)#controller t1 2/1
Router(config-controller)#framing ?
esf Extended Superframe
sf Superframe
Router(config-controller)#framing esf
Router(config-controller)#
Next, you need to choose the type of T1 CAS linecoding you wish to use, with the linecode
command. The two options are AMI and B8ZS. We will congure our T1 for B8ZS:
Router#config t
Router(config-controller)#linecode ?
ami AMI encoding
b8zs B8ZS encoding
Router(config-controller)#linecode b8zs
Next, you need to determine how the T1 interface will receive its clocking for TDM,
using the clock source command. The two options are internal, which uses the routers
local clock, and line, which will receive clocking from the remote-side router. The
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64 Chapter 2

command to set one of these clocking options is clock source. We will congure internal
clocking for our T1 here:
Router#config t
Router(config-controller)#clock source ?
internal Internal Clock
line Recovered Clock
Router(config-controller)#clock source internal
Lastly, a very functional feature for a T1 CAS is the ability to break a 24-channel T1
into two or more DS0 groups. A DS0 is the name for a single 64 Kbps channel of a CAS.
The T1 can be logically split using the ds0-group command so different signaling can be
used and the different channels can ultimately serve multiple uses. To do this, you create
separate DS0 group numbers and specify the timeslots. You then congure the signaling
type that you wish these timeslots to utilize. The following conguration shows a new DS0
group (group 0) using timeslots 112 and the different signaling types available:
Router(config-controller)#ds0-group 0 timeslots 1-12 type ?
e&m-delay-dial E & M Delay Dial
e&m-fgd E & M Type II FGD
e&m-immediate-start E & M Immediate Start
e&m-wink-start E & M Wink Start
ext-sig External Signaling
fgd-eana FGD-EANA BOC side
fgd-os FGD-OS BOC side
fxo-ground-start FXO Ground Start
fxo-loop-start FXO Loop Start
fxs-ground-start FXS Ground Start
fxs-loop-start FXS Loop Start
none Null Signalling for External Call Control
<cr>
Basic ISDN PRI Configuration
Because a T1 PRI uses ISDN with Q.931 signaling, you must know what type of ISDN
switch you are connecting the port to. To set the ISDN type, use the isdn switch-type
command. Please note that conguring the switch type is a global conguration command
c02.indd 64 9/21/11 11:48:08 AM
and not a config-controller command. Listed in the following code snippet are the
various switch-type options available. In our example, we will congure the switch to use
primary-5ess.
Router(config)#isdn switch-type ?
primary-4ess Lucent 4ESS switch type for the U.S.
primary-dms100 Northern Telecom DMS-100 switch type for the U.S.
primary-dpnss DPNSS switch type for Europe
primary-net5 NET5 switch type for UK, Europe, Asia and Australia
primary-ni National ISDN Switch type for the U.S.
primary-ntt NTT switch type for Japan
primary-qsig QSIG switch type
primary-ts014 TS014 switch type for Australia (obsolete)
Router(config)#isdn switch-type primary-5ess
Router(config)#
The primary ISDN type used by PSTNs in the United States is the primary-5ess option.
Also note the primary-qsig option, which is used when connecting to PBX systems that
use proprietary signaling, as we discussed earlier in the chapter.
Next, just as with T1 CAS, the framing, linecoding, and clock source must be
congured on an ISDN PRI port. Here is an example of a T1 PRI congured for ESF
framing, B8ZS as the linecode type, and receiving the clocking from the remote router:
Router(config-controller)#clock source line
Lastly, an ISDN T1 PRI port must be congured to designate which timeslots of the PRI
will be used, by using the pri-group timeslots command. This setting is important when
you order a fractional T1 PRI from the PSTN. A fractional PRI is a circuit that looks and
acts like a standard 24-channel T1 but in fact has a smaller number of channels available
for use. PSTNs often offer fractional PRI circuits to customers who need between 5 and 15
digital lines but not a full 24 channels. Here is an example of how to congure a T1 PRI to
use all 24 channels:
Router(config-controller)#pri-group timeslots 1-24
c02.indd 65 9/21/11 11:48:08 AM
66 Chapter 2

Summary
Chapter 2 gave you a bit of a history lesson about how analog and digital voice circuits
function at a physical level, along with transport and signaling methods. You also went
through the three required steps necessary for the analog-to-digital transformation to
occur. Lastly, you were shown the basic conguration commands necessary to congure
various analog and digital voice ports that are available on a Cisco voice gateway.
While future chapters in this book will expand on conguration methods used in
conguring legacy ports, Chapter 2 gives a detailed insight into how analog and digital
circuits work. Everything from electrical transport to the various signaling methods used
is important for every CVOICE candidate to understand. The more you understand how
underlying protocols work, the better you can understand why you are conguring specic
settings and how the circuit will operate differently.
Exam Essentials
Know the three different analog ports used on Cisco gateways. The three analog ports
are FXS, FXO, and E&M ports.
Understand the three analog signaling types. Address signaling controls the transmission of
telephone numbers. Informational signaling instructs callers as to how a call is progressing
on the network. Supervisory signaling controls the on- and off-hook status of a telephone
connection.
Know the difference between loop-start and ground-start signaling. Loop-start signaling
is primarily used on ports that connect to analog endpoints and use a simple method for
telephone line seizure. It is also prone to glare. Ground-start signaling overcomes glare by
requiring an end-to-end grounding before the line is seized. Ground-start signaling is often
used between PBX systems or between a PBX and the PSTN.
Understand the six types of E&M physical wiring. E&M is a signaling protocol used
between PBX systems. While all six wiring schemes use RJ-48 connections, the wiring
differs in the number of pairs used for signaling and the method used for signaling on- and
off-hook transitions.
Know the three different types of E&M supervisory signaling. E&M immediate-start,
wink-start, and delay-dial signaling differ in the way the calling switch sends address
information to the receiving switch. Immediate-start does a simple pause before sending,
wink-start waits for an on-off-on signal from the receiving switch, and delay-dial checks
the status of the trunk to the receiving switch before sending DTMF digits.
Understand and be prepared to configure analog voice ports. Analog voice ports are
congured within the config-voiceport mode on a Cisco voice gateway. There are several
options that are unique to each analog voice port that you should familiarize yourself with.
c02.indd 66 9/21/11 11:48:09 AM
Know the three required and one optional step in the analog-to-digital conversion process. In
order to transport voice on a digital circuit, the analog signal must rst be sampled. Second,
each sample must be quantized. Third, the quantized sample must be encoded into binary.
And the fourth, optional step is compressing the encoded data.
Know the three different digital ports used on Cisco gateways. The three digital ports are
T1, E1, and ISDN BRI.
Understand the purpose of TDM. TDM is a multiplexing method that uses time-based
slots for transport of multiple voice signals across the same wire.
Understand the difference between SF and ESF. SF bundles 12 TDM cycles together on a
T1 and uses 12 bits for framing. ESF bundles 24 TDM cycles together on a T1 and uses 18
bits for framing. The additional 6 bits inside the ESF are used for CRC.
Understand how B8ZS overcame the 8-zero problem found in AMI linecoding. B8ZS
uses a unique positive/negative voltage pattern to represent 8 binary 0s in a row.
Know the primary difference between CAS and CCS signaling. CAS uses in-band or
robbed-bit signaling, while CCS uses out-of-band signaling. There are pros and cons to each
signaling type, although you are more likely to see CCS circuits in use today because of their
ability to transport both voice and data, whereas a CAS circuit can only transport voice.
Understand and be prepared to configure digital voice ports. Digital voice ports are
congured while within the config-controller mode on a Cisco voice gateway. There
are several options that are unique to each digital voice port that you should familiarize
yourself with.
Written Lab 2.1
1. What is the type of analog port that commonly connects a PBX to the PSTN?
2. Which supervisory signal type connects the ring to the tip?
3. What term describes an occurrence found in loop-start signaling in which a phone line
is seized several seconds before a ring noties the called party of an inbound call on
their phone?
4. Pulse dialing and DTMF are what type of signaling?
5. Which E&M signaling protocol waits 150 ms before sending address information to
the telephone switch?
6. Write the config-voiceport command used to change an FXS or FXO port to use
loop-start signaling.
7. Write the command used to increase the output strength of an analog signal by +2 dB
while in config-voiceport mode.
Written Lab 2.1 67
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68 Chapter 2

8. Write the command used to congure a T1 controller to use the remote-side gateway
for clocking while in config-controller mode.
9. Write the command used to congure Extended Super Frame (ESF) on a T1 while in
config-controller mode.
10. Write the command used to enable channels 124 on a T1 PRI port while in config-
controller mode.
c02.indd 68 9/21/11 11:48:09 AM
Review Questions
1. Which supervisory signaling type is most recommended for connecting a business PBX to
the CO?
A. Loop-start
B. E&M
C. Ground-start
D. Local-loop
2. What process does DTMF use to specify dialed digits?
A. Combining two audio tones to specify a single digit
B. Using rapid on-hook and off-hook transitions to specify a single digit
C. Using a single audio tone to specify a single digit
D. Grounding the tip to ring in rapid succession to specify a single digit
3. Which type of analog signaling is used to alert a phone user to the progress a telephone call
is making?
A. Address
B. E&M
C. Informational
D. Relation
E. Supervisory
4. Ground-start signaling requires what before the ring and tip circuits are connected for line
seizure?
A. The local side is grounded.
B. The remote side is grounded.
C. Both the local and remote sites are grounded.
D. The trunk line is grounded.
5. Which E&M signaling methods verify that the remote switch is ready to receive address
signaling? (Choose all that apply.)
A. Wink-start
B. Immediate-start
C. Monitor-start
D. Delay-start
Review Questions 69
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70 Chapter 2

6. An E&M switch sends a notification to the remote side switch indicating its off-hook
status. The switch then waits for an on-off-on hook response prior to sending DTMF digits
across the trunk. Which E&M signaling type is being used?
A. Delay-start
B. Immediate-start
C. Ground-start
D. Loop-start
E. Wink-start
7. What informational signaling type specifically indicates that the telephone network is
unable to complete the call?
A. Dial tone
B. Reorder tone
C. Receiver off-hook
D. Congestion
8. Which of the following E&M wiring types is not supported on Cisco voice gateways but is
very similar to type II?
A. Type III
B. Type IV
C. Type V
D. SSDC5
9. What is the config-voiceport command used to configure an analog voice port to use the
standard informational signals used in Taiwan (TW)?
A. signal TW
B. impedance TW
C. cptone TW
D. dial-type TW
10. What is the config-voiceport command used to adjust the volume on an incoming analog
voice signal?
A. input gain
B. input impedance
C. input attenuation
D. input cptone
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11. What FXS port command is used to adjust the resistive strength to 600r, which the
attached analog phone is expecting?
A. Router(config-voiceport)#input gain 600r
B. Router(config-voiceport)#impedance 600r
C. Router(config-voiceport)#input 600r
D. Router(config-voiceport)#impedance gain 600r
12. Which of the following is not an E&M signaling type?
A. Router(config-voiceport)#signal delay-dial
B. Router(config-voiceport)#signal groundStart
C. Router(config-voiceport)#signal wink-start
D. Router(config-voiceport)#signal immediate
13. In the analog-to-digital process, what is the name of the step that assigns a number to a
specific sample?
A. Compound
B. Encode
C. Quantize
D. Compand
14. In the analog-to-digital process, what is the name of the step that converts the samples into
binary?
A. Compand
B. Encode
C. Compound
D. Quantize
15. What is the name for the process that allows multiple call streams to be transported on a
single pair of wires?
A. Framing
B. Linecoding
C. Multiplexing
D. Companding
16. What problem found in AMI linecoding does B8ZS fix?
A. The problem with sending 8 binary 0s in a row on the wire
B. The problem with sending 8 binary 1s in a row on the wire
C. The problem with sending 24 binary 0s in a row on the wire
D. The problem with sending 24 binary 0s in a row on the wire
Review Questions 71
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72 Chapter 2

17. What ISDN signaling subtype is used when attempting to connect a PBX that uses
proprietary signaling?
A. B8ZS
B. Q.931
C. QSIG
D. Type II
18. Which of the following commands correctly configures Extended Super Frame on a T1 port?
A. Router(config-voiceport)#linecode esf
B. Router(config-controller)#linecode esf
C. Router(config-controller)#framing esf
D. Router(config-voiceport)#framing esf
19. You are configuring a T1 port on a voice gateway. Which of the following commands is not
correct?
A. Router(config-controller)#isdn switch-type primary-5ess
B. Router(config-controller)#framing esf
C. Router(config-controller)#linecode b8zs
D. Router(config-controller)#clock source line
20. Which of the following is the correct command used to configure an E&M port as a type
II interface?
A. Router(config-voiceport)#signal 2
B. Router(config-controller)#type 2
C. Router(config-controller)#signal 2
D. Router(config-voiceport)#type 2
c02.indd 72 9/21/11 11:48:11 AM
1. C. Ground-start signaling is recommended to avoid glare problems that occur with
loop-start signaling.
2. A. DTMF stands for dual-tone multi-frequency. Two audio tones are combined to represent
a single telephone number digit.
3. C. Informational signaling is used to notify end users of the status a call resides in.
4. C. Ground-start signaling requires that grounding be performed end-to-end prior to line
seizure.
5. A, D. Both wink-start and delay-start perform a check to ensure that the remote switch is
capable of receiving address signaling.
6. E. Wink-start uses an on-off-on sequence called a wink as notication that the remote
switch is ready to receive address signaling information in the form of DTMF digits.
7. D. Congestion is the informational signal type used to indicate that the call is unable to
be completed.
8. B. E&M type IV is technically not supported on Cisco hardware, but if you need to
connect type IV devices, you can congure them as type II and rewire them to work properly.
9. C. The correct way to change informational signaling tones is to use the cptone command
followed by the two-letter country/region code.
10. A. The input gain command controls the strength in decibels (dB) of the incoming signal
on the analog port.
11. B. The impedance command is used to adjust resistive strength in ohms. Cisco offers
several different choices depending on the strength the attached analog device uses.
12. B. Ground-start signaling can be congured on FXS and FXO ports. E&M signaling can
be congured for wink-start, immediate, or delay-dial.
13. C. Quantizing is the process of taking a sample and assigning it a number based on the
frequency within the sample.
14. B. Encoding is the process of converting the quantized samples into binary.
15. C. Multiplexing is the process used to transport multiple voice signals over a single
pair of wires. The most common multiplexing used on voice circuits is time-division
multiplexing (TDM).
16. A. AMI will give an error when a source attempts to send 8 binary 0s in a row using
electrical signals. B8ZS overcomes this problem by sending an alternating series of
positive and negative polarity to represent all 8 0s in a row.
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74 Chapter 2

17. C. Q signaling (QSIG) uses Q.931 as its underlying signaling protocol but modies the
signals so other proprietary ISDN signaling protocols can also be used and understood.
18. C. A T1 port is congured within config-controller mode. The correct command to
congure Extended Super Frame is framing esf.
19. A. The ISDN switch type is a global conguration command and is not set when within
config-controller mode.
20. D. An E&M port is analog, so you must be in config-voiceport mode. The correct
command is type 2, which species that the E&M interface type is II.
c02.indd 74 9/21/11 11:48:12 AM
1. FXO
2. Loop-start
3. Glare
4. Address
5. Immediate-start
6. signal loopstart
7. output attenuation 2
8. clock source line
9. framing esf
10. pri-group timeslots 1-24
c02.indd 75 9/21/11 11:48:12 AM
c02.indd 76 9/21/11 11:48:12 AM
VoIP Operation
and Protocols
CHAPTER:
Describe the basic operation and components involved in
a VoIP call.
Describe VoIP call flows.
Describe RTP, RTCP, cRTP, and sRTP.
Describe H.323.
Describe MGCP.
Describe Skinny Call Control Protocol.
Describe SIP.
Identify the appropriate gateway signaling protocol for a

given scenario.
Chapter
3
c03.indd 77 9/21/11 11:16:21 AM
The last chapter covered the process of taking an analog signal
and processing it so it can be transported over digital circuits.
This process gets us one step closer to Voice over IP. Because
voice packets are already in a digital format, all we have to do is wrap the voice payload in an
IP packet, and it is ready for transport on an IP network. That is the rst topic of discussion
for Chapter 3. This process of packetizing voice signals for transport over an IP packet is
accomplished using RTP and RTCP. In addition, there are extensions to RTP that can be
used to decrease the header size of an IP voice packet and to transport the payload in a secure
manner. Well discuss these extensions, cRTP and sRTP, in detail, and youll see how and
when they can be used to improve call quality and secure transmissions.
Next, we will cover the four voice gateway signaling protocols: SIP, MGCP, SCCP, and
H.323. That discussion will also include an introduction to H.323 gatekeeper hardware and
common components specically found in H.323 networks. Lastly, we will cover various
situations in which one gateway signaling protocol would be preferred over another.
Voice Media Transmission Protocols
When you have a voice sample that has been converted to a digital format, you need to
include additional information so the voice payload can be sent to the intended destination
over an IP network. The information needed includes details such as the source and
destination IP address and transmission protocol and port used. Also, real-time trafc such
as voice requires additional protocol assistance for proper transport to a destination over IP.
The primary two protocols that accomplish this goal are RTP and its helper protocol, RTCP.
In addition, there are certain situations where the information stored within an RTP
packet header can be reduced so it can be more efciently sent over low-speed serial
connections. This is an extension of RTP called cRTP. Finally, well discuss how to
congure voice gateways to provide for secure transport of IP voice packets using sRTP.
Introduction to the Real-Time Transport Protocol
The Real-time Transport Protocol (RTP) was originally dened in IETF RFC 1889
and revised to its current standard, which is RFC 3550. The protocol was developed to
transport streaming data. By streaming data, we are specically talking about
real-time transport of voice and video. Because real-time transport of streaming data
occurs instantly, lost or damaged packets have no need to be resent. If the packets were
c03.indd 78 9/21/11 11:16:22 AM
resent, they would arrive at their destination late and/or out of order, and would be essentially
useless by the time the packet arrived. Therefore, RTP was designed to be used with the User
Datagram Protocol (UDP) instead of the Transmission Control Protocol (TCP).
UDP is a transport mechanism for IP packets that, unlike TCP, does not attempt to
retransmit or reorder packets that never arrive or are late to the destination. For this reason
and because UDP packets, lacking these features, are smaller than in TCP, UDP is an
ideal Layer 4 transport mechanism for both voice and video. UDP also offers multiplexing
capabilities for easy replication using multicasting protocols at upper layers of the OSI
model. In addition, UDP provides error-detection mechanisms that help make it both fast
and efcient on an IP network.
RTP functions strictly as an end-to-end protocol. This means that the IP source and
destination devices communicate RTP directly with each other, unlike those voice signaling
protocols that communicate with intermediary systems. For example, Figure 3.1 shows a
small network with two IP phones attached to it. They are using the Cisco proprietary SCCP
signaling protocol. When IP-phoneA wants to call IP-phoneB, the phone communicates to
the Cisco call processing agent (a CUCM). The CUCM then nds the location of IP-phoneB
and is responsible for the call setup. But as soon as the CUCM has established an end-to-end
call, the actual transport of voice packets goes directly between endpoints.
FI GURE 3.1 RTP end-to-end transport
M
CUCM
Switch
RTP packet ow
IP-phoneB IP-phoneA
S
C
C
P

s
i
g
n
a
l
i
n
g
S
C
C
P

s
i
g
n
a
l
i
n
g
The RFC for RTP does not specify the actual UDP ports that RTP should utilize. The
one requirement stated is that the UDP port must be an even number. Most voice networks
are set to use default RTP settings, which use random even-numbered UDP ports in the
range of 16384 to 32767 for the purpose of RTP transport. The RFC species that RTP
must always use even-numbered ports while RTCP uses odd-numbered ports. When a
connection is made between IP voice endpoints such as two IP phones, an even-numbered
UDP port is selected for the RTP packets to use from the source IP to the destination IP.
c03.indd 79 9/21/11 11:16:23 AM
80 Chapter 3
VoIP Operation and Protocols

This UDP port is then used for the entire duration of the call. Once the call is disconnected,
that port is released and can be reused by another RTP stream later. Also keep in mind that
an RTP stream is only a one-way communication. Therefore, a voice call must have two
separate RTP connections established in order to achieve two-way communication.
Dont Firewall Your VoIP
Brett was experiencing a problem at a remote site where some VoIP calls were
connecting but others were not. After eliminating various voice gateway conguration
problems, he narrowed the root cause to a miscongured rewall that sits on the edge
of the WAN connection. A rewall rule was congured to allow only a portion of UDP
ports for RTP.
A quick rule change to allow the proper range of UDP ports, and the remote site no longer
had intermittent call connection problems.
Because RTP UDP port selection is random and uses a large range of UDP ports, you
must account for this if your RTP trafc traverses a rewall. You must take proper care to
ensure that your rewall is sufciently opened to allow RTP streams for a wide range of
UDP ports. Otherwise, you will nd yourself in a situation where some RTP sessions are
allowed and others are denied.
So now that you understand how RTP uses UDP as its upper-layer transport mechanism,
lets look at the information contained within RTP. The RTP header is a variable size and
has a minimum size of 12 bytes without any optional elds. Within the RTP header are
various elds that hold all kinds of information related to the proper transport of real-time,
streaming data. These specic RTP header elds are listed here:
Version The version eld is two bits in size and species the version of RTP that is being
used. While there are technically two versions of RTP, only version 2 is in use today. If
future versions of RTP are developed, you will be able to differentiate between version
numbers in this eld.
Padding The padding eld is one bit. If this bit is set (binary 1), it indicates that this RTP
packet has one or more octets at the end that are not part of the voice or video payload.
Padding is often used for encrypting RTP payloads using sRTP.
Extension The extension eld is one bit. If this bit is set (binary 1), it indicates that the
xed RTP header is followed by a single header extension.
CSRC Counter The CSRC counter is four bits in size. It tells you the number of CSRC
headers (if any) that will be in the xed header.
c03.indd 80 9/21/11 11:16:23 AM
Marker The marker eld is one bit. If this bit is enabled (binary 1), it indicates a unique
event that is identied by the application using the RTP stream. Depending on the
application, the marker eld can mean many different things.
Payload Type The payload eld is seven bits in size. This eld identies the type of RTP
data that is inside the payload. This eld allows RTP to communicate the transport of
voice, data, and other streaming protocols.
Sequence Number The sequence eld is 16 bits in size. This eld is a counter that
increments by one for each RTP packet in a particular stream. This information can be used
by upper layers of an application to detect packets that are lost or that arrived unordered.
Timestamp The timestamp eld is 32 bits in size. This eld holds the exact time (sourced
by NTP) when the voice payload was encapsulated in the IP packet. This information can
then be used by the application to better transport time-sensitive payloads and to avoid jitter.
Synchronization Source Identifier (SSRC) The SSRC is 32 bits in size. This eld uniquely
marks multiple RTP streams differently that are originating from the same source. The
SSRC can then be used to differentiate among multiple RTP streams.
Contributing Source (CSRC) The CSRC is 32 bits in size. This optional eld is similar to
SSRC in that it is used to identify the source of streaming data. The difference is that this
eld specically identies contributing sources to streams that come from multiple sources
as opposed to the source itself.
Introduction to the Real-time Transport
Control Protocol
The Real-time Transport Control Protocol (RTCP) is a supporting protocol for RTP.
RTCP is dened in the same RFC 1889 and 3050 standards along with RTP. RTCP is an
out-of-band protocol in the sense that RTCP information is sent in separate, independent
packets from RTP. In addition, RTCP packets never contain a voice payload. Instead,
RTCP contains information about the specic RTP stream it is paired with. Whereas RTP
chooses a random even-numbered UDP port within the range of 16384 to 32767, RTCP
will choose the next-highest odd-numbered UDP port after RTP has randomly chosen its
port. For example, if an RTP connection is established and the protocol selects the random
UDP port of 18408, RTCP will then use the next-highest odd number, which is 18409. The
RTP conversation is always set up rst, followed by the RTCP conversation.
There are several types of RTCP packets. These include the following:
Sender Report Provides reception quality feedback from the sending device of the RTP
stream.
Receiver Report Provides reception quality feedback from the receiving device of the RTP
stream. It contains the same information as the sender report except that it lacks a 20-byte
sender information section used by senders.
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Source Description Contains information about the source host including CNAME,
username, email address, location, and other identifying information that is available.
Goodbye RTCP Indicates that at least one source of an RTP stream is no longer active.
Application-Specific Used as experimental packets for new applications that utilizes RTP.
This allows developers to try new features easily without modication of the protocol.
The purpose of RTCP is primarily to collect the following information:

Packet count for a single RTP stream

Packet loss for a single RTP stream

Packet delay for a single RTP stream

Variation in time between packets (jitter)
Because RTCP can collect information for a single RTP stream, it is a powerful
mechanism that can pinpoint where quality of service (QoS) problems may reside. This
information can be used by upper-layer protocols so they can adjust settings such as the
codec type used for most efcient transport at any given time on a network.
Introduction to Compressed RTP
If you look at the size of an IP packet carrying voice trafc, you may be surprised to see just
how small the payload of the voice data is compared to the size of the IP packet headers.
Every voice packet contains 20 bytes of IP information, 12 bytes of RTP elds, and 8 bytes
of UDP information. That adds up to 40 bytes for a complete IP/UDP/RTP header. With
voice payloads ranging (depending on the codec and compression used) between 20 and 160
bytes on average, headers that are 40 bytes seems like a great deal of added bulk.
Compressed RTP (cRTP) was developed to shrink the size of this header information
down from 40 bytes to a much more manageable 25 bytes. Different aspects of cRTP are
described in IETF RFCs 2508, 2509, and 3545. It is important to note that cRTP doesnt
actually compress anything but instead relies on the fact that much of the information
contained within IP/RTP and UDP headers is static for a specic stream. cRTP will stop
sending this redundant information after the rst transmission to the destination. The voice
gateway accomplishes this by stripping out the static header information prior to sending it
out the outbound interface. The process is CPU intensive and should only be used on serial
links where bandwidth is sparse. In addition, cRTP operates on a Layer 2byLayer 2 basis,
meaning that you will have to congure cRTP at every Layer 2 hop that requires it between
two endpoints.
It is recommended that cRTP only be enabled on connections that are at T1 speeds or
lower. Also, cRTP can be used on serial connections that use ISDN, Frame Relay, HDLC,
and PPP connections. The ideal situation in which cRTP should be used is on a reliable
yet low-speed serial connection where voice packets use a high-compression voice codec,
which commonly shrinks payload sizes between 20 and 50 bytes. Otherwise, the amount of
CPU processing power required to run cRTP overshadows its benets. With the continuing
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growth of high-speed WAN connections, you will nd that cRTP is being used less and less
in the real world.
Introduction to Secure RTP
Secure RTP (sRTP) is a fairly new protocol compared to RTP. Similar to the RTP/RTCP
partnership, sRTP has a secure helper protocol in sRTCP, which performs all of the same
RTCP functions in a secure manner. It was developed as described in IETF RFC 3711. The
purpose of the protocol was to provide RTP packets with the following:
Authentication and Message Integrity HMAC-SHA1 authentication can be congured to
ensure the authentication and assure message integrity between two RTP endpoints.
Payload Encryption Single-cipher AES encryption is used to encrypt streaming data
payloads. One of two cipher modes can be utilized, either Segmented Integer Counter
Mode (Segmented ICM) or f8. The default AES cipher mode is Segmented ICM. A NULL
cipher can also be congured, which essentially disables encryption but allows use of other
sRTP features such as authentication.
Replay Protection In a replay attack, an unauthorized user captures RTP packets in
transit (such as a one-way RTP stream of a telephone conversation) with the intent to
play them and/or modify them to be placed back on the wire for eventual delivery to the
destination. Replay protection checks to ensure that voice packets have not been previously
played. If it determines that the voice packet has been intercepted and played in transit, the
packet is dropped and a log message is triggered to notify network administrators.
The RTP stream receiver keeps an index of previously received RTP packets and
compares each new packet against the index. If any new packets do not match sequencing
found within the index, those packets are assumed to be tampered with and are not sent to
the destination.
If sRTP and sRTCP are congured on a voice gateway, they replace the use of standard
RTP and RTCP. However, sRTP and sRTCP can be congured to use cRTP if desired.
Voice Gateway Signaling Protocols
Voice gateways are just one component in a Voice over IP solution. As you learned in
Chapter 1, An Introduction to Traditional Telephony and Cisco Unied Communications,
voice gateways are primarily responsible for bridging an IP network with the PSTN. From
an IP network perspective, the voice gateway must be integrated very closely with the IP
call-processing agent such as one of the Cisco Unied Communications Manager solutions.
If a voice network is small, a CUCM Express solution can be installed; this enables you
to have a fully integrated call-processing agent and voice gateway solution on one router
appliance, as shown in Figure 3.2.
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On larger networks that require the use of the CUCMBE or CUCM, the voice gateway
must be separated from the call-processing agent, as shown in Figure 3.3.
IP phone
Call agent/voice
gateway
Analog phone
PSTN
FI GURE 3. 2 Integrated call-processing agent and voice gateway
IP phone
Analog phone
PSTN
V
M
Call agent Voice
gateway
Signaling
protocol
FI GURE 3. 3 Separated call-processing agent and voice gateway
Because the two voice appliances are separated, they need to be able to communicate call
routing and other information to each other. This is where voice gateway signaling protocols
come into play. There are four voice gateway protocols used on Cisco call-processing agents
and voice gateways to communicate signaling information:

H.323

SIP

MGCP

SCCP
The following sections detail how each protocol works and provide a general understanding
of how these protocols came to be and how they function. This information should help you
to understand the architectural differences among the four protocols.
Always keep in mind that these voice gateway signaling protocols simply
provide signaling mechanisms so voice has the proper communications
path. The actual transport of the voice stream for each of the protocols
mentioned is handled by RTP/sRTP.
H.323
H.323 is the oldest of the signaling protocols. It is an ITU-T standard for the transport of
voice, video, and data using a peer-to-peer architecture. A peer-to-peer architecture means
c03.indd 84 9/21/11 11:16:25 AM
that sender and receiver are peers in the sense that both have intelligence to route calls
from one point to another. It is also a distributed call control architecture, meaning that
H.323 peers operate independently of each other and do not rely on any other peer for the
handling and control of call signaling.
The H.323 protocol paints in very broad strokes; because it doesnt provide a lot of
specics about what it transports, it is used in a much greater variety (compared to other
voice gateway signaling protocols described in this study guide) of applications, including
voice, video, and real-time data. H.323 also has the benet of interacting well in both IP
and PSTN environments. It can translate between IP and PSTN addressing. This function
is commonly referred to as an H.323 gatekeeper and is explored later in this chapter.
H.323 is still widely deployed and is the default signaling protocol on Cisco voice
gateways and call-processing agents. H.323 is known as a protocol suite because it
actually comprises multiple sub-protocols that are also dened by the ITU-T. For example,
the H.225 sub-protocol species call setup and codec negotiations for endpoints, H.245
performs call control signaling between endpoints. Table 3.1 lists some of the more
common H.323 sub-specications.
TABLE 3.1 H.323 sub-specifications
Sub-specification Description
H.225 Call Setup Call setup and teardown for H.323-speaking devices. Can also
reformat ISDN Q.931 messages to interoperate with H.225
messages.
H.225 Call Routing Uses the Registration, Administration, and Status (RAS)
protocol for call routing.
H.235 Security specification between H.323 gateway and gatekeeper
devices.
H.245 Logical multimedia transport channel. Also performs a
capabilities exchange between endpoints.
H.450 Controls H.323 supplementary services between H.323
speaking devices, including call divert (H.450.3), call hold
(H.450.4), call park/pickup (H.450.5), and call waiting (H.450.6).
H.323 is based on the ISDN Q.931 protocol and therefore can be integrated easily with
PSTN networks because they speak similar languages, which means that little translation
needs to be done for native H.323 devices and legacy PSTN devices to interoperate.
Session Initiation Protocol
The Session Initiation Protocol (SIP) is an Internet Engineering Task Force (IETF) standard
protocol. SIP has gone through numerous RFC revisions, beginning with RFC 2543.
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There are several new RFCs regarding SIP, but most people still usually
reference RFC 3261 as the SIP standard, while newer RFCs are considered
to be minor updates. To read the full RFC on SIP, use the following URL:
http://tools.ietf.org/html/rfc3261
SIP was designed to transport both voice and video over IP networks using either UDP
or TCP, although UDP is the preferred method. SIP transports messages by default on UDP
5060. It is considered to be a peer-to-peer protocol, which means that both endpoints offer
SIP routing intelligence. SIP endpoints are called SIP user agents (UA). User agent clients
(UAC) send INVITE messages when they wish to establish a connection with another UA.
User agent servers (UAS) reply to INVITE messages.
SIP messages are structured in a simple manner. The messages are actually sent in
ASCII format. As stated earlier, SIP by default runs over UDP. If you are concerned about
someone being able to read your SIP signaling messages because they are sent in cleartext
ASCII, you can implement Transport Layer Security (TLS).
Most of you are probably most familiar with TLS and its predecessor SSL,
which you encounter whenever you visit a secure website by entering
https://.
TLS most often runs on connection-oriented protocols. Because UDP is connectionless,
when running SIP with TLS, it will use TCP for transporting signaling information instead
of UDP.
SIP endpoints are also addressed as Uniform Resource Locators (URLs). In fact, SIPs
text-based format closely resembles that of HTTP, which is widely used by web browser
clients who wish to view web pages on web servers. This resemblance is not by coincidence
but instead uses the same addressing functions. For example, consider the following SIP UA
address:
sip:5555@10.1.1.100;user=phone
The rst part of the address (5555) is the unique number for a SIP endpoint. This is
commonly the telephone extension number of an endpoint such as a SIP-enabled IP phone.
The second part of the address, after the @ symbol, is the IP network address of the next-
hop location of where the endpoint can be located on a network. The user=phone portion
species that the number 5555 is a phone extension.
SIP uses a distributed call-processing architecture. Many components come together
to make a complete SIP network. When dealing with SIP from a call agenttovoice
gateway perspective, the two components are peers. Both of them require independent
congurations to properly send and receive SIP information to one another and, ultimately,
route RTP streams inbound and outbound.
Two additional components found in many SIP environments are SIP proxy servers
and SIP registrar servers. SIP proxy servers take the responsibility of forwarding INVITE
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messages for the UACs. SIP register servers maintain a database of UA locations (number-
to-IP mapping) on a SIP network. The Cisco CUCM lineup provides both of these
functions when using SIP on IP phone endpoints.
Media Gateway Control Protocol
A second IETF (RFC 3435) standard protocol for call signaling and call setup between a
packet network and the traditional PSTN is Media Gateway Control Protocol (MGCP).
MGCP is the newest protocol of the bunch and uses a client-server architecture that was
standardized in 2003. When you compare it to a protocol such as H.323, you will quickly
nd that MGCP is far more limited in the things it can do. It essentially handles only
multimedia call control. But if this is all you need, it is an excellent choice because it is
extremely simple to congure and maintain. Being a client-server architecture, MGCP has
two different roles of responsibility:
Call Control Device This is the call-processing agent (CUCM). All control of how calls
are routed across the network is handled at the CUCM. A voice gateway running MGCP
has no knowledge of call control.
IP to PSTN Translation This is the voice gateway that runs MGCP. It receives call-control
instructions and performs the translation between IP and PSTN components.
As you can see, the call-control information is completely contained on the call-processing
agent, so this is a centralized call-control structure. The voice gateway must communicate
with the CUCM to know where to route the call. This methodology differs greatly from
the H.323 and SIP protocols, in which multiple devices contain call-routing intelligence.
If the MGCP voice gateway cannot communicate with the call-processing agent, it has no
knowledge of where it should route calls.
MGCP endpoint addresses have two segments:
Local Name This is the unique name of the MGCP speaking endpoint such as a CUCM
or voice gateway.
Domain Name This is the universal domain name the MGCP endpoints belong to. This
domain name must match before MGCP endpoints can begin communicating.
A potential single point of failure is that the MGCP voice gateway may not
be able to communicate with the MGCP call-control device. To eliminate
this risk, a feature called MGCP fallback can be configured to let gateways
fall back to the H.323 protocol when communication is lost. H.323 can then
be configured with call-control information directly on the gateway and
can properly route calls between networks.
MGCP signaling messages are sent in cleartext and use TCP and UDP port 2427.
Both the call-control server and the voice gateway client send MGCP messages to each other.
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These messages must be acknowledged to ensure receipt of every message. The messages
communicate information such as the following:

Codec used

QoS settings

Cleartext or encrypted voice streams

Amount of bandwidth reserved for each call
Keep in mind that while call control for the IP network is strictly located on the call-
processing agent, off-network addressing information to the PSTN must still be located on
the voice gateway itself. Despite this, call control for the PSTN addressing dial peers is still
controlled by the call-processing agent, which in a Cisco environment is a CUCM.
Skinny Client Control Protocol
The Skinny Client Control Protocol (SCCP) is Ciscos proprietary voice signaling protocol.
It is primarily used as an endpoint-to-call-agent protocol for signaling. It can be used on
voice gateways, however, for various reasons, such as the conguration of a DSP farm
or communication with an analog-to-IP VG200, which only runs SCCP. In addition,
FXS/FXO ports can be congured on a voice gateway to act like any other Cisco IP
phone congured with SCCP. Obviously, since it is a proprietary protocol, both the call
processing agent (CUCM) and voice gateway must be Cisco equipment.
SCCP is a client-server architecture with centralized call control. In this regard, it is
similar to MGCP rather than SIP or H.323. SCCP messages are transported over TCP port
2000. Because SCCP uses TCP for transport, messages can utilize TCP functionality built
into the protocol, such as error correction and a guaranteed delivery of packets.
Voice Gateway Signaling Protocol Comparison
To summarize the differences between the voice gateway protocols weve just examined,
we can categorize them by their organizational standard, architecture, and call-control
method. Table 3.2 compares the four voice gateway protocols.
TABLE 3. 2 Voice gateway signaling protocols at a glance
Protocol Standard Architecture Call Control
H.323 ITU-T P2P Distributed
SIP IETF P2P Distributed
MGCP IETF Client-server Centralized
SCCP Cisco Client-server Centralized
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An Introduction to Gatekeepers and Other H.323 Components 89
An Introduction to Gatekeepers
and Other H.323 Components
Because H.323 is a distributed architecture, many devices that utilize H.323 have their
own intelligence. Much of this intelligence is ofoaded to specialized hardware that
handles specic functions. This section will cover the functions of gatekeepers and other
H.323 components and how they interact with each other.
Gatekeeper
A gatekeeper is most commonly found in very large enterprise H.323 environments.
Its primary function is to maintain a database of telephone extensionstoIP address
mappings. Why are gatekeepers mainly found in large enterprise environments? Lets
explore that question.
When a Cisco voice environment has thousands of users and/or uses a distributed call-
processing scheme, multiple CUCM servers and voice gateways will be used. These servers
will then be clustered together.
Having a distributed call-processing structure forces administrators to congure dial
information for each of the clustered servers/gateways. Figure 3.4 shows a cluster of
CUCMs that must be congured to know the telephone numbers and IP mappings of each
clustered location.
FI GURE 3. 4 A call-processing cluster
M
CUCM
M
CUCM
M
CUCM
M
CUCM
Sharing of dial
information
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H.323 gatekeepers are commonly implemented for call admission control (CAC).
A CAC-enabled gatekeeper keeps a database of telephone number extensions to authorized
destination mappings. For example, a lobby telephone number extension will likely only
be allowed to call internal extensions, while an employee desk phone can dial off-network
destinations. Additionally, call admissions can be further broken down to admit some
phones to dial long-distance or internationally while limiting others to local dialing.
Another point has to deal with the administration of a large telephone network with
hundreds or thousands of telephone numbers. As you can imagine, managing all these
extension numbertoIP address mappings can become a daunting and complex administrative
task. To remedy this issue, many organizations implement a gatekeeper, a single source of dial
information that any of the CUCM servers and H.323-speaking gateways can access. Once
it is implemented, the CUCMs and gateways access the gatekeeper when they are looking for
a specic location to send calls to other member clusters. The CUCM knows the extension
number but not the IP location of destination phone. The gatekeeper responds with the IP
address of where the phone is located within the network. Figure 3.5 shows multiple
call-processing agents communicating with a gatekeeper in a single zone.
FI GURE 3. 5 Call-processing cluster with a gatekeeper
M
CUCM
M
CUCM
M
CUCM
M
CUCM
Zone 1
Sharing of dial
information
Gatekeeper
The CUCM servers must register with the gatekeeper before it can begin querying the
gatekeeper database. Gatekeepers can divide voice networks into gatekeeper zones that help
to segment the locations of phones on the network.
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An Introduction to Gatekeepers and Other H.323 Components 91
Cisco routers can be congured as gatekeepers that do this exact process. The CUCM
servers/H.323 gateways and the gatekeeper will then use H.323 (H.225 RAS sub-protocol)
signaling to facilitate the queries and responses between the two. Gatekeepers also
provide more functionality than just extension-to-IP mappings. Table 3.3 lists some of the
additional services that Cisco gatekeepers provide.
TABLE 3. 3 Gatekeeper functions
Functions Description
Calling privileges Permits or denies calls based on source/
destination extension
Call admission control (CAC) Limits number of calls based on bandwidth usage
Endpoint management using zones Categorizes endpoints into zones for ease of
management
Networks that implement H.323 gatekeepers will also have other H.323-specic devices
to properly process steaming voice, video, and data on IP networks and bridging to PSTN
signaled networks. Some additional components you should be familiar with include the
following:

H.323 proxy server

H.323 Multipoint Control Unit (MCU)
Lets briey describe each of these and then paint a picture of what a typical H.323
network looks like.
H.323 Proxy Server
H.323 proxy servers are servers that work as a head end for call setup and teardown of one
or more H.323 endpoints. A proxy server provides the following benets:

It adds security by hiding the identity of H.323 endpoints.

It can provide Quality of Service (QoS) functionality.

It can provide bandwidth reservation, using the Resource Reservation Protocol
(RSVP).

It can be congured to route based on the application being used, a capability known
as application-specic routing (ASR).
H.323 Multipoint Control Unit
Another popular H.323 component that CVOICE candidates should be familiar with is the
H.323 Multipoint Control Unit (MCU). These devices are used to control and facilitate
multimedia content such as audio and video for point-to-multipoint communication.
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This type of communication is more commonly known as conference calling. An MCU
is actually two separate components. One of these components is called the Multipoint
Controller (MC). This is where the H.245 logical channels are set up and torn down. The
Multipoint Processor (MP) is the component of the MCU that performs the convergence
of audio/video streams, combining them into one stream. It also performs the translation
between various codecs for compatibility.
Cisco offers several Multipoint Control Units in its Videoconferencing 3500 series of
products. These devices are fully compatible with Cisco IOS H.323 gatekeepers.
Cisco also offers BRI and PRI MCU gateways that connect ISDN (H.320)-
compatible videoconferencing systems with H.323 or SIP-compatible
systems.
A Typical H.323 Network
Weve now covered the topics of H.323 gateways, gatekeepers, proxy servers, and MCUs.
Figure 3.6 shows an example network layout in which these components would reside and
interoperate with each other.
H.323
proxy
IP phone
IP phone
PSTN
IP WAN
V
H.323
gatekeeper
H.323
voice
gateway
H.323
MCU
H.323 video
H.323 terminal
Headquarters
V
H.323
voice
gateway
V
M
CUCM-1
M
CUCM-2
H.323
terminal
H.323 ISDN
terminal
H.323 ISDN
video
POTS
phone
H.323 video
FI GURE 3. 6 Common H.323 network components
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Choosing the Appropriate Voice Gateway Signaling Protocol 93
In this diagram, you see a CUCM cluster that maintains IP phones. In addition, the
headquarters has H.323 terminals and video hardware. An H.323 gatekeeper controls
addressing information, call-admission control (CAC), and calling permissions. CAC is a
feature that monitors the amount of bandwidth on a path and either permits or denies a
call being established, based on the amount of bandwidth available. Once the addressing
information is known via the gatekeeper, the H.323 devices at the headquarters use the
H.323 proxy to place calls on behalf of them. The MCU controls multimedia conferencing
applications, and the H.323 gateway communicates H.323 for transport to the remote site
and translates between H.323- and H.320-capable devices located on the PSTN.
Choosing the Appropriate Voice
Gateway Signaling Protocol
Choosing the right voice gateway signaling protocol for a particular environment is an
important decision to make. There are several factors to consider when making this
decision, including these:

Voice/video equipment and vendors that will be used

Call control distribution desired

Voice signaling architecture desired
Looking at the rst factor, you must take a look at the endpoints and voice/video
hardware that you will be using on your network. Do you have endpoints that will only
support H.323, such as some legacy videoconference equipment? Will this be an all-Cisco
implementation or a mixed-vendor environment? If it is a mixed environment, then you
should avoid using SCCP as your gateway signaling protocol, because it is proprietary
Cisco technology.
Second, do you want to have a centralized or distributed call-control design model?
Centralized call-control systems are nice because of their ease of management. Most large
voice networks use a centralized call-control model. On the other hand, distributed call-
control models help to prevent failure because call-control information is pushed out to
multiple voice gateways instead of being centralized. Distributed call-control models also
allow for more exibility and allow for call control to act differently for each area of your
voice network.
Finally, you must consider how you want your signaling protocol to act between devices.
In point-to-point architectures, the protocol treats every voice device as a client, and
signaling is performed directly between the two. By contrast, client-server architectures
require all devices to rst speak to a centralized server (usually a PBX or, in Ciscos case,
a CUCM) to receive the necessary information to talk to a peer device. Also keep in mind,
however, that the two P2P signaling protocols (H.323 and SIP) can be congured to act as
client-server architectures using proxy servers and gatekeepers.
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Summary
VoIP is the process of taking digitized voice, packetizing it, and placing it on an IP network
for transport. There are many different protocols required to packetize voice, as you
learned here. Each protocol has a role in either transporting the voice payload from the
source to the destination, monitoring the voice transport stream, or providing the necessary
signaling protocols required for a voice call setup and teardown.
On nishing this chapter, you should have a solid understanding of how digital voice
signals are transported in IP packet payloads using RTP and RTCP. You also learned about
the various information contained within header elds and its uses. You also learned that
in certain situations, you can use cRTP to shrink the IP packet header over low-speed serial
links and secure packet information and data using sRTP.
We then moved on to describe the four different voice gateway signaling protocols, how they
work, and how they differ from each other. You also learned about the various components
found within an H.323 network. Finally, you learned the various situations where one signaling
protocol would be preferred and should be implemented when compared to the other.
Exam Essentials
Understand RTP and RTCP with respect to voice transport. RTP and RTCP are IETF
standard protocols used for the purpose of transporting real-time data over IP networks.
Understand why UDP is a better Layer 4 protocol over TCP for the transport of real-time
data. UDP does not provide error correction by resending lost or corrupt packets. This
is ideal for voice because real-time streams cannot use retransmitted packets. In addition,
UDP has less header information than TCP and therefore is more efcient on heavily used
networks.
Understand how cRTP shrinks voice packets and when it should be implemented. IP/RTP/
UDP headers contain a certain amount of static information. cRTP shrinks packets by not
sending this information across the network. cRTP should be used only on serial links T1
in size or lower. Also, you must carefully watch CPU utilization of the router running cRTP,
because it can be very CPU intensive.
Understand how sRTP can protect voice data. sRTP provides authentication, message
integrity, encryption, and replay protection when enabled between two voice gateways.
Understand the two different call agent and voice gateway models. Call agents can have
either an integrated voice gateway or a separated voice gateway. In the case of a separated
voice gateway, a voice gateway signaling protocol must be used for communication between
the two devices.
Understand the four types of gateway signaling protocols. H.323, SIP, MGCP, and SCCP
are the four gateway signaling protocols found on Cisco voice gateways.
c03.indd 94 9/21/11 11:16:31 AM
Written Lab 3.1 95
Know the characteristics of H.323. H.323 is an ITU-T standard protocol that uses a
point-to-point architecture with distributed call control.
Know the characteristics of SIP. SIP is an IETF standard protocol that uses a point-to-
point architecture with distributed call control.
Know the characteristics of MGCP. MGCP is an IETF standard protocol that uses a
client-server architecture with centralized call control.
Know the characteristics of SCCP. SCCP is a Cisco proprietary protocol that uses a
client-server architecture with centralized call control.
Understand the purpose of gatekeepers. A gatekeepers primary function is to maintain a
database of telephone extensions to IP address mappings.
Understand the purpose of an H.323 proxy server. An H.323 proxy server provides
services for call setup and teardown.
Understand the purpose of an H.323 MCU. An H.323 MCU provides functionality used
for point-to-multipoint conference-calling features.
Know when to choose one voice gateway signaling protocol over another. Factors such
as hardware used, features needed, and ease of administration factor into selecting a voice
gateway signaling protocol for a particular environment.
Written Lab 3.1
1. What protocol provides out-of-band data collection for streaming data QoS purposes?
2. Which UDP ports are RTP packets commonly sent on?
3. If an RTP stream resides on UDP 20012, what is the most likely UDP port number
used by RTCP?
4. cRTP shrinks which three headers?
5. What is the H.323 sub-protocol that species call setup and codec negotiations?
6. What is the IETF standard protocol that uses a P2P signaling architecture with
distributed call control?
7. In a voice gateway congured to use MGCP, where does all of the call-control
information reside?
8. A voice network uses a mixture of Cisco and non-Cisco equipment for its voice
gateways. Which signaling protocol is automatically excluded from consideration when
choosing a gateway signaling protocol?
9. Which H.323 MCU component is responsible for combining multiple real-time streams
from multiple sources into a single stream for transport to recipients?
10. What voice functionality monitors destination extensions for new call requests and
permits or denies them based on the amount of bandwidth available?
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Review Questions
1. Which of the following are not information required to be inside a voice IP packet for
transport on an IP network? (Choose all that apply.)
A. Source and destination IP address
B. Source and destination MAC
C. RTP header information
D. QoS information
2. RTP was designed and developed by the IETF to be used with which Layer 4 transport
protocol?
A. Q.921
B. TCP
C. UDP
D. Q.931
3. How many RTP connections are required for a two-way voice call between IP phones?
A. One
B. Two
C. Four
D. Eight
4. Which of the following voice gateway signaling protocols uses a centralized call-control
model? (Choose all that apply.)
A. MGCP
B. SIP
C. SCCP
D. H.323
5. An RTCP stream has set up a connection on UDP port 1955. What is its companion RTP
stream UDP port likely to be?
A. 20
B. 22
C. 1954
D. 1956
c03.indd 96 9/21/11 11:16:32 AM
Review Questions 97
6. You have just implemented voice on your IP network and noticed that some calls properly
connected while others were never established. Of the following choices, which one is likely
to be the problem?
A. Misconfiguration of a voice gateway signaling protocol.
B. A firewall between the source and destination IP phones is blocking a portion of
possible RTP UDP ports.
C. cRTP is causing the router CPU to spike and is dropping calls.
D. The RTP connection is made but RTCP fails, which is responsible for the transport of
voice payloads.
7. Which of the following RTCP packets contains information about the host device,
including CNAME, username, email address, and other data?
A. Sender Report
B. Source Description
C. Application-Specific
D. Receiver Report
8. What size is an uncompressed RTP header?
A. 8 bytes
B. 8 bits
C. 12 bytes
D. 12 bits
9. Which of the following is not an RTCP packet?
A. Hello
B. Application-Specific
C. Source Description
D. Goodbye
10. RTCP is used for collecting all of the following except what?
A. Payload size
B. Packet count
C. Packet loss
D. Variation delay
11. When is it recommended to implement cRTP?
A. On LAN interfaces 10 Mbps and higher
B. On LAN interfaces 10 Mbps and lower
C. On WAN interfaces T1 speeds and higher
D. On WAN interfaces T1 speeds and lower
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98 Chapter 3

12. sRTP provides all of the following security features except what?
A. Authentication
B. Replay protection
C. Authorization
D. Payload encryption
13. If an administrator wants to use sRTP for authentication, message integrity, and replay
protection but does not want the payload data encrypted, what can he do?
A. Disable sRTP for all RTCP packets.
B. Enable cRTP, which forces encryption to be disabled.
C. Use a NULL cipher to disable encryption.
D. Encryption must be enabled when using sRTP.
14. If sRTP is enabled on a voice gateway, which of the following two cannot be used? (Choose
all that apply.)
A. RTP
B. RTCP
C. cRTP
D. SIP
15. What is the purpose of a voice gateway signaling protocol?
A. A signaling mechanism for call setup and teardown
B. To transport voice payloads
C. To track statistics for QoS tuning
D. A signaling mechanism used for call admission control
16. What H.323 sub-protocol performs a media channel setup for the transport of voice or video?
A. H.225
B. H.450
C. H.245
D. Q.931
17. Which of the following is true for a voice gateway configured with MGCP?
A. It monitors QoS statistics.
B. It manages phone extensiontoIP address mappings.
C. It performs call setup and signaling.
D. It shares information with the call processing agent in a distributed call control model.
c03.indd 98 9/21/11 11:16:33 AM
Review Questions 99
18. When an IP phone initiates a call, it communicates with an intermediary server that
handles the call setup and teardown of that communication. What type of voice equipment
is this server?
A. Gatekeeper
B. User Agent Server (UAS)
C. Proxy
D. MCU
E. User Agent Client (UAC)
19. A network is configured to use SCCP as its voice gateway signaling protocol. IP phone A
calls IP phone B across the voice gateway. How will the voice packets flow through the
network?
A. Voice packets leaving the source are sent to a proxy server and then to the destination
device.
B. Voice packets leaving the source are sent to a gatekeeper server and then to the
destination device.
C. Voice packets leaving the source are sent to a call processing agent (such as a CUCM)
and then to the destination device.
D. Voice packets leaving the source are sent directly to the destination device.
20. What is the name for the feature that monitors the amount of bandwidth on a path and
either permits or denies a call from being established based on the amount of bandwidth
available?
A. CAC
B. QoS
C. MCU
D. RTCP
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100 Chapter 3

1. B, D. MAC addresses are found in frames and not packets. Additionally, while QoS
information can help to ensure better call quality, it is not required information.
2. C. RTP was developed to work with UDP because of its small header footprint and lack of
retransmission functionality, and cannot be used for transporting real-time trafc.
3. B. A single RTP stream provides real-time transport between a source and destination. If
two-way communication is needed, two RTP streams must be established.
4. A, C. Both MGCP and SCCP keep the call control information on the call-processing
agent that centralizes call control.
5. C. RTP and RTCP ports are randomly selected between UDP 163854 and 32767. RTP
streams are, by default, always even numbered. The RTCP port then will choose the next-
highest odd-numbered port that the RTP port has selected.
6. B. The most likely problem is that a rewall has been miscongured and is blocking a
portion of the UDP ports that a RTP/RTCP session can randomly choose when establishing
new connections.
7. B. The Source Description packet contains various items of descriptive information about
the sending voice device that is transmitting an RTP stream.
8. C. An uncompressed RTP header is 12 bytes and contains 10 elds.
9. A. There is no Hello RTCP packet type.
10. A. RTCP tracks information useful to QoS tuning mechanisms. Keeping track of payload
sizes is not one of them.
11. D. cRTP is recommended only for T1 connections and lower.
12. C. sRTP does not use authorization security features.
13. C. sRTP can be congured to use a NULL cipher, which essentially disables encryption
and sends payload trafc in cleartext.
14. A, B. If sRTP is used, it automatically enables sRTCP. That means that standard RTP and
RTCP cannot be used.
15. A. Voice signaling protocols are protocols used to facilitate call setup and teardown
between voice gateways and between a voice gateway and a call processing agent such as
a CUCM.
16. C. H.245 is the H.323 sub-protocol responsible for establishing the media channel used for
the transport of voice/video.
17. C. MGCP congured on a voice gateway simply performs call setup and signaling.
c03.indd 100 9/21/11 11:16:34 AM
18. C. Proxy servers are responsible for the setup and teardown of calls for multiple endpoints
for management and security purposes.
19. D. Voice payloads are sent in RTP packets, which are sent from the source device directly
to the destination device regardless of the signaling protocol used.
20. A. Call admission control (CAC) is a feature in which a voice device such as a gatekeeper
or proxy server can monitor the amount of available bandwidth on various links. If the
amount of bandwidth required for a new call is not available, CAC will deny the call
from being made as opposed to attempting to connect the call only to have poor quality
as the result.
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102 Chapter 3

1. RTCP
2. Even numbered ports between 16384 and 32767
3. 20013
4. IP, UDP, and RTP
5. H.225
6. SIP
7. At the call processing agent
8. SCCP
9. Multipoint processor
10. Call admission control (CAC)
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The VoIP
Path-Selection
Process
OBJECTIVES ARE COVERED
IN THIS CHAPTER:
Describe a numbering plan.
Describe digit manipulation.
Describe path selection.
Describe calling privileges.

Describe dial peers and the gateway call routing process.

Configure dial-peers.
Configure digit manipulation.
Verify dial-plan implementation.
Chapter
4
c04.indd 103 9/21/11 11:17:03 AM
In this chapter we begin to explore just how it is that a voice
gateway makes call-routing decisions. When a call enters
a voice gateway, a router must have the intelligence to use
information such as source and destination telephone extensions to route the call properly
out of the voice gateway to the proper destination. In addition, these telephone numbers
may need to be modied on the voice gateway before forwarding the call setup information
to the next destination on a voice network.
This chapter will cover what dial plans and dial peers are and how to congure them.
Well then move on to examine the difference between dial peers and call legs and explore
the digit-manipulation techniques used on voice gateways.
Understanding the Dial Plan
Path-Selection Process
Dial plans are congured on voice gateways using dial peers to determine how calls are
directed through the IP and PSTN networks. In addition to path-selection responsibilities,
dial plans provide the following primary tasks:
Digit Manipulation The modication of dialed digits prior to routing a call out of the
voice gateway
Calling Privileges The permission or denial of a caller to certain destinations
This section will rst cover the different call types all voice calls can be categorized
under. Next, we will examine call routing and path-selection techniques and the process
of matching dial peers. Finally, we will look at path-selection strategies that can be used to
streamline dial plans for ease of use and cost-savings benets.
Understanding Voice Call Types
Voice calls are categorized into call types based on the location of the source and destination
phones relative to the IP and PSTN networks. Depending on the type of call being made,
dial plans must be congured differently to ensure optimal paths at the lowest cost. In
the following diagrams of voice call types, you will see the portion of the end-to-end calls
designated within a circle. Anything outside the circle is handled by the PSTN and other
voice networks outside of managerial control.
c04.indd 104 9/21/11 11:17:05 AM
Local Calls
When the source and destination phones are connected to the same call processing agent
or voice gateway, it is considered to be a local call. Figure 4.1 shows an example of an IP
phone calling an analog phone. Both of these endpoints use the same voice gateway, so the
call is considered to be local.
FI GURE 4.1 A local call
IP phone Analog phone
V
Switch
Gi1/0 FXS0/0/0
Voice GW
On-Net Calls
When the source and destination phones are on the same network but traverse more than
one voice gateway, it is considered to be an on-network or on-net call. Because the call
is carried over a private network as opposed to the PSTN, there is no per-minute cost
incurred. Figure 4.2 shows the path between two IP phones located at different locations
but interconnected through an IP WAN. A call made between these two phones must be
processed by two voice gateways.
Switch
IP phone IP phone
Switch
V V
S0/1 S0/1 Gi1/0 Gi1/0
Voice GW Voice GW
IP WAN
FI GURE 4. 2 An on-net call
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106 Chapter 4
The VoIP Path-Selection Process

Off-Net Calls
When the source and destination phones are on different networks and no IP WAN is
available, the calls must traverse the PSTN to complete the call. This is considered to be
an off-network or off-net call. Typically this is a situation where you have administrative
control over one phone but not the other. Figure 4.3 shows an IP phone, located within
a site you control, calling an analog phone located somewhere on the PSTN. The phone
could be an analog or IP phone, and it could literally be located anywhere in the world.
From a management point of view, you are simply responsible for the voice gateway
connection to the PSTN and the dial plan for your phones, which determines when to use
the PSTN connection for off-net calls. It also means that since you are using the PSTN,
a per-minute cost is incurred. Typically an off-net access code is used to signal to the voice
gateway or CUCM that you wish to make an off-network call. The most common access
code used in this situation is the number 9.
FI GURE 4. 3 An off-net call
Switch
IP phone
Analog phone
V
S0/1 Gi1/0
Voice GW
PSTN
On-Net-to-Off-Net Calls
In situations where an on-net call cannot be made because of a WAN failure or
congestion, you can congure your voice gateways to use the PSTN as a secondary
(fall-back) path to perform off-net calls. Figure 4.4 shows the two phones in different
locations. An IP WAN is the primary path, with the PSTN congured as a backup path
in case of a WAN failure. Because the WAN connection between the two sites has failed in
Figure 4.4, the voice gateway will automatically detect the failure and use the alternate
path out the PSTN.
c04.indd 106 9/21/11 11:17:06 AM
PBX-to-PBX Calls
In some situations, you may need to integrate legacy PBX equipment. In fact, you may nd
more than one PBX in use, or you may be in the process of migrating from a legacy PBX
to a CUCM. In either case, a PBX-to-PBX situation will exist, in which calls have to be
transported from one PBX to the other, depending on which PBX is controlling certain
phones. The PBX gear is commonly interconnected using private T1 circuits in either a tie
line or a trunk circuit. The difference between a tie line and a trunk is that a trunk line
is dedicated and continuously in an active state, while a tie line circuit is brought up only
when it is needed. Figure 4.5 shows an example of two analog phones that are supported
by separate PBX systems interconnected by a T1 tie line.
Switch
IP phone
WAN
failure
IP phone
Switch
V V
S0/1
S0/2 S0/2
S0/1 Gi1/0 Gi1/0
Voice GW Voice GW
PSTN
FI GURE 4. 4 An on-net to off-net call
Analog phone Analog phone PBX PBX
Tie line
FI GURE 4. 5 A PBX-to-PBX call
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108 Chapter 4

Intercluster Trunk Calls
Chapter 1, An Introduction to Traditional Telephony and Cisco Unied Communications,
discussed various CUCM deployment models. In the geographical diversity deployment
model, multiple CUCMs are segregated in a way similar to the inter-networking of
services model, but the CUCM call-processing agents are clustered together to function as
a single unit. Cisco IP phones register with the cluster, and the signaling information is sent
across an intercluster trunk. Figure 4.6 shows a typical intercluster trunk call made by a
phone at one site calling a phone at a second site over the IP WAN. The call setup signaling
is transferred between the CUCMs at each site in order to establish the call.
V V
Cluster
M M
IP WAN
FI GURE 4. 6 An intercluster trunk call
Path Selection and Call Routing
Now that you have an understanding of the different call classications on a network,
we can begin to explore how voice gateways can be congured to rst select a path for a
particular voice call and then route that call. Depending on the voice gateway used and the
source and destination endpoints, the voice gateway will have to choose between POTS
ports or IP LAN/WAN to route across. To enable the voice gateway to make the correct
choice, the voice network administrator congures POTS dial peers for analog/digital
interfaces and VoIP dial peers for LAN/WAN ports.
POTS and VoIP Dial Peers
Voice gateway routers that use distributed call control models such as H.323 and SIP are
congured with dial peers to instruct the router about where they need to send voice trafc
c04.indd 108 9/21/11 11:17:07 AM
based on the telephone number dialed. The two primary types of dial peers that can be
congured to route calls on voice gateways are POTS and VoIP.
POTS Dial Peers
A POTS dial peer provides routing information for connecting to traditional telephony
devices such as analog phones, fax machines, and any off-network calls that are routed
out to the PSTN using either analog or digital interfaces connected to the voice gateway.
Dial peers are congured using the dial-peer voice command, followed by an identifying
number that represents the rule. Keep in mind that the number indicated in the dial-peer
voice command does not have to match the destination-pattern command and is simply
used to distinguish between multiple dial peers congured on your voice gateway. The pots
keyword is then used to specify that this is a POTS dial peer. The destination-pattern
command is then used to identify the telephone number the rule is to match on. Finally, the
port command is used to specify the port where the voice trafc will exit. An example of
POTS dial peers looks like the following:
Router(config)#dial-peer voice 5001 pots
Router(config-dial-peer)#destination-pattern 5001
Router(config-dial-peer)#port 0/0/0
Router(config-dial-peer)#exit
Router(config-dial-peer)#end
Router#
Here we have congured two different POTS dial peers. The rst dial peer is identied
as 5001 and maps telephone extension 5001 to analog port 0/0/0. The second dial peer
(5002) maps telephone extension 5002 to analog port 0/0/1.
VoIP Dial Peers
VoIP dial peers connect to voice devices that are IP capable. A VoIP dial peer could point
directly to a voice IP endpoint such as an IP phone, or it could point to a second IP voice
gateway, call-processing agent, or gatekeeper connected through a LAN or WAN.
Like their POTS equivalents, VoIP dial peers also use the dial-peer voice command
followed by a numerical rule identier, but they use the voip keyword (instead of pots) to
identify the rule as an IP rule and not a POTS rule. The destination-pattern command
is identical to its use with the POTS dial peer. Finally, the session target command is
used to identify the location of the next-hop IP where the voice endpoint is known to reside.
Following is an example of VoIP dial peer congurations:
Router(config)#dial-peer voice 5002 voip
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110 Chapter 4

Router(config-dial-peer)# destination-pattern 5002
Router(config-dial-peer)#session target ipv4:192.168.1.100
Router#
Here we have congured two VoIP extensions. 5002 is the rst VoIP dial peer extension,
and it is mapped to an IP address of 192.168.1.100. The 5003 extension is mapped to
192.168.1.101. The router then uses the IP routing table to locate the LAN/WAN interface
where the IP address is known to be located.
You can see that in the VoIP dial peer, the session target address must be
specified as either ipv4:, ipv6:, or dns:.
The key thing to remember about POTS and VoIP dial peers is that for every voice
gateway that is traversed, a physical dial-peer conguration is required so the voice gateway
knows how to accept the calls coming into the gateway and where to send them as they
leave it. For example, lets use Figure 4.7 to show where dial peers are required to complete
and end-to-end call.
FI GURE 4. 7 POTS and VoIP dial peer example
V V
Analog phone Analog phone
POTS dial peer POTS dial peer VoIP
dial peer
VoIP
dial peer
PSTN IP WAN PSTN
From the analog phones to the voice gateways, one dial peer is required. Yet two
VoIP dial peers are needed to traverse the IP network. One VoIP dial peer is congured
outbound on the voice gateway so it knows where to send the call, and a second VoIP dial
peer is needed so the receiving gateway can properly accept the call.
Call Legs
If dial peers are the physical representation of call routing on voice gateways, then call
legs are the logical counterpart. Call legs are the one-way connection of a call setup
between two voice gateways. Just like dial peers, call legs are either POTS legs or VoIP legs,
c04.indd 110 9/21/11 11:17:08 AM
depending on the part of the network the logical leg represents. For example, in Figure 4.8
we have Analog-Phone-A making a call to Analog-Phone-B across both the PSTN and IP
networks.
V V
Analog-Phone-A Analog-Phone-B
Inbound POTS
call leg
Outbound POTS
call leg
Inbound
VoIP
call leg
Outbound
VoIP
call leg
PSTN IP WAN PSTN
FI GURE 4. 8 POTS and VoIP call legs
As you can see, the one-way call-leg communication follows the physical dial-peer
representation. That is to say, any voice gateway must have two call legs associated with it
for every call that is to be processed.
Path-Selection Strategies
A voice gateway must match two dial peers in order to receive and then transmit calls
to the proper destination. Both the inbound and outbound dial peers must match one of
the congured dial peers. In addition, the rules used to select the best dial peer have a
hierarchical order of precedence. In the event of a tie or when no match can be made using
the congured dial peers, there are tie breakers and a default dial peer that can be used to
make a nal path-routing decision.
Inbound Dial-Peer Rules
When a call arrives on an interface coming into a voice gateway, an inbound dial peer must
be matched. The inbound dial-peer rules that follow are checked in the order shown. As
soon as a match is made, the call is immediately routed and no other rules are checked.
DNIS Number Dialed Number Identication Service (DNIS) interfaces are checked in
an attempt to match a dial peer with the telephone number that was dialed. When a user
picks up a phone and dials 555-5555, this number is the destination telephone number that
the user wishes to reach. The voice gateway will use DNIS to match the dialed number to a
congured dial-peer rule.
The command used to configure dial peers to match DNIS information is incoming
called-number. This command is used only on inbound dial peers.
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112 Chapter 4

ANI Number Automatic Number Identication (ANI) is the exact opposite of DNIS.
The ANI eld is commonly used for billing purposes and cannot be modied. The
telephone number matched against dial-peer rules here is the calling partys number. That
means that when a user picks up a phone that has the number 555-4444, the originating
phone number is used for matching purposes. The answer-address command is used to
congure dial peers to match ANI information.
DNIS Number (again) This might sound confusing, but for inbound dial peers, DNIS
information can either be congured using the incoming called-number as stated earlier,
or it can be checked using the destination-pattern command. This command can be used
by both inbound and outbound dial peers, and this rule is checked after any ANI rules.
Inbound Port The port interface can be used to match POTS calls that come into
the voice gateway. The command used to congure dial peers to match port interface
information is port.
It is possible that there may be multiple dial peers configured that match,
creating a tie within the rule system. If this were to occur, the voice
gateway will use the dial-peer rule that is configured first. Because of this
top-down rule-tie selection process, you must be careful as to the order in
which you configure your various dial peers.
Default Dial Peer 0 If no inbound matches are made using congured dial-peer rules,
the voice gateway will use a built-in catch-all rule that is often referred to as dial peer 0.
While it is nice to have a default dial peer to ensure that calls are not dropped inbound, the
dial-peer rules are not optimal for your voice network. For example, consider the following
rules that the voice gateway will use in the event a call is only matched using dial peer 0:

It must use the g.729r8 codec for VoIP dial peers.

There is no Resource Reservation Protocol (RSVP) support for VoIP dial peers.

The QoS preference is 0.

Fax-relay is disabled.

It cannot use DTMF relay for either POTS or VoIP calls.

There is no direct inbound dial (DID) forwarding support.

There is no Interactive Voice Response (IVR) support for POTS dial peers.
As you can see from this list of dial-peer 0 rules, the voice gateway restricts many
important features. That is why it is important to ensure that your voice gateways are
properly set up to match real inbound dial peers.
Outbound Dial-Peer Rules
Compared to inbound dial-peer rules, outbound dial-peer rules are relatively straightforward.
Outbound dial peers are matched using the destination-pattern command only. In addition,
c04.indd 112 9/21/11 11:17:09 AM
dial peer 0 cannot be used. Again, the destination-pattern command matches the remote
called number with either the remote IP address destination or outbound POTS port.
Introduction to PSTN and Private Numbering Plans
Before you begin conguring a dial plan strategy for your voice network, it is important
to fully understand current PSTN numbering plan rules. These rules must be properly
followed if you wish to make off-net calls. PSTN numbering plans are telephone digit rules
that were created at national and international levels to ensure consistency and phone
number coverage where needed. As telephone systems grew over the decades, it was soon
discovered that a national and international hierarchy system was needed to organize
numbers for the following reasons:
Conformance to Standards For telephone networks to work properly at a regional,
national, and international level, dial plans must follow the same rules.
Simplicity of Provisioning The implementation process will go more smoothly with a
well-thought-out plan.
Ease of Routing Calls Having a hierarchy within numbering plans will limit the telephone
number routing tables. Numbers can be more easily routed geographically by nation or by
an area within a specic country.
Ease of Growth A well-planned numbering hierarchy can set aside blocks of numbers that
can be used for future growth where additional telephone numbers are required.
Ease of Management A hierarchical system provides for well-dened management
boundaries. Control within those boundaries can be handled independently.
Clarity to End Users Your dial plan should make it easy for users to understand how to
use the system.
The next two sections cover the current international and North American numbering
plans. You will learn the numbering structure and rules that these plans are based on.
Understanding these PSTN numbering plan concepts will also help you better understand
private dial plan design concepts.
The International Numbering Plan
The ITU developed a numbering plan that is currently used by all nations around the
world. The International Numbering Plan is also known as the E.164 standard. This
standard breaks a national telephone number into three different categories:
Country Code (CC) Denes the country of origin
National Destination Code (NDC) Denes an optional country- or region-specic code
Subscriber Code (SC) Denes a central ofce code
Because the E.164 standard is a hierarchical model, the three codes are built on each other
from least specic to most specic in pinpointing the location of the originating phone that
has the telephone number in question, as shown in Figure 4.9.
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114 Chapter 4

CC-NDC-SC
CC
Least specific
Most specific
NDC
SC
FI GURE 4. 9 The E.164 hierarchy
Every recognized nation is assigned a unique country code that has been allocated by
the ITU board. The remaining NDC and SC codes are the responsibility of each individual
country to manage and allocate as they see t. The format must meet the E.164 standards
for the nation in terms of minimum and maximum digits allowed for each segment. The
format rules are listed in Table 4.1.
TABLE 4.1 ITU E.164 formatting rules
Segment Digit Min/Max
Country Code (CC) 13
National Destination Code (NDC) 015
Subscriber Code (SC) 115
In addition to these formatting rules, the maximum number of dialed digits for any
international call must be less than or equal to 15 including the country code. Countries
must follow this rule when developing their own national numbering plans.
To see the latest list of E.164 numbers, you can visit the ITU website at
http://www.itu.int/pub/T-SP-E.164D-2011.
The North American Numbering Plan
As stated earlier, national plans are dictated by each individual nation as long as it falls
within the international guidelines outlined by the ITU E.164 standard. The CVOICE
c04.indd 114 9/21/11 11:17:10 AM
exam focuses on the North American Numbering Plan (NANP) as the example national
plan for use throughout the exam.
Structure of the NANP
Like the E.164 standard, the NANP also uses a three-step hierarchy scheme, as shown in
Figure 4.10.
FI GURE 4.10 The NANP hierarchy
Area-CO-Subscriber
Area
Least specific
Most specific
CO
Subscriber
The NANP uses a xed 10-digit format for all numbers throughout the region. Table 4.2
lists each of the three NANP hierarchy segments and numbering rules.
TABLE 4. 2 NANP structuring rules
Segment Description Number of Digits Number Formatting
Area code Defined by geographic
location
3 [29][08][09]
Central office
code
Defined by the CO the
phone terminates at
within a specific area
3 [29][09][09]
Subscriber code A number that is locally
unique within the CO code
4 [09][09][09][09]
You will notice that the number formatting for both the area code and CO code omits
some digits from use. Also, there is an NANP rule that states the CO code can never be
X11, where X is any number 29. This is because X11 codes are used for government
services. Table 4.3 lists the X11 services in use today.
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116 Chapter 4

The most important and widely known X11 code in the NANP is for emergency
services (E.911), which will be discussed in greater detail in Chapter 6, Conguring Voice
Gateway Ports and DSPs. For now, simply know that they are treated differently on the
PSTN.
Private Numbering Plan Considerations
Private numbering plans are also critical when planning a voice network. The way you
address endpoints with telephone numbers can impact dial-peer conguration complexity
and scalability on your private network. The following sections identify design strategy
characteristics that must be carefully addressed when planning a private numbering plan:
PSTN DID Support
When planning a private numbering plan, it is usually best to work from the outside
in. This means you should rst look at PSTN access into the private network. In many
situations, companies purchase blocks of publicly routable telephone numbers from their
PSTN. These numbers can often be truncated to use the last four to ve digits internally as
extensions. This is referred to as direct inward dial (DID). If DID is to be used, it should
be the rst piece of the private addressing puzzle to investigate because these numbers will
have to be broken into blocks as possible depending on the number of sites and endpoints
on the network.
TABLE 4. 3 NANP X11 services
Reserved Number Description
011 International access code
211 Community government information
311 City government information
411 Local/national directory assistance
511 Traffic and road conditions
611 Telephone repair service information
711 Hearing-disabled relay service
811 Underground pipe safety service
911 Emergency services
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Access Code Support
The next factor that should be planned out is access code support. Access codes are special
digit combinations that are used to signal the call-processing agent of a special-case dialing
instruction. For example, when most businesses wish to call an off-network phone, they
typically rst dial a 9 and then the telephone number. The number 9 therefore is an access
code, and internal numbers should never begin with this particular number.
Number of Sites
Next on the private numbering plan design list is determining the number of sites you need
to consider in a single voice network. Large enterprises may consist of multiple campus sites
and remote ofces. Each of these remote sites should be accounted for when determining a
private numbering plan.
Number of Endpoints at Each Site
Lastly, you must determine the number of endpoints at each site. If you have a mixed IP
and analog environment, all of these endpoints must be considered, because anything
that requires a telephone number will have an impact on the number of private extensions
reserved at every site.
When developing a private numbering plan, always be aware of future
growth on your network. It is advisable to get an impression from upper
management of possible business growth strategies to address situations
where your main campus site expands rapidly or additional remote sites
pop up. Armed with knowledge of future plans, you can better organize a
numbering plan that will work both today and into the future.
Using Wildcards to Simplify Dial-Peer Configurations
Dial peers can be congured to match exact telephone numbers or a range of numbers.
Conguring dial peers to match a single telephone extension does not scale well when you
have a network with hundreds or thousands of phones. If you were to attempt this on a
large voice network, you would quickly see the benets of dial-peer-matching multiple
telephone extensions in a single rule.
Cisco has developed a series of wildcard characters that can be used when conguring
the destination extensions with the destination-pattern command, which can match on
multiple extensions in a single rule. Using different wildcards results in different extensions
being matched, so it is important to understand what each of the wildcard characters mean.
Table 4.4 lists wildcard characters that are supported by the destination-pattern dial-
peer command.
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If there are dial peers that match multiple rules, the most explicit rule
is chosen over the one that utilizes more wildcard rules. Also, in most
situations (SCCP and Enhanced SIP) the dial peers are matched on a
digit-by-digit basis. As each digit is dialed, the voice gateway attempts to
match it with the first exact dial peer possible. This can be confusing if, for
example, you have 4000 as a destination pattern but are attempting to call
extension 40001.
Using this wildcard information, lets create some dial-peer rules to see how various
wildcards work using a few common examples:
Example 1: Matching Digits Using .
Lets say you are working on your primary sites voice gateway (VG1) and need to create
a VoIP wildcard to support telephone extensions between the numbers 4000 and 4999, as
TABLE 4. 4 Destination pattern wildcard characters
Wildcard Character Description
. A single digit wildcard. Digits can be 09 and *.
[ ] Either a consecutive range of digits using a hyphen (-) or a
nonconsecutive range using a comma (,). A combination of - and
, can be used in the same rule.
( ) Matches a specific pattern. This can be used along with the ?, %, or
+ characters.
? Used to show that the preceding digit occurred zero or one
time only.
% Used to show that the preceding digit occurred zero or
more times.
+ Used to show that the preceding digit occurred one or more times.
T Known as the interdigit timeout. It is used to show that the
router will wait a period of time to collect all digits entered.
The digits can be 09 and *. The router will collect digits for 15
seconds or until the # key is pressed.
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Because all extensions in the 4XXX range are in one location, you can use the . (period)
wildcard to create a single VoIP destination pattern as follows:
Router(config-dial-peer)# destination-pattern 4...
Router#
Example 2: Matching Digits Using [ ]
In our next example, you again are conguring a voice gateway (VG1) at your primary site.
This time, however, the 4XXX range of numbers is not exclusive to your remote site. Only
the digits 4300 to 4999 are available at the site, as shown in Figure 4.12.
IP phone
IP phone
IP phone
IP phone
IP phone
IP phone
V V
VG2 VG1
Extensions:
5XXX
Extensions:
4XXX
IP WAN
FI GURE 4.11 Wildcard example 1
IP phone
IP phone
IP phone
IP phone
IP phone
IP phone
V V
VG2 VG1
Extensions:
4000-4299
Extensions:
4300-4999
IP WAN
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In this case, you can use a combination of [ ] and . wildcards to designate the destination
pattern, as shown here:
Router(config-dial-peer)# destination-pattern 4[39]..
Router#
Example 3: Matching Digits Using the ( ) Wildcards with
?, %, and +
The ( ) wildcard matches various patterns found within a telephone number. The best way
to explain these patterns is to give you a few examples using the other wildcards that can
be used in conjunction with ( ). This rst example uses the ? character, which means the
pattern within the ( ) is matched zero or one time:
Router(config-dial-peer)# destination-pattern 54(11)?
Router#
The pattern 54(11)? means that the dial peer will trigger a match either with 54 or 5411
only.
The next example will use the % wildcard, which means that any digits inside the ( )
will occur 0 or more times:
Router(config-dial-peer)# destination-pattern 54(11)%
Router#
So now, the dial peer will be triggered on 54, 5411, 541111, 54111111, and so on, until 32
digits have been reached.
Lastly, well use ( ) with the + wildcard, which means that the dial peer must match on
the digits inside the ( ) at least one or more times:
Router(config-dial-peer)# destination-pattern 54(11)+
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Router#
In this example, a match will be made on 5411, 541111, 54111111, and so on until 32
characters have been reached.
Example 4: Matching Digits Using T
The T wildcard is a bit of a catch-all and is commonly used in situations where the
administrator wants to create a dial peer for off-network calls to the PSTN. This is because
PSTN calls often use varying-length telephone numbers. For example, local calls usually
may only be seven digits in length, while long-distance and international calls require
additional digits to complete a call. Figure 4.13 describes the physical setup of our voice
gateways and how we wish to use the 9 digit as our trigger for all off-network calls.
IP phone
IP phone
IP phone
V
Voice
gateway
9T
PSTN
Here is the conguration needed to forward all calls beginning with 9 out the T1 PRI
circuit to the PSTN:
Router(config-dial-peer)# destination-pattern 9T
Router(config-dial-peer)# port 0/0/0:23
Router#
Now, anytime a user dials 9 and then any number of digits (up to 32), the router will
pause and collect those digits for 10 seconds by default. The router will then pass those
digits on to the PSTN through the connected T1 PRI circuit. This technique is called site-
code dialing and is described in more detail in the next section. It is important to note that
this specic site-code dialing example is referred to as an off-network access code, because
it is strictly used for dialing telephones outside the local voice network.
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Site-Code Dialing
You may nd yourself in a situation where you have DID ranges that have overlapping
extensions. In instances such as this, a site code can be used to identify the site of the extension
you wish to connect to. For example, Figure 4.14 shows three remote sites in our voice network.
FI GURE 4.14 Site-code configuration example. This approach wont work with
overlapping extensions
IP phone
IP phone
IP phone
IP phone
IP phone
IP phone
IP phone IP phone IP phone
V
V
V
VG2 VG1
Site 3
VG3
Extensions:
3XXX
Extensions:
3XXX
Extensions:
3XXX
IP WAN
Site 2 Site 1
In addition to the diagram, Table 4.5 shows the DID ranges and four-digit extensions
that are to be used in the example.
TABLE 4. 5 DID ranges and extensions
Site Name DID Numbers Four-Digit Extensions
Site 1 2225553XXX 3XXX
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Clearly, there is a problem of overlapping extensions if the company wants to use
a shortened extension dialing dial plan. If a user at Site 1 were to dial 3434, the voice
gateway wouldnt know if the call is intended to be a local call or a remote call to either
Site 2 or Site 3. In situations like this, it is best to congure a one-digit site code that
represents each site. So calls intended for Site 1 would need to be dialed as 13XXX, and
calls going to Site 3 would be dialed as 33XXX.
Dial-Plan Digit Manipulation
There are several circumstances in which you will need to forward a different number string
than the number originally entered by the user. You have already seen two examples of this.
The rst was off-network dialing, when the user enters the number 9 to indicate an off-
network call followed by the actual phone number. The voice gateway cannot simply forward
that number onto the PSTN network. Instead, it must remove the 9 prior to forwarding the
digits. A second was illustrated by the site-code dialing example in the last section. Here
again, the site-code digit must be stripped off before sending the extension to the remote
voice gateway. Phone number strings can be manipulated to add, remove, and substitute
numbers for various purposes.
Voice gateways collect digits using dial-peer rules as described earlier. As youve seen,
these rules can be exact telephone extension matches or use any possible variation of
wildcards. Again, the collected telephone digits are matched using the destination-
pattern command within a dial-peer rule. Within this same rule, the digits can be
manipulated using a variety of commands:

Digit stripping

Forwarding the last X number of digits

Prex adding

Number substitution

Translation rules and proles
The manipulated digits are then forwarded on to the POTS port or IP address, depending
on whether the rule is a POTS or VoIP dial peer.
Digit Stripping
Digit stripping is the process of removing digits that are explicitly dened in dial-peer
rules. For example, consider the following POTS dial-peer rule:
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Router#
By default, digit stripping is enabled and only the wildcard digits will be passed on. If
a user were to dial 5777, the voice gateway would match on this dial peer but strip off the
5 and only pass on the digits 777 out the T1 PRI port. If you want to pass on all digits
including those explicitly dened, you must use the no digit-strip command as follows:
Router(config-dial-peer)# no digit-strip
Router#
Now when a user dials 5777, the string pattern will again be matched by this dial peer,
but this time the entire 5777 string will be forwarded out the voice gateway on the T1 PRI
port.
Forwarding the Last X Digits
In addition to using the no digit-strip command to forward explicitly dened digits in
your destination-pattern dial peer rule, you can use the forward-digits command. This
command species the number of digits that should be forwarded out of the voice gateway.
If the caller enters digits above the set number, it will send the last dialed digits. In the rst
example of this technique, you will see how the previous example can use the forward-
digits command to send all four digits including the three wildcard numbers:
Router(config-dial-peer)# forward-digits 4
Router#
So as you can see, the forward digits 4 command will accomplish the exact same goal
that the no digit-strip command did.
A second example shows that the digit forwarding command can strip off any number
of digits that the caller rst enters. Consider the following example:
Router(config-dial-peer)# destination-pattern 835.......
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Router(config-dial-peer)# forward-digits 7
Router#
In this example, we have a dial-peer pattern that will collect 10 digits. The rst 3
digits (835) are explicit and the other 7 digits are wildcards. Using the forward-digits 7
command, the voice gateway will only forward the last 7 wildcard digits. As you can
see from this example, the forward-digits command is more exible than the no
digit-strip command; it can strip off both explicit and wildcard digits instead of only
explicit digits.
Prefix Adding
In the opposite direction from the forward-digits command, digits can also be added
to the beginning of a number string before it is forwarded out the voice gateway. This is
known as prex adding. A very common example of this is when the voice gateway needs
to forward the destination pattern to a PBX, which in turn forwards the string out to the
PSTN. If the PBX requires an off-network access code such as the number 9, you could
educate your users to dial 9 for this situation. Alternatively, you can add the digit at the
voice gateway using the prefix command as shown here:
Router(config-dial-peer)# destination-pattern 1..........
Router(config-dial-peer)#no digit-strip
Router(config-dial-peer)# prefix 9,
Router#
Notice the comma in the prefix 9, command here. The comma is used to
trigger a pause when forwarding digits onto the destination. This is to help
the destination PBX have enough time to secure an outside line prior to
receiving the actual dial string.
This dial-peer rule will look for callers entering a 1 followed by a 10-digit wildcard. The
no digit-strip command is used so the explicit 1 is forwarded out of the voice gateway.
In addition, a 9 will be added to the beginning of the dial string with the prefix 9,
command. So, for example, if a user enters 15554441418, the dial-peer rule will match the
string. The voice gateway will then forward the following string: 9,15554441418.
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Number Substitution
The voice gateway can collect telephone numbers and then substitute others for them
before forwarding those digits to the next destination. To do this, use the num-exp
command, which is also referred to as number expansion. Keep in mind that this is a
global conguration command that is applied to all outbound calls. This command is often
useful for when on-net calls dial the last few digits of the DID to make internal calls. But
some of these calls may have to cross the PSTN and use the full-length extension. Because
of this, number expansion can be used to prepend the digits so the user can continue to use
shortened extension dialing. For example, lets say you have three remote users who work
from home on a full-time basis. Their PSTN numbers are as follows:
555-222-2XXX
555-242-3XXX
555-284-4XXX
Without number substitution, internal callers would have to dial a full 10-digit PSTN
number in order to call the number. Instead, you would like them to simply dial the last
4 digits within the network. You can use num-exp to prepend the rst 7 digits that are
required by the PSTN, as shown in Table 4.6.
TABLE 4. 6 Number substitution example
PSTN Number Internal Extension
555-222-2XXX 2XXX
555-242-3XXX 3XXX
555-284-4XXX 4XXX
Now that we have our internal extensions to PSTN numbers dened, its only a matter
of creating number expansions and matching dial peers for each site. To do this, we use
num-exp dialed-number substitution-number globally, where dialed-number is the internal
extension that users will dial, and substitution-number is the number the voice gateway
will actually forward on. Then we create a dial peer to match the 4-digit extension that
will be dialed, matched, and modied to the full 10-digit number:
Router(config)# num-exp 2... 9000
Router(config-dial-peer)# port 0/0/2
Router#
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This dial peer matches multiple extensions (2000 to 2999) and forwards them to a single
POTS line such as an operator or automated attendant.
Protect My Internal Numbers
There are some cases where you dont want the people you are calling to know a direct
telephone number but instead give a general business number. This was the case for
a call center that made calls to their customers. The call center wanted to present the
customer with the main telephone number to the business instead of the extension of the
call center employee who happened to be calling.
You can manipulate ISDN caller ID numbers using dial peers. Unique to ISDN networks
is the calling-line identication, or clid, command. The ISDN Q.931 protocol is unique
in that it can send two calling numbers to a voice gateway to use for caller ID purposes.
The rst number is the unscreened number, which the calling party actually entered
into their telephone. The second is the network-provided number. Several conguration
commands can be used, but the most common are shown in this example:
Router(config-dial-peer)#clid network-number 5555555678 second-number strip
The network-number number command changes the number sent to the number
provided. In this case the calling party number will be changed to 5555555678, which
happens to be the primary telephone number to the business where customers are
presented with an automated attendant. The second-number strip keywords will
remove the network-provided number and not send anything to the PSTN. In this sense,
if the customer later has a question and looks up the caller ID information, they have and
can use the number provided to reach the automated attendant.
Translation Rules and Profiles
A very powerful digit-manipulation rule strategy is to use translation rules and proles.
The combination of these two nested command sets lets a voice gateway administrator
convert dial strings either before they are matched against an inbound dial-peer rule or
after a dial-peer rule match prior to forwarding the digits out the voice gateway to the next
destination.
The rst stage is to congure translation rules, which is a two-step process. First, we
dene translation rule sets, which can contain up to 15 individual rules. Then, once the
rules are dened within a rule set, they can be called upon for either incoming digits before
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a dial peer is matched or outgoing digits within a dial-peer rule. The translation rule must
also specify which dial string (called or calling) is to be translated.
Translation rule regular expressions are used within translation rules to provide an
easy and structured method to match number strings. Table 4.7 lists the regular expression
characters and their uses.
TABLE 4. 7 Translation rule regular expressions
Regular Expression Description
^ Matches at the start of a string.
$ Matches at the end of a string.
/ Designates the start and end of matching and/or replacement
strings.
\ The next character in the expression rule not processed as the
special character regular expression.
- Used to indicate a range of digits.
[list] Used to match a single character in a list.
[^list] Used to not match a single character in a list.
. Matches a single character.
* Repeats the last regular expression zero or more times.
+ Repeats the last regular expression one or more times.
? Repeats the last regular expression zero or one time.
( ) Used to group digits.
& Indicates that all matched digits are to be added into the
replacement string.
The process of creating translation rules with regular expressions can be difcult to
understand at rst. However, once you work with them for a while, the regular expressions
begin to make sense and you can see the simplied structure and true power that these
rules offer. To help you better understand rule creation, we will examine some rule-
creation examples.
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Translation Rule Example 1
This rst example will be a simple extension-to-extension translation rule. The rst command
will specify the rule set identication number using the voice translation rule set-number
command. The set-number is a number used to identify a translation rule set that will later
be referenced by translation proles. The second command is rule rule-number followed
by the matched and translated numbers inside the / / matching and replacement regular
expressions:
Router(config)#voice translation-rule 1
Router(cfg-translation-rule)#rule 1 /3456/ /7890/
This rule will match the digit string 3456 and convert it to 7890.
Our next example will match a NULL string. A NULL rule is used as a catch-all on voice
networks to direct any unknown numbers received from the DNIS to a single extension,
which is typically an operator or autoattendant. Here is the conguration to match any
characters using the /^$/ regular expression that matches any digits. The rule then replaces
it with extension 3000:
Router(cfg-translation-rule)#rule 1 /^$/ /3000/
Now things will get a bit more complex. This example will match a string and replace the
middle of the collected digits. Consider the following translation rule:
Router(cfg-translation-rule)#rule 1 /^$...$555$....$/ /\1792\2/
In this example we will match a 10-digit number. The / is the rst expression used to
signify the start of the character-matching process. The ^ means that the voice gateway
will look at the rst character entered in by the caller for matching purposes. The \
character then informs the voice gateway router to ignore the next character, which is an (
expression. Because we are not looking for an ( in the digits we are collecting, we have to
tell the voice gateway to ignore it. The ( does mean the start of a digit grouping, however.
Three . expressions follow, to signify that this is a wildcard where any digit can be entered
for a match. That is followed by another \) set used to close the 3-digit group while telling
the voice gateway to again ignore the ) from the collection process.
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The next number grouping is simply looking for the exact match on the numbers 555.
Lastly, we have a second grouping of four . wildcards. The / expression indicates the end
of the matching statement. So in essence, our translation rule is looking to match on the
following characters that are rst entered by a caller:
(XXX)555(XXXX)
The replacement statement begins with the / expression as usual. It is then followed
by \1. As you know, the \ is used to show that we will not process the next character, which
is 1. The number 1 does have signicance in the replacement string expression, however.
This means that the rst set of numbers we have inside parentheses (XXX) are to be pulled
into the nal replacement string as they were entered by the caller. The next digits, 792, will
replace the second set of numbers, which is 555. Next we see another \ followed by a 2. This
is to indicate that the second set of numbers contained within parentheses (XXXX) are also to
be pulled in unmodied and are to be processed in the nal number string.
So if a caller were to enter in the following number:
2225554444
the output of this number would be
2227924444
If the user were to dial more than 10 digits as shown here:
112555445555
the output number would be:
1127924455
The 555 would be changed to 792 as normal, and the last two digits entered would be
dropped because the ^ regular expression indicates that we begin collecting digits from the
beginning of the string.
Verifying Translation Rules
A great tool to verify that the translation rules you have created actually work as you
want them to is the test voice translation-rule translation-rule-number test-string
command. Using translation rule example 1, we can test to verify that the translation-
rule 1 will translate 3456 into 7890 as shown here:
Router# test voice translation-rule 1 3456
Matched with rule 1
Original number: 3456 Translated number: 7890
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
Router#
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If we tried a number that did not match our matching rule (3456), we would receive the
following:
Router# test voice translation-rule 1 9999
9999 Didnt match with any of rules
Router#
Testing translation rules prior to implementing them on a production network is very
important because it is easy to make a mistake. Make sure to utilize these testing methods
and thoroughly test your translation rules!
Creating Translation Profiles and Applying Them
to Inbound or Outbound Calls
Now that you know how to congure translation rule sets and rules, you need to learn how
to congure translation proles and apply them to both inbound and outbound calls.
Configuring Translation Profiles
Translation proles reference translation rules and dene what number the translation
rules should attempt to match against. This is accomplished using two commands. The rst
is the voice translation-profile name command, which simply denes the translation
prole and uniquely identies it with a name.
Once you name the translation prole, you are then in cfg-translation-profile mode.
Here you use the translate command followed by the number string type you wish to
match and translate. There are three types to choose from on a voice gateway, as listed in
Table 4.8.
TABLE 4. 8 Translation profile number string types
Type Description
called Matches the called partys number (DNIS)
calling Matches the calling partys number (ANI/CLID)
redirect-called Matches the called partys redirect number
Lastly, you must reference the translation rule set number that contains the translation
rules you wish to use. Lets look at an example translation prole:
Router(config)#voice translation-profile PSTN-out
Router(cfg-translation-profile)#translate called 909
Router(cfg-translation-profile)#end
Router#
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Here we create a translation prole named PSTN-out. Within the translation prole
conguration mode, we choose to match and translate the dial string or strings listed in
translation rule set 909.
Applying Translation Profiles to Inbound and Outbound Calls
A translation prole can be assigned to POTS/VoIP dial peers or POTS interfaces. To
congure translation proles on dial peers, you must enter into dial-peer conguration
mode and then use the translation-profile direction name command. The direction
name can either be incoming or outgoing. The incoming option will apply the translation
prole to calls coming into the voice gateway, and outgoing will apply to calls leaving
the voice gateway. The name section is where you list the name of the translation prole
you wish to use. For example, lets congure POTS dial-peer 101 to use our PSTN-out
translation prole for outgoing calls:
Router# configure terminal
Router(config)# dial-peer voice 101 pots
Router(config-dial-peer)# destination-pattern 2..
Router(config-dial-peer)# translation-profile outgoing PSTN-out
Router(config-dial-peer)# end
Router#
Alternatively, a translation prole can be congured directly on a POTS port that is
installed on your voice gateway. The actual translation-profile command has the exact
same conguration setup and two options for application to incoming or outgoing calls on
the port. In this example, we will assign translation prole POTS-in to incoming calls of
voice port 1/0:1:
Router(config)# voice-port 1/0:1
Router(config-voiceport)# translation-profile incoming POTS-in
Router(config-voiceport)# end
Router#
Verifying Dial-Plan Configurations
With all of the dial-peer congurations and numerous manipulation methods available, you
need to be familiar with some show and debug commands that are useful for verifying and
troubleshooting dial plans congured on your voice gateways. These commands include the
following:
show dial-peer voice summary
This command displays useful information about all of the configured POTS and VoIP dial
peers. Here is an example of the output of this command:
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Router# show dial-peer voice summary
dial-peer hunt 0
OPER DEST PASS
TAG TYPE ADMIN PREFIX PATTERN PREF THRU SESS-TARGET PORT
10 voip up up 1.. 1 syst ipv4:192.168.10.1
100 pots up up 0 0 1/0/0
101 pots up up 0 0 1/0/1
Router#
Using this command, you can get an overview of all your dial peers including the
number tag, administrative and operational status, destination pattern, any preference
configurations, and the target IP or POTS port. To get a more detailed look at a specific
dial peer, you can also use the show dial-peer voice tag, where you specify the tag
number.
show dialplan number number-string This command displays which dial peer would be
matched against a specic telephone number. This will verify that the correct dial peer will
be used if you have multiple dial peers congured on a voice gateway. In this example, we
will check to see which dial peer will be matched when a user dials extension 3002:
Router# show dialplan number 3002
Macro Exp.: 3002
VoiceOverIpPeer6
information type = voice,
tag = 2, destination-pattern = `3002,
answer-address = `, preference=0,
group = 2, Admin state is up, Operation
state is up,
incoming called-number = `,
connections/maximum = 0/unlimited,
application associated:
type = voip, session-target =
`ipv4:192.168.1.2,
technology prefix:
ip precedence = 5, UDP checksum =
disabled, session-protocol = cisco,
req-qos = best-effort,
acc-qos = best-effort,
dtmf-relay = cisco-rtp,
fax-rate = voice,
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payload size = 20 bytes
codec = g729r8,
payload size = 20 bytes,
Expect factor = 10, Icpif = 30,
signaling-type = cas,
VAD = enabled, Poor QOV Trap = disabled,
Connect Time = 25610, Charged Units = 0,
Successful Calls = 11, Failed Calls = 0,
Accepted Calls = 11, Refused Calls = 0,
Last Disconnect Cause is 10 ,
Last Disconnect Text is normal call
clearing.,
Last Setup Time = 84427934.
Matched: 3002 Digits: 4
Target: ipv4:192.168.1.2
As you can see from the output, extension 3002 will match VoIP dial peer 6. The command
output lists information including:

Operational status

Codec and QoS settings

Successful and failed calls

Destination IP address
Debug voip dialpeer If you wish to troubleshoot dial-peer matching in real time, you can
use the debug voip dialpeer enable mode command to view actual calls being processed.
This is valuable when troubleshooting problems occurring on a production network.
Debug voice translation Similar to the debug voip dialpeer command, you can enable
debugging on voice translations in real time to identify problems currently occurring on a
production network. To turn on debugging, you can use the debug voice translation
command while in enable mode.
Dont forget to turn off debugging after you are finished with it. Debugging
can use a large percentage of your routers CPU, and it is best to disable
debugging after you are finished. To disable all currently enabled
debugging commands on a router, use either the no debug all or undebug
all command.
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Summary
As you can see from the information contained in this chapter, voice gateways play a key
role in the processing and forwarding of call setup information. Voice gateways that use a
distributed call control model must contain information regarding how to handle the setup
of calls that come into the router. This is done by conguring dial peers that match various
information contained within call-setup messages. Inbound and outbound dial-peer rules
dictate what information can be used for routing. In addition, you learned how to manipulate
digits before sending the information out of the voice gateway router when necessary.
Exam Essentials
Understand the purpose of dial plans. Dial plans are primarily used to determine the path
a voice call should take based on source and/or destination dial strings.
Understand the different VoIP call types. VoIP call types differ based on location of the
source and destination phones relative to the IP and PSTN networks. These include local,
on-net, off-net, on-net-to-off-net, PBX-to-PBX, and intercluster trunk calls. The call types
implemented typically are chosen based on the location of the source and destination
phones relative to the IP and PSTN networks.
Understand the difference between POTS and VoIP dial peers. POTS dial peers send
trafc outbound on POTS interfaces such as FXS, FXO, and T1 interfaces. VoIP dial peers
send trafc to IP network destinations.
Be able to describe and Identify call legs and dial peers on a diagram. Dial peers are the
physical conguration of an end-to-end call. There are VoIP and POTS dial peers. One
dial peer is needed to receive incoming calls and another to route calls to a destination.
Call legs, on the other hand, are logical, one-way representations of a call between two
endpoints. They are also either POTS or VoIP call legs. A voice gateway must have two call
legs associated for every call that is to be processed.
Know the steps a voice gateway takes when choosing inbound dial-peer rules. Inbound
dial peers can be matched on a number of characteristics, including DNIS, ANI, and port.
If no manually congured dial peer is matched against an incoming call, dial peer 0 is
matched as a last resort.
Know the steps a voice gateway takes when choosing outbound dial-peer rules. Outbound
dial-peer rules are matched using the destination-pattern only. There is no dial-peer 0
catch-all rule as with inbound dial-peer rules.
Understand the International and NANP numbering plans. Every country follows
the International Numbering Plan (E.164). This plan assigns country codes and sets the
maximum number of dialed digits at 15. The NANP is the numbering plan standard used
in North America, which divides a 10-digit number into three categories.
Exam Essentials 135
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136 Chapter 4

Be able to understand the different private numbering plan design considerations. Private
numbering plans should always consider various organizational considerations such as
DID support, access code support, number of current and future sites, and the number of
current and future voice endpoints.
Know how to interpret and configure dial peers using wildcards. Wildcards help to
simplify dial-peer congurations by using characters to signify various dialed-number
ranges. By doing so, they allow you to manually congure fewer dial peers.
Know how to interpret, configure, and verify dial-peer manipulation techniques. Often,
telephone number strings must be manipulated before sending them to the destination.
There are several dial-peer manipulation techniques that can be used to add, remove, and
replace digits to suit any need.
Written Lab 4.1
1. When source and destination phones are connected to the same call-processing agent
or voice gateway, it is considered to be what kind of call?
2. What are the two voice-signaling protocols that require dial peers to be congured on
voice gateways?
3. What keyword is added to the end of the dial-peer voice command when the dial
peer also contains the session target ipv4:192.168.1.1 command?
4. Dial peer 0 is used when there is no specic match for what kind of dial peer?
5. Which NANP segment is dened by the CO where the call terminates?
6. Which destination pattern wildcard is used to show that a preceding digit occurs zero
or one time?
7. Using destination pattern wildcards, how can you match for the number 5 plus the
number 54 that occur one or more times?
8. Which destination pattern digit and wildcard combination is often used so users can
dial off-net both nationally and internationally?
9. Given the following destination-pattern:
Router(config-dial-peer)# destination pattern 1..........
what digit-manipulation command could you use to add a 9 and a comma before the
dialed digits?
10. What command can be used to debug voice translations in real time?
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Review Questions
1. Which of the following is not a characteristic of a dial plan on a voice gateway?
A. It determines how calls are directed through IP and PSTN networks.
B. It can provide digit-manipulation techniques.
C. It determines how UDP is directed though IP networks.
D. It can provide calling privilege techniques.
2. Which voice call type best describes when a local user inside the organization calls their
home telephone number?
A. Off-net
B. On-net-to-off-net
C. PBX-to-PBX
D. Intercluster
3. Which of the following does a POTS dial peer not provide routing information for?
A. Local IP phones
B. Local fax machines
C. Off-net PSTN calls
D. Local analog phones
4. Given the following POTS dial peer command, what does 3030 mean?
Router(config)# dial-peer voice 3030 pots
A. An identifier for the dial peer
B. The destination pattern
C. The DNIS
D. The ANI
5. Which of the following is the correct way to route outbound POTS dial peers?
A. Router(config-dial-peer)# destination-pattern 4040
B. Router(config-dial-peer)# port 1/0/1
C. Router(config-dial-peer)# session target ipv4:192.168.1.1
D. Router(config-dial-peer)# session target 4040
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6. Which of the following is the correct way to route outbound VoIP dial peers to
192.168.1.1?
A. Router(config-dial-peer)# session target ipv4:192.168.1.1
B. Router(config-dial-peer)# session target ip:192.168.1.1
C. Router(config-dial-peer)# destination-pattern ipv4:192.168.1.1
D. Router(config-dial-peer)# destination-pattern ip:192.168.1.1
7. A VoIP call traverses a WAN connection; how many dial peers are needed to traverse
the IP WAN?
A. Zero
B. One
C. Two
D. Four
8. Which of the following is a one-way logical representation of a single hop along an
end-to-end voice call?
A. Dial peer
B. Dial plan
C. Call leg
D. Digit manipulation
9. A call is received at a voice gateway. At this point, how are inbound and outbound call legs
processed?
A. Only the inbound call leg is processed.
B. The outbound call leg must be matched first, followed by the inbound call leg.
C. Only the outbound call leg is processed.
D. The inbound call leg must be matched first, followed by the outbound call leg.
10. An IP call is received at a voice gateway. Which of the following are possible ways to match
outbound dial peers?
A. Default dial peer (dial peer 0)
B. answer-address
C. incoming called-number
D. destination-pattern
11. Which of the following number categories are required when using the E.164 numbering
plan? (Choose all that apply.)
A. Subscriber code
B. National destination code
C. Country code
D. Area code
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12. How many digits does a telephone number comprise within the NANP plan?
A. Up to 15 digits
B. 10 digits
C. 15 digits
D. Up to 10 digits
13. What digit-manipulation technique can be used within a dial-peer statement to ensure that
all digits are forwarded to the destination?
A. prefix 4
B. no digit-strip
C. forward-digits 7
D. destination-pattern 9T
14. What digit-manipulation technique is configured globally?
A. Digit stripping
B. Translation rules
C. Translation profiles
D. Number expansion
Router(config-dial-peer)# destination-pattern 33(22)+
A. 3322
B. 332222
C. 33
D. 33222
Router(config-dial-peer)# destination-pattern 413..[35].
A. 4132248
B. 4134551
C. 4135328
D. 4136678
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17. Which of the following are steps required to configure and implement translation rules
and profiles? (Choose all that apply.)
A. Configure a translation profile referencing the translation rule set.
B. Apply the translation profile to inbound and/or outbound calls.
C. Configure a new translation rule set with rules defined.
D. Configure an access group referencing the translation profile and apply the access
group a dial peer.
18. You want to look at all of your dial peers configured on a voice gateway and issue the show
dial-peer voice summary command and see the following output:
PASS
TAG TYPE ADMIN OPER PREFIX DEST-PATTERN PREF THRU SESS-TARGET PORT
1 voip up up 4... 0 syst ipv4:192.168.10.1
10 pots up up 3... 0 1/0/0
20 pots up up 54.. 0 1/0/1
Which dial peer will match 4545?
A. 1
B. 2
C. 10
D. 20
19. Which of the following verification and troubleshooting commands would show
information about the operational status, QoS settings, and codec used for a specific dial
string?
A. show dialplan number number-string
B. show dial-peer voice tag
C. test voice translation-rule rule-number number-string
D. debug voip dialpeer
20. Which translation rule regular expression is used to signify the start and end of a match or
translation string?
A. \
B. /
C. ^
D. [
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1. C. UDP is a Layer 4 protocol used in the transport of IP trafc on networks. It is not a
direct characteristic of dial plans.
2. A. An internal phone that calls a phone that is not connected to the local voice network
must use the PSTN. This is known as an off-net call.
3. A. POTS dial peers handle call routing for traditional telephony devices that do not use IP.
IP phones would need to use a VoIP dial peer.
4. A. When you congure a dial peer, it must have a unique tag identier associated with it
that is used to differentiate it from other POTS and/or VoIP dial peers.
5. B. When a call is matched, a POTS dial peer routes calls out traditional telephony
connections (FXS, FXO, T1, and so on) using the port command followed by the
interface number.
6. A. The session target command is used to route VoIP dial peers to the next IP destination.
Because this is an IPv4 address, it must be specied.
7. C. A dial peer is needed to send the trafc out the IP WAN interface, and a second dial
peer is needed to accept the trafc on the remote voice gateway.
8. C. Call legs represent the logical, one-way path a voice call takes. Call legs can either be
POTS or VoIP depending on the network the call is traversing.
9. D. Both inbound and outbound call legs are required. The inbound call leg is matched rst.
Then the outbound call leg is matched and sent to the proper destination.
10. D. Only the destination-pattern can be used to match outbound dial peers. The other
choices can be used by inbound dial peers.
11. A, C. Of the three categories dened within the E.164 plan, only the country code and
subscriber code are required. The national destination code is optional.
12. B. The NANP plan species that an NANP telephone number must be 10 digits in length.
13. B. The no digit-strip command ensures that all digits including those explicitly dened
are forwarded to the destination. Digit stripping is enabled by default.
14. D. Number expansion digit manipulation is congured globally on voice gateways.
15. A, B. The ( ) means that the numbers are contained in a group. The + means that the
previous digit or group will occur one or more times. That means that 3322 and 332222
will be the two choices that match this destination pattern.
16. A, B. The [ ] means that one of a range of numbers given can be used. The . means that
the next digit is a single-digit wildcard. That means that 4132248 and 4134551 will be the
two choices that match this destination pattern.
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142 Chapter 4

17. A, B, C. A translation rule set must rst be congured; it may contain up to 15 different
matching/translation rules. A translation prole is then congured, which references
the translation rule set. Finally, the translation prole can be applied to inbound and/or
outbound calls.
18. A. The TAG species each unique dial peer. Dial peer 1 is the only rule that will match the
string 4545, and therefore this dial peer will be used to forward the call to the correct
destination.
19. A. The show dialplan number number-string command gives you detailed information
about how the voice gateway will handle a particular dialed number string including dial-peer
tag, operational status, QoS settings, codec used, call success/failure, and destination port or
IP address.
20. B. The / expression signies both the start and end of a match or translation string.
c04.indd 142 9/21/11 11:17:25 AM
1. Local
2. SIP and H.323
3. voip
4. Inbound
5. Central ofce code
6. ?
7. 5(54)+
8. 9T
9. prefix 9,
10. debug voice translation
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VoIP Design Options
THE FOLLOWING CVOICE EXAM OBJECTIVES
ARE COVERED IN THIS CHAPTER:
a VoIP call.
Choose the appropriated codec for a given scenario.
Describe and configure VLANs.

Describe DSP functionality.
Describe different voice ports and their functionality.
Describe codecs and codec complexity.

Describe the need to implement QoS for voice and video.
Describe the causes of voice and video quality issues.
Describe QoS requirements for voice and video traffic.
Chapter
5
C05.indd 145 9/21/11 11:18:46 AM
When you begin the design process for a VoIP network, there
are many decisions that need to be made prior to implementa-
tion. First, you need to understand the full capabilities of voice
gateway DSPs and how they can be used to ofoad processor-intensive tasks from the call-
processing agent. To design a network you must also understand some unique factors found
in VoIP networks, including VAD and network-related issues such as latency, jitter, and
packet loss. You can then choose a voice codec based on the speed/stability of your net-
work as well as the delity of the voice signal you need. To help determine voice load on a
network, in this chapter you will learn how to calculate the size of a frame and bandwidth
consumption based on codec types and sample/payload sizes.
Voice Gateway DSP Functions
Voice gateways do much more than simply route calls between networks. They can also
be used to ofoad processor-intensive tasks from the call-processing agents. Specialized
processors called digital signal processors (DSP) are used to perform multiple voice duties:
PSTN Termination When voice calls must be bridged between an IP network and the
PSTN, trafc is routed to the voice gateway, where a router is used to convert IP voice
packets to PSTN signaling such as a T1 circuit. The conversion requires DSP processing
power to translate between the two networks.
Transcoding Transcoding is the process of translating between two different voice codecs.
There are multiple codecs available for use on voice networks. Codecs are typically chosen
based on hardware compatibility and bandwidth limitations. DSP resources are used in the
translation process, allowing end devices that use different voice codecs to communicate
with each other. A Cisco Unied Communications Manager can perform transcoding
locally, but these can be ofoaded to voice gateways with high-speed DSPs.
Media Termination Point A voice gateway can be congured to be used as a media
termination point (MTP) to relay voice calls that are incoming from either H.323-capable
endpoints or other gateways. An MTP is used to provide endpoints running these signaling
protocols with additional functionality, including:

Call hold

Call transfer

Call park

Conference calling
C05.indd 146 9/21/11 11:18:47 AM
MTPs must also be used in a Cisco environment when there are both SIP and SCCP
phones. SIP DTMF tones are sent inside the payload (in-band), while SCCP phones only
support out-of-band DTMF tones. An MTP can be configured to translate the two tones
between in- and out-of-band.
Conference Calling for Cisco Phones A conference call on a Cisco voice network is
nothing more than the mixing of multiple audio streams (one for each phone in the
conference call) into a single stream that is sent to each phone in the call. In order for this
mixing of audio streams to occur, they must terminate at one point and be processed in near
real time. Similar to transcoding and MTP, a Cisco Unied Communications Manager can
handle some conference-calling duties locally, but doing so is very processor intensive, and
for large implementations its recommended that conference calling be ofoaded to the voice
gateways where DSPs can be used to ofoad call-mixing duties, as shown in Figure 5.1.
In addition, networks using a distributed services deployment model can be configured so
the remote sites voice gateways DSPs are used for local conference calling. This prevents
conference calls from having to needlessly traverse the WAN while consuming bandwidth.
Configuring DSP settings, including DSP farms, will be covered in
Chapter 6, Configuring Voice Gateway Ports and DSPs.
Understanding Voice and VoIP
Quality Considerations
Running voice over an IP network adds some complexity to the task of maintaining the
overall clarity of a call. Because voice is a real-time transmission, network administrators
IP Phone
Voice gateway ofoading
conference calling
IP WAN
V V
M
FI GURE 5.1: Conference call offloading
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must fully understand the terminology and issues that arise on IP networks. This section
covers voice clarity terminology and quality issues that should be understood in order to
design an IP network and troubleshoot voice problems on it.
Audio Fidelity
You can think of a voice network as a networked photocopy machine for audio. When
you speak into the telephone handset, the voice network makes a copy of your voice and
processes it for transport to a destination phone where that copy is replayed. As you know,
some photocopy machines are better than others, and the nished product is close to
the original but not exact, because some of the ner details may be missing. This is also
true for voice networks. The accuracy of the copied signal on a voice network is known
as delity. When audio is sampled using narrowband (3003400 Hz) frequencies and is
then highly compressed, the audio is considered to be low delity, while voice samples
taken with a larger range of wideband frequencies (507000 Hz) and transported using
lower compression ratios are called high delity. The difference between narrowband and
wideband voice collection is displayed in Figure 5.2.
Wideband audio offers a clearer and fuller-sounding voice representation but at the cost of
higher bandwidth requirements.
Echo and Echo Cancellation
A second clarity issue that can cause problems in the transmission of voice is called echo.
Just like yelling into a rocky cavern, echo is the reection of sound that arrives to the listener
a period of time after the direct sound is heard. A certain amount of echo is experienced on
every voice call, but much of it goes unnoticed and therefore can be ignored.
However, when analog signals are converted to digital signals and then compressed
using codecs, echo is often amplied to the point where it severely degrades the quality
of the call. Cisco states that noticeable echo becomes a distraction when the caller hears
their own voice 25 milliseconds or longer after the words are spoken. Echo occurs on
50 300 3400 7000
Frequency (Hz)
G
a
i
n
FI GURE 5. 2 Narrowband and wideband frequency collection
C05.indd 148 9/21/11 11:18:48 AM
traditional voice networks primarily because of an impedance mismatch. Impedance is the
ratio between voltage and electrical current. This ratio can become out of balance when
voice is transported across different networks. On IP networks, echo is typically due to
network delay. Most low-bandwidth codecs have echo cancellation built in using DSPs for
processing power; with traditional PSTN connections, the DSP has rmware that handles
echo cancellation.
There are two primary types of echo. Talker echo occurs when the talking partys voice
is transmitted to the destination but is also picked up by the receive wires during a two-
wire to four-wire transfer. The result is that the voice signal is sent back to the originating
talker with a delay equal to the one-way delay from source to destination. Talker echo is
the most common type of echo found on voice networks.
Listener echo is the second common type of echo. It occurs when the talkers voice is
echoed twice, between two two-wire to four-wire transfers. The rst echo is similar to the
talker echo, in which the voice is leaked to the receiving pair of the originating speaker.
That echo is then echoed back toward the listening party. The result is that the listening
party hears the talking partys voice twice.
An echo canceller eliminates echo of a voice signal by capturing its electrical
characteristics. It stores this electrical ngerprint based on the voice signals that are
being received (Rx). It then uses the stored signal and subtracts it from the transmit (Tx)
signal leaving the circuit. This effectively cuts off any echo that may be occurring on the
line. The amount of time that an echo canceller waits to listen for echo on the Rx line of
the tail circuit is called the ringing time. The ringing time required for a circuit may vary
depending on the quality of the circuit and number of transfers. The following example
shows how to modify the echo cancellation timer to 32 ms on voice port 1/1:0:
Router(config)#voice-port 1/1:0
Router(config-voiceport)#echo-cancel enable
Router(config-voiceport)#echo-cancel coverage 32
The default echo cancellation timer differs on IOS versions. As of this
writing, echo-cancel coverage 64 is the default setting for IOS version
12.3(4)T and higher releases. Also, by default echo cancellation is enabled.
It is shown in the example just in case it has been disabled using the no
echo-cancel enable command.
Background Noise
If you were to record and eliminate the silence in a typical telephone call between two
parties, you might be surprised to discover that approximately 65 percent of the call has
actual audio that needs to be transmitted. The rest of the call is silence and therefore can
be eliminated in theory. A 35 percent reduction in the amount of useless background
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noise transported across an IP networks is a welcome opportunity but must be handled
appropriately. Background noise of some kind is needed to let the parties know that the
call is still in progress. Voice Activity Detection (VAD) can be implemented on an IP
network to eliminate the transport of background noise and therefore reduce the amount
of bandwidth a call consumes. VAD is built into several low-bandwidth codecs, including
G.729b, G.729ab, and G.723.1 Annex A. Other, higher-bandwidth codecs, including
G.711, G.726, and G.728, do not have VAD built-in.
VAD should be disabled when configuring voice gateways to handle fax
machine traffic or modems. It also does not operate with music on hold
(MOH). Therefore, you must choose codecs that do not have built-in VAD
for these situations.
It is also important to note that VAD can have some serious drawbacks. The rst potential
problem is voice clipping, where the rst few milliseconds of speech is not transported,
because of VAD. Recent VAD software is fairly good about avoiding clipping problems, but
the problems increase when a caller is in a loud background situation where the software
has difculty differentiating between a callers voice and background noise. By default, VAD
is enabled on dial peers but not on POTS interfaces. When the noise threshold is 78 dBm,
VAD kicks in. To enable or disable VAD on a dial peer (running H.323) or voice port, use
the vad and no vad command as shown here, where we turn off VAD on dial peer 100:
Router(config-dial-peer)#no vad
Router#
When VAD is enabled, the detection timer can be adjusted globally on voice gateways
to help to prevent clipping problems. By default, VAD waits 250 milliseconds after silence
is detected before it stops sending silent packets. Figure 5.3 shows a voice signal reaching
a point where only background noise is detected. VAD waits a period of time before being
activated. You also can see that when voice begins to be collected, a small portion of the
sound is clipped because of VAD.
G
a
i
n
Time
VAD
VAD timer
Background noise
FI GURE 5. 3 An example of VAD
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This number can be adjusted from 250 ms to 65536 ms. Here is an example of
adjusting VAD detection to 750 ms:
Router(config)#voice vad-time 750
Router(config)#end
Router#
A second problem with VAD is that it does not take the human element into the equation.
With VAD enabled, there are often long stretches of time where there is complete silence
because background noise is not being transported over the network. When a caller hears
no sound at all on the line, they cannot be sure if the call is still in progress or if it has been
terminated. To resolve this problem, VAD is commonly paired with white noise or a
comfort noise synthesis. This is essentially a soft static noise that is injected at the local ends
of the call. It can be enabled along with VAD by using the comfort-noise command. In this
way, the caller hears a soft hissing that informs them that the call is still in progress without
the need of actually transporting background noise across the network. Here is an example
of enabling comfort noise synthesis on POTS ports connected to a voice gateway:
Router(config-voiceport)#comfort-noise
Notice that most of the VAD and related commands are configured on
voice gateways primarily for POTS ports. This is because IP phones can
natively perform VAD and white-noise generation without the aid of a voice
gateway. VAD can be enabled/disabled and modified on the Cisco Unified
Communications Manager, but that topic is outside the scope of this book.
Voice over IP Quality Considerations
IP networks have a xed amount of bandwidth with which to transport voice, video,
and data. Each network is unique, and the maximum amount of bandwidth end to end
is only as great as the lowest bandwidth link. For example, you may have a 10 Gbps core
but only a 1.5 Mbps WAN connection to 40 remote site phones and computers. That 1.5
Mbps of bandwidth can ll up quickly and can cause voice communication problems.
When calls are made between the primary site and remote ofce, the 1.5 Mbps WAN
is the lowest bandwidth circuit and is referred to as a bottleneck. Bottlenecks on a
network are often related to the following conditions, any of which can cause IP voice
quality issues.
Network Delay
Propagation delay is the amount of time it takes a packet to travel from source to
destination on a network. Every network has a certain amount of delay, called xed delay.
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This is the amount of time it takes in an ideal situation where the only slowdown is in how
fast it takes electrical and optical signals to transport IP packets. Fixed delay cannot be
avoided and is so slight that it does not affect the quality of voice. The following factors
contribute to xed delay on a network.
DSP Delay The amount of time it takes for the DSP to sample, encode, and compress
voice streams.
Packetization Delay The amount of time it takes the router to place encoded voice inside
IP packets for transport on the wire.
Serialization Delay The amount of time it takes to transmit bits on a physical medium
such as Ethernet or a POTS line.
Variable delay, on the other hand, occurs when data transport is slowed down by
bottlenecks on the network. Data in bottleneck situations must be queued and wait a
period of time before being transported. A certain amount of delay (xed and variable
combined) can occur before voice quality is affected. The ITU-T G.114 specication
recommends that end-to-end delay should not exceed 150 ms.
Network Delay This type of delay is due to bottlenecks on the network that cause packets
to be delayed on the wire.
De-jitter Buffer Delay When packets are unevenly spaced on arrival to the input
interface, they must be slowed down and properly spaced before they are processed by
the DSP. The playout delay buffer is responsible for this task. It is also referred to as the
de-jitter buffer.
Queuing and Buffering Delay This is the amount of time it takes for packets to be placed
in a queue until they can be sent out of the outbound router interface.
Network Jitter
Jitter on a network refers to the variation in the time between the receipt of each voice
packet. Jitter is always calculated at the receiving end of the network. Variable delay
causes variation in the time between packets. For example, a voice packet that is sent from
the source phone to the destination phone with no variable delay may have a difference
between packets of 3050 ms. However, bottleneck conditions can create temporary
variable delay in which it might take over 100 ms between receipt of voice packets, as
1 2 3
130 ms
4 5
55 ms 40 ms 40 ms
FI GURE 5. 4 Jitter variation
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This variation in the receiving of packets can cause the voice stream to skip and stutter,
which can be very annoying to the listener. It is recommended that network jitter be
reduced to 30 ms or less on average.
Packet Loss
Jitter is caused by bottlenecks delaying voice packets from arriving. It occurs because at the
bottleneck interfaces, packets must be placed into queues and wait until it is their turn to
be sent across the congested pipe. But if bottlenecks are severe enough, interface queues can
ll up and packets are dropped because there is no place to store them. Packet loss not only
is annoying but also can cause calls to fail completely. While ideally no voice packets are
lost, it is recommended that the overall total of packets lost for a voice call never exceed 1
percent. Many voice codecs have whats known as packet loss concealment (PLC) methods
to assist with packet loss on voice calls. When lost packets are detected on a voice gateway,
PLC analyzes the packets that did arrive at the input interface and determines what the
voice payload should sound like. In essence, PLC is software that takes a guess at what the
missing packet is and places it on the wire for transport. This replacement packet is far
better than no packet at all.
The majority of voice quality concerns can be solved by choosing the optimal codec for
your network as well as properly conguring Quality of Service (QoS) settings to ensure
that real-time packet transmissions such as voice are prioritized ahead of data trafc. The
next section covers several voice codecs available on Cisco endpoints and voice gateways.
Quality of Service considerations and congurations will be covered in Chapters 11,
Introduction to Quality of Service, and 12, Conguring Quality of Service,
of this book.
Defining Voice Codecs
Weve talked a little bit about codecs but have never ofcially dened them. The term codec
is short for coder/decoder. A codec is an algorithm responsible for the conversion of analog
waves (such as the human voice) into a digital format. The G.711 codec is special in the fact
that it provides no compression of the audio sample as it is digitized. Other codecs provide
compression in one form or another that sacrices audio quality for bandwidth savings.
The codec chosen should be based on the end-to-end bandwidth for all call legs as well as
compatibility of endpoints involved.
There are a number of voice codecs that you should become familiar with. Each codec is
slightly different and can be used in various network environments to provide the optimal
balance between quality voice and bandwidth reduction.
Voice Codec Types
Literally dozens of voice codecs are available in the wild. It used to be that most Cisco IP
phones supported only the G.711 and G.729 codecs. Times are changing, and Cisco is now
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delivering phones that natively understand several more codecs. New phones are designed
with enhanced acoustics and the ability to understand advanced voice codecs. For example,
the Cisco 9951 phone currently supports the following voice codecs:

G.711a

G.711u

G.729a

G.729ab

G.722

iLBC
In addition, your voice gateway can understand multiple other voice codecs, because it is the
device that connects to the PSTN, Internet Telephony Service Providers (ITSP), legacy PBX
systems, and remote sites with low-speed WAN connections. Therefore, the list of codecs
Cisco voice gateway IOS software can understand and work with is large and is growing all
the time. This section describes some of the more commonly used codecs on voice networks
today. For the most part, voice codecs attempt to balance voice quality with lower bandwidth
requirements for a call. In addition, Cisco is now supporting wideband voice codecs.
Most codecs available today on PSTN and VoIP networks provide narrowband
communication, in which analog signals are collected within 300 and 3400 Hz. This range
of frequencies picks up most human speech and therefore has been sufcient. However, if
you want to transfer other sounds (such as music) clearly, then a wider range of frequencies
must be collected. Wideband codecs collect frequencies between 50 and 7000 Hz. While
this results in a much richer and more natural sounding call, it also increases the amount of
bandwidth required because of the larger sample sizes that are collected. Other wideband
codecs compress the payload sizes, but doing so requires additional DSP resources to handle
the increase of compression duties. The two wideband codecs discussed in this book are
G.722 and iSAC. While wideband codecs specications are not new (G.722 was standardized
in 1988), the advancement of higher-powered DSP chips makes wideband audio possible.
G.711
The G.711 ITU-T standard is the most popular codec used on voice LANs today. The codec
uses 8-bit samples at 8 kHz sampling rates and encodes audio signals in 64 Kbps streams.
There are two main subsets of G.711, which use slightly different encoding schemes. The
G.711u algorithm is used in North America and Japan, while the G.711a algorithm is used
in the rest of the world. PSTN PRI connections utilize one of these codecs, depending on
what part of the world you are in. The difference between the two is that the u-law (or mu-
law) algorithms encoding scheme is a little more complex than the a-law algorithm used.
G.711 is also commonly referred as Pulse Code Modulation (PCM).
G.723.1
The G.723.1 ITU-T standard codec uses compression to deliver very low bandwidth voice at
acceptable quality. In addition, there is a G.723.1 Annex A variation that has built-in VAD.
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The non-Annex A variation does not have built-in VAD support. There are two additional
options when conguring this codec on Cisco voice gateways. The G.723ar53 codec option
operates at 5.3 Kbps, while the G.723ar63 operates at 6.3 Kbps. Because of the loss in
audio quality, DTMF and fax relay should never be attempted using G.723.1 codecs.
G.726
The G.726 ITU-T standard codec uses Adaptive Differential Pulse Code Modulation
(ADPCM) to reduce bandwidth bit rates while maintaining a relatively high sound quality.
The G.726 protocol ofcially replaces the now obsolete G.723 and G.721 protocols that
were the rst to use ADPCM. G.726 uses samples of 2, 3, 4, or 5 bits, at data rates of 16,
24, 32, or 40 Kbps depending on the subset of the codec. The benet of the G.726 protocol
is that if the PSTN or PBX is congured to use the ADPCM codec (which sometimes
happens when PSTN channels are overloaded), a Cisco voice gateway can natively be
understood by both IP and legacy digital voice networks.
G.722 (Wideband)
The G.722 ITU-T standard codec provides improved audio quality by taking wideband
samples. The codec uses the same Adaptive Differential Pulse Code Modulation (ADPCM)
found in G.726, but instead of compressing the audio payload, G.722 maintains the same
data rate size but doubles the audio content found in each packet. This is known as Sub-
Band ADPCM (SB-ADPCM) and allows for a 16 kHz sample rates at data rates of 48, 56,
or 64 Kbps.
A newer variation of G.722 is the G.722.2 codec, also known as Adaptive MultiRate
Wideband (AMR-WB). This codec applies advanced compression techniques to the 64
Kbps stream when congestion is observed on the network. This compression requires the
use of additional DSP resources during this time. When congestion is alleviated, the less-
compressed (and therefore higher-quality) stream returns.
G.728
The G.728 ITU-T standard codec provides compressed voice streams running at a xed 16
Kbps. This codec uses a low-delay code excited linear prediction (LD-CELP) technique that
provides reasonable quality voice at lower bit rates.
G.729
The G.729 ITU-T standard codec uses 10 millisecond audio samples and compresses the
audio signal into a small 8 Kbps bit rate. Because of the low bit rate, the G.729 codec is
a very popular choice for transporting voice over low-speed WAN connections. Also, as
with the G.723.1 codec, DTMF and fax relay should never be attempted with this codec.
You will also not want to use this codec for MOH streams. There are many subsets of
the G.729 protocol, referred to as annexes. Here is a breakdown showing the differences
between the codec annexes used in Cisco networks:
G.729a Requires less DSP processing power but provides lower-quality audio.
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G.729b Higher-quality voice than G.729a but considered a high-complexity codec that
will require more DSP resources. Also provides built-in voice audio detection (VAD) to
limit the number of empty voice packets being sent.
G.729ab The combination of G.729a and G.729b features. The result is a medium-
complexity codec with built-in VAD.
A new ITU-T G.729 annex has been developed to support wideband audio
samples that are compressed into 832 Kbps bit rates. This new annex
is known as G.729.1. While this codec is not yet supported by Cisco IP
endpoints, it may very well be supported in the future to provide high-
fidelity audio at low bandwidth rates.
GSM Full Rate
The GSM Full Rate (GSMFR) codec was the rst digital codec used on GSM mobile
networks. Its a high-complexity codec that uses 20 millisecond frames at a bit rate of 13
Kbps. The codec is also used in voicemail systems, and this is where you are likely to run
across it on voice networks. The quality of the audio stream is fairly poor compared to
newer and more advanced codecs.
Internet Low Bit Rate Codec
The Internet Low Bit Rate Codec (iLBC) is an open-standard protocol that is heavily
backed by Cisco. This protocol is designed to deliver high-quality audio with a relatively
low-bandwidth footprint of 13.33 Kbps using either 20 or 30 millisecond frames. One
feature of iLBC that is benecial to VoIP networks is its built-in graceful degradation of
audio signals if network congestion or other issues cause dropped packets, unordered
packets, or jitter on the network.
Internet Speech Audio Codec (Wideband)
The Internet Speech Audio Codec (iSAC) is a proprietary codec developed by Global IP
solutions but supported on Cisco voice gateways and Cisco UBE platforms. The iSAC
protocol terminates at the voice gateway and can be used to communicate natively with
iSAC-capable devices. It is a popular codec used for voice applications over the Internet,
including AOL Instant Messenger (AIM) and Google Talk. This wideband codec uses
samples of either 30 or 60 ms at 16 kHz using a sampling rate between 10 and 32 Kbps.
Understanding Codec Complexity
Voice codecs use various algorithms used to compress audio signals. Some algorithms used
for compression use more processing power than others. You have already learned that DSPs
are used by voice gateways to ofoad codec processing from the main CPU. Codecs are
therefore categorized by the number of simultaneous calls a single DSP can process. On Cisco
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equipment, three different DSP chipsets are in use today. The older C549 DSP chipset (PVDM)
can support up to eight low-complexity codec calls, four simultaneous medium-complexity
codec calls, and two high-complexity codec calls per DSP. The C5510 DSP chips (PVDM2)
can support 16 low-complexity codec calls, eight medium-complexity codec calls, and a
maximum of six high-complexity codec calls simultaneously per DSP. Lastly, PVDM 3 chips
can handle 16 low-complexity, 12 medium-complexity, and 10 high-complexity codec calls.
The PVDM2 is used on all new Cisco voice gateway routers including the Cisco ISR, while
the PVDM3 is currently available only on the 2900 and 3900 series ISR2 lineup. PVDM2
chips can be installed in voice gateways with the congurations shown in Table 5.1.
The more advanced and higher-density PVDM3 capabilities are listed in Table 5.2.
TABLE 5.1 PVDM2 DSP capabilities
PVDM Type Low-Complexity
Calls
Medium-Complexity
Calls
High-Complexity
Calls
PVDM28 8 4 4
PVDM216 16 8 6
PVDM232 32 16 12
PVDM248 48 24 18
PVDM264 64 32 24
TABLE 5. 2 PVDM3 DSP capabilities
PVDM Type Low-Complexity
Calls
Medium-Complexity
Calls
High-Complexity
Calls
PVDM316 16 12 10
PVDM332 32 21 14
PVDM364 64 42 28
PVDM3128 128 96 60
PVDM3192 192 138 88
PVDM3256 256 192 120
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Because DSPs handle a wide variety of voice duties, it can be difcult to
understand how many DSPs you will need to install on your voice gate-
ways. Fortunately, Cisco has an online DSP calculator that can be used to
help identify the number of DSP resources that will be used by entering
information such as the router platform, IOS version, modules used, and
number of conference calls expected. If you have a Cisco account, you can
log in and access the DSP calculator here:
http://www.cisco.com/cgi-bin/Support/DSP/dsp-calc.pl
There are only a few low-complexity codecs, which require no compression at all,
including these:

G.711 a-law

G.711 u-law

Clear channel (used for transport of non-voice data such as fax/modem)
Some of the more popular voice codecs used today that fall under the medium-
complexity category are these:

G.726 (all variations)

G.729a

G.729ab
These common voice codecs utilize more DSP/CPU resources and are considered high
complexity:

G.723 (all variations)

G.728

G.729

G.729b

GSMFR

iLBC

iSAC
Voice complexity can be congured manually for DSP chipsets using the codec
complexity type command within config-voicecard mode. On C549 DSP chips, you can
congure each DSP as being either medium or high complexity, as shown here:
Router(config)#voice-card 1
Router(config-voicecard)#codec complexity ?
high Set codec complexity high. High complexity, lower call density.
medium Set codec complexity medium. Mid range complexity and call density.
<cr>
Router(config-voicecard)#codec complexity
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By default, the codec complexity for C549 chips DSPs is set to high. This is so you can use
any voice codec, because low- and medium-complexity codecs will work ne when the DSP
is congured for high-complexity mode. It does mean, however, that twice the resources
actually needed will be allocated for the low- or medium-complexity codec. If you plan to
use only low/medium-complexity codecs, you can statically congure the DSPs to use only
a medium complexity. The downside here is that if you try to use a high-complexity codec,
the call will fail.
Statically Assigning DSP Resources Can Cause Dropped Calls
When voice gateways are pushed to capacity from a bandwidth perspective, many busi-
nesses choose to move from higher-quality codecs such as G.711 to lower-quality codecs
like G.729. The benet of migrating to G.729 comes from the per-call bandwidth saving
that you can achieve. This situation was exactly what happened to Wes, our resident net-
work administrator.
Wes was asked to modify the codec used on his voice gateway from G.711 to G.729b. This
was to be a fairly simple modication and would be seamless to the end user. But when Wes
went ahead and changed the codec type, all DSP functionality ceased to work.
Wes struggled to gure out the cause of his DSP problem and had to fall back to using
G.711. As soon as he did this, DSP services began functioning properly. It was only after
reaching out to another network engineer that the problem was spotted. It turned out
that the DSP voice card installed in the DSP farm was statically congured to use medium
complexity. This was not a problem because the G.711 codec is considered to be of
low complexity, but G.729b is a high-complexity codec and can only operate with DSP
resources when the voice card is congured for high complexity or in variable ex mode.
The C5510 (PVDM2) and PVDM3 chipsets offer two additional conguration options
that can be set as shown here:
Router(config-voicecard)#codec complexity ?
flex Set codec complexity Flex. Flex complexity, higher call density.
high Set codec complexity high. High complexity, lower call density.
medium Set codec complexity medium. Mid range complexity and call density.
secure Set codec complexity secure.
<cr>
Router(config-voicecard)#codec complexity
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The flex option is different from the high and medium complexity settings because
ex mode allows for the oversubscription of calls and allows low-, medium-, and high-
complexity calls to be processed on a DSP. While ex mode is benecial because of the
ability to support more TDM interfaces than DSP resources, it also leaves the door open for
occasions when DSP resources will be exhausted, resulting in call failures. The flex option
provides the ability to support any codec by automatically adjusting between medium- and
high-complexity settings. For example, when a call requires G.711 codec processing, the
flex option will choose to process the call using low complexity. However, a G.729b call
will be processed using high complexity.
The secure codec complexity option allows for the secure transport of voice streams
using authentication and encryption through the support of sRTP. Adding this added layer
of security obviously uses more DSP resources than processing the codec unauthenticated
and unencrypted.
By default, PVDM2 and PVDM3 DSPs are congured as ex resources. The show voice
dsp command can be used to verify how your DSPs are currently congured, as shown in
the following example:
Router# show voice dsp
FLEX VOICE CARD 1
*DSP VOICE CHANNELS*
DSP DSP DSPWARE CURR BOOT PAK TX/RX
TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT
===== === == ======== ======= ===== ======= === == ========= == ==== ============
C5510 001 01 modem-re 4.5.909 busy idle 0 0 1/1/0 05 0 298/353
*DSP SIGNALING CHANNELS*
===== === == ======== ======= ===== ======= === == ========= == ==== ============
C5510 001 05 {flex} 4.5.909 alloc idle 0 0 1/1/3 02 0 15/0
C5510 001 06 {flex} 4.5.909 alloc idle 0 0 1/1/2 02 0 17/0
C5510 001 07 {flex} 4.5.909 alloc idle 0 0 1/1/1 06 0 31/0
C5510 001 08 {flex} 4.5.909 alloc idle 0 0 1/1/0 06 0 321/0
END OF FLEX VOICE CARD 1
As you can see from the output, these DSPs use the C5510 chipset and are set for ex mode.
Quantifying Voice Codec Clarity
As you have learned, voice codecs offer compression that conserves bandwidth at the cost
of the quality of the audio signal transferred. But to what degree are these codecs degrading
audio clarity? Because VoIP has the capability to compress audio signals to allow more calls
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to be transmitted and received on a nite amount of bandwidth, it was quickly discovered
that a method of quantifying the quality of voice was needed to show just how much codecs
give up in terms of clarity versus bandwidth savings. The ITU-T has been responsible for
implementing several subjective and objective methods over the years.
This section presents four of its more popular methods used to quantify the quality of a
voice signal:

Mean Opinion Score (MOS)

Perceptual Speech Quality Measure (PSQM)

Perceptual Evaluation of Speech Quality (PESQ)

Perceptual Objective Listening Quality Analysis (POLQA)
Because the G.711 codec is transferred uncompressed, this is considered to be the
optimal voice quality that can be achieved. All other voice codecs are compared against
the optimal codec. That being said, all of the voice quality measurement tools dont
consider G.711 to be 100 percent optimal, because analog voice signals do not pick up
the complete spectrum of audible tones. Therefore, a perfect score within voice quality
measurement tools is nearly impossible. Lets take a look at each of these tools and
compare them.
Mean Opinion Score
As the name indicates, the Mean Opinion Score (MOS) test is simply a human opinion on
the quality of various codecs. MOS is an ITU-T (P.800) audio quality recommendation that
used a group of trained listeners to rank the perceived voice quality after digitization for a
large group of audio codecs. Each listener gave the codec audio output a score between 1
and 5. A score of 1 means that the audio quality is bad, while a score of 5 means that the
audio quality after digitization is excellent. Table 5.3 shows the MOS listening quality
scale and subjective terminology used in the ofcial ITU-T MOS scoring tests.
TABLE 5. 3 MOS listening quality scale
MOS Score MOS Subjective Rating
5 Excellent
4 Good
3 Fair
2 Poor
1 Bad
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Table 5.4 shows the results of the MOS testing for several popular voice codecs.
While the ITU-T MOS scale was performed by the ITU-T using the most scientic methods
available, it still must be considered subjective because humans are being used to essentially
grade each codec. Therefore, this is probably not the most accurate method for determining
voice quality even though the scores are widely referenced today. But keep in mind that
user perception is important and measures success or failure of an implementation. The
next two voice quality methods attempt to address this problem by completely eliminating
subjective opinion and objectively grading codecs using mathematical algorithms.
Perceptual Speech Quality Measure
The Perceptual Speech Quality Measure (PSQM) was developed by the ITU-T (P.861) to
objectively calculate the sound quality of various audio codecs. By developing an algorithm
and computers to calculate scores, it eliminates subjective error inherent in MOS. This
means that the tests are highly reproducible and therefore the scores are highly reliable.
Without getting into the specics of how the PSQM algorithm calculates scores, the
audio is scored and graded immediately after the audio sample is digitized and compared
against the original analog signal. The difference between the initial and encoded audio
samples is then graded and given a number between 0 and 6.5.
The ITU quickly discovered that PSQM was not an optimal method for scoring codecs
on a VoIP network. This was because PSQM calculated the difference between the
original and coded signal immediately after the process occurred. Therefore it did not
have any way to account for QoS, packet loss, jitter, or out-of-sequence packets, which can
impact various codecs differently. So while the PSQM scoring method is a sound quality
TABLE 5. 4 MOS codec scores
Codec Bandwidth (Kbps) Score
G.711 64 4.2
G.726 AD-PCM 32 3.8
G.728 16 3.6
Internet Lob Bit Rate Codec (iLBC) 15.2 4.1
GSM Full Rate (FR) 12.2 3.5
G.729a 8 3.7
G.723 r53 5.3 3.6
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measurement tool in a vacuum, it cannot properly score codecs as they are used in real-world
network scenarios. The ITU-T realized this and withdrew the P.861 recommendation and
instead developed P.862, which is also known as Perceptual Evaluation of Speech Quality.
Perceptual Evaluation of Speech Quality
The Perceptual Evaluation of Speech Quality (PESQ) measure is an ITU-T (P.862)
recommendation that is the current standard used around the world for quantifying voice
codecs by telephone equipment manufacturers such as Cisco. PSEQ is an extension of and
a successor to PSQM since the same methods are used to score voice quality by comparing
the unencoded audio sample with the digitized sample. The testing method takes an
additional step, however, to incorporate common VoIP issues found in the transport of
voice end to end on an IP network, such as jitter, latency, and unordered packets. Adding
these calculations into the scoring mix provides a more realistic audio quality score for
todays networks. Another major difference between PSQM and PESQ is that the quality
range matches the range used by MOS. Because PSEQ and MOS scores are both given a
value between 1 and 5 (4.5 is actually the best a codec can achieve), PESQ and MOS scores
can be easily compared side by side.
Perceptual Objective Listening Quality Analysis
The Perceptual Objective Listening Quality Analysis (POLQA) is a new ITU-T standard
being developed (P.863) for next-generation voice networks. The recommendation standard
is targeted to be the replacement for PESQ. Among the primary differences between PESQ
and POLQA is the ability to offer more advanced benchmarking for high-delity wideband
codecs and voice codec operation over 3G and 4G networks.
Choosing the Right Codec
With all the different codecs available, how do you choose the right codec to use for a
particular environment? While there is no dened set of rules, there are characteristics of
a network environment that you should take into consideration when choosing a codec for
your network or network segment.
Hardware Compatibility
Cisco voice gateways support a wide variety of codecs for connecting to non-Cisco equipment
such as the PSTN or legacy voice gear. But internally, Cisco IP phones and other endpoints
commonly support only a handful of the most popular codecs. The two most popular codecs
in use today are G.711, which is recommended for use over Ethernet LANs, and G.729 and its
variations, which use less bandwidth and are therefore recommended for use in remote sites
over lower-speed WAN connections and for Cisco wireless phones that use Wi-Fi for transport.
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Network Capacity
As you learned earlier in this chapter, bottlenecks can cause all kinds of voice quality
problems. If a network has bottleneck and bandwidth congestion problems at any point on
a network, it would be wise to use a low-bandwidth codec so that more calls can be made
within that nite amount of bandwidth.
Codec Complexity
As stated previously in this chapter, codecs are categorized as either medium complexity
or high complexity. Those codecs that use high complexity require either additional
processing power at the call processing agent and/or additional DSP usage on voice
gateways that have DSP chips installed. If your DSP resources are limited, it is best to use a
lower-complexity codec.
Endpoint Uses
Earlier you were cautioned about using several of the highly compressed low-bandwidth
codecs in situations where fax machines, DTMF tones, or MOH would be sent in-band.
If this is a requirement on your network, you should use a higher-quality audio codec to
ensure that tones and fax signals are properly received at the remote end.
Call Clarity
Lastly, you should consider the clarity of the voice stream and choose the codec with the
best clarity that can safely run on your network. When deciding between two codecs that
can operate efciently on your network, you should use the codec with the higher voice
clarity quantization (such as MOS).
Calculating IP Voice Bandwidth
Consumption
Before you even begin thinking about Quality of Service for your network, you need to
take a step back and look at your network purely from a bandwidth point of view. Your
goal when designing a network is to build it with the hopes that QoS never has to be used.
In order to accomplish this goal, you need to understand how much bandwidth a voice call
will consume given various situations that affect packet and size. The conditions include
codec used, Layer 2 and 3 overhead, header compression, security, VAD, and other factors
that determine the overall size of each packet and the amount of bandwidth required for a
single call.
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Frame and Bandwidth Calculations
You need to understand two calculations. The rst calculation is to determine the size
of a single voice frame. Generally speaking, this calculation can be determined using the
following equation:
total_frame_size = layer_2_header + IP/UDP/RTP_header + voice_payload_size
We can also determine the number of packets per second (PPS) a call requires, using the
following equation:
PPS = codec_bit_rate / voice_payload_size
Finally, using the results of the total_frame_size and PPS calculations, we can determine
the amount of bandwidth consumed by a single voice call. The following calculation is used
to determine bandwidth consumption:
call_bandwidth = total_frame_size PPS
Lets rst look at the IP/UDP/RTP_header and layer_2_header information to see how we
go about nding those numbers. Then well look at the voice codec used to see how we get
the codec_bit_rate and voice_payload_size numbers needed to complete our calculations.
Determining Packet and Frame Size Information
The total frame size consists of the following elements:
IP/UDP/RTP Header Size All voice packets require IP/UDP and RTP headers for transport
across any IP network. These three headers add up to 40 bytes and are broken down as
the following:

IP header: 20 bytes

UDP header: 8 bytes

RTP header: 12 bytes
Layer 2 Header Size In addition, different Layer 2 mechanisms can add further overhead:

Ethernet: 18 bytes (14 bytes for Ethernet header info and 4 bytes for FCS/CRC checks)

Multilink Point-to-Point Protocol (MLP): 6 bytes

Frame Relay Forum Standard 12 (FRF.12): 6 bytes
Voice Payload Size The voice payload size is a multiple of the codec sample size. This
number represents the number of voice data bytes that are contained within a single packet.
For example, a codec sample size of 10 ms is
~1
100 second. That works out to be 100 PPS. A
20 ms sample means that you have two samples or
~2
100 second or 50 PPS. The size of the
payload can vary not only by codec but by settings within the codec. For example, the G.711
codec can be congured to have voice payload sizes of 80, 160, or 240 bytes. The default voice
payload size for G.711 is 160 bytes, which works out to be 2 10ms codec samples.
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Keep in mind that the larger the payload size, the more efficient your voice
stream will be. This is because you are lowering the total number of packets
(and overhead) needed to transport your voice signals. The downside to
increasing payload size is that the longer the packetization period, the larger
the payload and, therefore, the lower the voice bandwidth.
Additional Voice Packet and Frame Size Factors
Tunneling trafc through an IP network and using RTP compression techniques also add
or subtract from the overall packet/frame size.
Security or Tunnel Overhead Sometimes voice trafc needs to be tunneled through other
protocols for security and connectivity reasons. This tunneling adds signicantly to the
overall size of a voice packet. Here is a list of popular tunneling methods supported by
Cisco hardware and how much they add to the size of a voice packet:

IPSec VPN: 5057 bytes

L2TP/GRE: 28 bytes

MPLS tagging: 4 bytes per tag (may be more than one tag present)
Figure 5.5 shows an example of a voice packet being encapsulated with IPSec that adds
additional overhead to the head and tail of the packet.
Voice payload ESP tail
ESP
head
New
IP
IP UDP RTP
RTP UDP IP Voice payload
FI GURE 5. 5 An example of IPSec overhead
cRTP Header Compression If compressed RTP is enabled, it can reduce the 40-byte
IP/UDP/RTP headers to 2 or 4 bytes in size. cRTP compresses the headers to 4 bytes
when UDP checksums are used and 2 bytes when they are not sent. As you can see, cRTP
drastically reduces the overall voice stream size and is great for low-bandwidth links. Also
remember that cRTP cannot be enabled on multiaccess links.
Codec Bit Rate
The codec bit rate is the number of bits per second (bps) that the codec uses to transmit to
maintain a steady voice call. If you dont know the codec bit rate but do know the codec
sample size and sample interval, you can use the following equation:
codec_bit_rate = codec_sample_size / codec_sample_interval
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For example, G.711 has a codec_sample_size of 80 bytes (640 bits) and a codec_sample_
interval of 10 ms (0.01 seconds). Therefore, our codec bit rate can be calculated as follows:
codec_bit_rate = 640 / 0.01
codec_bit_rate = 64,000 bits per second
Typically, you know the codec bit rate for the codec you want to use. For example, Table 5.5
lists the bit rates for several commonly used voice codecs.
Frame and Bandwidth Calculation Example
Now lets pull together everything you have learned and calculate voice packet sizes and
bandwidth consumption. In our example, consider that we were given the following voice
call setting information and asked to calculate both PPS and call bandwidth for a call:

Codec: G.729a (8 Kbps)

Voice payload: 30 bytes (240 bits)

FRF.12: 6 bytes (48 bits)

IP/UDP/RTP: 40 bytes (320 bits)
Given this information, we can rst calculate the total frame size. Its easiest to rst
convert bytes into bits when doing these calculations as follows:
total_frame_size = layer_2_header + IP/UDP/RTP_header + voice_payload_size
total_frame_size = 48 + 320 + 240
total_frame_size = 608 bits
TABLE 5. 5 Codec bit/byte rates per second
Codec bps Kbps
G.711 64000 64
G.729 (all) 8000 8
G.723ar53 5300 5.3
G.723ar63 6300 6.3
G.728 16000 16
iLBC_20 15200 15.2
iLBC_30 13330 13.33
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Next, we can calculate the PPS rate for our call:
PPS = 8000 / 240
PPS = 33.33
Finally, we can determine the amount of bandwidth used per voice stream, as shown here:
call_bandwidth = 608 33.33
call_bandwidth = 20,264 bps
call_bandwidth = 20,264 bps / 1000
call_bandwidth = 20.264 Kbps
In example 2, we will use the following settings:

Codec: G.728 (16 Kbps)

Voice payload: 60 bytes (480 bits)

Ethernet: 18 bytes (144 bits)

IP/UDP/RTP: 40 bytes (320 bits)

L2TP/GRE: 28 bytes (224 bits)
Let us rst calculate the total frame size, as shown here:
total_frame_size = layer_2_header + IP/UDP/RTP_header + voice_payload_size +
tunneling_overhead
total_frame_size = 144 + 320 + 480 +224
total_frame_size = 1168 bits
Next, we can calculate the PPS rate for our call:
PPS = 16000 / 480
PPS = 30.33
Finally, we can determine the amount of bandwidth used per voice stream as shown here:
call_bandwidth = 1168 33.33
call_bandwidth = 38,929 bps
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call_bandwidth = 38,929 bps / 1000
call_bandwidth = 38.93 Kbps
Dont forget that you can approximate VAD bandwidth savings that
account for approximately 35 percent of total throughput.
Cisco has an online voice codec bandwidth calculator that can be used to calculate how
much bandwidth various codecs will use in network situations as explained above. Figure 5.6
shows the interface of the voice calculator.
If you have a valid CCO login, you can use the online voice codec calculator that can be
found here:
http://tools.cisco.com/Support/VBC/do/CodecCalc2.do
It is recommended that you learn how to calculate voice packet sizes and bandwidth
consumption manually for the exam. The online calculator is a great resource to verify
your manual calculations, however.
FI GURE 5. 6 The Cisco Voice Codec Bandwidth Calculator
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Summary
A man who does not plan long ahead will find trouble right at his door.
Confucius
Voice networks can be implemented properly or poorly. You need to analyze your current
network to determine how to best deploy voice with the landscape you are given. Every
IP network is unique, and therefore every voice deployment is unique. In this chapter we
covered the technology, terminology, codec scoring techniques, and how to calculate voice
packet and bandwidth requirements. These topics can be used to help you to plan for and
prepare your IP network for the transport of voice.
Exam Essentials
Understand the primary functions of DSPs. DSPs can be used to perform PSTN
termination, transcoding, MTP functions, and call conferencing.
Know the concerns you need to address when operating voice over an IP network. Issues
such as delity, echo, background noise, network delay, jitter, and packet loss must be
addressed and managed to optimize a voice network.
Understand the primary differences between popular codecs. Most codecs can be
differentiated by their audio clarity, compression techniques, and compatibility with types
of endpoints.
Understand the difference between narrowband and sideband codecs. Narrowband
codecs capture audio between 300 and 3300 Hz, while wideband frequencies capture audio
between 50 and 7000 Hz. Wideband audio provides higher delity but at the cost of added
bandwidth requirements.
Understand the purpose of VAD. VAD is software used to identify when packets contain
no voice audio and instead contain only background noise. VAD will not send these
empty voice payloads across the network, which can reduce bandwidth for a call by 35
percent on average.
Know which codecs are medium complexity and which are high complexity. Codecs
differ in the amount of processing power they require by DSP resources. Most codecs that
use a complex algorithm and high rate of compression fall in the high-complexity category.
Understand the various methods to quantify voice codec quality. MOS is a subjective
scoring method, while PSQM and PESQ are objective. PESQ provides more accurate scores
than PSQM, because its tests include network latency, jitter, and packet loss. POLQA is a
new standard used to better grade next-generation codecs.
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Know the characteristics used to choose a voice codec. Different codecs are optimal based
on their network environment. Characteristics such as hardware compatibility, network
capacity, codec complexity, endpoint uses, and call clarity factor into the decision-making
process.
Understand how to calculate the size of a voice frame. A voice frame is calculated by the
adding the Layer 2 and Layer 3 headers plus the voice payload size.
Understand how to calculate the bandwidth requirements for a voice call. A voice call
bandwidth is calculated by taking the total size of a single voice frame multiplied by the
number of packets per second the stream will operate at.
Know the different factors that can add or reduce a voice frame size. Layer 2 transport
mechanisms, VPN tunneling, and cRTP can add to or reduce a voice frame size.
Written Lab 5.1
1. What is the name for audio samples that are collected between 50 and 7000 Hz?
2. When VAD is used, this adverse effect can prevent the rst few milliseconds of a per-
sons voice from being sent.
3. What IOS command can be used in global conguration mode to adjust the VAD
detection timer to 500 milliseconds?
4. What are the minimum network recommendations for delay, jitter, and packet for
voice networks?
5. Which popular voice codec has a bandwidth stream of 64 Kbps and uses no compres-
sion?
6. What voice card conguration command is used to set codec complexity to high?
7. What show command can be used to see what complexity settings your DSPs are cur-
rently congured for?
8. How many bytes are IP/UDP/RTP headers that are uncompressed?
9. What two pieces of information need to be multiplied together to determine the
amount of bandwidth required for a call?
10. Your voice codec bit rate is 32 Kbps and the payload is 20 bytes. What is the packet
per second (PPS) rate?
(The answers to Written Lab 5.1 can be found following the answers to the review questions
for this chapter.)
Written Lab 5.1 171
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Review Questions
1. Why are DSP resources needed to process DTMF tones between SIP and SCCP devices?
A. SCCP does not use DTMF.
B. SIP does not use DTMF.
C. SCCP DTMF tones are sent in-band and SIP tones are sent out-of-band.
D. SIP DTMF tones are sent in-band and SCCP tones are sent out-of-band.
2. On an IP network, what is the most common reason for echo?
A. VAD clipping
B. Glare
C. Network delay
D. Impedance
3. Under normal conditions, VAD can eliminate what percentage of bandwidth on a network?
A. 10 percent
B. 40 percent
C. 5 percent
D. 35 percent
4. When configuring a POTS port, how can you enable both VAD and comfort noise
synthesis? (Choose two.)
A. vad
B. VAD is enabled by default on POTS interfaces.
C. comfort-noise
D. voice vad-time 750
5. What are the two different types of network delay? (Choose two.)
A. Fixed
B. Variable
C. Queuing
D. Static
6. What is the definition of network jitter?
A. The amount of time it takes a packet to travel from its source to the destination
B. The variation of the amount of time it takes when sending packets from the source
endpoint
C. The variation of the amount of time it takes when receiving packets at the destination
D. The amount of time it takes a voice gateway to transcode a packet from one codec to
another
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7. All Cisco IP phones support what two voice codecs?
A. G.729
B. G.722
C. iLBC
D. iSAC
E. G.711
8. Which G.729 codec operates at medium complexity and has built-in VAD?
A. G.729a
B. G.729b
C. G.729i
D. G.729ab
9. Which of the following is not a codec complexity option on C5510 DSP chipsets?
A. Secure
B. High
C. Medium
D. Low
E. Flex
10. You configure the following:
Router(config-voicecard)# codec complexity secure
What did you enable?
A. IPSec
B. sRTP
C. GRE
D. SCCP
11. By default, C5510 DSP chips are configured for what type of codec complexity?
A. High
B. Secure
C. Medium
D. Flex
12. Which voice codec quality measurement tool is objective in testing but does not account for
network problems such as latency or jitter?
A. PESQ
B. PSQM
C. MOS
D. POLQA
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13. Which of the following codecs would you likely choose when using Cisco 7921 wireless IP
phones?
A. G.711
B. iSAC
C. G.729
D. G.726
14. You have two voice gateways separated by a low-speed WAN. The gateways must support
several simultaneous low-bandwidth calls and therefore only have DSPs to support
medium-complexity codecs. Which two of the following codecs could be used?
A. G.729a
B. G.729b
C. G.729ab
D. G.711
15. You are using the G.728 codec with 40-byte payloads. The traffic is going across a frame-
relay network. What is the bandwidth size for a single voice frame?
A. 64 bytes
B. 16 bytes
C. 118 bytes
D. 86 bytes
16. You are using the G.711 codec with 80-byte payloads. cRTP (without checksums) is enabled.
The traffic is going across a frame-relay network. What is the bandwidth size for a single
voice frame?
A. 116 bytes
B. 108 bytes
C. 88 bytes
D. 122 bytes
17. The G.729 codec uses 20-byte payloads. How many PPS will one call require?
A. 20
B. 33.33
C. 50
D. 10
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18. Your voice network uses the following network and voice codec:
G.711 (64 Kbps)
160 bit (20 byte) voice payload samples
Ethernet transport
How much bandwidth is required to support ve simultaneous voice calls?
A. 160 Kbps
B. 88 Kbps
C. 436 Kbps
D. 240 Kbps
19. Your voice network uses the following network and voice codec:
G.723 (6.3 Kbps)
72 byte voice payload samples
Ethernet transport
IP+GRE Tunneling
Approximately how much bandwidth is required to support three simultaneous
voice calls?
A. 41.7 Kbps
B. 32.8 Kbps
C. 22.73 Kbps
D. 18.97 Kbps
20. After you calculate the total bandwidth for a voice stream, what is one other factor that
you may need to take into consideration from a bandwidth usage perspective?
A. Delay
B. Jitter
C. Packet loss
D. VAD
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1. D. SIP sends DTMF tones inside the voice packet payload while SCCP sends them in
separate packets. An MTP (using DSPs) can be configured to translate DTMF
tones between incompatible devices.
2. C. While impedance is commonly the reason for echo on traditional telephone networks,
delay is the cause on IP networks.
3. D. VAD can detect silence on the line and prevent empty payload packets from being
sent. Up to 35 percent of a call is silence, and this is the percent of bandwidth that can be
eliminated from being sent on the network.
4. A, C. By default, VAD is disabled on POTS interfaces and must be enabled with the vad
command. Once VAD is enabled, it is wise to also enable comfort noise synthesis for user
feedback using the comfort-noise command.
5. A, B. Fixed delay is built into the network because of physical layer and protocol
limitations. Variable delay is caused by slowdowns on the network because of queuing of
packets in bottlenecks.
6. C. Jitter is the time it takes between the receipt of one voice packet and the next voice
packet in the same voice call, which varies because of network delay. A long gap between
the receipt of voice packets at the destination can cause the voice stream to stutter.
7. A, E. While newer Cisco IP phones support multiple voice codecs, low-end and older
phones typically support only G.729 and G.711.
8. D. The G.729ab codec takes the medium-complexity quality for Annex A and the built-in
VAD from Annex B.
9. D. There is no option to configure low codec complexity, because all codecs are
considered to be of medium or high complexity.
10. B. Using the secure codec complexity option enables secure RTP (sRTP).
11. D. The newer C5510 PVDM2 chips are configured for flex codec complexity type for
oversubscription.
12. B. Perceptual Speech Quality Measure is objective because the scoring methods are
computerized and are highly replicated. The problem is that the test does not factor
network problems into the scoring equation.
13. C. Wi-Fi is a shared medium, and therefore a low-bandwidth codec is recommended. The
two most popular codecs (and the only ones that the 7921 is compatible with) are G.711 and
G.729. G.729 uses only 8 Kbps of bandwidth compared to 64 Kbps per call.
14. A, C. All G.729 codecs are low bandwidth, but only the G.729a and G.729ab codecs can
operate at medium complexity.
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15. D. total_frame_size = 48 + 320 + 320 = 688 bits = 86 bytes
16. C. total_frame_size = 48 + 16 + 640 = 704 bits = 88 bytes
17. C. PPS = 8000 / 160 = 50
18. C. total_frame_size = 144 + 320 + 1280 = 1744 bits
pps = 64000 / 1280 = 50
call_bandwidth = 1744 50 = 88,000 bps = 87.2 Kbps
87.2 5 = 436
19. A. total_frame_size = 144 + 320 + 576 + 224 = 1264 bits
pps = 6300 / 576 = 10.93 = 11
call_bandwidth = 1264 11 = 13,904 bps = 13.9 Kbps
13.9 3 = 41.7 Kbps
20. D. If VAD is used on your calls, it can reduce bandwidth up to 35 percent more than was
calculated.
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1. Wideband
2. Clipping
3. voice vad-time 500
4. 150 ms, 30 ms, and 1 percent
5. G.711
6. codec complexity high
7. show voice dsp
8. 40 bytes
9. Total packet size and packets per second (PPS)
10. 5 PPS
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Configuring Voice
Gateway Ports
and DSPs
CHAPTER:
Configure analog voice ports.
Configure digital voice ports.
Configure dial peers.
Verify a dial plan implementation.
Describe different voice ports and their functionality.
Chapter
6
c06.indd 179 9/21/11 11:19:30 AM
This is the chapter where we begin to pull in everything
youve learned up to this point about voice networks and voice
gateways and really understand how to congure our Cisco
voice gateways for operation on IP and PSTN networks. In this chapter, well go through
the full conguration process to set up analog and digital interfaces in various scenarios.
In addition, we will go through the process of conguring a digital signal processor (DSP)
farm that ofoads services from a CUCM. At the end of this chapter, we will examine
several show, test, and debug commands used to verify congurations and troubleshoot
voice gateways.
Analog Port Configurations
In this section youll see how to congure FXS, FXO, and E&M ports and dial peers using
various example scenarios, including situations such as PLAR, DID, and CAMA.
Configuring an FXS and an FXO PLAR OPX Port
Our rst example will show how to congure our voice gateway to connect a single FXS port
for an analog telephone with a single FXO port that connects to the PSTN. Because we have
a single phone with a single FXO port, we will use off-premises extension (OPX) Private Line
Automatic Ringdown (PLAR) so that the telephone connected to the FXS interface must be
answered before the FXO interface answers the call, as shown in Figure 6.1.
FI GURE 6.1 An Example of FXS and FXO PLAR OPX
Remote ofce
Ext: 2222
PLAR OPX
2222
555-321-1234
0/1/0
FXO
0/0/0
FXS
V
PSTN
FXS interfaces commonly connect analog telephones or fax machines to voice gateways.
To congure an FXS port, you need to enter into config-voiceport mode by choosing the
slot/port number you wish to congure. For example, if we wanted to congure FXS port
0/0/0 on our router, we would issue the following commands:
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Once we are in config-voiceport mode, the FXS ports can be congured for various
signaling. By default, FXS ports are congured to operate identically to a POTS line in the
United States. Some of the default conguration settings will need to be modied to have the
ports operating properly based on locale. For example, lets say you have a voice gateway
that needs to connect FXS port 0/0/0 for a single analog phone. The phone and voice
gateway are located in Thailand. You should consider modifying the following options:
signal You can change the signaling from the default loopstart to groundstart. Loop-
start signaling has no current owing through it unless it is in use. Therefore it is cheaper
to use and commonly found in residential homes. Ground-start signaling uses an alternate
method to help eliminate glare, as you learned, but also uses more electrical current, which
makes it more expensive to run. Therefore, ground start is more commonly found in
businesses and costs extra. In our example conguration, we will choose to congure loop-
start signaling because it is more common.
cptone This command changes the call progress tones based on the locale of the phone.
You can see the different two-letter ISO-3166 country codes by issuing the cptone ?
command, as shown here:
Router(config-voiceport)#cptone ?
locale 2 letter ISO-3166 country code
AR Argentina IN India PE Peru
AU Australia ID Indonesia PH Philippines
AT Austria IE Ireland PL Poland
BE Belgium IL Israel PT Portugal
BR Brazil IT Italy RU Russian Federation
CA Canada JP Japan SA Saudi Arabia
CN China JO Jordan SG Singapore
CO Colombia KE Kenya SK Slovakia
C1 Custom1 KR Korea Republic SI Slovenia
C2 Custom2 KW Kuwait ZA South Africa
CY Cyprus LB Lebanon ES Spain
CZ Czech Republic LU Luxembourg SE Sweden
DK Denmark MY Malaysia CH Switzerland
EG Egypt MX Mexico TW Taiwan
FI Finland NP Nepal TH Thailand
FR France NL Netherlands TR Turkey
DE Germany NZ New Zealand AE United Arab Emirates
GH Ghana NG Nigeria GB United Kingdom
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GR Greece NO Norway US United States
HK Hong Kong OM Oman VE Venezuela
HU Hungary PK Pakistan ZW Zimbabwe
IS Iceland PA Panama
Router(config-voiceport)#cptone
Since our example router is in Thailand, we will use TH as our country code.
ring cadence This command modies the pulse and interval times when your analog
phone rings. This command rst looks to the cptone locale and uses the default specied
for that specic locale. If you want to modify this, you can use the ring cadence
patternXX command or ring cadence define pulse interval command to either choose
one of the precongured patterns or dene your own pulse/interval settings. In our case,
we will choose to modify our ring cadence to use pattern08.
ring frequency It is rare nowadays, but sometimes phones in different parts of the world
are triggered to ring using different frequencies. If you nd yourself wondering why a
phone is not ringing or emits a soft buzzing sound when it should be ringing, you may need
to adjust the ring frequency. In our example, well modify the default frequency from 25
Hz to 50 Hz.
station-id The station-id number and station-id name are often used on FXS
interfaces to add caller ID information to analog phones. In our example, we congure
the FXS port that is attached to our phone to the extension 2222 and a name of
Remote-Ofce.
no shutdown This command activates the FXS port we are conguring if it is in a
shutdown state. Typically, interfaces default to a shutdown state, and you must manually
enable them using this command.
So our nal conguration for a single FXS port in Thailand would look like this:
Router(config-voiceport)#cptone TH
Router(config-voiceport)#ring cadence pattern08
Router(config-voiceport)#ring frequency 50
Router(config-voiceport)#station-id number 2222
Router(config-voiceport)#station-id name Remote-Office
Router(config-voiceport)#no shutdown
Router(config-voiceport)#end
Router#
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Now that we have our FXS port congured, we can congure our FXO port 0/1/0.
Remember that FXO ports can only send DNIS information out to the PSTN. ANI can be
used to route calls internally.
Conguring an FXO port is similar to conguring an FXS port. The FXO port can be
congured for either loop-start or ground-start signaling. The vast majority of the time,
you will congure FXO ports going out to the PSTN to use ground-start signaling, which
is what we will do in our setup. There are two FXO-specic commands when conguring
your voice port: dial-type and ring number.
You can choose to congure your FXO port to use DTMF or pulse dialing using the
dial-type command. We will congure DTMF dialing, which is the default.
Remember that with FXO ports, the line terminates at the router and not a telephone
endpoint; therefore, the port must answer the call for you. By default, the port will answer
immediately (as soon as ring 1 is detected). You can change this setting to have the port
answer the call from anywhere between 1 and 10 rings using the ring number command.
This command is useful when your FXO port is split and you are sharing it between a
telephone and a fax machine or an automated attendant. You can congure ring number 4
so it gives a person the chance to answer the phone. If the phone is not manually answered
by three rings, a soon as the fourth ring is detected, the FXO port answers the call and
handles it according to the conguration. In this scenario, we will stick with the default
ring number of 1.
PLAR stands for Private Line Automatic Ringdown. This is an autodialing mechanism
that is used to associate a port with a single destination. PLAR can be congured on FXS
interfaces in locations such as public lobbies that will automatically match the extension
number attached to the PLAR command with a precongured dial peer that points to the
location of the information operator. In this situation, as soon as the PLAR-congured
phone goes off-hook, the remote extension is dialed. To congure PLAR on FXS interfaces,
you would use the connection plar extension command.
From an FXS bat phone perspective, PLAR is easy to understand. But PLAR can also
be used on FXO interfaces that connect to the PSTN as shown in our example. In this case,
we want to automatically forward inbound calls from the PSTN to our single telephone
with the extension 2222. Technically, we could use the same connection plar extension
command that we use with FXS ports. But this would cause the call to be terminated rst
at the voice gateway, and then a second ring would initiate to our analog phone at 2222.
This can cause problems with situations such as carrier billing records and a call ringback
cadence hiccup. Instead, the connection plar opx extension command bypasses the
rst termination at the router and simply waits until the FXS port goes off-hook.
Even though many of these conguration options are default settings, this example
congures them for learning purposes. Therefore, the nal FXO port conguration looks
like this:
Router(config-voiceport)#dial-type dtmf
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184 Chapter 6

Router(config-voiceport)#ring number 1
Router(config-voiceport)#connection plar opx 2222
Router#
Now that we have our physical ports congured, we need to congure dial peers so our
router will know where to route calls that are destined to either the FXS or FXO interface.
To do that, we need to rst determine what should trigger the dial peer. Then we need to
tell the dial peer from which POTS port to forward the call.
For the PSTN connection, we have it set up to operate as a PLAR line that automatically
triggers extension 2222, which is our analog phone extension. We need to congure a
POTS dial peer to let the router know that the device connected to voice port 0/0/0 is
extension 2222. To do that, we simply use the destination-pattern command. The
complete conguration for our rst POTS dial peer looks like this:
Router#
Lastly, calls made from the FXS port out to the PSTN require a dial peer. We will
use the number 9 as our dial-peer trigger. We then want the user to be able to dial any
combination of numbers so they can dial locally, nationally, and internationally. The
following dial-peer conguration triggers when the rst digit entered by the user is 9,
collects any number of digits for up to 10 seconds, and nally forwards all digits (except
the 9) out the FXO port to the PSTN:
Router(config-dial-peer)#destination-pattern 9T
Router#
The result of these congurations is that when a caller on the PSTN dials 555-321-1234,
the voice gateway automatically routes the off-premise call to extension 2222, which belongs
to the analog phone attached to our FXS port. In addition, internal phones such as our
analog phone can dial 9 and then a PSTN number to make off-network calls.
Configuring FXS/DID Inbound and FXO Outbound
Our second example will demonstrate a different method for using FXS portsthey have
the ability to function as inbound-DID ports. This is often useful in small to medium-size
c06.indd 184 9/21/11 11:19:33 AM
ofces that have a block of DIDs from their PSTN and a number of high-density FXS/DID
cards. DID allows callers on the PSTN or a separate PBX to dial direct telephone extensions.
This means they dont need to rst call an operator or automated attendant that switches
the call internally on a call-processing agent such as the CUCM. The alternative would be to
have a single PSTN line for each telephone. Obviously, a one-to-one PSTN-to-internal phone
scheme does not scale well and certainly is not cost effective, so the DID option is a great
alternative. One caveat to using FXS ports for DID support is that the interfaces cannot be
used for outbound calling to the PSTN. In that case, separate FXO interfaces and dial peers
must be congured to handle outbound calling. Figure 6.2 shows our example scenario, in
which an FXS/DID interface will be used for inbound DID calling.
FI GURE 6. 2 FXS/DID inbound and FXO outbound
Ext: 3005
3005
Inbound calls
Outbound calls
PSTN
strips of
555-441
555-441-3005
1/0/0
FXO
0/0/0
FXS/DID
0/0/1
FXS
V
PSTN
Lets say we have a small ofce with 10 internal phones. Our PSTN has given us the
following block of numbers: 555-441-3000 to 555-441-3009. When someone on the PSTN
dials 555-441-3005, the PSTN will strip off all but the last four digits and send 3005
to our voice gateway on the FXS/DID 0/0/0 port using DID wink-start signaling. The
four-digit extension is matched against the 3005 dial peer, and the call is routed out to the
analog phone congured on FXS interface 0/0/1.
First, we will congure our FXS/DID port that connects to the PSTN and the FXS port
that connects to the analog phone:
Router(config-voiceport)#signal did wink-start
Router(config-voiceport)#voice-port 0/0/1
Router#
Next, we will congure the inbound dial peer for the FXS/DID port. To accomplish
this, we will use the direct-inward-dial command to enable DID for this dial peer. This
command matches the DNIS destination number in the dial peer. In our case, we will use
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the incoming called-number .... command so we can match any four digits that come in
from the PSTN:
Router(config-dial-peer)#direct-inward-dial
Router(config-dial-peer)#incoming called-number....
Router#
Now we can congure the dial peer for our analog phone at extension 3005:
Router#
DID configurations such as the previous dial-peer configuration
highlight a classic example of one-stage dialing. When DID is configured
on an inbound dial peer as you see here, our local voice gateway never
terminates the call or presents a dial tone to the calling party. What
happens is that the digits are simply collected using the direct-inward-
dial command and matched against a dial peer. The dial peer in turn
then sends the call setup signaling information out port 0/0/1. If we did
not use the direct-inward-dial command and instead had an FXO port
connected to our PSTN, we would need to use two-stage dialing. This
means that a caller would dial an extension that terminates at the FXO
port. Then the router would present a secondary dial tone at which the
caller would have to dial an internal extension to reach the desired phone.
Alternatively, an autoattendant can be configured on the voice network to
route calls using a two-stage dialing method.
At this point, people on the PSTN can dial 555-441-3005 and reach our analog
phone. However, we still need to congure off-network dialing on our FXO interface. To
accomplish this goal, we will congure our physical FXO 1/0/0 interface for ground-start
signaling and create a dial peer that matches a 9 followed by the common wildcard digit-
matching conguration used to dial nationally within the NANP. Be sure to strip off only
the 9 and send all other digits to the PSTN:
c06.indd 186 9/21/11 11:19:34 AM
Router(config-voiceport)#exit
Router(config-dial-peer)#destination-pattern 9[28].........
Router(config-dial-peer)#forward-digits 10
Configuring E&M to Bridge Legacy PBX with VoIP
Networks
If you need to congure analog E&M trunks on a Cisco voice gateway, it is highly likely
that the other end of that connection connects to a legacy PBX. Figure 6.3 shows a voice
gateway that bridges an IP-based voice network with a legacy PBX phone system.
FI GURE 6. 3 E&M to bridge legacy and VoIP networks
Extensions
4xxx
Extensions
5xxx
2/1
Fa 4/0
2/0
PBX
PBX E&M settings:
Type I
4-wire
Immediate-start
E & M
Trunks
V
IP network
To IP: 192.168.10.2
As you can see from the diagram, the IP and legacy voice networks are bridged by two
E&M ports. In addition, the PBX requires this type of E&M setup:
E&M port type 1
Four-wire operation
Immediate-start signaling
First, we will congure and enable our two E&M ports:
Router(config)#voice-port 2/0
Router(config-voiceport)#signal immediate-start
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188 Chapter 6

Router(config-voiceport)#operation 4-wire
Router(config-voiceport)#voice-port 2/1
Router(config-voiceport)#signal immediate-start
Next, we will congure dial peers that match our four-digit PBX extension numbers and
forward them to our two E&M ports:
Router(config-dial-peer)#destination-pattern 4...
Router(config-dial-peer)#forward-digits all
Router(config-dial-peer)#port 2/0
Router(config-dial-peer)#dial-peer voice 4001 pots
Lastly, we can congure a dial peer for calls coming inbound from the E&M ports that
are destined to our IP phones. Note that this conguration assumes that our voice gateway
is properly congured for IP routing between the two voice gateways shown in Figure 6.3.
Router(config-dial-peer)#dial-peer voice 5000 voip
Configuring CAMA
In North America, there are special port congurations that connect to
Centralized Automatic Messaging Accounting (CAMA) trunks. While CAMA trunks can
be used for a variety of reasons, they are primarily used for dedicated access to Enhanced
911 (E911) services. Some U.S. states require businesses over a certain size to connect
directly to the E911 service. Businesses are commonly required to do so because they
have large buildings or campus areas, and the CLID for outbound calls oftentimes is a
centralized number as opposed to a specic extension. Therefore, the Automatic Number
Identication (ANI) is used by emergency services to better pinpoint where a caller is
within a business.
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Normally, a PSTN call is routed based on the destination telephone number. E911, on
the other hand, routes calls based on the calling partys ANI. The ANI is for E911 call
routing to help pinpoint the location of the call using a database and, ultimately, where the
emergency is. Having the ANI is also useful if the emergency call has been disconnected
and the E911 operator needs a callback number.
Cisco FXO interface cards can be used to congure CAMA trunks such as the four-
port VIC-4FXO. There are ve CAMA signaling options to choose from, as shown in this
example:
Router(config-voiceport)#signal cama ?
KP-0-NPA-NXX-XXXX-ST Type 2 CAMA Signaling
KP-0-NXX-XXXX-ST Type 1 CAMA Signaling
KP-2-ST Type 3 CAMA Signaling
KP-II-NPA-NXX-XXXX-ST-KP-NPA-NXX-XXXX-ST Type 5 CAMA Signaling
KP-NPD-NXX-XXXX-ST Type 4 CAMA Signaling
<cr>
The different signaling types primarily have to do with ANI and the number of digits
the PSAP is requesting as directed by your local emergency services. For example, when
using KP-0-NPA-NXX-XXXX-ST, the PSAP expects to see all 10 of the E.164 digits, while KP-
0-NXX-XXXX-ST signaling will drop the area code prior to forwarding digits. The type of
signaling used depends on your local PSAP. Make sure you congure your CAMA for the
proper signaling it is expecting.
FI GURE 6. 4 CAMA trunk to E911 services
V
PSTN
PSAP
KP-0-NPA-
NXX-XXXX-ST
911 with ANI for
reverse lolkup
CAMA
0/1
Dial peer
matched and
911 digits
sent to
PSAP.
911 OR 9911
is dialed.
E911
operators
Internal voice
network
These CAMA trunks then terminate at a local Public Safety Answering Point (PSAP).
When a 911 call is made from one of the IP phones, as shown in Figure 6.4, the voice
gateway should route the call out the CAMA trunk to the PSAP.
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If emergency services in your area require you to use KP-NPD-NXX-XXXX-ST
signaling, one additional step is required. The NPD in the signaling name
stands for Numbering Plan Digit. This is a single digit that represents the
three-digit NANP area code. Therefore, the PSAP requires that our voice
gateway send the NPD plus the three-digit central office and four-digit
subscriber code. Because of this, we must manually configure a table that
maps area codes to NPD codes that are specified by emergency services in
our area. For example, NPD digit 0 represents area code 555; therefore we
can use the ani mapping command as shown here:
Router(config-voiceport)#ani mapping 0 555
A full example of this configuration is found in Hands-On Lab 6.2 at the end
of this chapter.
In our example, the PSAP requires that we congure KP-0-NXX-XXXX-ST signaling. In
addition to the port setup, we will congure two dial peers to be routed out our CAMA
port. One dial peer will be for when users dial 911 and the other for users who dial 9911,
which is commonly done in environments where users are trained to dial 9 to reach an
outside line. For this particular dial peer, make sure you forward only the 911 digits. The
conguration of the voice port looks like this:
Router(config)#voice-port 0/1
Router(config-voiceport)#signal cama KP-0-NPA-NXX-XXXX-ST
Note: need to shut/no shut to complete the CAMA signal type configuration.
Router(config-voiceport)#shutdown
You can see that after we changed signaling to CAMA, the router gave us a console
message stating we must perform a shutdown and no shutdown on the FXO port to put the
interface into CAMA mode. Once the FXO port is congured, we can create the dial peers
to send 911 and 9911 calls out the CAMA interface. Heres how to accomplish this task:
Router#
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Notice that when we changed the signal type, the router asked us to perform a shut/no
shut to complete the CAMA signal type conguration. Also, notice that in both dial peers,
we used the forward-digits all and forward-digits 3 commands to forward either
all dialed digits or just the last three digits in the case a user dials 9911. As you are aware
already, different digit-manipulation methods can be used to achieve the same goal. For
example, instead of the forward-digits all command, we could have chosen to use no
digit-strip. Or in both the 911 and 9911 congurations, we could have used the prefix
911 command. There is no right or wrong way to do this as long as you accomplish your
goal to forward 911 out the CAMA interface.
Digital Port Configurations
Now we will move on to discuss various digital port congurations. Digital PSTN
connections are commonly used as multichannel trunks to the PSTN. In this section,
we will congure a T1 CAS, T1 PRI, and an ISDN BRI port to the PSTN in a couple of
different real-world scenarios.
Configuring a T1 CAS to Analog Cross-Connect
In our rst scenario, we are going to congure a digital T1 CAS to interoperate with
analog phones. This is useful for situations where analog devices such as fax machines are
still used. Figure 6.5 shows an example of a site that has one fax machine that requires a
dedicated analog line.
To accomplish this goal with our T1 CAS, we can take one of the 24 T1 CAS digital
(DS0) channels and place it into what is known as a ds0-group. Once we carve off our
single CAS channel timeslots, we can congure it to operate in a channel bank mode,
where the digital circuit cross-connects to analog lines.
PSTN
PSTN CAS settings:
ESF
B8ZS
Clock from PSTN
0/0/0
FXS
1/0 T1
CAS
Fax
V
FI GURE 6. 5 A CAS channel bank example
To congure the T1 CAS card for channelization support, you should rst identify the
T1 CAS slot/port and enter into config-controller mode. All T1s and E1s on a Cisco
router are congured by entering config-controller mode. Compare this to how we
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congure analog ports, by entering config-voiceport mode. In our example, we will rst
congure the T1 CAS settings as follows:
Extended Superframe (ESF) framing type
B8ZS line coding
Clock source from the PSTN (line)
ds0-group 0 that has one DS0 congured as FXO with ground-start signaling
The framing type sets the framing that your PSTN provider has congured on their end.
You can see the options listed here while in config-controller mode:
Router#config t
Router(config-controller)#framing ?
esf Extended Superframe
sf Superframe
We will congure ESF framing for the T1 CAS:
The line coding type you choose again depends on your PSTN provider. You have to set
your linecode to match whatever coding they provide to you on the circuit. Your options
are AMI or B8ZS. B8ZS is by far the most common linecoding these days, and we will
congure it here:
Router(config-controller)#linecode ?
ami AMI encoding
b8zs B8ZS encoding
The T1 CAS is a digital TDM circuit, as you have already learned. And digital circuits
must use precise clocking to determine when bits are sent across the wire. You have two
conguration options for clocking, as shown here:
Router(config-controller)#clock source ?
internal Internal Clock
line Recovered Clock
In this example, we will choose to receive clocking from the PSTN, so we will select the
line option:
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One of the great advantages of a CAS circuit is the ability to break up DS0s into DS0
groups that can be used for different purposes. In our example, we will assume that we
want to congure only the rst timeslot for a fax machine. The fax machine uses an FXS
port with loop-start signaling. A second analog port will be used to connect to a legacy
PBX using two-wire E&M wink start. The T1 channel for our fax will use loop start for
signaling to the CO. The channel used to connect to the legacy PBX will be congured for
E&M immediate start. First, lets look at all the different ds0-group port types available:
Router(config-controller)#ds0-group 0 timeslots 1 type ?
e&m-delay-dial E & M Delay Dial
e&m-fgd E & M Type II FGD
e&m-immediate-start E & M Immediate Start
e&m-wink-start E & M Wink Start
ext-sig External Signaling
fgd-eana FGD-EANA BOC side
fgd-os FGD-OS BOC side
fxo-ground-start FXO Ground Start
fxo-loop-start FXO Loop Start
fxs-ground-start FXS Ground Start
fxs-loop-start FXS Loop Start
none Null Signalling for External Call Control
<cr>
Router(config-controller)#ds0-group 0 timeslots 1 type
Now we will congure ds0-group 0 to include the rst timeslot with loop-start
signaling. Ds0-group 1 will then be congured for E&M immediate start. We then
individually congure our logical DS0 ports by using the T1 slot/port number, followed by
a colon (:) and the ds0-group number, which is either 0 or 1 in our case:
Router(config-controller)#ds0-group 0 timeslots 1 type fxo-loop-start
Router(config-controller)#ds0-group 1 timeslots 2 type e&m-immediate-start
Router(config-controller)#exit
Router(config-voiceport)#signal loop-start
Router(config-voiceport)#signal wink-start
Router#
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That completes our setup of our T1 CAS port for cross-connecting an analog fax and a
PBX using the DS0 channel bank. The nal steps require us to congure the FXS ports that
our analog fax machine and legacy PBX connect to along with the appropriate dial peers
that map the analog extension to the FXS ports and FXO for outbound PSTN dialing. The
one new global conguration command we need to use here to complete the channel bank
cross-connect is connect name voice-port analog-port T1 digital-port ds0-group-
number, which species that we will be mapping our DS0 channels to our two FXS ports.
The name species a unique name identier, while the analog-port and digital-port options
specify the slot/port of the connections we are cross-connecting. The ds0-group-number is
the number we use to identify the one channel DS0 group conguration, as shown here:
Router#config t
Router(config)#connect fax1 voice-port 0/0/0 t1 1/0 0
Router(config)#connect pbx1 voice-port 0/0/1 t1 1/0 1
And to complete the conguration we congure dial peers for our FXS and E&M
analog ports. We will also congure the necessary inbound and outbound dial peers. The
dial peer for the FXS port is self-explanatory. A second inbound dial peer for the E&M
port will route calls in the 555200XXXX range to the legacy PBX. On the T1 port, our
inbound dial peers accept calls from any number. And the outbound dial peer uses 9 as the
trigger digit for the voice gateway plus 10-digit national dialing in the United States. Notice
that we specify the DS0 group number when we tell the router to route off-network calls to
the PSTN:
Router(config-dial-peer)#destination-pattern 555200....
Router(config-dial-peer)#incoming called-number.
Router(config-dial-peer)#port 1/0:0
Router(config-dial-peer)#incoming called-number.
Router(config-dial-peer)#incoming called-number 555200....
Router(config-dial-peer)# dial-peer voice 9 pots
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Router(config-dial-peer)# dial-peer voice 99 pots
Router#
We can now verify our channel bank setup by issuing the show connection all
command, which displays the list of connections, their mappings, and their current state:
Router# show connection all
ID Name Segment 1 Segment 2 State
==========================================================================
1 fax1 voice-port 0/0/0 T1 1/0 01 UP
2 pbx1 voice-port 0/0/1 T1 1/0 02 UP
Configuring a T1 PRI
The conguration of T1/E1 PRI circuits is similar to that of T1/E1 CAS circuits. Well
point out several differences throughout this example. For this conguration, we are
asked to congure a fractional T1 circuit consisting of 12 channels. We are only asked
to congure the T1 connection to the PSTN and the outbound dial peer for off-network
calling, as shown in Figure 6.6.
FI GURE 6. 6 An ISDN T1 PRI example
PSTN
ISDN switch: Primary-NI
PSTN PRI settings:
Channels: 1-12
ESF
B8ZS
Clock from PSTN
3/0 T1
PRI
V
Dont forget that BRI and PRI ISDN carry Q.921 and Q.931 signaling out of band.
These two signaling protocols are used between the voice gateway and ISDN switch.
Several different switch types are in use on PSTNs around the world. When you are
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conguring a voice gateway, you must know which type of ISDN switch your PSTN is
using. You congure the switch type globally on the voice gateway by issuing the isdn
switch-type type command. The following output shows the different IDSN switch types;
after reviewing them, well congure the voice gateway to use primary-ni:
Router(config)#isdn switch-type ?
primary-dms100 Northern Telecom DMS-100 switch type for the U.S.
primary-dpnss DPNSS switch type for Europe
primary-net5 NET5 switch type for UK, Europe, Asia and Australia
primary-ni National ISDN Switch type for the U.S.
primary-ntt NTT switch type for Japan
primary-qsig QSIG switch type
primary-ts014 TS014 switch type for Australia (obsolete)
Router(config)#isdn switch-type primary-ni
Once we have set our PSTNs switch type globally, we can enter config-controller
mode and begin conguring our T1 circuit. When conguring a PRI, we use the pri-
group command to specify the timeslots we will be using. In our scenario, we will be using
only 12 channels for voice. We also must remember to include the 24th timeslot for our
signaling channel. The D timeslot is always 24 on a T1. But always remember that timeslots
of a T1 are numbered 124. However, channels are numbered 023. That means that the
D timeslot is 24 and the D channel is 23. And the E1 D timeslot is 16, while the D channel
is 15. Here is how we congure our T1 PRI to identify the rst 12 timeslots for voice
transport and our 24th timeslot for Q.921 and Q.931 signaling. We will also congure
framing, linecoding, and the clock source:
Router(config-controller)#pri-group timeslots 1-12,24
Router(config-controller)#end
Router#
We can verify that our channels were properly congured by issuing the show voice port
summary command, as shown here:
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Router1#show voice port summary
IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
=============== == ============ ===== ==== ======== ======== ==
0/1/0:23 01 xcc-voice up dorm none none y
PWR FAILOVER PORT PSTN FAILOVER PORT
================= ==================
Router#
The second column in the output of the show voice port summary command is CH,
for channel, which lists all 12 of our usable voice circuits. You can also see that our logical
D channel has been created as serial 0/1/0:23. We need to go into our D channel interface
and congure the isdn incoming-voice voice command to specify that this T1 circuit will
be used only for voice. Keep in mind that ISDN can transport voice, data, or both on a T1
or E1. If the channels are congured for voice, the voice gateway directs calls to be processed
by the DSP. If they are congured for data, the voice gateway bypasses the DSPs:
Router(config)#interface serial 0/0:23
Router(config-if)#isdn incoming-voice voice
Router(config-if)#end
Router#
Our last step is to congure an outbound dial peer for off-network national calls. Notice
that since all of our signaling is handled by channel 23, we send all this information to the
logical 0/1/0:23 port:
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198 Chapter 6

Router(config-dial-peer)#port 0/1/0:23
Router#
Configuring DSP Resources
Youve already learned how to congure your DSPs manually, to handle the termination of
voice codecs that operate in low, medium, or high complexities. In addition you know what
flex and secure modes are and when they should be used. In this section, we will explore
how to congure the ofoading of transcoding, conferencing, and media termination
points (MTP) services from the call-processing agent such as the CUCM.
Enabling a DSP Farm on a Voice Gateway
DSP chips can be included directly on voice card modules, or they can be independently
installed onto the router motherboard on chips that look similar to PC RAM. Once you
have sufcient DSP resources installed for the job, your next task is to congure your
router as a DSP farm. As the name implies, the router will work to ofoad, or farm out,
tasks such as transcoding and conferencing from the CUCM. Thus the CUCM must be
congured to allow DSP farming to proceed. The only way to do this with Cisco call-
processing agents and voice gateways is to use either the Cisco proprietary SCCP protocol
or MGCP. SCCP allows for more advanced conguration, and we will use this in our
conguration example. Figure 6.7 shows the communication process between the call-
processing agent and the voice gateway acting as a DSP farm.
DSP farm
S
C
C
P
c
o
m
m
u
n
ic
a
t
io
n
Switch
Fa 4/0
V
CUCM v8.0
FI GURE 6. 7 SCCP communication between CUCM and DSP farm
SCCP is used so that when the call-processing agent receives a request for transcoding,
conferencing, or MTP services, it can notify the DSP farm gateway and direct trafc away
from the CUCM and instead to the DSP farm.
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A DSP farm can function to support one service, such as transcoding, or a combination of
services depending on what is required on the network. To enable a DSP farm operation
on a voice gateway, rst navigate to the DSP card you wish to use and then issue the dsp
services dspfarm command, as shown here:
Router(config-voicecard)#dsp services dspfarm
Router(config-voicecard)#exit
Router(config)#
Creating DSP Profiles
Once DSP farm services are enabled, its time to create DSP proles, which are used to
allocate DSP resources and set their terms of usage. DSP proles are broken into the three
services that DSP farms can handle. A prole can be made for transcoding, conferencing,
and MTP. Also, each prole is given a unique identier, so it is possible to congure more
than one transcoding prole, for example, if you require different prole settings to be used.
Once you have chosen the prole type and given it a unique prole identier number,
you will be placed into config-dsp-farm-profile mode. Here you can congure the
unique prole rules, such as the codec types that can be used and the number of maximum
sessions the prole can handle at one time. Another required setting is to associate the
prole to SCCP for communication to the CUCM using the associate application SCCP
command. One last thing to remember is that the prole must be enabled by issuing no
shutdown. This will activate the prole, and the DSP resources required will be allocated.
For example, the following is transcoding prole 10, which species a number of codecs
that are allowed to be transcoded between one another. The maximum number of
simultaneous sessions is set to 5, and the prole is associated with SCCP:
Router(config)#dspfarm profile 10 transcode
Router(config-dspfarm-profile)#codec g711ulaw
Router(config-dspfarm-profile)#codec g711alaw
Router(config-dspfarm-profile)#codec g729ar8
Router(config-dspfarm-profile)#codec g729abr8
Router(config-dspfarm-profile)#codec g729r8
Router(config-dspfarm-profile)#maximum sessions 5
Router(config-dspfarm-profile)#associate application SCCP
Router(config-dspfarm-profile)#no shutdown
The maximum sessions command default is 0, so this number must be
changed before DSP resources are allocated for a profile.
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The conguration of transcoding and conferencing resources is identical except for the
prole commands that identify the two. When conguring MTP resources, the maximum
sessions command requires an additional keyword to be set. The hardware setting
species that DSP resources are used, while the software setting actually uses the router
processor and performs MTP in software. If you congure maximum sessions hardware,
keep in mind that MTP will only work for G.711 a-law and mu-law. If you have already
congured a hardware prole and realize you need to support other codecs, you must rst
remove the command by issuing a no maximum sessions command. Also remember that
since MTP software proles do not use DSP resources (they use the routers CPU instead),
you could exhaust your processing power if you terminate too many MTPs in software.
So if we wanted to congure a DSP prole 15 for MTP to terminate two calls using
hardware and two calls using software, we would congure something similar to the
following:
Router(config)#dspfarm profile 15 mtp
Router(config-dspfarm-profile)#codec g711ulaw
Router(config-dspfarm-profile)#maximum sessions hardware 2
Router(config-dspfarm-profile)#maximum sessions software 2
Router(config-dspfarm-profile)#associate application SCCP
Router(config-dspfarm-profile)#no shutdown
Configuring SCCP Communications
Now that we have our DSP farm enabled and our proles created, we can congure our
DSP farm router to communicate with our CUCM, as was shown in Figure 6.7.
To accomplish the SCCP conguration of the router, we rst must enable SCCP on the
router. To do this, we need to identify the IP address and physical port that the router will
use to communicate to our CUCM. In our case, the CUCM is at 10.10.10.100 and fa4/0
is the interface that will be used. When conguring the CUCM IP addresses, you can use
either the IP address or the domain name. You can congure the router to do a domain
lookup to retrieve the IP address using this method. Additionally, you should congure the
following settings:
identifier A unique identier to specify that this conguration is used between the
router interface and a specic CUCM.
priority If there is a redundant pair of CUCM call-processing agents, you can use
priority to create a primary connection and a backup connection in the case of a failure.
version Used to identify the version of software that the CUCM is running. As this book
goes to press, a CUCM running either version 7.0 or 8.0 software will use the version 7.0+
setting.
Next, it is time to identify the port that will be used to communicate to the CUCM.
The command we use to identify the port is sccp local. Finally, we can bring up SCCP on the
router by simply issuing the sccp command. Here is how to congure SCCP for our example:
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Router(config)#sccp ccm 10.10.10.100 identifier 1 priority 1 version 7.0+
Router(config)#sccp local FastEthernet 4/0
Router(config)#sccp
Router(config)#
We now have successfully identied our CUCM to the DSP farm gateway. The next
conguration step on the voice gateway is to create a DSP farm prole that associates the
farm with a CUCM call-processing agent group. The rst step is to use the sccp cucm group
command and give it a unique number. Once you create a group, you will be placed into config-
sccp-ccm mode, where you can congure settings that must match your CUCM conguration
on the call-processing agent. The bind interface command species the interface on which
the group will be active. Next, you need to congure two associate commands to specify the
CUCM the group pertains to and set the priority of the call-processing unit. The rst command
will be associate ccm identifier-number priority priority-number. The identifier-
number must match the identier that we congured previously in the sccp ccm 10.10.10.100
identifier 1 priority 1 version 7.0+ command. In our case, the identifier-number is
1. The priority-number species which CUCM is preferred if multiple units are congured for
redundancy. Up to four CUCM servers can be congured, where 1 is the most preferred and 4
is the least. The associate profile command sets the group to use the prole we previously
congured, which is 15. The nal group command is the register device-name command. The
device-name is a unique name that must be identical on both the voice gateway conguration
and the CUCM conguration. In our example, we use TXDSPFARM1 as our device-name. All
of the SCCP group conguration commands are shown here:
Router(config)#
Router(config)#sccp ccm group 1
Router(config-sccp-ccm)#bind interface FastEthernet4/0
Router(config-sccp-ccm)#associate ccm 1 priority 1
Router(config-sccp-ccm)#associate profile 15
Router(config-sccp-ccm)#register TXDSPFARM1
Configuring the CUCM
At this point our DSP farm on our voice gateway has been congured and points to one
or more CUCM call-processing agents. Now we must congure the CUCM to ofoad the
media resources we want the DSP farm to handle. While the conguration of a CUCM
is outside the scope of this book, you can nd where to congure the ofoading of media
services when using the CUCM version 8.0 GUI by rst logging into the Cisco Unied CM
Administration portion of the server. Next, navigate to the Media Resources tab and select
one of the three resources to ofoad:
Conference Bridge
Transcoder
Media Termination Point
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202 Chapter 6

Figure 6.8 shows the Media Resources tabs drop-down list choices.
FI GURE 6. 8 The CUCM Media Resources tab
From here, you can congure the CUCM to utilize the DSP farm located on your voice
gateway to handle these media resource services, which will free up CPU and memory
resources on the call-processing agent to handle more calls.
Match Up Those Names!
Sara and Mitch were setting up a new CUCM and DSP farm. Sara was responsible for
conguring the CUCM and Mitch was responsible for conguring the DSP farm router.
When the two nished their respective congurations, they discovered that the CUCM
and DSP farm would not cooperate when attempting to ofoad conference-calling duties.
Network connectivity was working properly, so the two administrators rechecked their
respective congurations. As it turns out, the CUCM conguration used the following as
the conference bridge name:
ConfDSP1
And the DSP farm IOS listed the following when they issued a show running-
configuration command:
register CnfDSP1
The conference bridge names must match exactly on the CUCM and DSP farm
congurations. The conguration was small, only one character off, but it was the
difference between a working conference bridge farm and a nonworking one.
c06.indd 202 9/21/11 11:19:41 AM
Voice Port and Dial-Peer Verification
Commands
In this section, you will get to know some useful show, test, and debug commands to verify
analog and digital voice port congurations and status. Some of the commands have been
used previously in this book, and others are new to you. The commands you should be
familiar with are these:
show voice port
show controller [t1|e1]
show voice dsp
test voice port
csim start
debug dialpeer
Lets take a quick look at each of these to see what information they can provide.
show voice port
The show voice port command is probably one of the most useful ways to verify
congurations and for troubleshooting. The command can be used on its own or with one
of several command options to display various port information. The show voice port
command on its own displays detailed information about all voice ports installed on the
voice gateway. You can also specify a specic port, as shown here, where we view the
information for a single port 0/0/0 (in this case, an FXS port):
Router#show voice port 0/0/0
Foreign Exchange Station 0/0/0 Slot is 0, Sub-unit is 0, Port is 0
Type of VoicePort is FXS
Operation State is DORMANT
Administrative State is UP
The Interface Down Failure Cause is 0
Alias is NULL
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to 0 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 16ms
Connection Mode is Normal
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204 Chapter 6

Connection Number is
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Analog Info Follows:
Region Tone is set for northamerica
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 25 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Hook Flash Duration Timing is set to 600 ms
This command is useful to verify operational information such as operation and
administration state, on/off-hook status, and ring status. In addition, you can verify
conguration settings such as signal type, cptone, and ring frequency.
If you want to get a quick glance at all of your voice interfaces with just a few details,
you can use the show voice port summary command, as shown here:
Router#show voice port summary
IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
============== == ============ ===== ==== ======== ======== ==
0/0/0 fxs-ls up dorm on-hook idle y
0/1/0 fxo-gs up dorm idle on-hook y
Router#
Here, you can see that this particular voice gateway has one four-port FXS and one
four-port FXO card installed. The -ls in the SIG-TYPE column tells us that the FXS
ports are congured with loop-start signaling, and the -gs tells us that the FXO ports are
c06.indd 204 9/21/11 11:19:42 AM
congured for ground-start signaling. The other columns tell us the status of our ports to
ensure that they are working properly.
show controller
The show controller command displays conguration and status information for digital
T1 or E1 trunks. The following example output shows the status and setup of T1 1/0:
Router# show controller T1 1/0
T1 1/0 is up.
Applique type is Channelized T1
Cablelength is long gain36 0db
No alarms detected.
alarm-trigger is not set
Framing is ESF, Line Code is B8ZS, Clock Source is Line.
Data in current interval (180 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
From the output you can see that the circuit is up and congured for ESF/B8ZS. In addition,
the clocking is set for Line, which will use clocking from the PSTN switch.
show voice dsp
For troubleshooting voice problems that may be related to DSP chips, you can us the show voice
dsp command to verify the codec conguration (medium, high, ex, or secure) and to view the
current state of all DSP resources. Here is an example of typical output using this command:
Router#show voice dsp
TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK COUNT
FLEX VOICE CARD 0
*DSP VOICE CHANNELS*
CURR STATE: (busy)inuse (b-out)busy out (bpend)busyout pending
LEGEND : (bad)bad (shut)shutdown (dpend)download pending
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206 Chapter 6

===== === == ========= ======= ===== ======= === == ========= == ====
*DSP SIGNALING CHANNELS*
===== === == ========= ======= ===== ======= === == ========= == ====
C5510 003 01 {flex} 9.4.7 alloc idle 0 0 2/0/0 02 0 81/0
C5510 003 02 {flex} 9.4.7 alloc idle 0 0 2/0/1 02 0 81/0
C5510 003 03 {flex} 9.4.7 alloc idle 0 0 2/0/2 06 0 80/0
C5510 003 04 {flex} 9.4.7 alloc idle 0 0 2/0/3 06 0 81/0
C5510 003 05 {flex} 9.4.7 alloc idle 0 0 2/0/4 10 0 80/0
C5510 003 06 {flex} 9.4.7 alloc idle 0 0 2/0/5 10 0 81/0
C5510 003 07 {flex} 9.4.7 alloc idle 0 0 2/0/6 14 0 80/0
C5510 003 08 {flex} 9.4.7 alloc idle 0 0 2/0/7 14 0 81/0
C5510 003 09 {flex} 9.4.7 alloc idle 0 0 2/0/8 18 0 12/1
C5510 003 10 {flex} 9.4.7 alloc idle 0 0 2/0/9 18 0 12/1
C5510 003 11 {flex} 9.4.7 alloc idle 0 0 2/0/10 22 0 12/1
C5510 003 12 {flex} 9.4.7 alloc idle 0 0 2/0/11 22 0 12/1
C5510 003 13 {flex} 9.4.7 alloc idle 0 0 2/0/12 26 0 12/1
C5510 003 14 {flex} 9.4.7 alloc idle 0 0 2/0/13 26 0 12/1
C5510 003 15 {flex} 9.4.7 alloc idle 0 0 2/0/14 30 0 12/1
C5510 003 16 {flex} 9.4.7 alloc idle 0 0 2/0/15 30 0 12/1
END OF FLEX VOICE CARD 0-
Router#
Interesting information that you can see in this example includes the DSP type, whether
the resource has been allocated, and the current state of each resource.
test voice port
If you are having problems with specic voice ports acting erratically, you can use the test
voice port commands to run specic tests to verify proper operation of your congured
voice interfaces. The commands entered dont actually run tests, but instead they congure
voice ports or DS0 groups into various testing states. The ve possible testing states that
can be congured are shown in Table 6.1.
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TABLE 6.1 Testing states for the test voice port command
State Description Possible Values
detector How the port detects ground on
the voice port or DS0 group. This
depends on the signaling type
used on the port.
[m-lead, loop, battery-reversal,
ring, tip-ground, ring-ground,
ring-trip] [on, off, disable]
inject-tone Places a local or network testing
tone on the voice port or DS0
group.
[local, network] [1000hz, 2000hz,
200hz, 3000hz, 300hz, 3200hz, 3400hz
500hz, quiet, disable, sweep]
loopback Places the port or DS0 group
into loopback testing mode so
either the local or remote end will
receive the test tone coming back
to verify end-to-end operation.
local, network, disable
relay Enables and sets the relay for
testing purposes.
[e-lead, loop, ring-ground,
battery-reversal, power-denial,
ring, tip-ground] [on, off, disable]
switch Places the port or DS0 group into
fax mode for specifically testing
fax transmissions.
fax, disable
As an example, let us assume that you want to generate a sweep of inject tones on FXO
port 0/1/0. The inject-tone test is used for a variety of reasons, including determining ideal
impedance settings. As you have learned, impedance on a call can introduce annoying
hissing, clicking, and volume problems because of timing issues. Analog voice ports can
adjust impedance settings to help combat audio problems using the impedance command
while in config-voiceport mode. We can set our voice port at different impedance levels
and test them using a sweep of tones to determine echo return loss (ERL) levels. ERL
measures the ratio between the power level of the transmitted signal and the power level
detected in the echo signal. The lower the ERL dB levels are, the better quality your voice
will be. So lets set our FXO 0/1/0 to use an impedance setting of 600r and run our inject-
tone sweep. Also note that we must disable echo cancellation and issue a shutdown and no
shutdown on the port prior to running these tests:
Router(config)#voice port 0/1/0
Router(config-voiceport)#no echo-cancel enable
Router(config-voiceport)#impedance 600r
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208 Chapter 6

Router#
Router#test voice port 0/1/0 inject-tone local sweep 200 0 0
Freq (hz), ERL (dB), TX Power (dBm), RX Power (dBm)
104 25 -8 -33
304 19 -7 -26
504 17 -8 -25
704 19 -8 -27
904 19 -8 -27
1104 20 -8 -28
1304 20 -8 -28
1504 21 -8 -29
1704 22 -8 -30
1904 22 -8 -30
2104 22 -8 -30
2304 22 -8 -30
2504 22 -8 -30
2704 22 -8 -30
2904 22 -8 -30
3104 22 -8 -30
3304 22 -8 -30
3404 22 -8 -30
Router#
And now lets change the impedance to 900r and run another test:
Router(config)#voice port 0/1/0
Router(config-voiceport)#impedance 900r
Router#test voice port 0/1/0 inject-tone local sweep 200 0 0
Freq (hz), ERL (dB), TX Power (dBm), RX Power (dBm)
104 26 -7 -33
304 20 -7 -27
504 17 -8 -25
704 20 -7 -27
904 20 -7 -27
1104 20 -7 -27
1304 20 -8 -28
1504 20 -8 -28
c06.indd 208 9/21/11 11:19:44 AM
1704 20 -8 -28
1904 20 -8 -28
2104 20 -8 -28
2304 20 -8 -28
2504 20 -8 -28
2704 20 -8 -28
2904 20 -8 -28
3104 19 -8 -27
3304 19 -8 -27
3404 19 -8 -27
Router#
The average ERL when impedance is set to 600r is 21.11, and it is 22.055 when impedance
is set at 900r. Therefore, the better choice for this particular FXO connection (lower ERL) is
600r. Keep in mind that this type of testing is commonly done with TAC support assisting you
in the process as well as suggesting optimal adjustments that you should make.
csim start
You can simulate an outbound telephone call directly from the voice gateway by issuing the
csim start extension enable mode command. This command is useful for testing dial-peer
matching, testing translation rules, and to verify that a phone can properly make calls. For
example, lets test a phone number using this command:
CME#csim start 4488
csim: called number = 4488, loop count = 1 ping count = 0
csim err:csim_do_test Error peer not found
The result of the test call is an error stating that no peer was found. If you know that
this call should have been possible, then you need to look at the dial peer and adjust it so it
matches the outgoing dialed number.
debug dialpeer
To review dial-peer matching in real time, you can use the debug dialpeer command. This
command is useful when youre connected to the console for troubleshooting purposes.
It can also be used when verifying the number that you will pass onto the next hop in the
case where you used translation rules to modify the original number. Here is an example of
the debug dialpeer command in use:
Router# debug dialpeer
Router#
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210 Chapter 6

09:22:18: Inside dpMatchCore:
09:22:18: destination pattn: 5552003333 expanded string: 5552003333
09:22:18:MatchNextPeer:Peer 11 matched
From the output, you can see that this is an outbound dial peer that was matched
against dial peer 11. The destination pattern is the pattern that was matched correctly,
and the expanded string displays the number we are forwarding. Always keep in mind
that when you use a debug command such as this one, you should disable it after you have
nished looking at it. In our case, we would issue a no debug dialpeer to prevent any
unwanted CPU processing on the router.
Summary
In this chapter we covered how to congure analog and digital voice ports in a wide range
of scenario settings including PLAR, DID, CAMA, CCS, and ISDN. In addition, we went
through the process of setting up a DSP farm to ofoad transcoding, conferencing, and
MTP services from the CUCM using SCCP as the communication signaling protocol.
Finally, we went through several commands that can be used on a voice gateway to test
analog/digital ports, DSP resources, and dial peers.
All of these tasks will come in handy when we begin conguring the various signaling
protocols in Chapter 7, Conguring Voice Gateway Signaling Protocols. By the end of
that chapter, you will have all the tools necessary to congure, monitor, and troubleshoot
a voice gateway not only to communicate with IP and analog phones internally but also to
connect to the PSTN using analog and digital circuits.
Exam Essentials
Know how to configure analog FXS, FXO, and E&M ports. FXS ports connect to analog
endpoints, FXO ports connect to the PSTN, and E&M ports interconnect two PBX systems.
Understand and know how to configure PLAR. PLAR is an autodialing mechanism that
is used to associate a port with a single destination.
Understand and know how to configure DID. DID is a feature a PSTN uses to strip off
digits prior to sending them to a private voice gateway.
Know the difference between one-stage and two-stage dialing. With a one-stage dialing
setup, the call is not terminated and does not present the caller with a second dial tone. With
two-stage dialing, a caller dials digits, which are accepted by a voice gateway, and the call
terminates at a second hop along the connection, where a second dial tone is given. The caller
must then enter a second series of digits to complete the intended call.
Understand CAMA and know how to configure it. CAMA is often used in North
America for E911 dialing. A separate CAMA port must be congured strictly for outbound
calling for emergency services.
c06.indd 210 9/21/11 11:19:44 AM
Written Lab 6.1 211
Know how to configure digital ports including CAS and ISDN CCS circuits. T1/E1 CAS
circuit DS0 timeslots can be congured to use different signaling types. CAS signaling is
in-band signaling. T1/E1 PRI and ISDN BRI circuits use out-of-band signaling, and their
respective timeslots must be congured to transport Q.921 and Q.931 signaling.
Know how to configure a DSP farm between a voice gateway and CUCM. DSP farms
can be congured on voice gateways to ofoad services that are handled by DSP hardware
chips.
Understand how to view and test voice port configurations on a voice
gateway. Commands such as show voice port, show controller, and test voice port
can be used for conguration verication and testing purposes.
Written Lab 6.1
1. What is the config-voiceport command used to congure the extension 555-1234 for
caller ID services?
2. While in config-voiceport mode, you want to congure PLAR on the port that
automatically forwards calls to extension 4875 as soon as the phone goes off-hook.
What command performs this function?
3. You are conguring a dial peer for an FXS port that uses DID. What command is used
to enable this functionality while in config-dial-peer mode?
4. You are conguring a dial peer for a CAMA port and have destination-pattern 9911
congured. What command is required to forward only 911 to the PSAP?
5. You are conguring a T1 CAS and want to congure group 0 so that the rst 12
timeslots use E&M immediate-start signaling. While in config-controller mode,
what command will you enter?
6. A T1 PRI that you are conguring will only utilize voice services. You navigate to
config-if mode and enter what command?
7. What command do you use to enable a DSP card for DSP farm functionality while in
config-voicecard mode?
8. You are in the middle of conguring a DSP prole for MTP services and are in config-
dspfarm-profile mode. What command is used to set the maximum number of
hardware sessions to 5?
9. What show command can be used to get a quick glance at all voice port interfaces
installed on a voice gateway?
10. You wish to simulate a phone call to extension 5555 on your voice gateway. What
command can accomplish this?
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212 Chapter 6

Hands-On Labs
To complete the labs in this section, you need a router with a voice-capable IOS, T1 PRI
interface, and one FXS port to be used as a CAMA port. Each lab in this section builds on
the previous one and will follow the logical voice gateway design shown in Figure 6.9.
FI GURE 6. 9 Voice gateway lab diagram
V
PSTN
PSAP
CAMA
0/0/1
S0/1/0
Outbound
NANP calls
911 and
9911
E911
operators
Internal voice
network
Here is a list of the labs in this chapter:
Lab 6.1: Conguring a T1 PRI
Lab 6.2: Conguring a CAMA Port for E911 Services
Lab 6.3: Conguring Outbound Dial Peer to the PSTN
Lab 6.4: Conguring Outbound Dial Peer to the PSAP
Hands-On Lab 6.1: Configuring a T1 PRI
In this lab, were going to congure a voice gateway that has a single ISDN PRI interface out
to the PSTN. The voice gateway has been partially congured. The task here is to congure the
T1 logical and physical port sections according to the PSTN requirements found in Table 6.2.
TABLE 6. 2 T1 PRI settings and PSTN requirements for Hands-On Lab 6.1
T1 PRI Settings PSTN Requirements
ISDN switch type Primary-NI
Framing Extended Superframe
Linecoding B8ZS
Clock source From the PSTN
1. Log into your voice gateway and go into conguration mode by typing enable and then
configure terminal.
2. Congure the ISDN switch type by typing isdn switch-type primary-ni.
c06.indd 212 9/21/11 11:19:45 AM
Hands-On Labs 213
3. Enter into config-controller mode by typing controller t1 0/1/0.
4. Congure a new PRI group specifying all T1 timeslots by typing pri-group timeslots 124.
5. Congure the T1 PRI to for ESF framing by typing framing esf.
6. Congure the T1 PRI to for B8ZS linecoding by typing linecode b8zs.
7. Congure the T1 PRI to use clocking from the PSTN by typing clock source line.
8. Exit config-controller mode by typing exit.
9. Enter into the T1 config-if mode by typing interface serial 0/1/0:23.
10. Congure the T1 PRI to direct all T1 voice channels to the DSP by typing isdn
incoming-voice voice.
11. Exit config-if mode by typing end.
Hands-On Lab 6.2: Configuring a CAMA Port for E911
Services
In this next lab, we will focus on conguring FXO port 0/0/1 for our CAMA connection
to the PSAP. Emergency services in our area want us to send a numbering plan digit (NPD)
plus the three-digit CO code and four-digit subscriber code. ANI mappings were given to
us as shown in Table 6.3.
TABLE 6. 3 Area codes and numbering plan digits
Area Code Numbering Plan Digit
312 0
773 1
630 2
850 3
configure terminal.
2. Enter into config-voiceport mode by typing port 0/0/1.
3. Congure the correct signaling type by typing signal cama KP-NPD-NXX-XXX-ST.
4. Reset the port by typing shutdown and then no shutdown.
5. Congure NPD to area code mappings by typing
ani mapping 0 312
ani mapping 1 773
ani mapping 2 630
ani mapping 3 850
6. Exit config-voiceport mode by typing end.
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214 Chapter 6

Dial Peer to the PSTN
An outbound dial peer needs to be congured for NANP calls that will point out of our T1
PRI. A dial peer needs to be created to trigger on the number 9. It will then collect a
10-digit E.164 string and forward those 10 digits onto the PSTN.
configure terminal.
2. Create a new POTS dial peer (we will use 9) by typing dial-peer voice 9 pots. This
will take you to config-dial-peer mode.
3. Congure a dial-string mapping to match 9 plus a 10-digit number according to
NANP guidelines by typing destination-pattern 9[28]. . . . . . . . .
4. Congure the dial peer to forward the last 10 digits to the PSTN by typing
forward-digits 10.
5. Congure the dial peer to send matched calls out our T1 port by typing port 0/1/0:23.
6. Exit config-dial-peer mode by typing end.
Dial Peer to the PSAP
In our nal hands-on lab for this chapter, we will congure outbound dial peers for E911
calling to the PSAP. Because we have internal callers dial 9 for an outbound call, we will
congure emergency calls to trigger on both 911 and 9911 so callers who dial the 9 trigger
prior to 911 will properly be connected. We also need to send 911 to the PSAP using
translation patterns.
configure terminal.
2. Create a new POTS dial peer (we will use 911) by typing dial-peer voice 911 pots.
This will take you to config-dial-peer mode.
3. Congure a dial-string mapping to match 911 by typing destination-pattern 911.
4. Congure the dial peer to forward all three digits to the PSAP by typing forward-digits all.
5. Congure the dial peer to send matched calls out our CAMA port by typing port 0/0/1.
6. Create a new POTS dial peer (we will use 9911) by typing dial-peer voice 9911 pots.
This will take you to config-dial-peer mode.
7. Congure a dial-string mapping to match 9911 by typing destination-pattern 9911.
8. Congure the dial peer to strip off the rst 9 so it only sends the last three digits to the
PSAP by typing forward-digits 3.
9. Congure the dial peer to send matched calls out our CAMA port by typing port 0/0/1.
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Review Questions
1. How can DNIS be used on FXO ports?
A. Inbound only.
B. DNIS cannot be used in either direction on FXO ports.
C. Outbound only.
D. Both inbound and outbound.
2. What is the name for an autodialing feature that is used to associate a port with a single
destination?
A. PSAP
B. PLAR
C. CAMA
D. OPX
3. You are configuring a PLAR on an FXO port that is connected to the PSTN. When a user
calls from the PSTN, you want to automatically forward that call to extension 455. Which
of the following commands will accomplish this task?
A. plar opx 455
B. connection opx plar 455
C. opx plar 455
D. connection plar opx 455
4. You have configured an FXS/DID port and associated dial peer as shown in the following
output. What additional dial-peer command is required if you want to accept only four
digits from the PSTN?
Router(config-voiceport)#signal did wink-start
Router(config-dial-peer)#incoming called-number . . . .
A. prefix 4
B. forward digits 4
C. direct-inward-dial
D. did
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216 Chapter 6

5. What is the name of the dialing process where a caller hears a dial tone, enters phone
digits, receives another dial tone, and must enter a second set of digits to reach the intended
destination?
A. Direct inward dial (DID)
B. One-stage dialing
C. Private Line Automatic Ringdown (PLAR)
D. Two-stage dialing
6. You are configuring an FXO port as CAMA interface to be used for E911 services. What alert
will the voice gateway send to the command line after configuring the CAMA signal type?
A. A notice to verify that the signaling type configured is what the PSAP is expecting
B. A notice to verify that dial peers for 911 and 9911 should be configured on the voice
gateway
C. A notice that the interface must be administratively disabled and reenabled
D. A notice that only 911 should be sent to the PSAP
7. How are E911 calls routed to the PSAP?
A. By DNIS
B. By destination address
C. By ANI
D. By area code
8. In situations where CAMA is required, where should internal calls to 911 be routed?
A. A CUCM
B. The PSAP
C. The PSTN
D. The SRST
9. Your office requires 12 analog FXO ports to the PSTN with ground-start signaling and 12
E&M ports to the PSTN with immediate-start signaling. Which of the following is best
suited to meet your needs?
A. T1 PRI
B. E1 PRI
C. T1 CAS
D. ISDN BRI
10. What channel/timeslot is used for signaling on a T1 PRI?
A. Channel 23 and timeslot 24.
B. Timeslot 23 and channel 24.
C. Timeslot 24 and channel 24.
D. Channel 23 and timeslot 23.
E. PRI uses in-band signaling.
c06.indd 216 9/21/11 11:19:47 AM
11. You are reviewing a T1 PRI configuration and are looking at interface serial 0/0:23. You
see the following command:
isdn incoming-voice voice
What is the purpose of this command?
A. To ensure that Q.931 signaling is sent on T1 0/0 channel 23
B. To ensure that Q.921 signaling is sent on T1 0/0 channel 23
C. To ensure that all channels are processed by DSPs
D. To ensure that Q.921 and Q.931 signaling is sent on T1 0/0 channel 23
12. You are asked to create a dial peer to send all NANP calls outbound on T1 PRI 1/0. The
following commands are already configured:
Router(config-dial-peer)#destination-pattern 9[28]. . . . . . . . .
Which of the following commands will properly complete the configuration?
A. port 1/0
B. port 1/0:24
C. port 1/0:1
D. port 1/0:23
13. Which of the following can be used to offload MTP?
A. SCCP
B. DSP
C. SRST
D. H.323
14. Which of the following commands is used to enable a DSP card for DSP farming services?
A. Router(config-voicecard)#dsp services dspfarm
B. Router(config-dial-peer)#service dspfarm
C. Router(config-dial-peer)#dsp services dspfarm
D. Router(config-voicecard)#service dspfarm
15. You have created a DSP profile and given it a unique identifier number. At what point in the
DSP farm-configuration process will the DSP profile identifier number be required?
A. While in config-dsp-farm-profile mode
B. When identifying the sccp ccm
C. When in config-sccp-ccm mode
D. When configuring the CUCM
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218 Chapter 6

16. Reviewing a DSP farm profile, you see the following entry:
maximum sessions hardware 4
What type of profile is this?
A. Translation
B. MTP
C. Conferencing
D. Transcoding
17. Given the following SCCP CCM group configuration, what must be configured identically
on the CUCM itself?
sccp ccm group 1
bind interface FastEthernet4/0
associate ccm 1 priority 1
associate profile 15
register TXDSPFARM1
A. TXDSPFARM1
B. Profile 15
C. Group 1
D. Priority 1
18. You issue the following voice gateway show command:
Router#show voice port 0/0/0
Foreign Exchange Station 0/0/0 Slot is 0, Sub-unit is 0, Port is 0
Type of VoicePort is FXS
Operation State is DORMANT
Administrative State is UP
The Interface Down Failure Cause is 0
Alias is NULL
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to 0 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 16ms
Connection Mode is Normal
Connection Number is
Initial Time Out is set to 10 s
c06.indd 218 9/21/11 11:19:48 AM
Interdigit Time Out is set to 10 s
Analog Info Follows:
Region Tone is set for northamerica
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 25 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Hook Flash Duration Timing is set to 600 ms
Given this information, which of the following statements is true?
A. The phone is currently not in use.
B. The phone is configured for use in France.
C. This is an E&M port.
D. Signaling is configured for ground start.
19. You issue a show voice dsp command on your voice gateway. Which of the following
might you see under the CODEC section of the output?
A. g711ulaw
B. g729r8
C. flex
D. encoded
20. What does the csim start 5551234 command do when run on a voice gateway?
A. It simulates a conference call originating from any source endpoint to a destination of
555-1234 to test DSP resources.
B. It simulates a conference call originating from the voice gateway to a destination of
555-1234 to test DSP resources.
C. It simulates a call originating from any source endpoint to a destination of 555-1234.
D. It simulates a call originating from the voice gateway to a destination of 555-1234.
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220 Chapter 6

1. C. FXO ports can only send DNIS information outbound toward the PSTN.
2. B. PLAR is a feature that can be congured on a voice port to automatically dial a specic
destination as soon as the phone goes off-hook.
3. D. The proper syntax for this command is connection plar opx extension.
4. C. The direct-inward-dial command is required to accept DIDs from the PSTN.
5. D. Two-stage dialing requires that the voice network receive two different sets of numbers
to reach the nal destination. The rst digits that are entered are processed and forwarded
to a second gateway that presents the caller a second dial tone at which to enter a second
set of digits.
6. C. When conguring an FXO port to be used as a CAMA interface for E911 calling, the
port must go through a shutdown and no shutdown; otherwise, outbound calling will not
function.
7. C. The calling partys number (ANI) is used to route calls to the PSTN. By using this
method, emergency services can use the ANI number and match it with their internal
database to nd the location of the calling party. That location is then used to route the
call to the nearest PSAP.
8. B. When a business is required to use CAMA interfaces for E911, dial peers should route
911 calls to the PSAP.
9. C. A T1 CAS can be logically broken into DS0 groups. These groups can be congured to
utilize different analog FXS, FXO, and E&M signaling types.
10. A. When referring to T1 signaling channels and timeslots, you congure Q.921 and Q.931
signaling on channel 23 and timeslot 24.
11. C. The isdn incoming-voice voice command tells the router that all channels on
T1 0/0 are to be used for voice and not data. All channels will then be directed to DSP
resources for transcoding.
12. D. When conguring an outbound dial peer for a T1 PRI, you must specify the logical T1
channel that is used. In the case of a T1, that channel is always 23. Therefore, when
conguring port information, you specify the T1 slot/port followed by a colon (:) and then
the channel number used for ISDN signaling.
13. B. A DSP chip can be congured to be used as a DSP farm to ofoad CUCM tasks such as
transcoding, conferencing, and MTP.
14. A. To enable a DSP card for DSP farming services, you must be connected to that
particular voice card (in config-voicecard mode) and enter the dsp services dspfarm
command.
c06.indd 220 9/21/11 11:19:49 AM
15. C. The DSP prole number is used while in config-sccp-ccm mode to identify the prole
needed by using the associate profile 15 command.
16. B. The DSP prole that includes either the hardware or software keyword is used only
when conguring MTP DSP farm proles.
17. A. The register TXDSPFARM1 command species the CUCM media resources device
name, which must be identical on the CUCM and voice gateway congurations.
18. A. From the show command output, you can see that the phone hook status is on-hook.
19. C. When viewing DSP resources, the CODEC species how the DSP resource is congured.
Possible options include medium, high, flex, and secure.
20. D. The csim start 5551234 command can simulate a voice call that originates from the
voice gateway to any dial string destination specied.
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222 Chapter 6

1. station-id number 5551234
2. connection plar 4875
3. direct-inward-dial
4. forward-digits 3 (prefix 911 is acceptable as well)
5. ds0-group 0 timeslots 112 type e&m-immediate-start
6. isdn incoming-voice voice
7. dsp services dspfarm
8. maximum sessions hardware 5
9. show voice port summary
10. csim start 5555
c06.indd 222 9/21/11 11:19:50 AM
Configuring Voice
Gateway Signaling
Protocols
CHAPTER:

a VoIP call.
Describe H.323.
Describe SIP.
Describe MGCP.
Identify the appropriate gateway signaling protocol for a

given scenario.
Chapter
7
c07.indd 223 21/09/11 11:23 AM
One of the most extensively covered topics on the CVOICE
exam deals with voice gateway signaling protocolswhen
to use them and how to congure them. In Chapter 3, VoIP
Operation and Protocols, you got a brief overview of the four signaling protocols used on
Cisco networks: H.323, SIP, MGCP, and Ciscos proprietary SCCP. The latter is commonly
used on Cisco networks within a LAN environment, while the other protocols are used for
signaling between networks.
In this chapter, we will dive in deeper to cover the ins and outs of H.323, SIP, and
MGCP, which are all commonly used and congured on Cisco IOS gateways that
attach to the PSTN. Well rst explore some more details about each signaling protocol,
including how to congure them and make common adjustments and how to use IOS show
commands to verify proper conguration and for troubleshooting.
Configuring H.323
By default, Cisco voice gateway dial peers are congured to operate using the H.323
signaling protocol. As you learned in Chapter 3, H.323 is a peer-to-peer protocol, which
means that the voice gateway must be congured with dial-peer information so the voice
gateway knows where to route various VoIP calls that come into it. When connecting to
the PSTN via an ISDN connection such as a T1 PRI, the H.323 signaling protocol is used
between the voice gateway and the CUCM, while ISDN Q.921 and Q.931 signaling is used
between the voice gateway and the PSTN, as shown in Figure 7.1.
FI GURE 7.1 H.323 and ISDN signaling
PSTN
Internal IP
network
H.323 Q.921
Q.931
Voice
gateway
V
CUCM
M
c07.indd 224 21/09/11 11:23 AM
From an operational standpoint, H.323 has two modes for call initiation, called slow
start and fast start.
H.323 slow start initiation mode processes a call through the following signaling stages:

Call setup

Call proceeding

Alerting

Connect

H.245 negotiation
This process is illustrated in Figure 7.2.
FI GURE 7. 2 The H.323 slow start process
IP
network
H.323 signaling
1. Call setup
2. Call proceeding
3. Alerting
4. Connect
5. H.245 negotiations
V V
Calling
party
Called
party
As soon as the connect signaling step is complete, H.323 proceeds to perform H.245
(voice control channel) negotiation, and the call is then in progress. The H.245 channel is
then responsible for the following:

Exchanging capabilities information such as encryption, ow control, and jitter
management

Opening and closing media stream channels, which transport voice and video

Determining the master or responder endpoints

Handling modication requests to change the mode or capability of open media
streams
So as you can see, the faster we get to H.245, the faster our media channels can be
opened between two endpoints.
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226 Chapter 7
Configuring Voice Gateway Signaling Protocols

With H.323 fast start, on the other hand, the H.245 negotiation process kicks off during
the call-setup stage and does not wait for the other three signaling stages before allocating
resources for the voice transport channel. This process is illustrated in Figure 7.3.
FI GURE 7. 3 The H.323 fast start process
IP
network
H.323 signaling
1. Call setup and H.245 negotiations
2. Call proceeding
3. Alerting
4. Connect
V V
Calling
party
Called
party
As you can imagine, the H.323 fast start method is more efcient than slow start. It
should be noted that fast start is available only in equipment that supports H.323 version 2
or higher, so some legacy equipment may not be compatible. H.323 fast start is the default
mode on newer Cisco voice gateways.
One H.323 feature that can utilize fast start is called H.323 Early Media. This feature
can be used when two voice gateways connect to each other using H.323 fast start. When
the Early Media feature is also used between gateways, it allows the gateways to open
up media transport channels prior to H.225 negotiation and thus prior to the call being
accepted between the two parties. These early channels can be used for streaming of media
such as broadcast announcements or music on hold (MOH). Figure 7.4 shows the Early
Media feature in action.
FI GURE 7. 4 The H.323 fast start process with Early Media
IP
network
H.323 signaling
1. Call setup and H.245 negotiations
2. Call proceeding
4. Alerting
5. Connect
V V
Calling
party
Called
party
3. RTP stream for early media
c07.indd 226 21/09/11 11:23 AM
Configuring an H.323 Gateway
As stated previously, H.323 is the default signaling protocol for Cisco voice gateways.
Dial peers are required so that when a call comes in, the voice gateway can match the call
based on inbound dial peers and send it to the appropriate destination using outbound dial
peers. To create a simple H.323 dial-peer connection to a remote H.323 gateway, it is just a
matter of performing the following tasks:
1. Enter the dial-peer voice command with a unique identier number followed by
the voip option to specify that this is a Voice over IP dial peer as opposed to a POTS
dial peer.
2. Congure a destination pattern to identify how to reach remote phones on the other
side of an IP connection.
3. Congure a destination IP address to point to the next-hop voice gateway along the path
to the destination phones using the session target ipv4: command followed by the IP
address of the remote voice gateway interface that your voice gateway is aware of.
Why Is My VoIP Gateway Not Processing VoIP?
Did you congure your VoIP gateway but nd that it doesnt seem to be operating? It
is possible that voice services have been disabled on your router. To verify this, you
can issue a show gateway command to see if the gateway signaling protocol you are
running is enabled or shut down. You can enable the VoIP service on IOS routers with
voice software by issuing the voice service voip global conguration command.
This brings you to conf-voi-serv mode. You can then issue shutdown to disable the
service or no shutdown to bring it back up. One additional option is to use shutdown
forced, which brings the VoIP service down regardless of any calls that may be using
VoIP service on the router. Without the forced keyword, the router will stop accepting
additional VoIP connections but also wait for any currently active calls to complete.
VoIP is enabled by default, so typically it is not necessary to enter this command when
conguring VoIP unless someone has explicitly shut down the service.
To show you how dial peers are created between two H.323 gateways, we will use
Figure 7.5 as our example network.
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228 Chapter 7

We want to route all calls between the two sites across the IP WAN using H.323
signaling. We will start by conguring the New York router using 4000 as our locally
signicant dial-peer identier, as follows:
NewYork(config)#dial-peer voice 4000 voip
NewYork(config-dial-peer)#destination-pattern 4...
NewYork(config-dial-peer)#session target ipv4:192.168.2.1
NewYork(config-dial-peer)#
Next, lets congure the Boston router so we can properly reach phones in both
directions using dial peer 3000:
Boston(config)#dial-peer voice 3000 voip
Boston(config-dial-peer)#destination-pattern 3...
Boston(config-dial-peer)#session target ipv4:192.168.1.1
Boston(config-dial-peer)#
This example conguration setup shows you how to congure H.323 signaling using the
default settings. But sometimes you will want to modify some of the defaults, depending on
certain network differences that you may come across. These include settings such as:

Fast or slow start connections

Codec preference

Session transport mode

Adjusting H.225 settings

Adjusting H.225 timers

Binding a virtual H.323 gateway address
The next few sections show how to modify these H.323 gateway signaling settings.
Configuring H.323 Fast or Slow Start Connections
If you need to connect to a legacy H.323 that supports only slow start setups, you need to
use the call start config-serv-h323 command to specify that the voice gateway uses
FI GURE 7. 5 An example H.323 network
IP
WAN
New York
H.323 gateway
Boston
H.323 gateway
V V
Extensions
3000-3999
Extensions
4000-4999
192.168.2.1/24 192.168.1.1/24
c07.indd 228 21/09/11 11:23 AM
slow start signaling, so it is compatible with equipment at the other end. To get to this
mode, you must rst use the voice service voip command. There are actually several
voice service options for POTS, VoIP, and others, as shown here:
Router(config)# voice service ?
pots Telephony
voatm Voice over ATM
vofr Voice over Frame Relay
voip Voice over IP
Router(config)# voice service
Once you choose voice service voip, the IOS will place you into conf-voice-serv
mode. You then must specify h323 as the signaling protocol you wish to modify. The
following example shows the steps required to force an H.323 gateway to use slow
start globally:
Router(config)# voice service voip
Router(conf-voi-serv)# h323
Router(conf-serv-h323)#call start slow
Router(conf-serv-h323)#end
Router#
Alternatively, you can congure H.323 slow start on a specic dial peer instead of
globally on the voice gateway. To accomplish this, you must rst create an H.323 voice
class that species slow start signaling. The voice class can then be applied to any dial
peer that you choose. The following example shows how to congure an H.323 voice class
(labeled 1) for slow start connections. The voice class is then applied to dial peer 10:
Router(config)#voice class h323 1
Router(config-class)#call start slow
Router(config-class)#exit
Router(config-dial-peer)#voice-class h323 1
Router#
Configuring Codec Preference
You can use signaling protocols to specify the preferred codec and other codec-specic
parameters to be used in the H.323 conguration as well. To do so, you create codec voice
classes and specify your preferences. To set preferences for codecs, you can use the codec
c07.indd 229 21/09/11 11:23 AM
230 Chapter 7

preference command followed by a preference number and the codec in question. A lower
preference number is preferred. Optionally, you can also set the following codec settings:
bytes The voice packet payload size specied in bytes
fixed-bytes Whether the bytes specied are nonnegotiable between the voice gateways
transparent To specify that the codec capabilities be passed transparently to the remote
voice gateway
As an example, we will congure preferences (voice class 30) for the following three
codecs and specic settings:

Highest preference: g711ulaw with a 160-byte payload

Middle preference: g726r32 with an 80-byte xed payload

Lowest preference: g729br8
Here is how to congure the codec preferences on a voice gateway:
Router(config)#voice class codec 30
Router(config-class)#codec preference 1 g711ulaw bytes 160
Router(config-class)#codec preference 2 g726r32 bytes 80 fixed-bytes
Router(config-class)#codec preference 3 g729br8
Router(config-class)#end
Router#
Once the codec preferences are specied, they can be applied to VoIP dial peers as
shown in this example, where we apply voice class 30 to dial peer 15:
Router(config-dial-peer)#voice-class codec 30
Router#
The voice class codec commands are not H.323 specific but can be used
to set codec preferences and settings for any voice gateway signaling
protocol, including H.323, MGCP, and SIP.
Keep in mind that the codec preferences should be congured on both sides to ensure
proper codec selection.
The codec selection will be used only when the local gateway is used to initiate
signaling. If a call is initiated on a different H.323 gateway inbound to the local router,
the preferences will be dictated by the remote router. In addition, if no codec preferences
c07.indd 230 21/09/11 11:23 AM
are specied for the remote inbound dial peers, the default codec of G.729r8 with
20-byte voice payloads will be used. If no inbound dial peer is specied on the remote
voice gateway, dial peer 0 is triggered, which supports all voice codecs.
Configuring the H.323 Session Transport Mode
A third H.323 setting that can be modied is the H.323 session transport mode. By
default, H.323 is transported over TCP. You can change the transport layer protocol for
H.323 dial-peer messages to UDP by issuing the session transport udp command. This
command is run while in config-serv-h323 mode.
You might choose to go with UDP for transport instead of TCP, because UDP has less
overhead. If you are seeking to reduce H.323 signaling bandwidth utilization, you can
congure your voice gateway for UDP, as shown in the following example:
Router(conf-voi-serv)#h323
Router(conf-serv-h323)#session transport udp
Router#
It is not recommended that you adjust the transport method from the
default TCP unless necessary. Changing from TCP to UDP not only
prevents concurrent H.323 sessions on a voice gateway but also limits the
number of adjustments that can be made using the H.225 TCP commands,
described next.
Modifying H.225 Settings
There are a couple of H.225 settings that sometimes need to be adjusted for an H.323
gateway depending on the network you are residing on. The rst H.225 command we will
look at can adjust the maximum number of concurrent calls on an H.225 TCP connection.
The command is session transport tcp calls-per-connection followed by a numerical
value for maximum concurrent calls. The default maximum is 15, and you can set the
number between 1 and 9999. We will adjust this value to support 20 concurrent calls in the
following example:
Router(conf-serv-h323)#session transport tcp calls-per-connecton 20
Router#
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232 Chapter 7

Another useful H.225 command will let you adjust the number of seconds that an H.225
idle connection will remain intact before tearing it down. This timer is modied using the
h225 timeout tcp call-idle command, followed by either value and then the number
of seconds you wish the timer to use (the range is 0 to 1440 seconds) or the keyword never
to keep the connection established indenitely. The default timer is set to 10 seconds. The
following example shows how to adjust the H225 TCP call-idle timer to 5 seconds:
Router(conf-serv-h323)#h225 timeout tcp call-idle value 5
Router#
Adjusting H.225 Timers
There are two H.225 timers that can be adjusted in conguring an H.323 gateway to help
the H.225 connection process according to the physical transport medium and utilization.
Over-utilized links may not function well with the default timer settings. In some
situations, an increase in these timers may be benecial. H.225 timers are adjusted by rst
using the voice class h323 command followed by a unique number tag. The IOS will then
place you in config-class mode, where you can adjust various H.323 settings, including
H.225 timers. Here is an example of how to enter into the correct voice class mode for
H.323 using 100 as our unique identier:
Router(config-class)#
The two H.225 timeout settings you must be familiar with are these:
h225 timeout tcp establish This command species the amount of time VoIP dial
peers will wait to hear an H.225 receive response from the remote gateway. The default is
15 seconds, but this timer is often shortened when there are backup gateways that take over
the transport of H.323 in the event of a connection failure on the primary connection. By
shortening the timer from 15 seconds to something more practical like 3 seconds, you make
it less likely that your end users will notice that a network failure has occurred. Here is an
example of how to adjust the H.225 timeout to 3 seconds:
Router(config-class)#h225 timeout tcp establish 3
Router#
c07.indd 232 21/09/11 11:23 AM
h225 timeout setup This command adjusts the number of seconds that a voice gateway
will wait in response to an H.225 call setup message. The default value is 15 seconds.
Again, this is sometimes reduced to speed up the backup process in the event of a network
failure. The following example adjusts the timeout to 3 seconds:
Router(config-class)#h225 timeout setup 3
Router#
Any H.225 timer adjustments congured must then be applied to corresponding H.323
dial peers using the voice class h323 command followed by the unique identier we used.
For example, we will apply our H.323 voice class 100 settings to VoIP dial peer 999, as
shown here:
Router(config)#dial-peer voie 999 voip
Router(config-dial-peer)#voice-class h323 100
Router#
Binding a Virtual H.323 Gateway Address for Redundancy
The nal H.323 conguration we will look at is interface binding; we will congure it to
bind the H.323 gateway source address to a virtual interface for redundancy purposes. The
H.323 gateway IP address is the address used by remote routers for forwarding H.323 calls
to us. If we were to use physical interfaces for this purpose, it would create a single point of
failure. To prevent this, we can use multiple physical interfaces to connect to remote networks
and bind them to a virtual interface weve created on our voice gateway. Figure 7.6 shows an
example of how we can use a loopback interface congured with the IP of 10.10.10.100 to
use as our H.323 voice gateway IP address:
FI GURE 7. 6 An H.323 loopback interface
IP
WAN
Local
H.323 gateway
Switch
Remote
H.323 gateway
V
V
Multiple physical
interfaces
Interface loopback 0:
10.10.10.100/24
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234 Chapter 7

By conguring a loopback interface and specifying it as our H.323 gateway IP for
this router, we ensure that the address will be used as the sole-source IP address for any
outbound H.323 connections. Here is how to congure loopback 0 with our IP and as our
virtually bound H.323 gateway source interface:
Router(config)#interface loopback0
Router(config-if)#ip address 10.10.10.100 255.255.255.0
Router(config-if)#h323-gateway voip bind srcaddr 10.10.10.100
Router#
H.323 show Commands
Now that you know how to congure H.323 signaling between two gateways, you need to
familiarize yourself with two IOS show commands used to verify proper conguration and
operational status.
show gateway
The show gateway command displays operational information regarding signaling
protocols such as H.323. Here is an example of the output of this command if congured
to connect to another H.323 gateway as in previous examples:
Router#show gateway
H.323 ITU-T Version: 4.0 H323 Stack Version: 0.1
H.323 service is up
This gateway is not registered to any gatekeeper
Alias list (CLI configured) is empty
Alias list (last RCF) is empty
Router#
From the output you can see that the H.323 service is up but not registered to a
gatekeeper. This is because our particular setup does not require a gatekeeper. Conguration
of H.323 gatekeepers is discussed in Chapter 10, Conguring and Managing CUBE and
H.323 Gateways, of this study guide.
show h323 gateway h225
The show h323 gateway h225 command gives us H.225 signaling protocol setup information.
Here is an example of the output you might see on a voice gateway:
c07.indd 234 21/09/11 11:23 AM
Router#show h323 gateway h225
H.225 STATISTICS AT 00:46:12
H.225 REQUESTS SENT RECEIVED FAILED
Setup 12 53 0
Setup confirm 48 0 0
Alert 30 0 0
Progress 29 0 0
Call proceeding 53 0 0
Notify 0 0 0
Info 0 0 0
User Info 0 0 0
Facility 32 0 0
Release 44 43 1
Reject 0 0 0
Passthrough 0 0 0
H225 establish timeout 0
RAS failed 0
H245 failed 0
Router#
You can see that the command output displays counter information regarding H.225
messages that have been sent and received between H.323 gateways. It also shows counters
for any failures. These counters can be of great use while troubleshooting H.323 signaling
problems. You can clear the counters on the voice gateway by issuing the clear h323
gateway h225 command, as shown here:
Router#clear h323 gateway h225
H.225 stats cleared at 00:48:29
Router#show h323 gateway h225
H.225 STATISTICS AT 00:48:34
H.225 REQUESTS SENT RECEIVED FAILED
Setup 0 0 0
Setup confirm 0 0 0
Alert 0 0 0
Progress 0 0 0
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236 Chapter 7

Call proceeding 0 0 0
Notify 0 0 0
Info 0 0 0
User Info 0 0 0
Facility 0 0 0
Release 0 0 0
Reject 0 0 0
Passthrough 0 0 0
H225 establish timeout 0
RAS failed 0
H245 failed 0
Router#
Configuring SIP
Session Initiation Protocol (SIP) is a widely popular protocol thanks in part to its open-
standard nature, ease of conguration, and wide support from almost all voice hardware
manufacturers. In this section, we will explore the inner workings of SIP signaling and how
to congure a Cisco IOS router to signal the setup of voice calls over an IP network.
As you learned in Chapter 3, SIP is a peer-to-peer protocol. When connecting to the
PSTN via ISDN such as a T1 PRI, SIP signaling is used between the voice gateway and the
CUCM, while ISDN Q.921 and Q.931 signaling is used between the voice gateway and the
PSTN, as shown in Figure 7.7.
FI GURE 7. 7 SIP and ISDN signaling
PSTN
Internal IP
network
SIP Q.921
Q.931
Voice
gateway
V
CUCM
M
Specically, the SIP voice gateway signaling protocol is responsible for the following tasks:

Determine the location of target endpoints.

Determine the capabilities of target endpoints.
c07.indd 236 21/09/11 11:23 AM
Configuring SIP 237

Determine whether the destination endpoint is available for a call.

Establish a session between the originating and target endpoints and handles the
transfer and/or termination of the call.
Well look at each of these in order.
Determine the Endpoint Locations
SIP must learn where the destination devices are located, including IP address, extension,
and name. This is done in either of two ways, depending on how the voice network is set
up. The methods for acquiring and resolving destination addresses are these:

A locally stored IP addresstodomain-name table.

A proxy server that seeks out SIP table information stored in SIP registrar, redirect, and
location servers. In a Cisco environment, these tasks are all typically handled by a CUCM.
Determine the Endpoint Capabilities
SIP performs signaling for a wide range of audio and video devices, and the protocol must
determine what capabilities are compatible between endpoints that wish to communicate.
This capability verication is accomplished using the Session Description Protocol
(SDP). SDP is an RFC 2327 protocol that uses standard ASCII codes for describing and
negotiating multimedia sessions. The protocol collects information from SIP endpoints in
ASCII format and sends that information to the target in the form of a SIP invite message. The
target device then receives the multimedia characteristics, determines which are compatible
with itself, and then sends that information back to the originating SIP endpoint in a SIP
response message. This method of exchanging SDP messages is known as SIP early offer
and is shown in its basic form in Figure 7.8.
FI GURE 7. 8 A SIP early-offer SDP exchange
IP
network
SIP signaling
1. SIP invite (SDP ofer message)
2. SIP OK (SDP media answer)
V V
Calling party Called party
Here are the
multimedia
characteristics
I can work with.
Here are the
multimedia
characteristics
that match
what you can
work with.
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There is a second SDP method for exchanging multimedia capabilities called SIP delayed
offer. Using this method, the initial SIP invite message from the initiating device does not
contain SDP information. Instead, the initiating device sends an invite without the SDP
capabilities. The delayed-offer process then dictates that the target device is the one responsible
for sending an SDP message to the initiating device in a SIP OK message. Figure 7.9 shows the
basics of the delayed-offer SDP method.
FI GURE 7. 9 A SIP delayed-offer SDP exchange
IP
network
SIP signaling
1. SIP invite
2. SIP OK (SDP media ofer)
V V
Calling party Called party
Here are the
multimedia
characteristics
I can work with.
3. SIP ACK (SDP media answer)
Here are the
multimedia
characteristics
that match
what you can
work with.
The delayed-offer method is recommended on ITSP trunks because it forces the service
provider to choose the optimal audio/video codecs for their service. The default method on
Cisco voice gateways is to use early offer.
Regardless of the SIP offer method used, the SDP messages contain the same information,
including audio/video capabilities and endpoint ownership. An example SDP description
looks like the following:
v=0
o=ssmith 5557843 IN IP4 192.168.10.102
s=test1
c=IN IP4 192.168.10.102
t=0 0
m=audio 3456 RTP/AVP 18 0
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Configuring SIP 239
From the example SDP message provided, you can see the following, according to Table 7.1.
TABLE 7.1 SDM example message contents
SDP Symbol Description Value
v= Protocol version 0
o= Owner ssmith
s= Session name test1
c= Connection information Network=IN, IP=192.168.10.102
t= Time start=0, stop=0
m= Media codes audio, 18=G.729a/8000, 0=G.711 PCM
(others include 8=PCMA, 96=G.72624/8000,
97=G.72640/8000, 98=G.72616/8000)
Determine Endpoint Availability
If the intended destination phone is busy or ofine or nobody picked up the ringing phone
(ring-no-answer), SIP is responsible for letting the calling party know that the call cannot
be completed. SIP uses various informational messages to notify the calling party.
Establish a Session
If the phone is capable of accepting calls and the user picks up the ringing phone, SIP is
responsible for making the initial connection between endpoints. After the connection
is established, the phones use RTP streams independent of SIP for the actual voice
communication. SIP is still monitoring the call, however, for any disconnects or call
transfers during the call. For transfers, SIP will be responsible for establishing a secondary
connection session to the new target phone. Once the transfer connection is established,
SIP tears down the old connection automatically.
Now that you understand what SIP needs to do when signaling, the next section will
cover how to congure SIP and modify some default settings between two voice gateways
on an IP network as well as when interoperating with an ISDN PSTN circuit.
Configure SIP between IP Voice Gateways
Our rst SIP conguration example will be to connect to a remote SIP gateway in an ITSP
situation. We will use Figure 7.10 as our network setup for this example.
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In order to properly connect our SIP gateway, we will need to congure our gateway as
a SIP user agent (UA). Once the basic SIP foundation is set up to point to the ITSP, we will
create VoIP dial peers that will direct calls out to our SIP neighbor.
Lets rst tackle our SIP UA conguration. For this we need to have some basic
information about our internal network, namely, the IP address of the registrar/SIP server
and a username/password combination so that we can perform secure MD5 authentication
on the network. The username and password are entered while in config-sip-ua
conguration mode by issuing the authentication username name password 0 password
command. The 0 in the command indicates that the password is entered in clear text. If the
password is already encrypted, you should change the 0 to 7 to let the router know not to
encrypt the password a second time. In our example we will enter ssmith as our username
and an unencrypted password, mypassword.
Because our internal IP network is running SCCP, we will need to set our SIP UA
to point to a SIP server and registrar server, which is the IP address of our CUCM, as
indicated in Figure 7.10.
Lets rst congure our UA settings as shown here:
Router(config)#sip-ua
Router(config-sip-ua)#authentication username ssmith password 0 mypassword
Router(config-sip-ua)#registrar 192.168.99.99
Router(config-sip-ua)#sip-server 192.168.99.99
Router(config-sip-ua)#end
Router#
FI GURE 7.10 A SIP network configured between IP gateways
PSTN Internet
Extensions
40004999
192.168.99.99
SCCP
signaling
10.1.1.100
Voice
gateway
SIP signaling
ITSP voice
gateway
V V
M
IP
c07.indd 240 21/09/11 11:23 AM
Configuring SIP 241
Once we complete those changes, basic SIP is congured on our network. Next, we will
need to congure inbound and outbound VoIP dial peers, which must specify that we use
SIP signaling. Remember, if we dont specify SIP signaling, then the dial peer defaults to
H.323. To set the dial peer to SIP, we use the session protocol sipv2 command.
We will rst congure VoIP dial peer 4000, which will match any 4XXX destination
number. We will then point the call to our SIP proxy server using the session
target sip-server command. Note that we could also congure session target
ipv4:192.168.99.99, but since weve already identied that IP address as our SIP proxy
server with the sip-server 192.168.99.99 command while in config-sip-ua mode, we
simply need to tell the dial peer to send the call to our SIP proxy to let it complete the call.
If we did not have a SIP server specied, then we would be required to enter the IP address
of the CUCM.
Router(config-dial-peer)#session protocol sipv2
Router(config-dial-peer)#destination pattern 4...
Router(config-dial-peer)#session target sip-server
Router#
Next, we will congure outbound dial peer 9 so that all off-network calls are routed out
to our SIP peer, which is the ITSP. Here is the correct conguration for this dial peer:
Router(config-dial-peer)#destination pattern 9T
Router(config-dial-peer)#session target 10.1.1.100
Router#
Thats really all there is to congure a basic SIP network to an ITSP. Different ITSPs
have various requirements, however, so you will have to work with them on a case-by-case
basis because they may require you to modify additional settings for your SIP connection
to work properly. Some of those modications that you may need to set are discussed in the
next two sections.
Configure Secure SIP Communications
SIP calls can be congured to secure the signaling protocol, the voice transmission, or both. It
is highly recommended that you secure both SIP signaling and the RTP voice call. To secure
SIP signaling, you must enable SIP secure (SIPS), an authentication and encryption mechanism
for SIP using the Transport Layer Security (TLS) protocol. This protocol runs on top of TCP.
You can enable SIPS either globally on the voice gateway or on an individual dial-peer basis.
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Support for SRTP is fairly new and requires a voice gateway with IP voice
IOS 12.4(15)T or higher.
In order for SIP secure to work, it must be congured end to end. To congure SIPS
globally, you must enter into config-serv-sip mode and enter the url sips command.
This command species that the router generate universal resource locators (URLs) in SIPS
format for all VoIP calls on this voice gateway. Here is an example of how to congure
SIPS globally:
Router(config)#voice service voip
Router(config-voi-serv)#sip
Router(config-serv-sip)#url sips
Router(config-serv-sip)#end
Router#
Conguring SIPS on an individual dial-peer basis requires that you enter into
config-dial-peer mode for a specic dial peer and enter the voice-class sip url
sips command, as shown here with dial peer 100:
Router(config-dial-peer)#voice-class sip url sips
Router#
Individual dial-peer SIPS configurations take precedence over any globally
configured SIPS setup. Therefore, you can configure SIPS globally on the
voice gateway and then disable it on specific dial peers if needed. To do
this, simply enter into config-dial-peer mode for the desired dial peer
and enter no voice-class sip url to disable SIPS.
Next, we want to secure the actual voice packets that run over RTP. To accomplish this,
you can use SRTP in conjunction with SIPS. As with SIPS, you can congure SRTP globally
or on an individual dial-peer level. To enable SRTP, you enter into config-voi-serv mode
and enter the srtp command. Additionally, it is highly recommended that you also enter
srtp fallback. This command species that if an RTP peer cannot support SRTP, the
voice gateway can fall back to unencrypted RTP streams. If you dont enter this command
and you run across a device that does not support SRTP, the RTP stream cannot be set up
and the call will fail. The conguration commands look like this:
c07.indd 242 21/09/11 11:23 AM
Configuring SIP 243
Router(config-voi-serv)#srtp
Router(config-voi-serv)#srtp fallback
Router(config-voi-serv)#end
Router#
To congure SRTP on a dial peer basis, you must enter the srtp and srtp fallback
commands within config-dial-peer mode, as shown here with dial peer 100:
Router(config-dial-peer)#srtp
Router(config-dial-peer)#srtp fallback
Router#
Modify SIP Voice Gateway Settings
There are several modications that CVOICE candidates should be familiar with.
Commonly changed default settings include these:

Conguring inbound and outbound SIP transport protocols

Modifying SIP signaling timers

Modifying SIP signaling retries

Modifying the maximum number of proxy and redirect servers

Binding a SIP source IP address

Conguring SIP for ISDN interoperation
Lets take a closer look at how to congure each of these on a Cisco voice gateway.
Configuring Inbound and Outbound SIP Transport Protocols
SIP accepts UDP messages coming inbound from potential SIP peers. This can be changed
so that TCP messages are accepted. To do this, you enter into config-sip-ua mode and
issue the transport tcp command. This will change the routers inbound SIP message
behavior globally. Here is an example of how to modify the SIP inbound transport method:
Router(config-sip-ua)#transport tcp
Router#
You can also modify the transport protocol that your voice gateway uses to send
SIP message to peers. Again, by default, the protocol used is UDP. You can change SIP
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244 Chapter 7

outbound message protocol behavior to TCP within individual dial peers, as shown in this
example using VoIP dial peer 88:
Router(config-dial-peer)#session transport tcp
Router#
The inbound and outbound transport methods must be the same on a
voice gateway. So if you change the inbound transport method to TCP, you
must also change each SIP dial peer to utilize TCP as well.
Modifying SIP Signaling Timers
Numerous SIP timer settings can be modied, as shown in the following output:
Router(config-sip-ua)#timers ?
buffer-invite Time to buffer the INVITE while waiting for display info
connect Time to wait for confirmation a session connected
connection Connection related timers
disconnect Time to wait for confirmation a session disconnected
expires Time to wait for the expiration of an INVITE request
hold Time to wait during hold before disconnecting
info Time to wait before INFO retransmission
keepalive Options keepalive related timers
notify Time to wait before NOTIFY retransmission
options Time to wait before OPTIONS retransmissions
prack Time to wait before starting PRACK retransmission
refer Time to wait before REFER retransmission
register Time to wait before REGISTER retransmission
rel1xx Time to wait before starting reliable 1xx retransmission
trying Time to wait for provisional response to INVITE
update Time to wait before starting UPDATE retransmission
Router(config-sip-ua)#timers
While the default settings work in most situations, some slower, congested, and less-
reliable networks might be better off if the timers are expanded. Table 7.2 lists several of the
most important SIP timer types, with each ones purpose and default setting.
c07.indd 244 21/09/11 11:23 AM
Configuring SIP 245
TABLE 7. 2 SIP timers
Timer Type Purpose Range Default Setting
trying Time to wait for an INVITE
response
1001000 ms 500 ms
connect Time to wait for an ACK
response
1001000 ms 500 ms
disconnect Time to wait for a BYE
response
60000300000 ms 180000 ms
expires Time that an INVITE message
is valid
1001000 ms 500 ms
To modify SIP timers, you must be in config-sip-ua mode and then issue the timers
command followed by the timer type and new time in milliseconds (ms). The following
example adjusts the trying, connect, and expires timers to the maximum 1000 ms:
Router(config-sip-ua)#timers trying 1000
Router(config-sip-ua)#timers connect 1000
Router(config-sip-ua)#timers expires 1000
Router#
Modifying SIP Signaling Retries
Similar to the SIP timer options, multiple SIP retry types can be adjusted, as shown in the
following output:
Router(config-sip-ua)#retry ?
bye BYE retry value
cancel CANCEL retry value
info INFO retry value
invite INVITE retry value
keepalive KEEPALIVE retry value
notify NOTIFY retry value
options OPTIONS retry value
prack PRACK retry value
refer REFER retry value
register REGISTER retry value
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246 Chapter 7

rel1xx Reliable 1xx response retry value
response Response Methods retry value
subscribe SUBSCRIBE retry value
update UPDATE retry value
Router(config-sip-ua)#retry
Again, depending on your network, it might be helpful to increase SIP retries on
networks that are prone to congestion and/or dropped packets. Table 7.3 lists each retry
type, its purpose, and the default setting.
TABLE 7. 3 SIP retry types
Retry Type Purpose Default Setting
Invite Max number of INVITE message retries 6
Response Max number of RESPONSE message retries 6
Bye Max number of BYE retries 10
Cancel Max number of CANCEL retries 10
In our example, we will modify our invite and response timers from the default of 6
retries to 8, as shown here:
Router(config-sip-ua)#retry invite 8
Router(config-sip-ua)#retry response 8
Router#
If you want to reset your SIP UA configuration back to default settings,
you can issue the default command followed by any of the following:
max-forwards, retry {invite, response, bye, cancel}, sip-server, timers
{trying, connect, disconnect, expires}, transport.
Modifying the Maximum Number of Proxy and Redirect Servers
If you use SIP proxy and redirect servers on your network and have multiple servers (CUCMs
in Cisco networks) for redundancy, you may need to adjust the maximum number of proxy
and redirect servers. The max-forwards command species the maximum number of
proxy or redirect servers that can forward requests. The default maximum is 70. To make
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Configuring SIP 247
these SIP modications you must be in config-sip-ua mode. Here is an example of how to
change the max proxy/redirect servers to 10:
Router(config-sip-ua)#max-forwards 10
Router#
Binding a SIP Source IP Address
As youve learned, SIP has two parts, a signaling path and a media path for the transport
of voice. Many times, the two media types have different path source IP addresses. This can
cause difculties when attempting to route through a rewall that requires you to know the
source IP addresses. To help eliminate conguration problems, the SIP source-bind feature
can be used to statically assign an IP address to a specic voice gateway interface to be used
for the signaling, media, or both signaling and media source IP addresses. To use this feature,
you must be in config-serv-sip mode and issue the bind command followed by either
control, media, or all. You must then specify the interface you wish to bind SIP to. This
interface needs to be congured with either an IPv4 or IPv6 address for proper operation.
Here is an example to congure source interface FastEthernet 0/1 to be bound for both SIP
signaling and media:
Router(conf-voi-serv)#sip
Router(conf-serv-sip)#bind all source-interface fa0/1
Router(conf-serv-sip)#end
Router#
Configuring SIP for ISDN Interoperation
SIP often has to interoperate with PSTN circuits such as ISDN BRI and PRI circuits. An
example would be a private network that utilized SIP over WAN connections as well as out
to the PSTN using an ISDN PRI, as shown in Figure 7.11.
FI GURE 7.11 SIP and ISDN interoperation
WAN
V
Ext: 5555 Detroit
voice gateway
SIP signaling
Chicago
voice gateway
S1/0
Call coming into the Chicago
voice gateway that is destined for
Ext: 5555
PSTN
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Calls that come in from the PSTN over ISDN circuits contain valuable information,
including caller ID number and display name. By default, SIP will only send the caller ID
number that the call originated from. There is a two-step process to forward display names
on to the terminating gateway. If you wish to receive the calling name, you must rst use
the signaling forward command in conf-voi-serv mode, followed by either the none
or unconditional keyword. The signaling forward command tells the voice gateway to
forward signaling information to the terminating voice gateway. The none keyword blocks
the gateway from forwarding the information, while the unconditional keyword forwards the
information. You want to forward the information on to the terminating gateway as shown
in Figure 7.11, so you will congure the following:
Chicago#configure terminal
Chicago(config)#voice service voip
Chicago(conf-voi-serv)#signaling forward unconditional
Chicago(conf-voi-serv)#end
Chicago#
Step 2 requires that you navigate to the ISDN interface on our voice gateway and issue
the isdn supp-service name calling command. This species that you wish to forward
the calling name information sent out of the ISDN circuit. Here is an example of how to
congure this on ISDN PRI serial 1/0:
Chicago(config)#interface s1/0:23
Chicago(config-if)#isdn supp-service name calling
Chicago(config-if)#end
Chicago#
Optionally, if the display name is not available, you can issue the clid substitute name
command to show the number in its place. You congure this command in conf-voi-serv
mode, as shown here:
Chicago(config)#voice service voip
Chicago(conf-voi-serv)#clid substitute name
Chicago(conf-voi-serv)#end
Chicago#
All of the caller-ID configurations can also be configured at the dial-peer
level instead of globally. The command syntax is identical.
c07.indd 248 21/09/11 11:23 AM
Configuring SIP 249
ISDN and SIP Caller-ID Blocking
ISDN supports caller-ID blocking at the network level. While the number will be blocked
at the destination phone by SIP, the message could potentially still be read elsewhere on the
network in SIP message requests. If you want to completely block incoming caller ID from
ISDN calls on your network, you can issue the clid strip pi-restrict command. This will
permanently remove caller-ID information from SIP messages that come in as private, instead
of simply hiding it from the destination endpoint. Caller-ID blocking is congured on each
dial peer. Here is an example of how to congure CLID blocking on VoIP dial peer 101:
Chicago(config)#dial peer voice 101 voip
Chicago(config-dial-peer)#clid strip pi-restrict
Chicago(config-dial-peer)#end
Chicago#
SIP show Commands
There are several useful show commands to determine if SIP is operational as well as for
troubleshooting purposes. The following is a list of commonly used SIP show commands
with examples of their output so you can see what information can be gleaned.
show sip-ua statistics
This command is great for troubleshooting because it displays counters for various SIP
successes and failures, as shown in this example output:
Router# show sip-ua statistics
SIP Response Statistics (Inbound/Outbound)
Informational:
Trying 0/5, Ringing 0/4,
Forwarded 0/0, Queued 0/0,
SessionProgress 0/0
Success:
OkInvite 0/4, OkBye 0/0,
OkCancel 0/0, OkOptions 0/0,
OkPrack 0/0, OkPreconditionMet 0/0,
OkSubscribe 0/0, OkNotify 0/0,
OkInfo 0/0, 202Accepted 0/0
Redirection (Inbound only):
MultipleChoice 0, MovedPermanently 0,
MovedTemporarily 0, UseProxy 0,
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AlternateService 0
Client Error:
BadRequest 0/0, Unauthorized 0/0,
PaymentRequired 0/0, Forbidden 0/0,
NotFound 0/0, MethodNotAllowed 0/0,
NotAcceptable 0/0, ProxyAuthReqd 0/0,
ReqTimeout 0/0, Conflict 0/0, Gone 0/0,
ReqEntityTooLarge 0/0, ReqURITooLarge 0/0,
UnsupportedMediaType 0/0, BadExtension 0/0,
TempNotAvailable 0/0, CallLegNonExistent 0/0,
LoopDetected 0/0, TooManyHops 0/0,
AddrIncomplete 0/0, Ambiguous 0/0,
BusyHere 0/0, RequestCancel 0/0,
NotAcceptableMedia 0/0, BadEvent 0/0,
SETooSmall 0/0
Server Error:
InternalError 0/0, NotImplemented 0/0,
BadGateway 0/0, ServiceUnavail 0/4,
GatewayTimeout 0/0, BadSipVer 0/0,
PreCondFailure 0/0
Global Failure:
BusyEverywhere 0/0, Decline 0/0,
NotExistAnywhere 0/0, NotAcceptable 0/0
SIP Total Traffic Statistics (Inbound/Outbound)
Invite 5/0, Ack 4/0, Bye 0/4,
Cancel 0/0, Options 0/0,
Prack 0/0, Comet 0/0,
Subscribe 0/0, Notify 0/0,
Refer 0/0, Info 0/0
Retry Statistics
Invite 0, Bye 2, Cancel 0, Response 4,
Prack 0, Comet 0, Reliable1xx 0, Notify 0
SDP application statistics:
Parses:5, Builds 4
Invalid token order:0, Invalid param:0
Not SDP desc:0, No resource:0
Last time SIP Statistics were cleared:<never>
Router#
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Configuring SIP 251
Note the last line of the output, which indicates when the SIP statistics were last cleared.
If you are troubleshooting with this command, you can clear out the statistics using the
clear sip-ua statistics command.
show sip-ua status
This command gives you a quick view of the SIP conguration settings so you can easily
see if there have been any modications. The following output tells you what SIP options
are enabled/disabled and shows you settings such as your max-forwards for SIP proxy and
redirect servers. Additionally, you can see what SDP information is supported and required:
Router#show sip-ua status
SIP User Agent Status
SIP User Agent for UDP: ENABLED
SIP User Agent for TCP: ENABLED
SIP User Agent for TLS over TCP: ENABLED
SIP User Agent bind status(signaling): DISABLED
SIP User Agent bind status(media): DISABLED
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards: 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
SIP support for ISDN SUSPEND/RESUME: ENABLED
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED
SDP application configuration:
Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported: audio image
Network types supported: IN
Address types supported: IP4
Transport types supported: RTP/AVP udptl
Router#
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show sip-ua timers
If you made changes to the SIP timers, you can verify them using this command. Here is an
example of its output, displaying all of the default SIP timer settings:
Router#show sip-ua timers
SIP UA Timer Values (millisecs unless noted)
trying 500, expires 180000, connect 500, disconnect 500
prack 500, rel1xx 500, notify 500, update 500
refer 500, register 500, info 500, options 500, hold 2880 minutes
tcp/udp aging 5 minutes
Router#
show sip-ua retry
SIP retry values can also be scanned and veried using this command. Here is an example
of its output, showing the default SIP retry settings:
Router#show sip-ua retry
SIP UA Retry Values
invite retry count = 6 response retry count = 6
bye retry count = 10 cancel retry count = 10
prack retry count = 10 update retry count = 6
reliable 1xx count = 6 notify retry count = 10
refer retry count = 10 register retry count = 6
info retry count = 6 subscribe retry count = 6
options retry count = 6
Router#
show sip-ua calls
This command is useful for seeing calls being made in real time. Helpful information
includes source and destination E.164 numbers, source and destination IP addresses, and
codecs being used, as shown in this example output:
Router# show sip-ua calls
SIP UAC CALL INFO
Call 1
SIP Call ID: A06260318EC511DF-A260B9C09A07DD7B@192.168.1.77
State of the call: STATE_ACTIVE (6)
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Substate of the call: SUBSTATE_NONE (0)
Calling Number: 4010
Called Number: 5510
Bit Flags: 0x12120030 0x220000
Source IP Address (Sig ): 192.168.1.77
Destn SIP Req Addr:Port: 192.168.10.66:5063
Destn SIP Resp Addr:Port: 192.168.10.66:5063
Destination Name: 192.168.10.66
Number of Media Streams: 1
Number of Active Streams: 1
RTP Fork Object: 0x0
Media Stream 1
State of the stream: STREAM_ACTIVE
Stream Call ID: 27
Stream Type: voice-only (0)
Negotiated Codec: g711ulaw (160 bytes)
Codec Payload Type: 0
Negotiated Dtmf-relay: inband-voice
Dtmf-relay Payload Type: 0
Media Source IP Addr:Port: 192.168.1.77:19465
Media Dest IP Addr:Port: 192.168.10.66:16890
SIP UAS CALL INFO
Number of UAS calls: 0
Router#
Configuring MGCP
Compared to H.323 and SIP, conguring MGCP is a breeze. This is primarily thanks to
the fact that MGCP is a client-server architecture with centralized call control. The call
control agent in the Cisco world is a CUCM, which manages all dial plans and the setup
and teardown of connections between the IP network and the PSTN. When connecting to
the PSTN via ISDN, MGCP is different from H.323 and SIP in that all Layer 3 signaling
is controlled by the call agent. Therefore, both MGCP and Q.931 signaling is backhauled
between the PSTN and the CUCM, while Q.921 signaling is used between the voice
gateway and the PSTN. MGCP creates a separate channel that is used to backhaul the
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254 Chapter 7

Like SIP, MGCP uses SDP for session initiation between endpoints. The MGCP gateway
is responsible for converting voice signals between traditional PSTN connections and the
IP network. MGCP signaling is used to report events such as off-hook status or DTMF
occurrences. These events are sent up to the call agent as notication event messages. And
because the call agent handles all call-routing decisions, no dial peers are congured on the
voice gateway (unless you have analog phones directly attached). This is what is meant by
centralized call control.
Because MGCP relies so heavily on the call agent, if connectivity is lost,
MGCP does not have the capability on its own to independently route calls
between the IP and PSTN. Fortunately, you can configure H.323 fallback and
Survivable Remote Site Telephony (SRST) on the voice gateway to handle
these situations. Both of these techniques are defined and described in
Chapter 9, Advanced Voice Gateway Features, of this study guide.
MGCP uses cleartext communication between the voice gateway and call agent. The
commands are sent from the call agent to the voice gateway using UDP port 2427 by
default. The actual MGCP conguration on voice gateways primarily denes where the
call agent is located on the network so a communications channel can be started. MGCP
denes two distinctly different voice gateway types.
Residential Gateways
MGCP residential gateways are responsible for providing signaling between the IP network
and analog voice ports including FXS, FXO, and E&M. Figure 7.13 shows an example of
what a residential gateway network might look like.
FI GURE 7.12 MGCP and ISDN signaling
PSTN
Internal IP
network
MGCP Q.921
Q.931
Voice
gateway
V
CUCM
M
Q.931 information between the call agent and voice gateway. MGCP communication
boundaries are shown in Figure 7.12.
c07.indd 254 21/09/11 11:23 AM
While this example shows the MGCP voice gateway supporting analog telephone
endpoints, keep in mind that the analog endpoint could be a key system or analog PBX.
Trunking Gateways
As the name suggests, MGCP trunking gateways differ from residential gateways because
they connect an IP network to the PSTN using trunk ports such as T1s and E1s. Figure 7.14
shows an example of what a trunking gateway network might look like.
FI GURE 7.13 An MGCP residential gateway
Voice
gateway
Switch
Analog ports
V
CUCM
M
G
C
P

s
i
g
n
a
l
i
n
g
M
FI GURE 7.14 An MGCP trunking gateway
PSTN
Voice
gateway
Trunk
Switch
V
CUCM
M
Regardless of the gateway type, an MGCP gateway exchanges control commands
between itself and the call agent. These messages are simplistic in nature and are used to
either notify the call agent of things occurring on the PSTN or to take orders from the call
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256 Chapter 7

agent. Each of these messages is sent with a four-letter acronym to dene the command.
Responses to the commands begin with a three-number response code. The nine MGCP
communications messages and their functions are listed in Table 7.4.
TABLE 7. 4 MGCP command acronyms
MGCP Command
Acronym Acronym Meaning Command Function
AUEP Audit endpoint Used to audit the status of endpoints.
AUCX Audit connection Used to audit the status of endpoint
connections.
CRCX Create connection Used to create RTP connection that
terminates on the voice gateway.
DLCX Delete connection Used to delete RTP connection that is
terminated on the voice gateway.
MDCX Modify connection Used to modify existing RTP
connection that is terminated on the
voice gateway.
RQNT Request for notification Used by the call agent to request the
voice gateway to begin monitoring
for signaling events.
EPCF Endpoint configuration Used by the call agent to remotely
send a configuration command to the
voice gateway.
NTFY Notify Used by the voice gateway to notify
the call agent of an event the call
agent has requested (in an RQNT
command message) it monitors for.
RSIP Restart in progress Used by the voice gateway to inform
the call agent that it is in the process
of restarting.
The MGCPs events and signaling messages are bundled into various MGCP
conguration packages to provide simplicity when setting up your MGCP gateway. Each
package serves a specic MGCP function. You enable the packages you wish to utilize
and leave the other packages disabled. Packages are categorized into groups; for example,
the line-package group contains all of the signaling and event packets when operating
a residential gateway. The trunk-package provides all the default events and signals for
PSTN trunking gateways. Other packages provide more specic events and signaling such
c07.indd 256 21/09/11 11:23 AM
as a secure RTP (srtp-package) package and a fax transmission (fxr-package) package.
Package types are congured on voice gateways using the mgcp package-capability
command while in global conguration mode. Following is a listing within the Cisco IOS
showing possible MGCP package types and their descriptions:
Router(config)#mgcp package-capability ?
as-package Select the Announcement Server Package
dtmf-package Select the DTMF Package
fm-package Select the FM Package
fxr-package Select the FXR Package
gm-package Select the Generic Media Package
hs-package Select the Handset Package
it-package Select the IT Package
lcs-package Select the Line Control Signaling Package
line-package Select the Line Package
mdr-package Select the MDR Package
mf-package Select the MF Package
pre-package Select the PRE Package
res-package Select the RES Package
rtp-package Select the RTP Package
script-package Select the Script Package
srtp-package Select the SRTP Package
sst-package Select the SST Package
trunk-package Select the Trunk Package
Router(config)#mgcp package-capability
Configure an MGCP Residential Gateway
As stated previously, MGCP residential gateways connect analog ports to an IP network.
The rst IOS conguration step that must be performed is simply to enable MGCP on
your voice gateway. To do that, just type in mgcp while in global conguration mode.
Additionally, if your call agent is a CUCM, you need to add the ccm-manager mgcp
command. If your call agent is from a different vendor, this command is not needed.
Next, you need to inform your local voice gateway where your call agent (or call agents
if there are more than one) is located on the IP network. To accomplish this, you use
the mgcp call-agent command in global conguration mode and specify either the IP
address or the hostname (using DNS) of your CUCM. To nish the command, you use the
service-type mgcp keyword to indicate you are using MGCP signaling. This command
can be used multiple times in the conguration to add additional call agents. The voice
gateway will send requests out to all congured call agents and will use the rst one that
responds to a MGCP message request.
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Finally, you will select the MGCP event and signaling packages you will utilize on your
network using the mgcp package-capability command. Note that the line-package is
enabled by default for residential gateway congurations. In our example we will use the
line-package, dtmf-package, gm-package, and rtp-package.
Using Figure 7.15 as our example network, we enable MGCP on our voice gateway,
congure the two CUCM call agents, and enable the following MGCP packages for our
residential gateway:

line-packageThe default package used with MGCP residential gateways for message
information such as caller ID, hook ash, and reorder tones.

dtmf-packageUsed to generate DTMF tones.

gm-packageUsed to generate media events and signals for a wide range of generic
events including the congestion tone, fax tones, and ringback.

rtp-packageUsed to generate RTP event messages such as continuity tones and tests,
jitter buffer modication messages, and RTP/RTCP timeouts.
The following commands show how to congure an MGCP residential gateway:
Router(config)#mgcp
Router(config)#ccm-manager mgcp
Router(config)#mgcp call-agent 192.168.10.100 service-type mgcp
Router(config)# mgcp package-capability line-package
Router(config)# mgcp package-capability dtmf-package
Router(config)# mgcp package-capability gm-package
Router(config)# mgcp package-capability rtp-package
Router(config)#end
Router#
FI GURE 7.15 An MGCP residential gateway network
Voice
gateway
Switch
V
FXS 0/0/1
Secondary
CUCM
192.168.20.100
M
Primary
CUCM
192.168.10.100
M
c07.indd 258 21/09/11 11:23 AM
If your CUCM is listening for MGCP signaling on a UDP port other than
the default 2427, you can change the port when defining call agents.
For example, if you want to change the port to 3000 to a call agent at
192.168.1.10, you would configure:
mgcp call-agent 192.168.1.10 3000 service-type mgcp
Alternatively, if the MGCP wishes to communicate to the voice gateway
on a different port, this can be changed using the mgcp command used to
enable MGCP on the voice gateway. To change the voice gateway listening
port to 3000, you would issue mgcp 3000 on the voice gateway in global
conguration mode.
The remainder of an MGCP residential gateway conguration is performed within
individual POTS dial peers. In our example, we will congure a POTS dial peer (300) for
FXO port 0/0/1 and issue the application mgcpapp command to associate the dial peer
and port with MGCP:
Router(config-dial-peer)#application mgcpapp
Router#
Notice that we do not include any destination-pattern commands in our POTS dial
peer; thats because the call agent controls this information.
Configure an MGCP Trunking Gateway
The primary steps for enabling, specifying call agents, and choosing MGCP package groups
youve seen for the conguration of residential gateways are identical for trunking gateways,
although the specic MGCP packages you select will be different. Also note that the trunk-
package MGCP conguration group will be enabled by default when conguring a trunking
gateway. In our example we will use trunk-package, dtmf-package, gm-package, and
rtp-package.
The remainder of the MGCP trunking conguration is very different from the MGCP
residential conguration, however. When conguring trunking gateways, you must specify
the controller of the T1/E1 you wish to enable MGCP on. Then you must use the PRI ds0-
group command, specify the timeslots, specify a type of none, and then issue the service
mgcp key phrase.
Using Figure 7.16 as our example network, we enable MGCP on our voice gateway,
congure a CUCM call agent, and enable our required MGCP group packages. Also note
in the conguration example how the MGCP line package that was used in our residential
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260 Chapter 7

gateway is replaced with the trunk package for our trunking gateway. The trunk package
includes trunk-specic events and tones.
FI GURE 7.16 An MGCP trunking gateway network
PSTN M
Voice
gateway
V
CUCM
10.165.1.100
Switch
T1 2/0
Router(config)#mgcp
Router(config)#ccm-manager mgcp
Router(config)# mgcp package-capability trunk-package
Router(config)# mgcp package-capability dtmf-package
Router(config)# mgcp package-capability gm-package
Router(config)# mgcp package-capability rtp-package
Router(config)#end
Router#
Next, we will congure T1 PRI port 2/0 to enable MGCP on all 24 timeslots of DS0
group 0, which is already congured:
Router(config-controller)#ds0-group 0 timeslots 124 type none service mgcp
Router(config-controller)#end
Router#
MGCP show Commands
Now that you know how to congure both residential and trunking MGCP gateways, lets
look at some of the IOS show commands used to verify our conguration and to be used
for troubleshooting MGCP signaling problems. The next section covers show commands
commonly used for gathering MGCP information.
show mgcp profile
This command is great when verifying conguration settings and when comparing
timeout values, as shown this example output:
c07.indd 260 21/09/11 11:23 AM
Router#show mgcp profile
MGCP Profile default
Description: None
Call-agent: 10.165.1.100 Initial protocol service is MGCP 0.1
Tsmax timeout is 20 sec, Tdinit timeout is 15 sec
Tdmin timeout is 15 sec, Tdmax timeout is 600 sec
Tcrit timeout is 4 sec, Tpar timeout is 16 sec
Thist timeout is 30 sec, MWI timeout is 16 sec
Ringback tone timeout is 180 sec, Ringback tone on connection timeout is
180 sec
Network congestion tone timeout is 180 sec, Busy tone timeout is 30 sec
Network busy tone timeout is 0 sec
Dial tone timeout is 16 sec, Stutter dial tone timeout is 16 sec
Ringing tone timeout is 180 sec, Distinctive ringing tone timeout is 180 sec
Continuity1 tone timeout is 3 sec, Continuity2 tone timeout is 3 sec
Reorder tone timeout is 30 sec, Persistent package is ms-package
Max1 DNS lookup: ENABLED, Max1 retries is 5
Max2 DNS lookup: ENABLED, Max2 retries is 7
Source Interface: NONE
T3 endpoint naming convention is T1
prefer active call-agent is DISABLED
CAS Notification Digit order is Default
Router#
show mgcp
This command can tell you the current operational status of the voice gateway. In the
following output, you can see that MGCP is both administratively and operationally active.
This means that the voice gateway is ready to communicate with the call agent located at
IP 10.165.1.100. Additionally, you can see information such as codec type and supported
MGCP packages:
Router#show mgcp
MGCP Admin State ACTIVE, Oper State ACTIVECause Code NONE
MGCP call-agent: 10.165.1.100 Initial protocol service is MGCP 0.1
MGCP validate call-agent source-ipaddr DISABLED
MGCP validate domain name DISABLED
MGCP block-newcalls DISABLED
MGCP send SGCP RSIP: forced/restart/graceful/disconnected DISABLED
MGCP quarantine mode discard/step
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MGCP quarantine of persistent events is ENABLED
MGCP dtmf-relay for VoIP is SDP controlled
MGCP dtmf-relay for voAAL2 is SDP controlled
MGCP voip modem passthrough disabled
MGCP voaal2 modem passthrough disabled
MGCP voip modem relay: Disabled
MGCP T.38 Named Signalling Event (NSE) response timer: 200
MGCP Network (IP/AAL2) Continuity Test timer: 200
MGCP RTP stream loss timer: 5
MGCP request timeout 500
MGCP maximum exponential request timeout 4000
MGCP gateway port: 2427, MGCP maximum waiting delay 3000
MGCP restart delay 0, MGCP vad DISABLED
MGCP rtrcac DISABLED
MGCP system resource check DISABLED
MGCP xpc-codec: DISABLED, MGCP persistent hookflash: DISABLED
MGCP persistent offhook: ENABLED, MGCP persistent onhook: DISABLED
MGCP piggyback msg ENABLED, MGCP endpoint offset DISABLED
MGCP simple-sdp DISABLED
MGCP undotted-notation DISABLED
MGCP codec type g711ulaw, MGCP packetization period 20
MGCP JB threshold lwm 30, MGCP JB threshold hwm 150
MGCP LAT threshold lwm 150, MGCP LAT threshold hwm 300
MGCP PL threshold lwm 1000, MGCP PL threshold hwm 10000
MGCP CL threshold lwm 1000, MGCP CL threshold hwm 10000
MGCP playout mode is adaptive 60, 40, 1000 in msec
MGCP Fax Playout Buffer is 300 in msec
MGCP media (RTP) dscp: ef, MGCP signaling dscp: af31
MGCP default package: trunk-package
MGCP supported packages: gm-package dtmf-package trunk-package line-package
hs-package ms-package dt-package mo-package
mt-package
fxr-package md-package
MGCP Digit Map matching order: shortest match
SGCP Digit Map matching order: always left-to-right
MGCP VoAAL2 ignore-lco-codec DISABLED
MGCP T.38 Max Fax Rate is DEFAULT
MGCP T.38 Fax is ENABLED
MGCP T.38 Fax ECM is ENABLED
MGCP T.38 Fax NSF Override is DISABLED
c07.indd 262 21/09/11 11:23 AM
MGCP T.38 Fax Low Speed Redundancy: 0
MGCP T.38 Fax High Speed Redundancy: 0
MGCP Fax relay SG3-to-G3: ENABLED
MGCP control bind:DISABLED
MGCP media bind:DISABLED
MGCP Upspeed payload type for G711ulaw: 0, G711alaw: 8
MGCP Dynamic payload type for G.72616K codec
MGCP Dynamic payload type for G.Clear codec
MGCP Dynamic payload type for NSE is 100
MGCP Dynamic payload type for NTE is 99
MGCP rsip-range is enabled for TGCP only.
MGCP Comedia role is NONE
MGCP Comedia check media source is DISABLED
MGCP Comedia SDP force is DISABLED
MGCP Guaranteed scheduler time is DISABLED
MGCP DNS stale threshold is 30 seconds
Router#
show mgcp statistics
This command is useful for troubleshooting purposes because it displays incrementing counts
of sent and received MGCP messages between the voice gateway and call agent. Counters
increment for successful and failed message packets, so you can quickly see if your MGCP
messages are being properly transmitted across your network. You can verify information
including unrecognized, duplicate, or failed packets. Here is an example of the output of this
show command:
Router#show mgcp statistics
UDP pkts rx 20, tx 21
Unrecognized rx pkts 0, MGCP message parsing errors 0
Duplicate MGCP ack tx 0, Invalid versions count 0
CreateConn rx 10, successful 10, failed 0
DeleteConn rx 4, successful 4, failed 0
ModifyConn rx 10, successful 10, failed 0
DeleteConn tx 0, successful 0, failed 0
NotifyRequest rx 10, successful 10, failed 0
AuditConnection rx 0, successful 0, failed 0
AuditEndpoint rx 0, successful 0, failed 0
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RestartInProgress tx 1, successful 1, failed 0
Notify tx 0, successful 0, failed 0
ACK tx 20, NACK tx 0
ACK rx 0, NACK rx 0
IP address based Call Agents statistics:
IP address 10.165.1.100, Total msg rx 20, successful 20, failed 0
DS0 Resource Statistics
-
Utilization: 0.00 percent
Total channels: 0
Addressable channels: 0
Inuse channels: 0
Disabled channels: 0
Free channels: 0
Router#
You can see in this example that weve successfully communicated with our call agent
at 10.165.1.100, because it is listed in the statistics command output and MGCP packets
are being transmitted and received. The statistics counters can be reset by issuing the clear
mgcp statistics command, so you can easily see if any changes to the conguration had
an impact on success transmits and receipts of MGCP messages.
show ccm-manager
You can easily verify that you are properly registered to a CUCM using this command.
Information at the beginning of the output lists primary and backup CUCM call agents,
their current status, and the host/IP address of the call agent, as shown in this example
where we have our primary call agent (10.165.1.100) registered but have no backups
congured:
Router#show ccm-manager
MGCP Domain Name: CCVP
Priority Status Host
============================================================
Primary Registered 10.165.1.100
First Backup None
Second Backup None
Current active Call Manager: None
Backhaul/Redundant link port: 2428
Failover Interval: 30 seconds
c07.indd 264 21/09/11 11:23 AM
Summary 265
Keepalive Interval: 15 seconds
Last keepalive sent: 09:33:22 CST Jan 22 2011
(elapsed time: 02:17:46)
Last MGCP traffic time: 10:51:08 CST Jan 22 2011
(elapsed time: 00:00:00)
Last failover time: None
Last switchback time: None
Switchback mode: Graceful
MGCP Fallback mode: Not Selected
Last MGCP Fallback start time: None
Last MGCP Fallback end time: None
MGCP Download Tones: Disabled
TFTP retry count to shut Ports: 2
Backhaul Link info:
Link Protocol: TCP
Remote Port Number: 2428
Remote IP Address: 10.165.1.100
Current Link State: OPEN
Statistics:
Packets recvd: 0
Recv failures: 0
Packets xmitted: 0
Xmit failures: 0
PRI Ports being backhauled:
Slot 0, VIC 0, port 0
FAX mode: cisco
Router#
Summary
In Chapter 7 you learned about the three primary voice gateway signaling protocols:
H.323, SIP, and MGCP. Each of these protocols operates slightly differently on IP and
PSTN networks, and we explored their differences and how to congure and tweak their
settings. Additionally, it is important to know several show commands used to verify proper
conguration of your voice gateway and to use them in cases where troubleshooting is
needed. We will continue to explore voice gateway congurations in various situations that
utilize these signaling protocols in Chapter 9 of this study guide.
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266 Chapter 7

Exam Essentials
Understand H.323 voice gateway signaling boundaries between the CUCM and
ISDN. H.323 signaling operates between the CUCM and the H.323 voice gateway. ISDN
Q.921 and Q.931 signaling operates between the H.323 gateway and the PSTN.
Know the difference between H.323 slow and fast start initiation methods. With slow
start, the H.245 channel is created after the call setup, call proceeding, alerting, and connect
phases complete, while fast start sets up the H.245 channel during the call setup stage.
Know how to configure basic H.323 between two voice gateways. H.323 is the default
signaling protocol for Cisco voice gateways. Conguring signaling between two voice
gateways is just a matter of conguring the proper dial peers that point to the remote
gateway.
Understand the primary H.323 settings that can be modified. Sometimes it is necessary
to modify H.323 from its default settings. Options such as conguring slow start initiation,
codec preference, and transport mode and adjusting H.323 timers are often changed to
meet the needs of the network.
Understand H.323 interface binding. If you want to eliminate a physical single point
of failure on your H.323 network, congure interface binding to bind multiple physical
interfaces together to create a single logical interface.
Know how to verify H.323 configurations. There are several show commands that are
useful for verication and troubleshooting H.323, including show gateway and show h323
gateway.
Understand SIP voice gateway signaling boundaries between the CUCM and ISDN. SIP
signaling operates between the CUCM and SIP voice gateway. ISDN Q.921 and Q.931
signaling operates between the SIP gateway and the PSTN.
Understand the responsibilities of SIP. SIP is an end-to-end protocol that can be
responsible for knowing the location of endpoints and their capabilities, determining if
endpoints are available, and the establishment and teardown of a SIP call.
Know how to configure basic SIP between two voice gateways. Conguring SIP between
voice gateways on a network is a matter of enabling SIP version 2 on the router, conguring
UA settings, and creating the proper VoIP dial peers to point to the remote SIP gateway.
Know how to configure secure SIP communications. You can congure SIP itself by
enabling SIPS. Additionally, you can secure the voice channel by enabling SRTP and
SRTP fallback.
Understand the primary SIP settings that can be modified. Sometimes it is necessary to
modify SIP from its default settings. Options such as transport protocol method, signaling
timers, retry limits, and proxy and redirect server settings are often changed to meet the
needs of the network.
c07.indd 266 21/09/11 11:23 AM
Written Lab 7.1 267
Know how to configure SIP to interoperate with ISDN. There are a few steps that should
be taken so that SIP properly interoperates with ISDN. Commands such as signaling
forward unconditional, isdn supp-service name, clid strip-pi-restrict, and clid
substitute name all deal with how SIP can properly handle caller-ID functions that are
transported across ISDN networks.
Know how to verify SIP configurations. There are several show commands that are useful
for verifying and troubleshooting SIP, including show sip-ua statistics, show sip-ua
status, show sip-ua timers, show sip-ua retry, and show sip-ua calls.
Understand MGCP voice gateway signaling boundaries between the CUCM and
ISDN. MGCP signaling operates between the CUCM and MGCP voice gateway. ISDN
signaling Q.921 operates between the MGCP gateway and the PSTN, while ISDN Q.931
operates between the PSTN and CUCM.
Understand the difference between MGCP residential and trunking gateways. MGCP
residential gateways use analog ports such as FXS, FXO, and E&M, while trunking
gateways use PSTN connections that can be trunked such as an ISDN T1/E1 PRI.
Know how to configure basic MGCP between two voice gateways. Conguring MGCP
between voice gateways on a network is a matter of enabling MGCP on the router,
notifying the MGCP gateway where the call agent resides, and choosing the MGCP
signaling packages your gateway requires. If your gateway has analog ports directly
congured, you must add the application mgcpapp command to the dial peer. If your
gateway has trunking ports, you must enable the MGCP service on the physical controller
interface.
Know how to verify MGCP configurations. There are several show commands that are
useful for verication and troubleshooting MGCP including show mgcp profile, show
mgcp, show mgcp statistics, and show ccm-manager.
Written Lab 7.1
1. What is the default H.323 initiation method?
2. You wish to congure a voice gateway but nd that the VoIP service has been manually
shut down. While in global conguration mode, you enter voice service voip, which
brings you to the conf-voi-serv conguration mode. What must you enter to start the
VoIP service?
3. By default, how many concurrent calls are supported using the H.323 protocol on
Cisco IOS gateways?
4. You are conguring a VoIP dial peer and are in config-dial-peer mode. What
command do you use to enable this dial peer to utilize the SIP signaling protocol?
5. In a SIP environment, what is a SIP voice gateway known as?
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6. Once a SIP call is connected and RTP connections are established independently
between two endpoints, SIP continues to monitor the call to provide what two
services?
7. What SIP show command lets you see if SIP is enabled to operate on both UDP
and TCP?
8. When connected to the PSTN through an ISDN BRI, MGCP backhauls what type of
signaling between the PSTN and local CUCM?
9. You want to enable fax transmission services on your MGCP router for outbound fax
service to the PSTN. What MGCP conguration package must you enable?
10. What protocol and port does MGCP operate on by default?
Hands-On Labs
To complete the labs in this section, you need a Cisco router with a voice-capable IOS and
two FastEthernet ports to use as our simulated WAN connection and a second connection
to our internal network. Each lab in this section builds on the last and will follow the
logical voice gateway design shown in Figure 7.17.
PSTN
Extensions: 3XX
10.154.79.101
Fa1/0
172.16.3.10
Local SIP
gateway
ITSP
gateway
V V
M
IP
Internal
network
Fa2/0
FI GURE 7.17 Voice gateway lab diagram
c07.indd 268 21/09/11 11:23 AM
Hands-On Labs 269
This lab assumes that basic IP networking and the local voice network are operational.
Additionally, we assume the ITSP voice gateway is already properly congured. We are
only concerned with conguring the local SIP gateway.
Lab 7.1: Conguring Basic SIP
Lab 7.2: Modifying SIP Timers and Retries
Hands-On Lab 7.1: Configuring Basic SIP
In this lab, were going to congure a voice gateway that has a connection to our ITSP
for outbound calling over Fa1/0. Our phones receive E.164 information from the remote
SIP server located at the ITSP. The task here is to enable SIP and point it to the SIP registrar
using the information found in Table 7.5.
TABLE 7. 5 Sample configuration values for basic SIP
SIP Network Information PSTN Requirements
SIP registrar address 10.154.79.101
Username cucmsipregistrar
Password password101
Bind interface Fa2/0
Remote SIP address 172.16.3.10
Outbound dial peer Dial 9 for off-net calls and allow all E.164 numbers
1. Log into the local voice gateway and go into conguration mode by typing enable and
then configure terminal.
2. Enter into conf-voi-serv mode by typing voice service voip.
3. Enter into config-serv-sip mode by typing sip.
4. Congure SIP source binding by typing bind all source-interface fa2/0.
5. Return to global conguration mode by typing exit and exit.
6. Enter into config-sip-ua mode by typing sip-ua.
7. Congure the SIP registrar server by typing registrar 10.154.79.101.
8. Congure SIP authentication to the local SIP registrar server by typing authentication
username cucmsipregistrar password 0 password 101.
9. Return to global conguration mode by typing exit.
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10. Enter into config-dial-peer mode and create VoIP dial peer 300 by typing dial-peer
voice 300 voip.
11. Enable SIP on this dial peer by typing session protocol sipv2.
12. Congure a destination pattern to match the internal IP phones by typing
destination-pattern 3...
13. Point the dial peer to our internal SIP proxy (the CUCM) by typing session target
10.154.79.101.
15. Enter into config-dial-peer mode and create VoIP dial peer 9 by typing dial-peer
voice 9 voip.
16. Enable SIP on this dial peer by typing session protocol sipv2.
17. Congure a destination pattern to match all external phones by typing destination-
pattern 9T.
18. Point the dial peer to our internal SIP proxy (the CUCM) by typing session target
10.154.79.101.
Hands-On Lab 7.2: Modifying SIP Timers and Retries
In Lab 7.1 we congured basic SIP. But now were beginning to experience some problems
because of congestion on the network. It has been suggested that we modify the following
timers and retries on our voice gateway according to Table 7.6.
TABLE 7. 6 Sample configuration values for modifying SIP timers and retries
SIP Timers/Retries PSTN Requirements
trying timer 1000 ms
connect timer 1000 ms
disconnect timer 200000 ms
expires timer 1000 ms
invite retries 10
response retries 10
bye retries 15
cancel retries 15
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Hands-On Labs 271
2. Enter into conf-sip-ua mode by typing sip-ua.
3. Modify the SIP trying timer by typing timers trying 1000.
4. Modify the SIP connect timer by typing timers connect 1000.
5. Modify the SIP disconnect timer by typing timers disconnect 200000.
6. Modify the SIP max invite retries by typing retry invite 10.
7 Modify the SIP max response retries by typing retry response 10.
8. Modify the SIP max bye retries by typing retry bye 15.
9. Modify the SIP max cancel retries by typing retry cancel 15.
10. Exit config-sip-ua mode by typing end.
11. Verify your SIP timers by typing show sip-ua timers.
12. Verify your SIP retries by typing show sip-ua retry.
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Review Questions
1. A what stage of the H.323 slow start initiation process does the voice gateway perform
H.245 negotiation?
A. After the call setup stage
B. During the call proceeding stage
C. After the connect stage
D. After the alerting stage
2. Which signaling protocol does not need dial peers configured on the voice gateway in order
to operate?
A. SIP
B. SIP v2
C. H.323
D. MGCP
3. What two functions are the responsibility of H.245 when using H.323 signaling?
A. Exchange capabilities information between endpoints.
B. Transport SDP information between endpoints.
C. Transport voice or video streams within an H.245 channel.
D. Open and close media channels.
4. Which of the following is the correct IOS command mode and command used to enable
H.323 slow start signaling globally on a voice gateway?
A. Router(conf-voi-serv)#call start slow
B. Router(conf-serv-h323)#h323 call start slow
C. Router(conf-voi-serv)#h323 call start slow
D. Router(conf-serv-h323)#call start slow
5. Which of the following might be a valid reason to modify the H.323 initiation procedure
from the default fast start to slow start?
A. If the network is prone to dropped packets and/or congestion
B. If you have an H.323 endpoint that does not support fast start
C. If you plan to use media features such as MOH
D. If you wish to utilize the H.323 early media feature
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6. You have finished configuring codec preference 10 on your voice gateway and want to apply
it to dial peer 101. Which of the following commands correctly shows this?
A. Router(config)#dial-peer voice 101 pots
B. Router(config)#dial-peer voice 101 pots
Router(config-dial-peer)#voice-class preference 10
C. Router(config)#dial-peer voice 101 voip
D. Router(config)#dial-peer voice 101 voip
Router(config-dial-peer)#voice-class preference 10
7. You have been asked to eliminate a physical single point of failure on your H.323 network
by configuring interface binding. Which of the following properly configures a virtual
address with the IP address of 172.16.3.1/24 and binds the interface to be used as the
H.323 source address?
A. Router(config)#interface loopback0
Router(config-if)#h323-gateway voip bind srcaddr loopback 0
B. Router(config)#interface loopback0
Router(config-if)#h323-gateway voip bind srcaddr 172.16.3.1
C. Router(config)#interface loopback0
Router(config-if)#gateway voip bind srcaddr 172.16.3.1
D. Router(config)#interface loopback0
Router(config-if)#h323-gateway voip interface-bind 172.16.3.1
8. Which H.323 show command would be most useful when troubleshooting H.323
call-signaling problems?
A. show gateway
B. show h323 gateway h225
C. show h323 gateway
D. show h323 gateway statistics
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9. In what format are SDP messages sent?
A. Hexadecimal
B. ASCII
C. Binary
D. Cisco proprietary format
10. What type of SIP message contains SDP data when using the SIP early-offer method?
A. Ack
B. Response
C. Invite
D. OK
11. Which of the following is not a task that is handled by SIP?
A. Knowing the capabilities of target endpoints
B. Providing the setup and teardown of voice calls
C. Knowing the location of target endpoints
D. Opening a backhaul channel for the transport of Q.931 signaling between the PSTN
and CUCM when the SIP gateway is connected to an ISDN circuit from the PSTN
E. Determining if the destination endpoint is available for a call
12. SIP must signal back to the calling party when the calling phone is all of the following
except what?
A. Busy
B. Incompatible firmware
C. Offline
D. Ring-no-answer
13. Which of the following commands is used to enable SIP on a VoIP dial peer?
A. Router(config-dial-peer)#session protocol sipv2
B. Router(config-dial-peer)#gateway protocol sipv2
C. Router(config-dial-peer)#session protocol sip
D. Router(config-dial-peer)#gateway protocol sip
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14. Which of the following commands configures SIP secure globally on a voice gateway?
A. Router(config)#voice service voip
Router(config-serv-sip)#udp sips
B. Router(config)#voice service voip
Router(config-serv-sip)#sip secure
C. Router(config)#voice service voip
Router(config-serv-sip)#url sips
D. Router(config)#voice service voip
Router(config-serv-sip)#tcp sips
15. When configuring SRTP, what optional configuration step is not required but
recommended to ensure interoperation with multiple SIP endpoints?
A. Router(config-voi-serv)#session transport tcp
B. Router(config-voi-serv)#url sips
C. Router(config-voi-serv)#timers expires 1000
D. Router(config-voi-serv)#srtp fallback
16. Which of the following show commands is used to view SIP calls that are being made
through the voice gateway in real time?
A. show sip-ua connections
B. show sip connections
C. show sip calls
D. show sip-ua calls
17. Which three functions does MGCP handle?
A. Send messages between the MGCP voice gateway and CUCM using SDP messages.
B. Provide a separate channel for backhauling Q.931 traffic from the PSTN to the
CUCM.
C. Use communications messages between the MGCP and CUCM.
D. Provide in-band signaling to backhaul Q.921 signaling between the PSTN and CUCM.
E. Learn the location of target endpoints.
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18. What MGCP configuration is operational by default when configuring an MGCP
residential gateway?
A. rtp-package
B. pre-package
C. dtmf-package
D. fxr-package
E. line-package
19. You are reviewing a Cisco voice gateway configured with MGCP. You perform a show run
and see the following two configuration commands:
mgcp call agent 10.10.100.100 service-type mgcp
ccm-manager mgcp
Based on this, which of the following is true?
A. This is an MGCP residential gateway.
B. The voice gateway connects to a CUCM.
C. The voice gateway connects to a call agent that may or may not be a CUCM.
D. This is an MGCP trunking gateway.
20. Which MGCP show command shows if MGCP is enabled and operationally active?
A. show mgcp statistics
B. show ccm-manager
C. show mgcp profile
D. show mgcp
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1. C. When using H.323 slow start, H.323 must wait for the call setup, call proceeding,
alerting, and connect stages to nish.
2. D. MGCP lets the call agent (CUCM) handle dial-peer functions.
3. A, D. H.245 exchanges information about capabilities such as encryption, ow control,
and jitter data between H.323 endpoints. Additionally, the protocol is responsible for
opening and closing media streams but is not responsible for the actual transport.
4. D. The correct command to modify H.323 initiation is call start slow within
conf-serv-h323 mode.
5. B. H.323 fast start operates on devices that support H.323 version 2 or higher. If you have
legacy equipment that does not support at least H.323 version 2, you must congure slow
start initialization.
6. C. The correct config-dial-peer command is voice-class codec 10. Also remember
that codec preference commands can be used only on VoIP dial peers.
7. B. While in config-if mode for the loopback interface, you must rst congure the
correct IP address and then bind it to H.323 by issuing the h323-gateway voip bind
srcaddr 172.16.3.1 command.
8. B. The show h323 gateway h225 command displays H.323 call-control messages that
are the responsibility of H.225. The details such as number of sent, received, and failed
messages increment for H.225 requests such as setup, alert, and call proceeding.
9. B. SDP messages are sent and received in ASCII format.
10. C. When SIP is congured for early offer, SDP messages are transported in invite messages
from the initiating device to the target device.
11. D. SIP voice gateways terminate both ISDN Q.921 and Q.931 at the voice gateway
interface so that signaling is transferred between the voice gateway and the PSTN. Q.931
signaling never reaches the CUCM when SIP is used for signaling.
12. B. SIP signals a calling phone when the called phone is busy or ofine or the calling party
does not answer (ring-no-answer).
13. A. By default, dial peers use H.323 signaling. This must be modied. The correct config-
dial-peer command to enable SIP on a dial peer is session protocol sipv2.
14. C. SIP secure (SIPS) encrypts SIP message transmissions. This feature can be enabled while
in config-serv-sip mode by issuing the url sips command.
15. D. It is recommended that when you enable secure RTP, you also use the srtp fallback
command so that the voice gateway falls back to using unencrypted RTP if an end device
does not support it.
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16. D. The show sip-ua calls command displays calls that are being made through the local
voice gateway in real time. Information includes source and destination IP address, E.164
numbers, and codec(s) used.
17. A, B, C. MGCP is a client-server protocol. It uses special communications messages with
the CUCM using SDP on UDP port 2427.
18. E. By default, line-package is operational on MGCP residential gateways. This package
contains the signaling necessary to operate analog voice ports.
19. B. The rst command is used by both residential and trunking gateways, so you cannot
determine if the router is a MGCP residential or trunking gateway. The ccm-manager mgcp
command is only needed when the call agent is a Cisco UCM.
20. D. The show mgcp command shows the MGCP administrative and operational status on
the rst line of the output.
c07.indd 278 21/09/11 11:23 AM
1. Fast start
2. no shutdown
3. 15
4. session protocol sipv2
5. User Agent (UA)
6. Call transfers and disconnects
7. show sip-ua status
8. Q.931
9. fxr-package
10. UDP 2427
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c07.indd 280 21/09/11 11:23 AM
Configuring and
Managing CUCM
Express
OBJECTIVES ARE COVERED IN
THIS CHAPTER:
Implement Cisco Unified Communications Manager
Express to support endpoints using CLI.
Describe the appropriate software components needed to

support endpoints.
Configure DHCP, NTP, and TFTP.
Describe the differences between the different types of

ephones and ephone-dns.
Configure Cisco Unified Communications Manager

Express endpoints.
Chapter
8
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The new CVOICE 8.0 exam requires that test candidates
understand how to congure a basic voice network using a
Cisco Unied Communications Manager Express router.
In addition, the candidate must understand the infrastructure required to support IP
endpoints. This chapter covers the current options for powering IP phones on a network,
and it shows how to congure VLAN trunks and VLAN voice access ports and
network infrastructure services that support voice, including DHCP, NTP, and TFTP.
The remainder of the chapter covers using the CUCM Express IOS command-line software
to congure and verify voice settings and operational status for both the SCCP and
SIP protocols.
Voice Network Infrastructure
Considerations
There are several network infrastructure factors that you must consider when implementing
a CUCM Express or any voice system over an IP network. In this section we will cover IP
phone power options, voice VLAN congurations, and network services such as DHCP
and NTP that support the use of voice on a network.
Power Options for IP Phones
Cisco IP phones, being much more than simple analog telephones of old, require a power
source to operate. Currently there are three ways of providing power to Cisco IP phones:

Power brick

Powered patch panel/power injector

Power over Ethernet (PoE) switch
Lets briey review each of these IP phone power methods.
Power Brick
The power brick is the simplest to understand. It connects to a power port on the back of
the phone and plugs into a standard 110v AC wall outlet. You then connect a Category 5 or
higher Ethernet cable into a switch to provide network connectivity.
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The power brick option may be useful in situations where you will use only a handful
of phones. Otherwise, you may want to investigate a PoE option, because it can be more
cost effective and, quite simply, its nicer to combine power and Ethernet in one cable to
eliminate the need for a second connection to the phone.
Powered Patch Panel/Power Injector
A second power option is to have a device that sits between your IP phone and switch
(which is not PoE capable). This is known as a midspan method, because the power
sits in the middle of the connection. A powered patch panel can terminate nonpowered
Ethernet on one end and a powered Ethernet termination point on the other. These patch
panels allow the power to be connected back at the wiring closet, so no power brick is
required and the phone receives both power and Ethernet over a single Category 5 or 6
Ethernet cable.
You can also purchase a Cisco power injector. These devices provide the same midspan
sit-in-the-middle power function as the powered patch panel but only for a single phone
per injector.
Power over Ethernet Switch
The most streamlined and efcient method of providing power to phones (and other
PoE-capable devices) is the Power over Ethernet (PoE) switch. The switch is responsible
for detecting and outputting the required power on each switchport. By adding PoE
functionality to the switch, you have fewer devices that need UPS protection in the event
of a power outage.
There are a couple of gotchas that you need to be aware of when powering Cisco
phones with any PoE option. The rst is to be sure of the type of inline power and quantity
that the phone supports. The second thing to watch out for is ensuring that your switch can
properly handle the power load. Lets look rst at the two inline power methods for Cisco
switches and then at switch power capacities.
Inline Power Method 1: Cisco Inline Power
In its typical fashion, Cisco began offering a proprietary inline-power option to customers
before an open standard was available. In early 2000, Cisco began selling Catalyst switches
with the proprietary inline power (ILP) functionality. ILP uses RJ-45 pins 1, 2, 3, and 6
to provide power to the phones. Using the same wiring that Ethernet uses to transmit and
receive is called phantom power.
Ciscos proprietary inline power provides a xed 6.3W of power to any device that
requires it. ILP detects a capable device by sending a very-low-voltage AC signal across
the transmit pairs and expects to receive the same signal back on the receive pairs.
This is because the ILP-capable phones have a low-pass lter that bridges the specic
voltage signal from TX to RX. Once the switch receives the voltage back on the receive
pair, it knows that the device requires power and sends the 6.3W on that specic
switchport.
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Configuring and Managing CUCM Express

Inline Power Method 2: Cisco IEEE 802.3af
In mid-2003 the IETF released the 802.3af PoE standard. This became the de facto
standard for powering Ethernet over 10/100 and 1000Base-T. The standard states that
power can be sent across the Category 5/6 cabling either on an active transmit/receive pair
or over the inactive pairs for 10/100Base-T. 1000Base-T has the requirement of using either
pins 1, 2, 3, and 6; or 4, 5, 7, and 8 for power. Cisco uses pins 1, 2, 3, and 6 on its
802.3af-supported PoE switches.
The 802.3af standard handles endpoint detection using a different method than ILP. It
uses a low-power DC signal sent across a copper pair. Just as in ILP, the voltage is looped
back to the switch by a slightly more advanced lter to signal that the end device is capable
of receiving power. Unlike ILP, 802.3af has ve different classes of power that it can
transmit. It knows the power level the end device requires by the voltage strength that it
receives back during the detection phase. Table 8.1 lists the 802.3af power classications.
Class 0 is the default class and allocates a full 15.4W of power to any device that falls
into the category. This class is for devices whose vendor did not choose to implement a
power classication. Youll commonly nd this in inexpensive PoE products. Moving up,
a device that declares itself as class 1 will have a max power requirement level between
0.44W and 3.84W. The switch allocates 4.0W of power for these devices. Class 2
allocates 7.0W for devices requiring a maximum power level of 3.84W to 6.49W. Class 3
is for any device that requires 6.49W to 12.95W, and the switch allocates 15.4W of power.
Class 4 is not currently in use but was set aside so an additional power level can be added
in the future.
Cisco Inline Power Switch Backward Compatibility
Because Cisco jumped the gun a few years early with its pre-standard ILP, it now faces
the need to support the newer 802.3af as well as its own proprietary ILP standard on its
TABLE 8.1 IEEE 802.3af classifications
Class Usage
Min Power Level at
the Switch (in Watts)
Max Power Levels at
the Device (in Watts)
0 Default 15.4 0.4412.95
1 Optional 4.0 0.443.84
2 Optional 7.0 3.846.49
3 Optional 15.4 6.4912.95
4 Reserved for
future use
N/A N/A
c08.indd 284 9/21/11 11:24:38 AM
Catalyst line of PoE switches. The methods of power detection are fairly different between
ILP and 802.3af, and Cisco has come up with a method that allows its switches to detect
the power requirements of Cisco phones. Here are the steps the PoE switch goes through
for powering Cisco IP phones:
1. The switch detects that the IP phone is PoE capable and sends the ILP power
amount of 6.3W.
2. If the phone is only capable of using the ILP proprietary inline-power method, the
phone boots normally and the process ends. If the phone uses the 802.3af power
method, the phone will boot into low-power mode using the 6.3W of power
provided to it on the port.
3. Once the phone boots into low-power mode, it exchanges Cisco Discovery Protocol
(CDP) messages with the switch and negotiates which 802.3af class the phone should
reside in. CDP is a Cisco proprietary Layer 2 messaging protocol that is commonly
used between Cisco devices to determine who their neighbor is and what their
capabilities are.
4. When the negotiation process is complete, the switch provides the necessary power
to fully boot the IP phone.
CDP must be enabled on both the switch and switchport for it to negotiate
power with Cisco PoE endpoints. If CDP is disabled, the switch will have no
choice but to allocate the maximum amount of power for the 802.3af class
the device belongs to.
Cisco PoE Intelligent Power Management
Depending on the types of endpoints you deploy and the type of switch and power supply
used, you need to be aware that you can eventually exhaust the amount of power available
to the switch. If you add too many PoE phones to a switchport, the switch may have
allocated all the available power, so that your device will not receive the necessary electricity
to power the phone. There is also the fact that the 802.3af classication system can often
set aside more power than is necessary, which can unnecessarily limit the number of PoE
devices that can be powered. That is why Cisco offers multiple congurable modes on its
PoE-capable switches. This is known as Intelligent Power Management (IPM). This section
shows how to congure IPM modes.
The power inline IOS commands allow you to change PoE settings on a port-by-port
basis. Lets look at the PoE interface commands available to us:
4506-switch(config-if)#power inline ?
auto Automatically detect and power inline devices
never Never apply inline power
static High priority inline power interface
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The auto setting is the default. If the endpoint is a Cisco device such as a Cisco 7965
IP phone, the power settings will be negotiated automatically. To prove this, lets look at a
show power inline command output on the 4500 switch:
4506-switch#show power inline gigabitEthernet 3/2
Interface Admin Oper Power(Watts) Device Class
From PS To Device
--------- ------ ---------- ---------- ---------- ------------------- -----
Gi3/2 auto on 13.5 12.0 Cisco IP Phone 7965 3
Interface AdminPowerMax
(Watts)
---------- ---------------
Gi3/2 15.4
You can see that the Admin setting is set to auto. Using CDP, the switch detected the
Cisco 7965 phone. The switch placed it into 802.3af power settings as a class 3 device
and thus allocated 15.4W of power to it. However, the switch went one step further and
dropped the power output to the device to 12.0W.
You can use the never option if you choose to not provide any power on that specic
port. Finally, the static option can be useful if you have non-Cisco phones that you
know use only a specic amount of power. This helps with power budgeting and reduction
of surprises.
Configuring VLANs and Voice VLANs
CVOICE candidates should have a thorough understanding of VLANs and the benets
they provide for devices on a network. Those benets include limiting broadcast messages,
improved security, and logical segmentation based on business processes. These benets
can be extended into the VoIP realm as well. Because CVOICE candidates must know how
to install and congure CUCM Express endpoints on an IP network, we will review VLAN
trunking and how to assign both voice and data VLANs to access-layer ports.
Configuring VLAN Trunks
Lets take a very simple network example that has two VLANs. VLAN 10 is the Sales
VLAN and VLAN 20 is the Marketing VLAN. Our network consists of two Cisco
switches that are connected through a single Fast Ethernet port. What if you have Sales and
Marketing employees connected to multiple switches, but you would like them to reside in
the same logical VLAN? The solution to this problem is to use the link connecting the two
switches as a VLAN trunk port. Figure 8.1 shows our new network topology with
two switches that have VLANs 10 and 20 trunked between them.
c08.indd 286 9/21/11 11:24:39 AM
A trunk port is a link between two Layer 2 switches that can transport trafc from
multiple VLANs. It keeps the trafc between the VLANs separate by tagging each frame.
VLAN tagging essentially places a VLAN identier on each frame. In our example,
frames on VLAN 10 that need to go from one switch to the other are tagged as belonging
to VLAN 10. By far the most common trunk method is 802.1Q, which is what we will
implement in detail here.
Using Figure 8.1 as our example, lets congure a trunk link between switches A and
B using the 802.1Q trunking protocol on port Fa0/1. For simplicitys sake, we will
assume that both switch A and switch B have been identically congured to switches
VLAN 10 and 20. Conguring an 802.1Q trunk between the switches requires the
following steps.
VLAN 10
Switch B
Fa0/1
Fa0/1
Trunk VLAN
10 and 20
VLAN 20
Switch A
FI GURE 8.1 A VLAN trunk
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288 Chapter 8

Step 1: Configure the Trunk Encapsulation Type
The rst step is to set the encapsulation type for the trunk interface. The command for
conguring trunk encapsulation is:
switchport trunk encapsulation [dot1q|isl]
Step 2: Configure the Trunk Mode
There are several options for the trunks operational mode, including dynamic desirable
and dynamic auto. But since we know we want to congure the port as a trunk, we can
simply hard-code the port using the switchport mode trunk command. Lets congure
Switch-A and Switch-B for trunking on port Fa0/1.
Here is the conguration for Switch-A:
Switch-A#configure terminal
Switch-A(config)#interface fa0/1
Switch-A(config-if)#switchport trunk encapsulation dot1q
Switch-A(config-if)#switchport mode trunk
Switch-A(config-if)#end
Here is the conguration for Switch-B:
Switch-B#configure terminal
Switch-B(config)#interface fa0/1
Switch-B(config-if)#switchport trunk encapsulation dot1q
Switch-B(config-if)#switchport mode trunk
Switch-B(config-if)#end
So now we have a two-switch network with two VLANs that are properly trunked
together. A Sales department user on a PC on Switch-A, VLAN 10, can communicate with
another Sales department user attached to Switch-B on VLAN 10. The same is true for the
Marketing department users on VLAN 20.
While it is true that users on VLAN 10 of Switch-A can communicate with
users on VLAN 10 of Switch-B, users on VLAN 10 cannot communicate
with any users on VLAN 20 or vice versa. Layer 3 routing is required to
perform this functionality. This topic is outside the scope of the CVOICE
exam and has not been included in this study guide.
Configuring and Verifying Voice VLANs
Many Cisco mid- and high-range phones such as the 7945G give users the ability to plug a
PC into an Ethernet port on the phone to provide network connectivity. The phone essentially
becomes a three-port switch at that point. One port (port 0) connects the phone to the access-
layer switch, the second virtual port (port 1) is for voice trafc to the phone, and the third port
(port 2) is to connect to a PC for standard data transport. Figure 8.2 shows how a PC is plugged
directly into the phone, which is essentially trunked with both a voice and a data VLAN.
c08.indd 288 9/21/11 11:24:40 AM
As you can see, the connection between the switch and the Cisco phone is an 802.1Q
trunk link. It is necessary to have a trunk because we have our voice and data separated
on two different VLANs. When conguring the trunk on the switchport that connects
to the phone, we use a slightly different method. The Cisco IOS has a unique method to
identify a VLAN specically as a voice VLAN. In all actuality, this trunk link between our
switch and the Cisco phone is not a full-edged 802.1Q trunk like those we have practiced
conguring between two switches and a switch and router. Instead, the Cisco switch and
Cisco IP phone use CDP to implement this quasi-trunk. The VLAN that is congured
as the voice VLAN is marked with an 802.1Q tag, while the data VLAN is considered
to be the native VLAN and is left unmarked. This trunk is capable of handling only two
VLANsone tagged VLAN for voice and one untagged VLAN for data.
It used to be that the trunk link between the access switch and Cisco IP
phone was indeed a full-blown 802.1Q trunk. Unfortunately, it was easy to
fool this setup, and PCs could easily join the voice VLAN and use sniffers
to collect and re-create voice calls. Because the new quasi-trunk setup
uses CDP to identify which devices can join the voice VLAN, the new
method is much more secure. But the 802.1Q trunks to phones are still
used when third-party phones make use of built-in switch functionality
but cannot use Ciscos CDP messaging protocol. You also may run into
installations with older Cisco equipment (such as the NM-16ESW) that
cannot use voice VLANs and instead must rely on 802.1Q trunks.
With that understanding of the voice VLAN, lets go back to our two-switch network
with data VLANs 10 and 20 and say we have congured a new VLAN (VLAN 100) for
voice. We attach an IP phone to port fa0/5. We then need to congure the port to access
our new voice VLAN (using the switchport voice vlan command) and the Sales VLAN
to the Cisco phone according to Figure 8.2.
With this information, we can congure the switchport to quasi-trunk our voice and
data VLANs to our Cisco IP phone:
Switch#configure terminal
Switch(config)#interface fa0/5
Switch(config-if)#switchport voice vlan 100
Switch(config-if)#switchport access vlan 10
Switch(config-if)#end
FI GURE 8. 2 A Cisco IP phone switch
Switch
Fa0/5 Trunk link
Cisco phone
PC
Voice VLAN
Data VLAN
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To verify, we can run the show vlan brief command to ensure that our port fa0/5 is in
both VLAN 10 and VLAN 100:
Switch#sh vlan brief
VLAN Name Status Ports
---- -------------------------------- --------- -------------------------------
1 default active Gi0/1, Gi0/2
10 Sales active Fa0/1, Fa0/2, Fa0/3, Fa0/4
Fa0/5, Fa0/6, Fa0/7, Fa0/8
Fa0/9, Fa0/10, Fa0/11, Fa0/12
Fa0/13, Fa0/14, Fa0/15, Fa0/16
Fa0/17, Fa0/18, Fa0/19, Fa0/20
Fa0/21, Fa0/22, Fa0/23, Fa0/24
20 Marketing active Fa0/25, Fa0/26, Fa0/27, Fa0/28
Fa0/29, Fa0/30, Fa0/31, Fa0/32
Fa0/33, Fa0/34, Fa0/35, Fa0/36
Fa0/37, Fa0/38, Fa0/39, Fa0/40
Fa0/41, Fa0/42, Fa0/43, Fa0/44
Fa0/45, Fa0/46, Fa0/47, Fa0/48
30 Management active
100 Voice active Fa0/5
1002 fddi-default act/unsup
1003 trcrf-default act/unsup
1004 fddinet-default act/unsup
1005 trbrf-default act/unsup
Switch#
Sure enough, port Fa0/5 belongs to both the Sales (VLAN 10) and Voice (VLAN 100)
VLANs.
Network Infrastructure Services for VoIP Support
A CUCM Express or other Cisco Layer 3 device such as a router or multilayer switch can
be congured to provide Dynamic Host Control Protocol (DHCP) services to your phones
to assign IP addresses and other network information to the phones dynamically. One of
these devices can also serve as a centralized point for synchronizing your UC equipment
clocks by being the Network Time Protocol (NTP) point of reference. Lets look at how we
congure both of these network services for our voice network.
Configuring DHCP for Voice Functionality
DHCP allows an endpoint device (such as a Cisco IP phone) to boot up on the network
and request network information, which it dynamically receives from a DHCP server. This
section will show you how to congure DHCP on your CME router for your end devices.
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DHCP server functionality is considered to be a service on your IOS router and is
enabled by default on IOS versions 12.0(1)T and later. The rst step in your DHCP server
conguration process is to ensure that specic IP addresses on your network are never
handed out to endpoints. IPs such as default gateways and other static interfaces that are
already in use must be specically excluded. To do this, you use the ip dhcp excluded
address command followed by the beginning and ending IP addresses you wish to exclude.
In our example, we will exclude the rst 20 addresses of the pool.
The next step is to actually create our DHCP pool and give it a name using the ip dhcp
pool command. The following shows how we dene a pool and exclude specic addresses:
Router(config)# ip dhcp excluded-address 192.168.100.1 192.168.100.20
Router(config)# ip dhcp pool voip-pool
Router(dhcp-config)#
As soon as you give a name to your DHCP pool, you are placed into dhcp-config
conguration mode. This is where you actually create your IP scope with the network
command and any additional DHCP information you want to give to the endpoints.
Following are the common parameters for endpoints.
Default-router
This parameter is mandatory for all endpoints. It tells the endpoint what IP address it
should use for its default-gateway.
Domain-name
Species the domain name you want your endpoints to use.
DNS-server
Informs the endpoints about the IP addresses of their DNS servers for name resolution. You
can specify up to eight DNS servers with a single command.
Lease
This command allows you to specify how long an endpoint is to maintain the dynamically
assigned IP address. You can specify the number of days, hours, minutes, or even if it can
maintain the address innitely.
Option
Another critical parameter that you will want to congure when setting up DHCP for your
Cisco IP phones is the IP address of the TFTP server where the Cisco phone conguration
les are located. All Cisco phones (SIP and SCCP) must download a conguration le when
they rst boot. This le contains important information required for the phone to function
properly with the CUCM Express. The IP phones must know the location of the TFTP
server so they can request the conguration le. The DHCP option command is followed
by a specic numeric parameter for the additional DHCP information you wish to send to
endpoints. In the case of TFTP, that parameter number is 150. You can then specify the IP
address or domain name of the TFTP server.
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Here is an example of DHCP conguration parameters, using the following information:

Network: 192.168.100.0/24

Default router: 192.168.100.1

Domain name: ccnavoice1.com

DNS server: 192.168.10.5

TFTP server: 192.168.100.10

Lease time: 6 hours
Router(dhcp-config)#network 192.168.100.0 255.255.255.0
Router(dhcp-config)#default-router 192.168.100.1
Router(dhcp-config)#domain-name ccnavoice1.com
Router(dhcp-config)#dns-server 192.168.10.5
Router(dhcp-config)#option 150 ip 192.168.100.10
Router(dhcp-config)#lease 0 6 0
Router(dhcp-config)#end
Once those conguration steps are complete, when an IP phone congured on the correct
voice VLAN boots, it will request an IP address from the DHCP server. Our DHCP server
will return an IP address, default gateway, DNS server, and TFTP server along with a lease
time of 6 hours for the information to be valid for the phone.
On large networks with multiple VLANs, there typically is a server (or
cluster of servers for redundancy) that is responsible for all DHCP requests
on a network. Because DHCP clients use broadcasts to attempt to find the
DHCP server, it would be necessary to configure a DHCP server on each
and every VLAN that required DHCP services. Obviously, this does not
scale well. Instead, we can use the ip-helper address command followed
by the IP address of the remote DHCP server. The command is configured
on Layer 3 interfaces such as a router interface or switched virtual inter-
face on a multilayer switch. This is known as DHCP relay. This command
enables the Layer 3 interface to listen for DHCP broadcast requests and
send a separate broadcast message to the DHCP server identified. When
the DHCP server replies to the Layer 3 interface, the router/multilayer
switch then relays that information onto the original requestor.
Monitoring and Troubleshooting the DHCP Service
You can monitor your DHCP service with the following useful show commands:
show ip dhcp binding
Use this command to display the dynamic IP-to-MAC address mappings. It also lets you
know when a specic lease will expire. The following example shows the binding for the
DHCP leased IP address 192.168.100.101:
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Router# show ip dhcp binding 192.168.100.101
IP address Hardware address Lease expiration Type
192.168.100.101 00a0.9802.32de Mar 01 2009 12:00 AM Automatic
show ip dhcp conflict
This command lists any IP address conicts and the time the detection occurred. It also
indicates the method of conict detection. The next example shows a conict for the IP
192.168.100.101:
Router# show ip dhcp conflict
IP address Detection Method Detection time
192.168.100.101 Ping Mar 01 2009 12:28 PM
Configuring the Network Time Protocol
NTP should be congured on every single piece of network equipment in a production
network. It is very important to have synchronized times for all of your logging information.
It is good practice to specify two devices on a network that have access to a public time
source from the National Institute of Standards and Technology (NIST). Keep in mind that
NTP runs over UDP port 123, so make sure you have this port opened on your rewall
rule-set to allow access.
The conguration of a time source on a Cisco IOS device is quite simple. This conguration
is used to point your CUCM Express at a designated NTP server that resides on your network.
First, you should specify the time zone that your local equipment resides in, using the clock
timezone command. Next, you issue the command ntp server and specify an IP address of
one of the public time servers. Then all of your other network devices can be congured to
peer with the device receiving an external clock. For example, heres the conguration of a
router for an external NTP server that uses the external time source IP of 192.5.41.41:
Router(config)#clock timezone CHICAGO -6
Router(config)#ntp server 192.5.41.41
Router(config)#end
An Overview of CUCM Express
CUCM Express is a unique offering from Cisco. Instead of the server- and Linux-based
solutions of the CUCM and CUCM Business Edition platforms, CUCM Express runs
on a specialized version of IOS software and Cisco ISR hardware. The ability to have
a fully functioning router, voice system, and optional voicemail messaging system in a
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single device greatly appeals to small businesses and branch ofces that wish to limit the
amount of hardware and administration duties. This section will cover the primary voice
capabilities found in CUCM Express along with its supported hardware and maximum
number of IP phones. Lastly, we will look at the software licenses required to legally
operate CUCM Express.
Understanding CUCM Express Capabilities
Cisco Unied Communications Manager Express is geared toward small and medium-size
environments. CUCM Express combines all of the following IOS software capabilities into
a signally managed device:

Call processing agent

Call setup and routing

Dial plan administration

Phone feature administration

Telephone directory services

Services such as music on hold (MOH), paging, intercom, hunt groups, interactive
voice response (IVR), and call detail records (CDR)

Voice gateway

Translation between PSTN and IP networks, transcoding, and compression

Ability to provide SRST

Direct termination of PSTN ports

DSP service capabilities

Optional voicemail

A Unity Express voicemail module can be installed directly into the router,
providing voicemail services
CUCM Express can be managed either by using the command-line
interface (CLI) or through an optional web-based GUI.
For internal IP-to-IP voice calls, CUCM Express uses the concept of virtual dial peers
instead of the static dial peers that we are accustomed to conguring on voice gateways.
These virtual dial peers are created automatically as we dene telephones and directory
numbers. You will learn how to congure IP phones in a CUCM Express environment later
in this chapter.
CUCM Express can support a wide range of Cisco and third-party vendor phones and
common voice codecs as IP voice endpoints, including G.711, G.729, and iLBC. CUCM
Express supports SCCP and SIP signaling protocols to end devices. In addition, if you are
operating Cisco IP phones in SIP mode, CUCM Express can offer additional voice features
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that are not typically found in generic SIP environments. The features available depend on
the Cisco phone model being used.
Understanding CUCM Express Hardware Requirements
From a hardware perspective, CUCM Express has the capability to support up to 450
IP phones on a single-system deployment. A multisite deployment design can also be
implemented by interconnecting multiple CUCM Express systems using a signaling
protocol such as H.323. The maximum number of supported IP phones depends on the
CUCM Express ISR hardware model, as shown in Table 8.2.
TABLE 8. 2 IP phones supported by ISR and ISR G2 CUCM Express
Router Model Maximum Supported IP Phones
1861 ISR 15
2801 ISR 25
2811 ISR 35
2901 ISR G2 35
2821 ISR 50
2911 ISR G2 50
2851 ISR 100
2921 ISR G2 100
2951 ISR G2 150
3825 ISR 175
3845 ISR 250
3925 ISR G2 250
3945 ISR G2 350
3925E ISR G2 400
3945E ISR G2 450
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Understanding CUCM Express Software Licensing
One of the more complex tasks required when ordering Cisco voice equipment is the way
Cisco handles licensing structures. There are three Cisco licenses needed to run your
CUCM Express system and Cisco phones on your network:

Cisco IOS license for voice capabilities

CUCM Express feature license

Individual user licenses for the total number of Cisco phones
In this section well review each of these so you can properly license and run a CUCM
Express system and Cisco IP phones.
IOS Licenses for Voice
The rst license allows you to download and operate a version of Cisco IOS that has
CUCM Express functionality. When you purchase a router, it comes with an IOS feature
set with which it can run the router. It also allows you to download and install new
versions of this IOS feature set when they become available.
CUCM Express Feature Licenses
Just because you own the license to run the voice-capable IOS image doesnt mean you can
start adding Cisco IP phones! The second license you need is the CUCM Express feature
license. This license determines how many phones you can run on the CUCM Express. It is
sold in bundles; the smallest bundle is for 25 Cisco IP phones. Table 8.3 shows the current
CUCM Express feature license bundles available.
TABLE 8. 3 CUCM Express 7965 feature license bundles
License Description
FL-CCME-250 CUCM Express support for up to 250 IP phones
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As you can see, you are given several ordering choices for a single phone! The CP-
7965G= is simply a spare phone. It does not come with a license. These are most commonly
purchased to serve as cold spares at businesses. If a licensed phone on the network
were to break, it could be replaced with the unlicensed spare as a one-to-one trade. These
unlicensed phones are less expensive but can be used only as replacements.
The other two license options are for either the CUCM/CUCMBE or the CUCM
Express call-processing systems. The pricing is slightly different for these two parts. The
CH1 licenses are more expensive than the CCME licenses, but the CH1 licenses can
legitimately be used by the larger CUCM system. By contrast, the CCME licenses cannot
be used for the CUCM/CUCMBE systems. So if you think you may upgrade from a CUCM
Express system to one of the bigger CUCM systems, you may want to go ahead and
purchase the CH1 licenses so you wont have to purchase phone user licenses twice.
New Software-Activated Voice Licensing
With the Cisco Generation 2 ISR platforms, there will soon be a new licensing approach that
uses a software activation process to license voice package options. The various voice licenses
can be purchased in bundles for different voice network sizes. The licenses are activated and
enforced within the IOS software. Additional licenses can easily be purchased and added to
the currently activated licenses. Also, if you currently have the older right-to-use licenses,
dont despair because they will be fully transferrable to the new licensing structure. At the
time of this writing, the new software activation method is not yet operational, but be on the
lookout for it soon!
Initial CUCM Express Configuration
Cisco IP phones rely on external sources to receive information such as the rmware
and conguration les. This section details the les that the phones require and how to
Cisco Phone User Licenses
Finally, you need the Cisco phone user license. When you place an order for Cisco phones,
you are given license options for each Cisco phone. For example, Table 8.4 lists the part
numbers and descriptions for the 7965G phone.
TABLE 8. 4 Cisco IP phone part numbers
Part Number Description
CP-7965G= Spare phone w/o license
CP-7965G-CH1 Phone w/ CUCM user license
CP-7965G-CCME Phone w/ CUCM Express user license
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congure them on the CUCM Express router. First, you will see how to turn CUCM
Express into a TFTP server to offer up specic Cisco phone rmware les. Then well move
on to the four mandatory CUCM Express system congurations needed to support IP
phones. Finally, you will see how to congure and generate individual phone conguration
les to allow each Cisco phone to have unique functionality within the voice system. After
all these steps are completed, your Cisco phone can successfully connect to its host CUCM
Express and use the information gathered to function as a VoIP phone.
Configuring CUCM Express as a TFTP Server
When a Cisco IP phone initially powers up, it will use CDP to determine the voice VLAN
it should belong to and then request and receive, at a minimum, an IP address/subnet
mask and gateway IP address via DHCP. It also must have the all-important option 150 IP
address, which is the location of the TFTP server. As youve already learned, for voice the
TFTP server is responsible for delivering Cisco phone rmware and conguration les to
the phones when requested. The TFTP server can be located anywhere on your network,
but in smaller environments, the CUCM Express router is congured for TFTP. This is
the rst server the IP phone gets its information from. One group of les that our Cisco
IP phone will request is its rmware, which is specically tailored to the type of Cisco
phone hardware. If you are using your CUCM Express router to handle TFTP server
functionality, you must congure the IOS to serve up the rmware that your phones will
request. All you need to do is gure out what Cisco phones you will want to allow on your
network and then congure the router to serve the appropriate les. You can see all of the
rmware le directories by issuing the dir flash:/phone command:
Directory of flash:/phone/
47 drw- 0 Apr 7 2009 18:18:28 +00:00 7945-7965
56 drw- 0 Apr 7 2009 18:18:56 +00:00 7937
58 drw- 0 Apr 7 2009 18:19:24 +00:00 7914
60 drw- 0 Apr 7 2009 18:19:26 +00:00 7906-7911
69 drw- 0 Apr 7 2009 18:19:52 +00:00 7920
71 drw- 0 Apr 7 2009 18:19:58 +00:00 7931
79 drw- 0 Apr 7 2009 18:20:24 +00:00 7942-7962
88 drw- 0 Apr 7 2009 18:28:46 +00:00 7921
96 drw- 0 Apr 7 2009 18:29:30 +00:00 7940-7960
101 drw- 0 Apr 7 2009 18:29:38 +00:00 7970-7971
110 drw- 0 Apr 7 2009 18:30:06 +00:00 7975
118 drw- 0 Apr 7 2009 18:30:34 +00:00 7941-7961
511664128 bytes total (395001856 bytes free)
Lets assume that we are going to be conguring Cisco 7945 and 7965 phones in our
environment. Therefore, we need to congure our TFTP server to offer all of the les
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within the flash:/phone/7945-7965 directory. Notice that some of the rmware les work
for multiple phones, including the rmware for the 7945 and 7965.
Conguring the CUCM Express router to serve as a TFTP server for the rmware
les is quite simple. Each le needs to have its own tftp-server flash:/phone/
<firmware_file> command. Also note that because our CUCM Express les are organized
with a directory structure, we must provide a directory alias for the Cisco phones. Cisco
phones are unintelligent devices for the most part. They know only the name of the rmware
les but not where they are located. Because our CUCM Express phone rmware software
is organized into directories, we must create aliases so that when the Cisco phone asks for a
le, it knows in which subdirectory the le is located.
We can look at all of the 7945-7965 phone rmware les inside the directory by issuing
the dir flash:/phone/7945-7965 command:
Router#dir flash:/phone/7945-7965
Directory of flash:/phone/7945-7965/
48 -rw- 2496963 Apr 7 2009 18:26:30 +00:00 apps45.8-5-3S.sbn
49 -rw- 585536 Apr 7 2009 18:26:34 +00:00 cnu45.8-5-3S.sbn
50 -rw- 2453202 Apr 7 2009 18:26:44 +00:00 cvm45sccp.8-5-3S.sbn
51 -rw- 326315 Apr 7 2009 18:26:46 +00:00 dsp45.8-5-3S.sbn
52 -rw- 555406 Apr 7 2009 18:26:48 +00:00 jar45sccp.8-5-3S.sbn
53 -rw- 638 Apr 7 2009 18:26:50 +00:00 SCCP45.8-5-3S.loads
54 -rw- 642 Apr 7 2009 18:26:50 +00:00 term45.default.loads
55 -rw- 642 Apr 7 2009 18:26:52 +00:00 term65.default.loads
These phones will need all eight les to function properly using SCCP. If you are
going to run the phones in SIP mode, you need to install the necessary SIP rmware
les. To offer these les up for downloading to the phones, you need to congure
the following:
Enter configuration commands, one per line. End with CNTL/Z.
Router(config)#tftp-server flash:/phone/7945-7965/apps45.8-5-3S.sbn alias
apps45.8-5-3S.sbn
Router(config)#tftp-server flash:/phone/7945-7965/cnu45.8-5-3S.sbn alias
cnu45.8-5-3S.sbn
Router(config)#tftp-server flash:/phone/7945-7965/cvm45sccp.8-5-3S.sbn alias
cvm45sccp.8-5-3S.sbn
Router(config)#tftp-server flash:/phone/7945-7965/dsp45.8-5-3S.sbn alias
dsp45.8-5-3S.sbn
Router(config)#tftp-server flash:/phone/7945-7965/jar45sccp.8-5-3S.sbn alias
jar45sccp.8-5-3S.sbn
Router(config)#tftp-server flash:/phone/7945-7965/SCCP45.8-5-3S.loads alias
SCCP45.8-5-3S.loads
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Router(config)#tftp-server flash:/phone/7945-7965/term45.default.loads alias
term45.default.loads
Router(config)#tftp-server flash:/phone/7945-7965/term65.default.loads alias
term65.default.loads
Router(config)#
At this point, if you were to add one of these phones to your network, it would receive
all the necessary IP information and download the phone rmware les from the TFTP
server. The phone will not register to CUCM Express, however. It is still missing vital
congurations that must be set up on CUCM Express for the registration process to
occur. The next few sections of this chapter show how to congure CUCM Express to
allow Cisco phones to work with the call processor, and how to identify and serve up
default conguration les to your Cisco IP phones. First youll see how to congure SCCP
signaling, and then youll explore the differences when conguring SIP signaling between
the CUCM Express and IP phones.
Configuring the Mandatory CUCM Express System
Settings Using SCCP Signaling
The majority of CUCM Express conguration tuning happens while in config-telephony
conguration mode. There are four conguration steps that must be accomplished to get
the system to properly register phones for call processing using SCCP signaling:
1. Congure the source IP address for CUCM Express.
2. Congure the maximum number of ephones and ephone-DNs (directory numbers)
allowed on the CUCM Express.
3. Identify and set the rmware load les that Cisco IP phones should request based on
the Cisco phone model.
4. Generate and serve up default phone conguration les via TFTP to the Cisco
IP phones.
Heres a detailed look at each of these steps:
Step 1: Configure the Source CUCM Express IP Address
The source IP address denes the location of the CUCM Express call-processing agent.
All of the Cisco IP phones on the network will use this address for all communications
with the CUCM Express hardware. After a Cisco phone downloads the correct rmware
used via TFTP, it requests and receives generic information about CUCM Express. One
item is the source IP address where CUCM Express can be found. In the example shown
in Figure 8.3, well assume that all of our IP phones reside on the voice VLAN
of 192.168.10.0/24.
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Were going to use the 192.168.10.1 IP address as our source IP for the CUCM Express.
The conguration of the CUCM Express source IP address is as follows:
Router(config)#telephony-service
Router(config-telephony)#ip source-address 192.168.10.1
Router(config-telephony)#end
Router#
Youll see later how this information is eventually packaged within a default
conguration le and sent to all Cisco IP phones on the network.
Step 2: Configure Max Ephones and DNs
Step 2 of our CUCM Express system conguration involves setting the maximum number
of ephones and ephone-DNs. Ephones represent physical phones. They are the way you
identify a particular device within the IOS. Ephone-DNs, by contrast, are the telephone
Switch
Data
VLAN 1
192.168.1.0/24
Voice
VLAN 10
192.168.10.0/24
Cisco phone
Cisco phone
Cisco phone
CUCM Express
Trunk VLAN
1 and 10
Telephony source IP:
192.168.10.1
FI GURE 8. 3 A sample CUCM Express network
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number extensions congured on each phone. Figure 8.4 shows a Cisco phone with buttons
for multiple ephone-DNs. This particular Cisco phone has buttons to handle up to eight
ephone-DNs.
FI GURE 8. 4 Cisco IP phone extension buttons
By default, the maximum number of both ephones and ephone-DNs is 0. You might
wonder why Cisco sets the defaults to 0 if you still have to set them to 1 or more to get a
single phone to work. The answer has to do with memory allocation. When a maximum
number of ephones and ephone DNs is set, the router sets aside memory for each one.
For example, if you set max-ephones to 10 and max-dn to 50, the router allocates memory for
each of the 10 ephones and all 50 ephone-DNs regardless of whether you actually use them
or not. Keep this in mind, because you dont want to set the maximums too high, which
could overtax your router. In our example, were going to set our max-ephones to 8 and our
max-dn to 20:
Router(config-telephony)#max-ephones 8
Router(config-telephony)#max-dn 20
Router#
The maximum number of ephones and ephone-DNs that can be congured depends on
hardware, because different devices have different amounts of memory installed in them.
To show you what happens when you try to go over the maximum setting, lets say that
your max-ephones is 8 and you attempt to add a ninth phone to CUCM Express. When this
occurs, the phone will not be allowed to register and will display a Registration Rejected
message, as shown in Figure 8.5.
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Also, if you exceed the max-dn number, you will receive an error when attempting to
congure the maximum +1 ephone-DN. The following example has a max-dn set to 20, so on
the 21st ephone-DN conguration youll see this log message on the CUCM Express console:
Router(config)#ephone-dn 21
dn 21 exceeds max-dn 20
Router(config)#
Step 3: Identify and Set Firmware Load Files
Step 3 of the CUCM Express system conguration process deals with how you handle the
distribution of rmware for your Cisco phones. In the previous steps, we identied the les
that our Cisco phones need and have congured our router to serve them using TFTP. The
CUCM Express telephony processes must also be congured to set the rmware les you
choose to dene for each phone hardware type. As mentioned earlier, when the phones rst
communicate, they have very little information to begin with and must be told virtually
everything. One piece of information a phone does possess is the hardware type of Cisco
phone it is. This information is then used by CUCM Express to determine which rmware
load le it should request. The rmware load le basically tells CUCM Express what
rmware to tell the Cisco phones to download. It can be a bit difcult to gure out which
rmware load le you need to congure for each phone. The best way to nd out which
load les you need is to search on the cisco.com website for CME X.X rmware, where
X.X is the version of the CUCM Express software you are running.
FI GURE 8. 5 Registration Rejected message
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Once you locate the rmware, you can click the link to display a table listing the
correct load le for each phone that needs to be congured within config-telephony
configuration mode. In Figure 8.6, you can see that you need to congure SCCP45.9-1-
1SR1S.loads as our load le.
FI GURE 8. 6 Cisco phone load file table
You know that you need to use SCCP45.9-1-1SR1S.loads as your key load le because:
1. You are using SCCP as your signaling protocol.
2. The le marked with an asterisk (*) is the load le.
In this example, we are conguring CUCM Express to tell our Cisco 7945 and 7965
phones which rmware load les they should request:
Router(config-telephony)#load 7945 SCCP45.9-1-1SR1S.loads
Updating CNF files
CNF files update complete for phonetype(7945)
Router(config-telephony)#load 7965 SCCP45.9-1-1SR1S.loads
Updating CNF files
CNF files update complete for phonetype(7965)
Step 4: Generate and Serve Default Phone Configuration Files
The default phone conguration le is the XML conguration le that informs a Cisco
IP phone of all the general information it needs to communicate with the CUCM Express
system. Included in this default phone conguration are the source IP address and the port
through which the phones can communicate with the call-processing agent. It also includes
the load conguration lenames we just nished setting up.
At this point, Im referring to the phone conguration les as default because there
is nothing unique about the congurations right now. Once we begin conguring phone
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extensions and other settings unique to the phones, this information will be compiled and
stored as a single phone conguration le. But since none of that information is congured
at this time, the conguration les have only the default information that all the Cisco
phones share.
The phone conguration le is automatically updated every time a change is made that
affects the conguration. For example, if you need to add additional load les for a Cisco
phone, as soon as an addition, subtraction, or change occurs in the telephony-service
conguration prompt, the conguration le updates itself. You can also manually update
the phone conguration le by issuing the create cnf-files command within the
config-telephony command structure. Here is an example of this command:
Router(config-telephony)#create ?
cnf-files create XML cnf for ethernet phone
Router(config-telephony)#create cnf-files
Creating CNF files
Once these four steps have been completed, you can back out of config-telephony
conguration mode and nish your basic SCCP conguration by conguring ephones and
ephone-DNs. Before you do that, however, we will go over the slightly different procedure
for conguring CUCM Express for SIP signaling.
Configuring the Mandatory CUCM Express System
Settings Using SIP Signaling
If you want to use SIP signaling instead of SCCP, then a few of the mandatory CUCM
Express system settings are slightly different. This section goes through the conguration
procedure, noting the differences between commands.
IP Telephony Digit-Collection Methods
While we are discussing SCCP and SIP endpoint protocols, its a good time to talk about
the different digit-collection methods supported by the two. Digit collection refers to the
method that the signaling protocol uses to collect telephone numbers that are entered
using an endpoint such as a Cisco IP phone. There are two primary methods of digit
collection. In the digit-by-digit method, the user picks up the phone and immediately
receives a dial tone. Each digit the user presses into the telephone endpoint is immediately
transferred using the signaling protocol in use. This is how analog telephones operate. It is
also how SCCP and Enhanced SIP phones operate.
The other digit-collection method is called en-bloc. The user enters all the required
telephone digits before receiving a dial tone. Once all digits are entered, the user presses
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1. The rst thing that you need to do is to enable the SIP process to allow SIP calls to be
made between separate networks on CUCM Express. This is a matter of enabling SIP
while in conf-voi-serv conguration mode and issuing the allow-connections sip
to sip command as shown here:
Router#configure trminal
Router(conf-voi-serv)#allow-connections sip to sip
2. Next, you need to specify that your local CUCM Express system is the SIP registrar
server for SIP phones that attempt to register with it. To do so, you must enter into
conf-serv-sip configuration mode and issue the registrar server command.
Then the bind control source-interface command includes the interface that your
source-address is congured on so that CUCM Express will not use the IP layer to
determine the source address for SIP signaling. In the following example, we will bind
our CUCM Express source address to loopback 0 for all SIP control signaling:
Router(conf-voi-serv)#sip
Router(conf-serv-sip)#registrar server
Router(conf-serv-sip)#bind control source-interface loopback0
Router(conf-serv-sip)#end
Router#
3. Now you need to enter into config-register-global structure by issuing voice reg-
ister global and then enter the mode cme command. These two commands are the
SCCP command equivalent of the voice service voip command, and this is where
the majority of the CUCM Express phone settings are congured:
Router#configure trminal
Router(config)#voice register global
Router(config-register-global)#mode cme
the call button, and at that point, a dial tone is received and all digits are sent in one large
block using the signaling protocol. This is how current mobile phone digit collection
works, and it is also how Simple SIP phones operate.
It is important to know what digit-collection type your voice network is using, because
it can affect when dial-peer patterns are matched. For example, say you have two dial
peers. Dial peer 1 has a destination pattern of 123, and dial peer 2 has a destination
pattern of 12345. When digit-by-digit collection is used, dial peer 2 could never be
matched because it will always match after the third digit is sent. However, if en-bloc
digit collection is used, if the user enters the exact extension of 12345, it will indeed
trigger on dial peer 2.
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4. Just as with the SCCP phones, you need to congure your source CUCM Express IP
address, phone rmware load les, your maximum number of hardware phones, and
your maximum number of directory numbers. Following is an example showing how
to congure these settings using the same parameters as our SCCP CUCM Express
conguration so you can see the differences:
Router(config-register-global)#source-address 192.168.10.1
Router(config-register-global)#max-pool 8
Router(config-register-global)#max-dn 20
Router(config-register-global)#load 7945 SIP45.8-5-3S.loads
Router(config-register-global)#create profile
As you can see when comparing the SIP and SCCP congurations, there are some
noticeable differences in syntax. The SIP source-address command is slightly different
from the SCCP ip source-address command but serves the same purpose as when
conguring SCCP. The max-pool command is identical to the max-ephones command, and
the max-dn command remains the same when setting the maximum number of directory
numbers CUCM Express can be congured with.
When working with SIP in a CUCM Express environment, there are some
differences in what physical phones and directory numbers are called
compared to an SCCP setup. When configuring IP phones for SCCP, you
call the physical IP phone an ephone, and when using SIP, it is called a
voice register pool. Also, SCCP ephone-DNs are known as voice register
DNs. You will see examples of how to configure voice register pool phones
and voice register DNs later in this chapter.
Next, the firmware load command is identical to the one for SCCP, but keep in
mind that you need to offer your phones SIP rmware as opposed to SCCP rmware.
Additionally, your TFTP server will need to be congured to offer up the required SIP
rmware les.
Finally, you need to generate your default conguration proles that are similar to the
SCCP cnf-les. This is accomplished using the create profile command, which generates
conguration les that are used for provisioning SIP phones.
Configuring SCCP and SIP Phones
and Directory Numbers
Now that you know how to congure a CUCM Express router for basic operation for both
SCCP and SIP, you need to congure the phones and directory numbers that will connect
to your CUCM Express system and use it as a call-processing agent.
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Configuring Basic SCCP Ephone and Ephone-DNs
Up until now, all of the conguration information that the Cisco IP phones receive from
the CUCM Express has been generic information that all of the phones share. Ephone and
ephone-DN congurations are the way the administrator can control the unique features
that belong to each phone. Well rst look at what an ephone-DN is and how to congure
the most basic type. Then youll learn about ephones and how to apply an ephone-DN to
an ephone.
Configuring an SCCP Ephone Directory Number
An ephone-DN is what we usually think of as a telephone number. This is the extension
that a user dials when they wish to call your phone. On CUCM Express, there are many
different ephone-DN conguration settings that can be used to add functionality, but for
now, all you want to do is add a single ephone-DN to a phone. Youll see different ways
to congure ephone-DN line options later in this chapter. From a directory number (DN)
standpoint, you need to rst create an ephone-DN logical tag. Then, once you are in the
config-ephone-dn conguration mode, you give the ephone-DN an extension number.
Lets congure your rst ephone-DN with an extension of 4001 and a second ephone-DN
with an extension of 4002:
Router(config-ephone-dn)#number 4001
Router(config-ephone-dn)#exit
Router(config-ephone-dn)#end
Router#
Now that you have two directory numbers congured, lets apply them to two Cisco
phones using the ephone conguration command.
Configuring an SCCP Ephone
An ephone conguration is the logical representation of a physical IP phone. This is where
you apply all the unique ephone-DNs and other settings that are ultimately pushed down
to the phone hardware. Every phone on CUCM Express has a unique ephone tag in which
all of the phone congurations are applied. CUCM Express maps the ephone conguration
to the unique MAC address of the phone. By using the MAC address, the phone can
physically move around the network and continue to maintain the same conguration
settings wherever it goes.
Since youre creating the most basic phone conguration, the only information youll
need to congure ephones is the MAC address of each phone and the ephone-DN tag you
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wish to apply. You will also congure two optional but important settingsthe physical
phone type and the preferred codec.
You will also specify the telephone model using the type command followed by the
model number. This lets CUCM Express explicitly know what phone rmware to offer this
particular ephone.
By default, CUCM Express is preconfigured with templates for most of
the popular Cisco IP phones. If the phone you wish to use is not included,
you can create your own ephone-type template using the ephone-type
global configuration command followed by a name to identify your new
phone type template. An optional addon keyword indicates that the phone
type includes the specified add-on module, such as a Cisco 7916 button
expansion side-car module. You will then be placed into config-ephone-
type mode, where you can enter a unique device-id. Additionally, you can
specify the device-type, num-buttons, and max-presentation settings,
which specifically state what your phone is capable of supporting.
You can also set the codec command followed by a codec supported by the phone. This
command sets your preferred codec when calling between two phones that utilize CUCM
Express as their call-processing agent. The default codec is G.711 and will be used if a
codec is not dened here. For phones that connect to other IP phones through VoIP dial
peers, you can use the dspfarm-assist keyword so CUCM Express can negotiate codec
preference for VoIP dial peer calls. Using this optional keyword, CUCM will direct calls
to a DSP farm that will transcode between the specied codec and G.711. Keep in mind
that you will need to make sure you have adequate DSP resources, based on the number of
simultaneous calls that will be transcoded.
Lets congure two Cisco phones with your ephone-DN extension numbers. Ephone 1
will be congured to use extension 4001 and ephone 2 will be congured with extension
4002. Both phones are 7965Gs and prefer to use the G.729r8 codec for local calls only:
Router(config)#ephone 1
Router(config-ephone)#mac-address 0014.1c4d.2589
Router(config-ephone)#type 7965
Router(config-ephone)#codec g729r8
Router(config-ephone)#button 1:1
Router(config-ephone)#exit
Router(config-ephone)#mac-address 0014.4c7f.a49b
Router(config-ephone)#end
Router#
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CUCM Express can automatically assign extensions to brand-new phones
that do not have a specific ephone-to-MAC-address mapping configured.
Using the auto assign command in the config-telephony command
structure, you can specify the hardware types eligible for auto-assign as
well as specify which ephone-DNs should be assigned. As soon as you
power up an eligible phone and it registers to CUCM Express, auto-assign
starts up and builds an ephone configuration. To do this it pulls in the MAC
address of the phone and configures the lowest unused tagged ephone-
dn from the range specified. This option is perfect for new environments
where it doesnt matter who receives a particular extension number or for
fast deployments where editing can come later.
The mac-address conguration is self-explanatory, but the button conguration needs
some explanation. The rst number of the button command indicates the Cisco IP phone
button that is being congured. For example, on a Cisco 7965G IP phone, there are six
extension buttons available, so this number could be 16. On the other hand, a 7945G
phone has only two buttons, so this number could only be 1 or 2. The colon (:) indicates
that you want a standard ring for this extension. There are many different types of audible
and silent rings that well sort out later on, but for now, you just want a standard ring for
your phone. The last number in the conguration species the ephone-DN to apply to
the physical phone. Since we specied that ephone 1 uses ephone-DN 1, the extension on
button 1 of ephone 1 will be 4001. Therefore ephone 2 will be congured to use ephone-DN 2
or extension 4002.
Configuring Basic SIP Voice Register Pools
and Voice Register DNs
Similar to SCCP, SIP setup in a CUCM Express environment requires that you congure
physical phones and logical directory numbers. The processes are similar to the conguration
of SCCP. In fact, some conguration commands are identical. With other commands, the
command syntax is slightly changed, as you will see in the next two sections.
Configuring SIP Voice Register DNs
The directory numbers are congured using the voice register dn command followed
by a unique DN tag, which is used to differentiate multiple directory numbers congured
on the CUCM Express system. After that, you use the number command followed by the
telephone number you wish to associate with a DN. Lets congure two voice register DNs
that are similar to the ephone DNs congured in the previous example:
Router(config)#voice register dn 1
Router(config-register-dn)#number 4001
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Router(config-register-dn)#exit
Router(config)#voice register dn 2
Router(config-register-dn)#number 4002
Router(config-register-dn)#end
Router#
Configuring SIP Voice Register Pools
Again, an easy way to understand the conguration of voice register pools is to compare
them to SCCP ephone-DNs. With voice register pools, you specify the MAC address of
your physical phone, the physical-phone type, and your preferred codec for local calls, as
shown in this example:
Router(config)#voice register pool 1
Router(config-register-pool)#id mac 0014.1c4d.2589
Router(config-register-pool)#type 7965
Router(config-register-pool)#codec g729r8
Router(config-register-pool)#number 1 dn 1
Router(config-register-pool)#exit
Router(config)#voice register pool 2
Router(config-register-pool)#id mac 0014.4c7f.a49b
Router(config-register-pool)#type 7965
Router(config-register-pool)#codec g729r8
Router(config-register-pool)#number 1 dn 2
Router(config-register-pool)#end
Router#
You assign extensions to telephone buttons using the number command followed by
the button number. Then you use the dn keyword to indicate that you wish to set a voice
register DN to the button.
SCCP Ephone-DN Line
Configuration Options
The CVOICE exam requires that you understand different SCCP ephone-DN line options
that are commonly used in both small and large environments. A key-system environment
is commonly used in small businesses where the vast majority of calls are coming from
the PSTN. The key-system model uses a single-line-extension-to-many-phones shared-line
design. Alternatively, a PBX modeled system uses an individual line approach with one
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This section will rst show a typical ephone-DN shared-line conguration, and one that
has two DNs on a single telephone; both are common in key-system environments. You
can then combine these options to congure dual-line and octo-line ephone-DNs, which
are also common in small-business environments. Finally, we will look at typical PBX line
congurations that commonly use single-, dual-, or octo-lines with individual extensions for
each phone.
Configuring Ephone-DN Shared Lines
In a key-system environment, you commonly see the entire PSTN extension congured on
the line instead of a truncated 4- or 5-digit extension. Furthermore, all phones must be
capable of answering any call. That means that all the ephone-DNs will be congured as
buttons on every phone. This is known as a shared line. One way of conguring this shared
line is to congure a single ephone-DN and apply it to multiple ephones. The following key-
system example conguration shows two ephone-DNs that represent two separate external
PSTN phone numbers. The DNs are assigned to both phones, and both will ring when the
number is dialed. The rst phone to answer gets the call:
Router(config-dn)#number 5555552121
extension to one phone model. There are several more directory number line options that
are congurable for SCCP or SIP and SCCP, as shown in Table 8.5.
TABLE 8. 5 SIP and SCCP ephone-DN line option compatibility
DN Option SCCP Compatible? SIP Compatible?
Single-line Yes Yes
Dual-line Yes No
Octo-line Yes No
Shared-line Yes Yes
Two DNs with one telephone Yes Yes
Dual-number line Yes Yes
Overlay line Yes No
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Router(config-dn)#ephone-dn 2
Router(config-dn)#number 5555559191
Router(config-dn)#exit
Router(config-ephone)#button 1:1 2:2
Router(config-ephone)#button 1:1 2:2
Router#
Figure 8.7 shows what ephone-DN 1 looks like after these congurations are made and
the phone is reset.
FI GURE 8. 7 A shared line
Lets say that a phone call is placed to 555-555-2121. Both ephone-DN 1 and ephone-
DN 2 will ring. If ephone-DN 2 answers the call rst, line 1 of ephone-DN 1 shows this
line as in use by lighting the extension button red and using the double-handset icon next
to the line number. Figure 8.8 shows line 1 of ephone 1 in use.
Because line 1 is in use, if the person using ephone-DN 1 needs to make a call, they must
choose to use line 2, which is currently not in use.
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Configuring Two Ephone-DNs with One Number
An alternative shared-line method is to congure two ephone-DNs with the same number.
You then can congure the ephones to use the separate ephone-DN congurations. You set
a preference on the ephone-DN conguration so that one particular phone will always ring
rst. If the preferred ephone is busy, then the next ephone with the lowest preference will
ring instead. This preference is set on the CUCM Express, and its value can be between 0
and 9. You can accomplish this multiple ephone-DN conguration with a shared line using
the preference conguration command. This next conguration example shows how to
congure two ephone-DNs with a single phone number. You can see that ephone-DN 1 has
a preference of 0, which means that when a call is made to this extension, it will ring the
phone that is congured to use ephone-DN 1 rst.
Router(config-ephone-dn)#preference 0
Router(config-ephone-dn)#ephone-dn 2
Line in use
FI GURE 8. 8 DN in use
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Router(config-ephone)#ephone 2
Router#
If you did not configure a preference on the ephone-DNs, or you set them
to be the same, the CUCM Express would round-robin the calls between
the two ephones. The preference command gives you control of where
CUCM Express routes calls.
If ephone-DN 1 is in use, any new call will also be sent to ephone-DN 1 because it is
the lowest preferred DN regardless of whether the phone is busy. So a second call placed to
our extension would receive a busy signal, and ephone-DN 2 would never receive any calls.
To get around this problem, you congure ephone-DN 1 with the no huntstop command.
The huntstop command tells the CUCM Express that it should look for the next preferred
ephone-DN if the most preferred phone is busy. Now when ephone-DN 1 is busy, a second
call placed on the shared extension will roll over and ring ephone-DN 2:
Router(config-ephone-dn)#no huntstop
Router#
Configuring Ephone-DN Dual- and Octo-lines
Another shared-line key-system conguration we need to look at is when the phone extensions
are congured as dual-line and octo-line DNs. So far, weve congured only single-line phones.
A single-line phone can only make and receive one call at a time. So if the line is already in
use, you cannot place the call on hold to make a second call. Likewise, if line 1 is in use, a
second phone call to the extension will receive a busy signal.
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Dual-line phones, on the other hand, allow the phone to place calls on hold or receive
a second call when in use. And octo-line phones are capable of handling up to eight
simultaneous calls on a single phone button extension. Dual- and octo-lines are congured
within the ephone-DN as shown here:
Router(config)#ephone-dn 1 ?
dual-line dual-line DN (2 calls per line/button)
octo-line octo-line DN (8 calls per line/button)
Conguring ephone-DNs with dual lines is extremely benecial because it allows for
additional functionality when your phone is in use. For now, lets assume that your small
business has a single PSTN line that is to be shared between two phones congured with dual-
line ephone-DNs. Just as in the previous conguration example, we want to ensure that the
rst call made to the extension is received on ephone-DN 1 and that a second call rolls over to
ephone-DN 2 if ephone-DN 1 is already in a call. Lets say you congure the following:
Router(config)#ephone-dn 1 dual-line
Router(config-ephone-dn)#ephone-dn 2 dual-line
Router#
In this situation, the rst call will always go to ephone-DN 1. But because the ephone-
DN is congured as a dual-line, a second call will also go to ephone-DN 1. Only a third
simultaneous call will make it to ephone-DN 2. To get around this dual-line problem, you
can use the huntstop channel command on ephone-DN 1. The huntstop command prevents
calls from hunting to the second channel of the ephone-DN. So if you combine the no
huntstop command with the huntstop channel command, you get the result that the rst
call always goes to ephone-DN 1, and if channel 1 of ephone-DN 1 is busy, the second call
will be sent to ephone-DN 2. Here is the full conguration example to accomplish your goal:
c08.indd 316 9/21/11 11:24:55 AM
Router(config-ephone-dn)#huntstop channel
Router#
There are additional phone button options that also expand the shared-line experience
for SCCP phones. The concept of overlay buttons will be explained in the Conguring
Ephone Button Options section of this chapter.
Configuring SCCP Individual Lines
PBX systems, which are more commonly found in larger ofce environments, assign a
unique phone extension to every phone. This allows the caller to reach a specic person
within an organization. Also, because of the size of the environment, a large percentage
of phone calls are on-network calls. To help make life easier for the phone users, phone
extensions are used instead of the full phone number. Typical extensions are four or
ve digits in length. These digits often correspond to the last digits of the full PSTN DID if
there is one. Also, you will nd that the phones almost always are congured as dual-line
ephone-DNs. This is because you need a second line to enable the additional functionality
that the PBX system offers. In the previous section, you learned how to congure the most
common key-system methods of sharing a single phone number with multiple phones. Here
is a very basic and common method of conguring two PBX system phones with separate
extension numbers:
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Router#
Now you have two phones with separate extensions. The CUCM Express system can
then be congured for additional features to tailor your system to your environment.
Next well look at how to congure ephone button options to enhance your SCCP phone
congurations.
Configuring Ephone Button Options
As you saw earlier, when its time to assign ephone-DNs to specic ephones, you use the
button command in ephone-config conguration mode:
button 1:1
The separator between the line button you wish to congure and the ephone-DN
identier is an ephone button separator. There are many different button separator options
available for use. Lets look at all the options available. Table 8.6 details what each of these
button separator functions does:
TABLE 8. 6 Button separator options
Separator Option Name Function
: Normal ring Phone rings normally with default ring tone. Also uses flashing
lights on line button and headset lamp to indicate ring.
s Silent ring No audible ring when calls come into the phone. Uses
flashing lights on line button and headset lamp to indicate
ring. No audible call-waiting beep.
b Silent with
beep
No audible ring when calls come into the phone. Uses
flashing lights on line button and headset lamp to indicate
ring. Call-waiting beep is audible.
f Feature ring Phone rings using an alternate ring tone from the default.
m Monitor line Used to monitor status (on- or off-hook) of a line.
Commonly used on receptionist phones to verify if an
employee is currently using the phone. No audible ring
when calls come into the phone, and the line cannot be
used to make or take calls.
w Watch phone Similar to the monitor mode except it allows the user to
monitor all ephone-DNs on a phone instead of a single
ephone-DN. This mode presents a more accurate picture of
user availability compared to using the m option separator.
c08.indd 318 9/21/11 11:24:56 AM
The ring phone button options (:, s, b, and f) are fairly straightforward and need no
more explanation. Well focus on when you would want to use the monitor and overlay
button options.
Monitor Line (m)
Lets say you have an administrative assistant who is tasked with taking your calls and
transferring them to your phone when you are not busy with other calls. The monitor line
button option allows your assistants phone to monitor your ephone-DN. That way, your
assistant can see if you are currently on a call using that ephone-DN. If you are already
busy on the line, the assistant knows you are busy and can hold all other incoming calls for
you. The line congured in monitor mode cannot make or receive any calls. Instead, it is
simply used as a visual aid to see if another line is being used. In this example, my phone
is assigned the number 4040. My administrative assistant has his own number of 4041
assigned to button 1. Also congured is button 2 to monitor my ephone-DN:
Router(config-ephone)#button 1:2 2m1
Router#
Now when I pick up my phone to make a call, my administrative assistant can see that
Im busy on that ephone-DN. Figure 8.9 shows the administrative assistants phone when
the 4040 line is in use.
Separator Option Name Function
o Overlay line Associates multiple ephone-DNs with a single line button.
No call-waiting functionality.
c Overlay with
call waiting
Same as the overlay line but with call-waiting functionality
added.
x Expansion line Another overlay line option. The difference is that if the line
button extension is in use, new calls are allowed to overflow
to additional line buttons to help prevent a busy signal.
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One of the drawbacks to this setup is what would happen if my phone were to be
congured with multiple ephone-DNs. Multiple monitor button operators would then need
to be created for each extension. A way around the monitor line limitation is to use the
watch phone (w) button separator.
Watch Phone (w)
The watch phone button option does exactly the same thing as the monitor line option,
except that it monitors all of the ephone-DNs of an entire ephone instead of just one
ephone-DN. You congure the button to watch the primary line of a phone, and it
monitors all lines on the phone. This is far more useful than the monitor line option,
because you can see if any of the lines on a phone are in use. Also just like the monitor line
option, a line congured with the watch phone option cannot make or receive any calls.
The status on the watching display button shows the phone in use when the following
conditions occur on the watched phone:

Off-hook and/or in use

The phone is not registered (unregistered or deceased)

In DnD (do not disturb) mode
Overlay Line (o)
Overlay lines allow you to congure multiple ephone-DNs to a single phone button on a
Cisco phone. Cisco phones have a nite number of phone buttons to use. You can use the
FI GURE 8. 9 The phone configured to monitor ext. 4040
Line
in use
c08.indd 320 9/21/11 11:24:57 AM
overlay button option to assign multiple ephone-DNs to a single physical phone button.
Ephone-DNs that are congured on a particular ephone with the overlay option must all be
single-line or dual-/octo-line phones. There cannot be a mix of single- and multi-line phones.
A common example of using an overlay line is when you have a main line that is
answered by anyone in a specic department. This overlay shared-line conguration is best
paired with the preference and no huntstop commands shown earlier in this chapter. In
our example, we have a department that has two employees. Each employee has a unique
extension for their phone. There is also a shared line number (5454) that is congured as an
overlay line on button 1. When we congure the ephone-DN, we make sure to congure the
unique extension rst. The rst ephone congured is the number that is displayed on the
phone display LCD panel. The overlay line is congured, but that number is never seen on
the phone button display. The shared line is congured on ephone-DN 3 and ephone-DN 4.
Ephone-DN 3 has the lower preference and will handle the rst call. It is also congured to
look for the next preferred ephone-DN with the same extension if the most preferred phone
is busy by using the no huntstop command. The complete conguration looks like this:
Router(config-ephone)#button 1o1,3,4
Router#
Phone button 1 of both the phones is congured with its unique number as well as the
shared-line number for the department. Calls placed to 6001 go only to ephone-DN 1.
Calls placed to 6002 go only to ephone-DN 2. But calls placed to 5454 are sent to both
phones. The conguration consumes only one phone button on each phone. Now other
buttons are open to be congured for additional lines or speed-dial capabilities if desired.
Here is a show ephone for our two congured ephone-DNs. As you can see, the rst
number assigned in the overlay conguration is bound to the phone and idle. The
shared number is visible but not the primary number.
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Router#show ephone
ephone-1 Mac:0021.A084.4F0C TCP socket:[3] activeLine:0 REGISTERED in SCCP ver
12/8
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9
button 1: dn 1 number 6001 CH1 IDLE overlay
overlay 1: 1(6001) 3(5454) 4(5454)
ephone-2 Mac:0021.A02E.7D9A TCP socket:[4] activeLine:0 REGISTERED in SCCP ver
12/8
overlay 1: 2(6002) 3(5454) 4(5454)
Lets say a call is placed to extension 5454, and ephone-DN 2 answers the call. Now a
show ephone looks like this:
Router#show ephone
12/8
overlay 1: 1(6001) 3(5454) 4(5454)
12/8
button 1: dn 3 number 5454 CH1 CONNECTED overlay shared
overlay 1: 2(6002) 3(5454) 4(5454)
Active Call on DN 3 chan 1 :5454 192.168.10.3 27418 to 192.168.1.100 24646 via
192.168.10.3
Jitter 7 Latency 0 callingDn 5 calledDn -1
At this point, ephone-DN 3, which is number 5454, is owned and controlled by ephone-
DN 2. A second call is made to 5454; this time, ephone-DN 3 is in use, so it rolls over to
the next ephone-DN, which is 4. Because ephone-DN 2 is congured with an overlay with
both ephone-DN 3 and 4, the phone rings on ephone-DN 2. A show ephone with both
ephone-DN 3 and 4 in use looks like this:
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Router#show ephone
12/8
overlay 1: 1(6001) 3(5454) 4(5454)
192.168.10.2
12/8
overlay 1: 2(6002) 3(5454) 4(5454)
192.168.10.3
As you can see, this shared-line overlay conguration is a very good option in many
ofce environments. It also highlights a combination of PBX and key-system capabilities
on CUCM Express. Situations that combine both PBX and key-system functionality are
commonly called hybrid systems.
Overlay with Call Waiting (c)
This button separator option is the same as the overlay except that it adds call-waiting
functionality. Call waiting is the ability for a phone to receive two or more simultaneous calls.
The user answering the call can place a currently engaged call on hold to answer the second call.
To see this difference, we will congure our CUCM Express router with the same conguration
as the overlay example except we will use the call-waiting button separator option. Well also
have to congure ephone-DN 3 as a dual-line phone so it can utilize call waiting:
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Router(config-ephone)#button 1c1,3
Router(config-ephone)#button 1c2,3
Router#
So what are the results of this conguration? The rst call to extension 5454 is handled
by ephone-DN 3 because of its lower preference. A second call rolls over to ephone-DN 4,
because the no huntstop option has been set. Ephone-DN 4 rings ephone-DN 1, but it also
sends the call-waiting beep to ephone-DN 2, which is currently in a call. This way, the user
on ephone-DN 2 is notied of a second call and can, if he/she wants to, place the rst
call on hold and answer the second.
Expansion Line (x)
The expansion button separator is used to expand line coverage for an overlay button (o).
It does not work when the overlay separator button is congured for call waiting (c). When
the extensions congured as overlay lines are in use, the expansion lines begin taking calls.
In this example, we have ephone-DN 1 congured to overlay ephone-DNs 1 and 2, which
are both 7001. Ephone-DN 1 is also a dual-line phone. We also have button 2 congured
as an overlay for line 1 on the phone:
Router(config-ephone)#button 1o1,2 2x1
Router#
So in this example, what happens? The rst call to 7001 goes to button 1. The second
call also goes to button 1, because it is a dual-line phone and channel 2 is free. The
third call will overow to button 2 because both lines are busy on button 1. Always
remember that overow lines will be used only when all other lines are occupied.
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Configuring CUCM Express Telephony
Service Features
The conguration steps for most telephony service features are performed while in
config-telephony conguration mode. These features provide multiple ways to tailor your
voice environment to better t the needs of your end users. This section will show you how
to congure several of the most important telephony service features. We will look at
how to change the language and ring tone settings to match the location where your
endpoints will reside. You will also see how to modify the date and time formats and
modify the phone handset system message to personalize your voice system. Keep in mind
that these features can be congured in either SCCP or SIP mode. The commands are
identical for both protocols except where explicitly indicated.
Configuring User Locale and Network Locale
By default, CUCM Express is set for the English (US) language for its location. What
happens if you need to deploy this system in Colombia, where Spanish is the native
language? To modify the language used on the Cisco phone handsets, including soft keys,
help, and other buttons, we can use the user-locale command. Lets see what language
options are currently available:
Router(config-telephony)#user-locale ?
<0-4> user locale index 0 to 4 (0 is default)
DE Germany
DK Denmark
ES Spain
FR France
IT Italy
JP Japan
NL Netherlands
NO Norway
PT Portugal
RU Russian Federation
SE Sweden
US United States
Using our Colombian deployment example, well choose ES for our locale, so Spanish
will be displayed on our handsets:
Router(config-telephony)#user-locale ES
Updating CNF files
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326 Chapter 8

CNF files update complete
Please issue create cnf command after the locale change
CNF file creation is already On
Updating CNF files
Whenever we make changes to the conguration of a telephone, we will need to reset the
phone in order to obtain all of the updated conguration and settings as manipulated.
The network-locale command modies tones and cadence differences between
geographic regions. Unlike user-locale, which changes language functions of the phones,
the network-locale settings are based on regional standards for telephone signaling.
Using our Colombia deployment example, we can use ES for the user-locale because
Colombians speak the same language as Spaniards. The network-locale settings differ,
however, because each region has different tones within its geographic regions:
Router(config-telephony)#network-locale ?
<0-4> network locale index 0 to 4 (0 is default)
AT Austria
CA Canada
CH Switzerland
CO Colombia
DE Germany
DK Denmark
ES Spain
FR France
GB United Kingdom
IT Italy
JP Japan
NL Netherlands
NO Norway
PT Portugal
RU Russian Federation
SE Sweden
US United States
Router(config-telephony)#network-locale CO
Updating CNF files
Please issue create cnf command after the locale change
Updating CNF files
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Can You Translate This for Me?
Jeff was an IT consultant who recently began installing CUCM Express solutions in businesses.
All of his implementations up to this point had been for local businesses in the United States,
where English is the dominant language. A recent client, however, called for a Canadian
deployment. Some employees had English as their primary language and others had
French. In addition, the company regularly had visits from consultants from Spain, which
required a third language. Since Jeff was new to the language-localization features of the
CUCM Express, he had to do a bit of research to gure out the best conguration method to
provide the three different language options to users. He learned that if the CUCM Express
is going to be in a mixed-language environment, his best option was to congure user-locale
and network-locale ephone templates. This is an example of how the ephone templates
were used to remedy this situation:
Router(config-telephony)#user-locale 1 ES
Router(config-telephony)#user-locale 2 FR
Router(config-telephony)#network-locale 1 ES
Router(config-telephony)#network-locale 2 FR
Router(config-telephony)#ephone-template 1
Router(config-ephone-template)#user-locale 1
Router(config-ephone-template)#network-locale 1
Router(config-ephone-template)#ephone-template 2
Router(config-ephone-template)#user-locale 2
Router(config-ephone-template)#network-locale 2
Router(config-ephone-template)#ephone 1
Router(config-ephone)#ephone-template 1
Router(config-ephone)#ephone-template 2
Router(config-ephone)#exit
Updating CNF files
Router(config-telephony)#restart all
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Configuring the Date and Time Format
Similar to user-locale is the date and time format. Different countries display the date
and time differently. In the United States, the date is displayed as mm/dd/yy. In other
regions, such as Europe, the date is displayed as dd/mm/yy. The default format is mm/dd/
yy. If you wish to change the format on your Cisco IP phones, you use the date-format
command. You can specify the following formats:
Router(config-telephony)#date-format ?
dd-mm-yy Set date to dd-mm-yy format
mm-dd-yy Set date to mm-dd-yy format
yy-dd-mm Set date to yy-dd-mm format
yy-mm-dd Set date to yy-mm-dd format
Lets change the date format to the European dd/mm/yy:
Router(config-telephony)#date-format dd-mm-yy
Router#
Now when we reset our phones, we get the date to display with the day rst.
When configuring date format settings for SIP, the commands are identical
but the date must use a slash (/) as opposed to a hyphen (-) to separate the
month, day, and year such as in the SIP configuration command:
date-format mm/dd/yy
We can congure the time format to use either a 12- or 24-hour clock, with the
time-format command followed by 12 or 24. In this example we set a 24-hour clock for
our phones:
Router(config-telephony)#time-format 24
Router#
This method sets up a very simple and streamlined way to congure ephones that ts the
needs of the local user. Note that by default, the English (US) locale is congured if you
do not specify a template. So, for example, ephone 1 is for English-speaking users because
there is no ephone template 1 or 2 specied. Ephone 2 is congured for user-locale 1,
which is Spanish, and ephone 3 uses the French language as specied in template 2.
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Modifying the Cisco IP Phone Keepalive Timer
Cisco IP phones are constantly informing the call-processing agent (in our case, CUCM
Express) that they are still active on the network. The phone uses keepalive messages for
notication, with an interval of 30 seconds by default. If CUCM Express misses three
keepalives in a row, it assumes the Cisco phone is no longer active.
In situations where your IP phones reside on a congested network, it might be advisable
to increase the keepalive timer to help reduce network load. To do this, you enter into
config-telephony conguration mode and use the keepalive command followed by the
number of seconds in between each sent notication, as shown here:
Router(config-telephony)#keepalive 60
Router#
Now your IP phones will send keepalive messages every 60 seconds, and CUCM Express
will declare the Cisco phone as deceased after 180 seconds or three missed keepalive
messages in a row.
Cisco IP Phone Restart versus Reset
When you make modications to a previously congured CUCM Express, there will be
some settings that need to be pushed to your connected Cisco IP phones using either a
restart or a full reset of the phone. If this is not done, the phones will not grab the new
conguration le. The next section covers the difference between a phone restart and reset
and when each should be used.
Restart
A restart is a partial reset of the Cisco IP phone. The phones connect to the TFTP server
and update any changes to the conguration le. This command will update the following
information:

Directory numbers (DNs)

Phone buttons

Speed-dial
You have the ability to restart either all of the connected phones or one at a time. If you
wish to restart all of the phones, you must be in config-telephony conguration mode and
issue a restart all command. Here is an example of the output of this command:
Router(config-telephony)#restart all
Reset 2 phones: at 5 second interval - This could take several minutes
per phone
Starting with 7960 phones
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Router(config-telephony)#
Reset/Restart-all looking for phones registered as type 30008 7902
[output omitted]
Reset-All: Requesting Restart for phone SEP0021A086D04D at 192.168.10.12
deviceType 436 Idle [count=1]
May 2 07:28:51.878: %IPPHONE-6-UNREGISTER_NORMAL: ephone-1:SEP0021A086D04D
IP:192.168.10.12 Socket:1 DeviceType:Phone has unregistered normally.
[output omitted]
Reset/Restart-all looking for phones registered as type 30016 CIPC
Reset-All: Requesting Restart for phone SEP001E68E1AFE9 at 192.168.1.15
deviceType 30016 Idle [count=2]
May 2 07:29:04.858: %IPPHONE-6-UNREGISTER_NORMAL: ephone-2:SEP001E68E1AFE9
May 2 07:29:05.250: %IPPHONE-6-REG_ALARM: 23: Name=SEP001E68E1AFE9 Load= 7.0.1.0
Last=Reset-Restart
May 2 07:29:06.122: %IPPHONE-6-REGISTER: ephone-2:SEP001E68E1AFE9
IP:192.168.1.15 Socket:3 DeviceType:Phone has registered.
Reset/Restart-all looking for phones registered as type 39999 none
[output omitted]
Reset/Restart-all looking for phones registered as type -1 Unknown Ephone type
Restart-All issued for 2 phones
To restart a single phone, you navigate into config-ephone conguration mode and
issue the restart command.
Reset
The reset command performs a full reboot of the Cisco IP phone. This process requires the
phone to go through both the TFTP download and DHCP renewal processes, so it takes more
time for the phone to become fully operational within the CUCM Express system. In addition
to handling the same three conguration updates that the restart command can perform, the
reset command updates the phone if any of the following were added/deleted or modied:

Date/time

Phone rmware

CUCM Express source IP address

TFTP download path

Voicemail access number
Just like the restart command, reset can be performed on all phones or a single phone.
To reset all phones, you must be in config-telephony conguration mode and issue a reset
all command. And for a single ephone, navigate to the ephone you desire and enter a reset
command. Here is the command-line output when we reset all the phones on the system:
c08.indd 330 9/21/11 11:25:01 AM
Router(config-telephony)#reset all
ITS configuration has been changed, selecting sequence-all reset
Reset 2 phones: sequentially with 240 second per-phone timeout to guarantee TFTP
access
- this could take several minutes per phone
you may abort this process using reset cancel
Starting reset sequence with 7960 phones
Router(config-telephony)#
[output omitted]
Reset-All: Requesting Reset for phone SEP0021A086D04D at 192.168.10.12 deviceType
436 7965 Idle [count=1]
Reset-All received Unregister from ephone-1 SEP0021A086D04D
May 2 07:56:31.941: %IPPHONE-6-UNREGISTER_NORMAL: ephone-1:SEP0021A086D04D
May 2 07:57:08.905: %MGCP-3-INTERNAL_ERROR: mgcp_cfg_commands: nvgen lawful-
intercept: should not happen
May 2 07:57:33.149: %IPPHONE-6-REG_ALARM: 25: Name=SEP0021A086D04D Load=
SCCP45.8-5-3S Last=Initialized
May 2 07:57:33.165: %IPPHONE-6-REGISTER: ephone-1:SEP0021A086D04D
Reset sequence-all, Ready to reset next phone (last 61 sec)
[output omitted]
Reset/Restart-all looking for phones registered as type 30016 CIPC
Reset-All: Requesting Reset for phone SEP001E68E1AFE9 at 192.168.1.15 deviceType
30016 CIPC Idle [count=2]
Reset-All received Unregister from ephone-2 SEP001E68E1AFE9
May 2 07:57:41.885: %IPPHONE-6-UNREGISTER_NORMAL: ephone-2:SEP001E68E1AFE9
Last=Reset-Reset
[output omitted]
Reset/Restart-all looking for phones registered as type -1 Unknown Ephone type
Reset-All issued for 2 phones
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You can also reset a single phone by navigating into config-ephone conguration mode
and issue the reset command.
You can also reset a Cisco phone using the handset unit by pressing the
Settings button followed by **#** on the keypad.
Using CUCM Express Verification
and Troubleshooting Commands
When setting up CUCM Express for the rst time, you may need some basic troubleshooting
skills. This section goes through some of the more common troubleshooting steps, including
how to gure out why a Cisco phone wont register and how to determine the state of an
ephone on your network.
Troubleshooting Cisco Phone Registrations
There will come a time when you add a new Cisco phone to your CUCM Express
environment and it simply will not register. Because you understand the boot process,
there is a methodical way of troubleshooting the problem. Here is the order in which
troubleshooting should be performed:
1. Troubleshoot DHCP issues.
2. Troubleshoot TFTP issues.
3. Troubleshoot ephone registration issues.
Troubleshooting these three items in order will help you to nd and x the vast majority
of phone registration problems youll encounter.
Troubleshooting DHCP Issues
When the phone boots up, one of the rst things it displays is a Conguring IP message.
This tells you that the phone is attempting to nd the DHCP servers so it can receive the IP
address and TFTP information needed to download the rmware and conguration les. If
the IOS device you are on is the DHCP server, you can verify that your phone is receiving
DHCP information by using the debug ip dhcp server events command. Heres an
example of the output you will receive when a device successfully receives an IP address
from the DHCP server that is congured on your CUCM Express router:
Router#debug ip dhcp server events
DHCP server event debugging is on.
May 17 18:18:54.303: DHCPD: Sending notification of ASSIGNMENT:
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Using CUCM Express Verification and Troubleshooting Commands 333
May 17 18:18:54.303: DHCPD: address 192.168.10.2 mask 255.255.255.0
May 17 18:18:54.303: DHCPD: htype 1 chaddr 0021.a086.d04d
May 17 18:18:54.303: DHCPD: lease time remaining (secs) = 86400
On a Cisco IP phone, you can verify that your phone received DHCP information
by pressing the Settings button and navigating to the Network Conguration area. If
your phone is not receiving an IP address, you should start by looking at a possible
misconguration of DHCP and not of the VoIP rmware or CUCM Express.
Troubleshooting TFTP Issues
If your phone is receiving DHCP information, the next thing it attempts to do is to
download the rmware and conguration les required to operate. If your phone is stuck
with the Registering notication on the screen, you can try to run the debug tftp
events command to see if your phone is requesting les that are not on your TFTP server.
Keep in mind that this command is useful only if your router is acting as the TFTP
server. Here is an example of the output of this command for a phone that successfully
receives some but not all of the requested rmware and conguration les:
Router#debug tftp events
TFTP Event debugging is on
Router#
May 17 18:51:36.855: TFTP: Looking for CTLSEP001E68E1AFE9.tlv
May 17 18:51:37.887: TFTP: Looking for SEP001E68E1AFE9.cnf.xml
May 17 18:51:37.887: TFTP: Opened system:/its/XMLDefaultCIPC.cnf.xml, fd 9, size
1056 for process 248
May 17 18:51:37.891: TFTP: Finished system:/its/XMLDefaultCIPC.cnf.xml, time
00:00:00 for process 248
May 17 18:51:42.315: TFTP: Looking for Communicator/LdapDirectories.xml
May 17 18:51:43.423: TFTP: Looking for Communicator/LdapDialingRules.xml
May 17 18:51:49.823: TFTP: Looking for SEP001E68E1AFE9.cnf.xml
May 17 18:51:49.823: TFTP: Opened system:/its/XMLDefaultCIPC.cnf.xml, fd 9, size
1056 for process 248
May 17 18:51:49.827: TFTP: Finished system:/its/XMLDefaultCIPC.cnf.xml, time
00:00:00 for process 248
May 17 18:51:50.043: TFTP: Looking for English_United_States/ipc-sccp.jar
May 17 18:51:50.063: TFTP: Looking for United_States/g3-tones.xml
Last=Initialized
Router#
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Any line that begins with Looking means that the Cisco phone is requesting the le. If
the TFTP server knows where a le is located, it will process the le, giving you the Opened
statement. Finally, once the le is transferred, you will receive a Finished message.
As you can see in the sample output, this phone registered to the CUCM
Express even though it did not receive all of the files it requested. Some of
the files, such as LdapDirectories.xml, are supplementary services that
do not affect phone registration. The TFTP server did manage to serve up
the required files for the phone to register on the system.
If your phones are not receiving the necessary rmware or conguration les, you
should make sure that your TFTP server is congured to serve up the les your phone is
requesting. To do so, you can issue a show telephony-service tftp-bindings command.
Heres a sample of typical output from this command:
Router#show telephony-service tftp-bindings
tftp-server system:/its/united_states/7960-tones.xml alias United_States/7960-
tones.xml
tftp-server system:/its/united_states/7960-font.xml alias English_United_
States/7960-font.xml
tftp-server system:/its/united_states/7960-font.xml alias English_United_
States/7920-font.xml
tftp-server system:/its/united_states/7960-dictionary.xml alias English_United_
States/7960-dictionary.xml
tftp-server system:/its/united_states/7960-kate.xml alias English_United_
States/7960-kate.xml
tftp-server system:/its/united_states/7960-kate.xml alias English_United_
States/7920-kate.xml
tftp-server system:/its/united_states/SCCP-dictionary.xml alias English_United_
States/SCCP-dictionary.xml
tftp-server system:/its/SEPDEFAULT.cnf alias SEPDefault.cnf
tftp-server system:/its/XMLDefault.cnf.xml alias XMLDefault.cnf.xml
tftp-server system:/its/ATADefault.cnf.xml alias ATADefault.cnf.xml
tftp-server system:/its/XMLDefaultCIPC.cnf.xml alias SEP001E68E1AFE9.cnf.xml
tftp-server system:/its/XMLDefault7965.cnf.xml alias SEP0021A086D04D.cnf.xml
If there are any les that are being requested and not listed by this command, you should
locate them on your ash storage and serve them up using the tftp-server conguration
command.
Determining the State of an Ephone
Once your phones are congured and registered on your CUCM Express system, youll
want to familiarize yourself with the show ephone command, because it provides a wealth
c08.indd 334 9/21/11 11:25:03 AM
of information that can prove very useful when troubleshooting. First, well look at the
different registration states you will see.
Ephone Registration States
There are three different states that an ephone can be in. Table 8.7 lists the states and what
each state means.
Lets look at all three of these states by issuing the show ephone command:
Router#show ephone
ephone-1[0] Mac:0021.A086.D04D TCP socket:[-1] activeLine:0 DECEASED
debug:0 caps:9
button 1: dn 1 number 4001 CH1 DOWN
ephone-2[1] Mac:0021.A02E.7D9A TCP socket:[5] activeLine:0 REGISTERED in SCCP ver
12/9
debug:0 caps:9
button 1: dn 2 number 4002 CH1 IDLE
ephone-3[2] Mac:001E.68E1.AFE9 TCP socket:[1] activeLine:0 UNREGISTERED
debug:1 caps:8
IP:192.168.10.14 1556 CIPC keepalive 127 max_line 8
TABLE 8. 7 Ephone registration states
State Meaning
REGISTERED Indicates the phone is registered to CUCM Express and is active.
UNREGISTERED Indicates the phone unregistered normally from CUCM Express
and is not active.
DECEASED Indicates the phone is unregistered abnormally because of a
keepalive timeout.
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There are three ephones congured on this CUCM Express system. Ephone-1 is in
a DECEASED state, which means that CUCM Express has lost contact with the switch.
CUCM Express uses keepalives to monitor the state of the phones. After six missed
keepalive messages, the phone is placed into a DECEASED state. This typically happens
when a phone loses power. Ephone-2 is in a REGISTERED state. This means that this phone
is operational on the network and is ready to make and receive calls. Lastly, ephone-
3 is in an UNREGISTERED state. This state means that the phone gracefully unregistered
from the CUCM Express. You can see the type of phone this is on the third line from
the bottom, where it says the phone hardware is CIPC. Given that ephone-2 is a Cisco IP
Communicator, the phone probably unregistered when the user exited the application.
Ephone Extension States
A second piece of information that can be gained from the show ephone command is the
state of a phone extension. There are six ephone extension states that an ephone extension
can have. Table 8.8 provides a description of each of these states.
TABLE 8. 8 Ephone extension states
State On- or Off-hook
Ephone Registration
State Description
DOWN N/A Unregistered/
Deceased
Ephone is not registered to CUCM
Express.
IDLE On-hook Registered Ephone is ready to make and
receive calls.
SEIZE Off-hook Registered Ephone handset has been picked
up but no call has been made.
RINGING Off-hook Registered Ephone is calling another
extension.
ALERTING On-hook Registered Ephone is receiving a call from
another extension.
CONNECTED Off-hook Registered An active call is in progress
between two or more extensions.
Lets look at the show ephone command to see what each of the ephone extension
states looks like while we go through the process of ephone registration and
call processing.
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Ephone Extension DOWN State
The two following examples of ephone extensions show that the ephone registration process
is in either a DECEASED or an UNREGISTERED state for the extensions to be in a DOWN state:
ephone-1[0] Mac:0021.A086.D04D TCP socket:[-1] activeLine:0 DECEASED
debug:0 caps:9
ephone-3[2] Mac:001E.68E1.AFE9 TCP socket:[1] activeLine:0 UNREGISTERED
debug:1 caps:8
IP:192.168.10.14 1556 CIPC keepalive 127 max_line 8
Ephone Extension IDLE State
A phone is ready to either make or receive calls when the extension is in an IDLE state. In
order for this to happen, the ephone must be properly REGISTERED to CUCM Express, as
shown here:
ephone-1[0] Mac:0021.A086.D04D TCP socket:[1] activeLine:0 REGISTERED in SCCP ver
12/9
debug:0 caps:9
Ephone Extension SEIZE State
When an end user on ephone-2 wishes to make a call, they pick up the handset of the
phone. As you know, this action changes the phone from an on-hook state to an off-hook
state. This is called a line seizure. When this happens, the show ephone command has the
ephone extension in a SEIZE state, as shown here:
12/9
debug:0 caps:9
button 1: dn 2 number 4002 CH1 SEIZE
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Jitter 0 Latency 0 callingDn -1 calledDn -1
Ephone Extension RINGING and ALERTING States
Lets say that a user picks up a phone and dials an extension. Once that process reaches
the CUCM Express, the phone where the user called from is put into a RINGING state.
At this point the CUCM Express sends back the audible ringing tone through the phone
handset to indicate that the call is being processed, and the user is just waiting for the
called party to pick up their handset to complete the call. At the same time, the called
phone goes into an ALERTING state. In this state the called phone is on-hook but ringing to
alert the end user that someone is attempting to speak with them. The show ephone output
looks like this:
12/9
debug:0 caps:9
button 1: dn 1 number 4001 CH1 RINGING
call ringing on line 1
12/9
debug:0 caps:9
button 1: dn 2 number 4002 CH1 ALERTING
Jitter 0 Latency 0 callingDn -1 calledDn 1
Ephone Extension CONNECTED State
The remote phone rings, and the end user picks up the phone to answer it. At this point,
the CUCM Express places both calls into a CONNECTED state. You can also see in the show
ephone command that it lists the source and destination IP addresses:
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Summary 339
12/9
debug:0 caps:9
button 1: dn 1 number 4001 CH1 CONNECTED
192.168.10.12
12/9
debug:0 caps:9
button 1: dn 2 number 4002 CH1 CONNECTED
192.168.10.13
Summary
At its core, the CUCM Express system is a voice gateway with the capability of providing
call-processing services for up to 450 phones. It is a unique system that can handle most
unied communications voice functionalities in a single IOS system. CUCM Express is
often found in small and medium-size businesses but also can be found in large-network
remote sites that use a distributed call-processing model.
This chapter covered many of the steps that must be accomplished before implementing
a voice system, including everything from power options to voice VLANs and even services
used by IP phones, including DHCP, NTP, and TFTP. The chapter continued to describe
CUCM Expresss hardware and software licensing requirements as well as how to congure
SCCP and SIP endpoints.
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Exam Essentials
Know the three different power options for IP phones. The power brick is attached to the
phone and plugs directly into the wall outlet. A power patch panel or power injector sits
between an IP phone and a standard non-PoE switch. Power is sent to the phone over the
same cable that voice trafc resides on. Finally, the PoE switch offers power directly from
the switch to the phone over an Ethernet cable.
Understand the different PoE proprietary and IETF standards for IP phones and PoE
switches. The Cisco proprietary method is ILP and the IETF standard is 802.3af.
Know how to manipulate power requirements for your IP phone deployment for PoE
switches. Using Ciscos intelligent power management, you can set switchports to
disable power, assign a static amount, or have the switch intelligently determine the power
requirements of the PoE endpoint.
Understand the difference between data and voice VLANs. Cisco switches use CDP to
identify Cisco IP phones on the network. Voice VLANs are congured differently at the
switchport level. Finally, voice VLANs are tagged on the switchport, while any PC that is
connected to a switch or Cisco IP phone is untagged.
Know how to configure DHCP for VoIP support. Cisco IP phones rely heavily on DHCP
servers for information such as IP address, default-router, DNS, and the location of IP
phone conguration les by dening the option 150 parameter for a TFTP server.
Understand the purpose of NTP and how to configure it. NTP is used to synchronize
time for all of your phone equipment on the network. Synchronization of time helps to
ensure proper operation and support of your VoIP network.
Understand CUCM Express hardware and software capabilities. The maximum number
of IP phones that can be supported depends on the ISR model you are working with. Once
you have an ISR router set up with CUCM Express IOS software, it can be congured to
be used as a call-processing agent, voice gateway, voicemail server, and IP router in a
single system.
Understand the three CUCM Express licenses and the new software activation
method. Cisco has three different licenses for CUCM environments. One license is for
the voice-capable IOS. The second is the CUCM Express feature license. The third is the
individual user license. Currently, software is licensed as a right-to-use license. The new
model will be one that is activated automatically in software.
Know how to configure the mandatory CUCM Express system configuration settings using
the command line. The mandatory conguration settings to get a CUCM Express router
ready for operation to run SCCP are to specify a source IP address of the call-processing
agent, set the max ephones and ephone-DNs, and set the rmware load les and default
conguration les. For SIP phones, you must rst enable SIP and SIP-to-SIP calls, then
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specify a source IP address of the call-processing agent, set the max pool and max DNs,
and set the rmware load les and default conguration les.
Understand what the auto-assign configuration command does. Auto-assign allows you
to set up a pool of ephone-DNs. When Cisco IP phones connect to CUCM Express for the
rst time, the auto-assign function registers the ephone. It maps an ephone-DN taken from
the pool to the MAC address of the phone. This functionality is a great way to partially
automate a phone rollout.
Understand the difference between the telephony-service restart and reset
commands. Restart is a quick reset of the phone. It is good to use when you make
changes to the conguration le, including changes to DNs, phone buttons, and speed dial.
Reset is a full boot of the phone. This command causes the phone to go through a DHCP
renewal process. It is also required when you change global parameters such as date/time,
CUCME source IP, and TFTP download path.
Know how to configure different DN line options. Key system phones are typically
congured identically and share DNs. PBX systems are congured with unique DNs on
each phone and are individually tailored to meet the needs of the user.
Understand the different types of ephone button options. Using the button separator
when conguring extensions lets you set various ring options, phone monitoring, and
overlay features.
Know how to configure your CUCM Express system to meet the needs of your
users. Depending on where you set up your CUCM Express, you may need to modify user
options to match the native language. In addition, you can modify the network options
to match the PSTN tone and cadence that are familiar to the area, and you can modify
CUCM Express to display the date and time in a familiar format.
Know how to troubleshoot CUCM Express registration and extension states. Understand
how to best troubleshoot registration and extension problems using command-line debug
and show commands.
Written Lab 8.1
1. What interface command assigns a switchport to voice VLAN 55?
2. What DHCP server command removes the rst 20 IP addresses from being included in
the DHCP pool on the 192.168.10.0/24 network?
3. What conguration command tells CUCM Express to serve up the flash:/
phone/7945-7965/SCCP45.8-5-2-27.sbn le via TFTP?
4. What cong-telephony command sets the SCCP source IP address for the CUCM
Express system to 172.16.55.100?
5. What config-telephony command sets the maximum number of ephones to 30?
6. What config-telephony command sets the maximum number of ephone-DNs to 50?
Written Lab 8.1 341
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7. What is the config-register-pool command to congure SIP voice register DN 5
on button 2?
8. What is the config-ephone command to congure ephone-DN 1 on button 2 and DN
2 on button 1?
9. What is the config-ephone command to assign button 2 to ephone-DN 8 and have it
use an alternate ring?
10. What is the config-ephone-dn command to set a DN to be more preferred than a DN
that has its preference set to 2?
Hands-On Labs
To complete the labs in this section, you need a CUCM Express router and two Cisco IP
phones. The phones used in this example are 7940s, but you can use any phone or the
IP Communicator softphone if you wish. The labs will follow the logical network design
Ext. 444
Ext. 555
CUCM Express
Telephony source IP:
192.168.10.1
FI GURE 8.10 CUCM Express lab diagram
These labs build on each other, so it is best to perform them in the order listed:
Lab 8.1: Conguring CUCM Express as a TFTP Server
Lab 8.2: Conguring CUCM Express for Basic SCCP Phone Operation
Lab 8.3: Verifying the Conguration and Status of your Ephones
as a TFTP Server
In this lab, we are going to add 7940 phones to our voice network. In order for them to
work properly, we need to congure the CUCM Express router as a TFTP server to serve
up the rmware les that the 7940 phones require.
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1. Log into your CUCM Express router and go into privileged execution conguration
mode by typing enable.
2. Check to see which rmware les the 7940 phones need by viewing the les on the
ash drive. To do this, type dir flash:/phone/7940-7960. You should see something
similar to the following output:
Directory of flash:/phone/7940-7960/
97 -rw- 129824 Mar 7 2011 18:14:31 +00:00 P00308000500.bin
98 -rw- 458 Mar 7 2011 18:14:31 +00:00 P00308000500.loads
99 -rw- 705536 Mar 7 2011 18:14:34 +00:00 P00308000500.sb2
100 -rw- 130228 Mar 7 2011 18:14:34 +00:00 P00308000500.sbn
3. Enter into conguration mode by typing configure terminal.
4. Congure the CUCM Express router to serve up the 7940 rmware les. Note that
because the les are organized in a directory structure, you need to include the
alias command:
tftp-server flash:/phone/7940-7960/P00308000500.bin alias P00308000500.bin
tftp-server flash:/phone/7940-7960/P00308000500.loads alias P00308000500.bin
tftp-server flash:/phone/7940-7960/P00308000500.sb2 alias P00308000500.bin
tftp-server flash:/phone/7940-7960/P00308000500.sbn alias P00308000500.bin
5. Exit conguration mode by typing end.
for Basic SCCP Phone Operation
In our second lab, we will go through the basic conguration necessary to get our two
Cisco 7940 phones up and running on the voice network. We will set our max-ephones to 5
and max-dn to 10. Additionally, we will set the preferred codec to g729r8.
1. Log into your CUCM Express router and go into privileged execution conguration
mode by typing enable.
2. Enter into cong-telephony conguration mode by typing configure terminal and then
telephony-service.
3. Congure the IP source address to the address given in the diagram by typing ip
source-address 192.168.10.1.
4. Congure the maximum ephones to 5 and maximum ephone-DNs to 10 by typing
max-ephones 5, pressing Enter, and then typing max-dn 10.
5. Set the rmware load les for the 7940 phones by typing load 7940-7960 PPPPPPPP.
loads, where PPPPPPPP is the load lename for your particular Cisco phone.
Hands-On Labs 343
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6. Exit config-telephony conguration mode by typing exit.
7. Congure ephone-DN 1 to have the number 444 by typing ephone-dn 1 and
then number 444.
8. Congure ephone-DN 2 to have the number 555 by typing ephone-dn 2 and
then number 555.
9. Congure the MAC address of ephone 1 by typing ephone 1, pressing Enter, and then
typing mac-address XXXX.XXXX.XXXX. Your MAC address will be unique.
10. Congure the MAC address of ephone 2 by typing ephone 2, pressing Enter, and then
typing mac-address XXXX.XXXX.XXXX. Your MAC address will be unique.
11. Congure button 1 of ephone 1 to use ephone-DN 1 by typing ephone 1, pressing
Enter, and then typing button 1:1.
12. Congure the phone type and codec preference by typing type 7940, pressing Enter,
and then typing codec g729r8.
13. Congure button 1 of ephone 2 to use ephone-DN 2 by typing ephone 2, pressing
Enter, and then typing button 1:2.
14. Congure the phone type and codec preference by typing type 7940, pressing Enter,
and then typing codec g729r8.
15. Exit config-ephone conguration mode by typing end.
Hands-On Lab 8.3: Verifying the Configuration
and Status of Your Ephones
Now that we have our phones properly congured, we can verify our conguration settings
and check to see if the phones are connected.
1. Log into your CUCM Express router and go into privileged exec conguration mode
by typing enable.
2. Verify the conguration and status of your ephones by typing show ephone and
reviewing the output of this command. An example of the output you should see
is listed here:
Router#show ephone
ephone-1[0] Mac: 0021.A086.D04D TCP socket:[5] activeLine:0 REGISTERED in SCCP
ver 12/9
debug:0 caps:9
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Preferred Codec: g729r8
ver 12/9
debug:0 caps:9
Preferred Codec: g729r8
3. From a conguration standpoint, you should verify that the CUCM Express properly
sees the following information:

Telephone model type (in this example, Cisco 7940)

Button and DN numbers (button 1: dn1 for ephone 1 and button 1: dn2 for
ephone 2)

Extension numbers (number 444 for ephone 1 and number 555 for ephone 2)

Preferred codec (g729r8)
4. From an operational standpoint, you should verify that the phones are in a REGISTERED
and IDLE state (assuming the phones are not in use).
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Review Questions
1. An ILP PoE switch can power devices of up to how many watts?
A. 6.0W
B. 15.4W
C. 6.3W
D. 7.0W
2. What protocol does a Cisco IP phone use to tell the PoE switch how much power it requires
for the phone?
A. PoE protocol
B. iLBC
C. VTP
D. STP
E. CDP
3. What Cisco power-saving method helps to negotiate the exact power requirements of a
Cisco IP phone?
A. IPM
B. 802.3af
C. ILP
D. CDP
4. Why is it important to configure DHCP option 150 for Cisco voice networks?
A. It defines the default gateway for the phone.
B. It defines the IP address of the TFTP server.
C. It defines the IP address of the communications manager.
D. It defines the IP address for CDP.
5. Which of the following is not a CUCM Express license?
A. Cisco SCCP license
B. Cisco IOS license for voice capabilities
C. CUCM Express feature licenses
D. Individual user license
6. What is the tftp-server IOS command used for?
A. To identify the IP address of the TFTP server
B. To set option 150 for DHCP clients
C. To identify files the router serves via TFTP
D. To enable Secure FTP
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7. What command is used to manually update the phone configuration load files on a CUCM
Express system configured for SCCP endpoints?
A. create firmware
B. create cnf-files
C. load cnf-files
D. load firmware
8. Which of the following required commands is missing from the CUCM Express initial
configuration for SCCP endpoints?
A. ip source-address 192.168.1.1
B. create profile
C. source-address 192.168.1.1
D. auto assign
9. When configuring SIP to control signaling to IP phone endpoints, which of the following
commands is used to set 120 as the maximum number of telephone extensions that can be
configured on the system?
A. Router(config-register-global)#max-pool 120
B. Router(config-telephony)#max-ephones 120
C. Router(config-telephony)#max-dn 120
D. Router(config-register-global)#max-dn 120
10. Which of the following CUCM Express dial number line options are not compatible when
the end device is running SIP? (Choose all that apply.)
A. Single-line
B. Dual-line
C. Octo-line
D. Shared-line
E. Two DNs with one telephone
11. What line type and button type does this configuration represent?
A. Shared-line, octo-line
B. Huntgroup
C. Two DNs with one telephone
D. Shared-line, overlay
E. Shared-line, extension-line
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12. When configuring SIP to control signaling to IP phone endpoints, which of the following
commands is used to create the default configuration load file?
A. Router(config-register-global)#create cnf-files
B. Router(config-telephony)#create cnf-files
C. Router(config-register-global)#create profile
D. Router(config-telephony)#create profile
13. Which command-line operation does a quick reset of all phones currently registered on a
CUCM Express system using a single command?
A. Router(config-telephony)#restart reset
B. Router(config-ephone)#restart all
C. Router(config-telephony)#restart all
D. Router(config-ephone)#restart reset
E. Router(config-ephone)#reset all
F. Router(config-telephony)#reset
14. When troubleshooting a Cisco phone that powers up and connects to the network but will
not register, what is the first logical thing to check?
A. Ensure that the proper firmware and configuration files are accessible to the phone.
B. Ensure that the ephone is properly configured in the CUCM Express configuration.
C. Make sure that the phone is receiving the correct IP address and other network
parameters through DHCP.
D. Check to see if the clock is properly synchronized with NTP.
15. When viewing show ephone output like the following, what does ALERTING mean on
the extension?
ver 12/9
debug:0 caps:9
button 1: dn 2 number 4002 CH1 ALERTING
A. The phone is currently in a call.
B. The phone is on-hook.
C. The phone is calling another extension.
D. The phone is receiving a call.
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16. With multiple ephone-DNs sharing a single number, what command can you use to
prioritize which ephone-DN will always receive the incoming call if it is not in use?
A. priority
B. preference
C. state
D. no huntstop
17. With multiple ephone-DNs sharing a single number when the phone preference for each
ephone-DN is the same, how is call routing handled for incoming calls?
A. Calls will be received on the ephone-DN with the lowest tag.
B. This configuration will not work. The ephone-DNs must be configured with
different priorities.
C. Calls will be received on the ephone with the lowest tag.
D. Calls will be handled round-robin style.
18. What ephone overlay button separator would you use if you want calls to come in on this
extension only when all other lines are busy?
A. o
B. c
C. w
D. x
E. m
19. What is the term used to describe the configuration of multiple ephone-DNs on a single
physical phone button?
A. Ephone
B. Ephone-DN
C. Dual-line
D. Call waiting
E. Overlay
20. What configuration option can you change so that Cisco phones will display information
on the screen in a different language?
A. network-locale
B. language-locale
C. user-locale
D. telephony-service-locale
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350 Chapter 8

1. C. An ILP PoE switch provides a xed 6.3W of power to capable devices.
2. E. The Cisco Discovery Protocol (CDP) is used to discover IP phones and negotiate
power options.
3. A. Cisco Intelligent Power Management works between Cisco PoE switches and Cisco IP
phones to negotiate and allocate the exact amount of power needed by the phone.
4. B. Option 150 denes the IP address of the TFTP server, where the phone can download
conguration les.
5. A. You need a license for the voice IOS, the CUCME software for a specic number of
endpoints, and the individual user licenses for endpoints.
6. C. The tftp-server command is used to specify les that the router can serve to clients
using TFTP.
7. B. You manually update conguration SCCP load les using the create cnf-files
command.
8. A. The initial conguration must include the ip source-address 192.168.1.1 command
to identify the IP address of the call-processing agent to the Cisco IP phones.
9. D. The max-dn command species the number of telephone extensions that can be
operated on CUCM Express. This command is entered while in config-register-global
conguration mode.
10. B, C. The dual- and octo-lines are possible only when SCCP is used between the CUCM
Express and IP phone.
11. D. The conguration shows two ephones with a shared-line and overlay button.
12. C. The create profile command is used to build the default conguration load le
for SIP endpoints. This command is entered while in config-register-global
conguration mode.
13. C. The restart all command within config-telephony conguration mode performs a
quick reset of all registered phones.
14. C. When a phone powers up and connects to the network, its rst task is to receive network
parameters such as an IP address, gateway, subnet mask, and the option 150 parameter. If
your phone is not receiving one or more of these, it will fail to register properly.
15. D. Alerting means that someone is trying to call that ephone-DN, but the user has not yet
picked up the handset.
16. B. The preference command allows you to set which ephone-DN will receive all calls
when not in use. The lower number is the more preferred ephone-DN.
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17. D. If preferences are the same, then calls will be handled in a round-robin manner.
18. D. The expansion (x) line button separator helps prevent a caller from receiving a busy
signal. The calls will go to this line only when all other lines are busy.
19. E. An overlay line is a phone button separator conguration option that allows you to
congure multiple ephone-DNs on a single phone button.
20. C. The user-locale option allows you to change the language displayed on the LCD
screens of Cisco IP phones.
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352 Chapter 8

1. switchport voice vlan 55
2. ip dhcp excluded address 192.168.10.1 192.168.10.20
3. tftp-server flash:/phone/7945-7965/SCCP45.8-3-2-27.sbn alias
SCCP45.8-3-2-27.sbn
4. ip source-ip 172.16.55.100
5. max-ephones 30
6. max-dn 50
7. number 2 dn 5
8. button 1:2 2:1
9. button 2f8
10. preference 1 (or preference 0, which is the default)
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Advanced Voice
Gateway Features
CHAPTER:
Describe path selection.
Describe calling privileges.

a VoIP call.
Describe VoIP call flows.

Describe dial peers and the gateway call routing process.

Configure digit manipulation.
Implement fax support on a gateway.
Chapter
9
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Were at the point in this study guide where we begin to
expand the capabilities of the IOS voice gateway and CUCM
Express to see what value-added features can be implemented.
In Chapter 9, we will investigate several scenarios that require you to go beyond basic
IP and voice conguration to further enhance the voice experience. We will begin by
conguring DTMF relay support to improve the reliability of DTMF tones on an
IP network. We will then cover the still-important topic of fax machines and modems.
These devices still need to be supported, and youll see how to implement that on
a voice gateway. Finally, youll learn how to implement failover, toll bypass, and
call-restriction techniques.
Configuring DTMF Relay Support
By default, H.323, SIP, and MGCP transport DTMF tones in band. This means the tones
are sent in standard RTP voice packets just as if they were part of a regular voice stream.
This method may work ne for you, but if you are using highly compressed codecs, the
tones may not be reconstructed accurately enough and youll run into connection problems.
For example, when using interactive voice response (IVR) services, it is critical that when
the calling party presses a number to direct them through the IVR menu system, the
number is correctly interpreted by the system so the call can be properly routed.
To make sure that DTMF tones are correctly interpreted, you can congure
DTMF tones to be sent out of band using specially crafted RTP packets, while using a
codec with lower compression to ensure that the digit tones are better replicated at the
opposite end. This section will show how to congure DTMF relay support for H.323,
SIP, and MGCP.
Configuring H.323 DTMF Relay
H.323 DTMF relay is congured while in config-dial-peer configuration mode. To
enable sending of DTMF tones out of band, you simply use the dtmf-relay command
followed by the DTMF method you wish to use. For H.323 the possible relay options
are these:
cisco-rtp This method uses a Cisco proprietary method of transporting DTMF tones in
special RTP packets.
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Configuring DTMF Relay Support 355
h245-alphanumeric This method uses the H.245 alphanumeric user input method for
specifying only dial-pad tones, namely, 09, *, #, and the AD buttons that are represented
as ASCII characters.
h245-signal This uses the H.245 tone signal method that sends the same dial-pad
tones in ASCII format as the h245-alphanumeric method does. The h245-signal
method also sends along the length of time that the button was pressed, which is
sometimes necessary.
rtp-nte This method uses the named telephone event dened in RFC 2883, which
species a standard method for transporting DTMF tones in RTP packets. One optional
keyword that is compatible with H.323 is digit-drop, which will explicitly drop the in-
band tones from being sent. Without this command, the DTMF tones will be sent both in
and out of band.
The remote end gateway that you are communicating with must also be congured
to use one of these out-of-band signaling methods. When you congure DTMF relay on
a dial peer, you can specify one or more DTMF relay methods. The order of priority is
determined by the router, and the order in which you congure them has no effect. Cisco
rates the order of priority as follows:
1. cisco-rtp
2. rtp-nte
3. h245-signal
4. h245-alphanumeric
As an example of how this works, we will congure VoIP dial peer 100 to use both
H245 alphanumeric and H245 signal methods:
Router(config-dial-peer)#dtmf-relay h245-alphanumeric h245-signal
Router#
So now our voice gateway is congured to use either H.323 alphanumeric or H.245
signal methods. But even though H.245 alphanumeric was entered in the command rst,
the gateway will still prefer to use H.245 signal.
Configuring SIP DTMF Relay
Conguration of DTMF relay using SIP is similar to the H.323 conguration, except that
your dial peer must specically have the session protocol sipv2 command to enable SIP;
by contrast, H.323 is enabled by default. One DTMF relay method is compatible with both
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SIP and H.323. The dtmf-relay rtp-nte command is the relay method the two signaling
protocols use. The optional digit-drop keyword is also a valid SIP command to drop the
in-band tones.
The second DTMF relay method is unique to SIP:
sip-notify This SIP-only DTMF relay method species that out-of-band DTMF tones
are sent in SIP-notify messages between remote voice gateways. Both DTMF relay methods
can be congured, with the priority method being rtp-nte. Heres an example of how to
congure SIP signaling on VoIP dial peer 200 to use only the rtp-nte method that will also
drop in-band DTMF tone digits:
Router(config-dial-peer)#dtmf-relay rtp-nte digit-drop
Router#
Configuring MGCP DTMF Relay
DTMF relay support for the MGCP voice gateway signaling protocol can be congured
in one of two ways. Remember that MGCP relies heavily on the call-processing agent and
in many cases simply does what the call-processing agent asks of it. Along the same lines,
DTMF relay and relay negotiation can be completely handled by the call agent (CA), so
the voice gateway is oblivious to the process. It simply passes messages between the DTMF
relay endpoints. If you want, however, you can congure the voice gateway (GW) to have
control of the DTMF relay conguration and negotiation process with the remote end
gateway. This is useful when you dont want to completely rely on the call-processing agent
for signaling.
To congure DTMF relay for MGCP, you must globally enter the mgcp dtmf-relay
voip codec command followed by either the low-bit-rate or all keyword to specify
what codecs can be used. You then issue the mode keyword, followed by either nte-gw
or nte-ca to specify whether the gateway or call agent handles DTMF relay functions.
Finally, you must enable the dtmf-package if you are using the voice gateway for DTMF
negotiations. The following example shows how to congure the RFC 2833 standard
DTMF relay method, letting the voice gateway negotiate the relay method using any
voice codec:
Router(config)#mgcp dtmf-relay voip codec all mode nte-gw
Router(config)#mgcp package-capability dtmf-package
Router(config)#end
Router#
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Configuring Fax Support
Voice on an IP network oftentimes uses codecs that provide voice payload compression.
The greater the compression, the less bandwidth is consumed on a network. While
many compression methods work ne for the human voice, the same cannot be said for
fax transmissions. Fax transmissions are far more sensitive to the loss of delity that
compression cannot replicate on the other end. Additionally, VAD and echo cancellation,
which save bandwidth and improve the quality of voice calls, have a negative impact on fax
transmissions. The result of using a fax machine over a compressed voice circuit with VAD
and echo cancellation is that faxes come out garbled and incomplete.
Fortunately, we can choose from three options that can utilize an IP network while
providing consistent and reliable transmissions from one fax machine to another. There are
three different fax service over IP methods that we can implement:

Fax relay

Fax pass-through

T.37 store-and-forward fax
In the following sections, we will cover the differences between the three fax service
methods and show how to congure them.
Understanding Fax Relay
Fax relay is the VoIP fax transmission technique that has been around the longest of the
three possible methods. The analog fax transmissions are terminated at the voice gateway,
which then demodulates, packetizes, and retransmits the packets to the remote voice
gateway. This process is accomplished using either the Cisco fax relay or T.38 fax relay
method, explained next.
Cisco Fax Relay
Ciscos proprietary fax relay method uses special RTP packets to transport the communication
stream between voice gateways. Cisco fax relay is the default relay method and works only
when both voice gateways are Cisco hardware and utilize T.30-compatible fax machines. T.30
fax machines are dened by the ITU-T and are also dened as group 3 fax machines. They
use digital formats and compression methods for compiling the fax transmission but transmit
the signal in an analog stream. Using Cisco fax relay, the transmitting voice gateway creates a
virtual T.30 fax machine interface, which is responsible for demodulating the analog T.30 fax
transmission (using DSPs) coming inbound on the analog port that the real fax is attached to.
It then packetizes the demodulated fax transmission for transport on an IP network using the
special RTP packets. Once the packets reach the destination voice gateway, the voice gateway
remodulates the signal and sends the reconstructed analog transmission to the destination fax
machine. Figure 9.1 shows how Cisco fax relay transport functions.
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Keep in mind that the fax machines are not aware of the demodulation and remodulation
process because they transmit and receive only analog signals. The conversion that occurs
at the voice gateway is completely transparent, and to them, it looks as if they are still using
a completely analog system from end to end. Cisco fax relay works with either the H.323 or
SIP signaling protocol.
ITU-T T.38 Fax Relay
The alternate to Cisco fax relay is to use the ITU-T T.38 fax relay standard method,
which is useful in situations where one voice gateway is a non-Cisco device. This method
is compatible with H.323, SIP, and MGCP voice gateway signaling protocols. One
main conguration difference between the Cisco and T.38 relay methods is that the
T.38 method requires you to create independent dial peers for the transmission of T.38
Internet fax packets (IFP). Also, just as with the Cisco method, T.38 relay requires that
your transmitting and receiving fax machines conform to the ITU-T T.30 standard for
transmitting digital streams and can handle digital compression natively.
The process of demodulating and remodulating a fax transmission between voice
gateways is similar to the Cisco fax relay method with the exception that the T.38 method
uses its own method to packetize and transport fax streams on an IP network, and it
doesnt use the concept of virtual fax interfaces. Figure 9.2 shows how the T.38 fax
transmission transport functions.
Voice
gateway
V
Voice
gateway
V
Fax
Ext: 3000
Fax to
4000
T.30 T.30 Cisco proprietary RTP
VoIP
Demodulation
and
packetization
Depacketization
and
remodulation
Fax
Ext: 4000
FI GURE 9.1 The Cisco fax relay process
Voice
gateway
V
Voice
gateway
V
Fax
Ext: 3000
Fax to
4000
T.30 T.30 ITU-T T.38
VoIP
Demodulation
and
packetization
Depacketization
and
remodulation
Fax
Ext: 4000
FI GURE 9. 2 T.38 fax relay
c09.indd 358 9/21/11 11:25:38 AM
Again, as in the Cisco fax relay method, the process is transparent, and the fax
machines never realize that they are using an IP network for transport.
Configuring Cisco Fax Relay
Fax relay can be enabled either on a global basis while in config-voi-serv conguration
mode or on an individual dial-peer level while in conf-dial-peer conguration mode. If
both methods are used on a voice gateway, the dial-peer setting takes precedence over the
global setting. Here is an example of conguring Cisco fax relay globally:
Router(config-voi-serv)#fax protocol
Router#
The same command in config-dial-peer mode will enable Cisco fax relay on an
individual dial peer.
Configuring T.38 Fax Relay
Because the Cisco fax relay method is the default on Cisco routers, it takes a few additional
conguration commands to enable T.38 fax relay.
Configuring T.38 Fax Relay with H.323 and SIP
Enabling T.38 fax relay either locally or globally with the H.323 or SIP protocol is a matter
of rst conguring all of the necessary and compatible signaling and DTMF tones for your
respective protocol. Then its a matter of issuing the fax protocol t38 command. There
are several optional keywords that you can enter at the end to modify how T.38 fax relay
operates; here is a summary of them:
nse Uses Cisco proprietary named serviceg events (NSE) for fax signaling. An additional
force keyword requires the use of NSEs; it is used for interoperability between H.323/SIP
T.38 congurations and MGCP T.38 congurations.
ls-redundancy Stands for low-speed redundancy and species the number of redundant
T.38 packets to be sent using the low-speed V.21-based T.30 protocol. The range is either
0 to 5 or 0 to 7 depending on the hardware platform you are using. The default number of
redundant packets is 0.
hs-redundancy Stands for high-speed redundancy and species the number of
redundant T.38 packets to be sent using the high-speed V.17, V.27, V.29, T.4, or
T.6 protocol. Range can be set between 0 and 3, and the default number of redundant
packets is 0.
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In addition to the T.38 conguration options, there is an alternate way to send fax
transmissions in the event that T.38 fax relay fails. There are two different options, and
they are enabled using the fallback keyword followed by one of the following:
cisco Attempts to use the Cisco proprietary fax relay method.
pass-through Attempts to use the fax pass-through method of transmitting fax messages.
The codecs that can be used are either G.711mu-law or G.711a-law. The fax pass-through
is explained in the next section.
The following example shows how to congure T.38 fax relay on VoIP dial peer 80
using SIP. We will also set the ls-redundancy and hs-redundancy to 3 and specify that the
dial peer attempt to use Cisco fax relay in the event the T.38 method fails.
Router(config-dial-peer)#fax protocol t38 ls-redundancy 3 hs-redundancy 3
fallback cisco
Router#
Always remember that fax relay can also be congured globally while in conf-voi-serv
conguration mode. The specic dial-peer congurations take precedence over the global
congurations if both are used.
Understanding and Configuring Optional Fax SIP
and H.323 Settings
CVOICE candidates should be familiar with a few optional conguration commands.
The rst command is fax rate, which is used to statically assign the transmission rate for
outbound T.38 transmissions. The second command is fax-relay, which is used either
to disable error checking or to suppress tones so the sending and receiving Super Group 3
(SG3) fax machine can negotiate a lower group 3 (G3) speed.
fax rate <rate> voice This command statically sets the fax transmission rate. The
default rate depends on the codec method in use and is always set to the maximum possible.
The value can be lowered, which alters the maximum possible rate to one of the following:

2400 bps

4800 bps

7200 bps

9600 bps

12000 bps

14400 bps
The command is congured while in config-dial-peer mode. Here is an example of how
to congure a static transmission rate of 7200 bps on VoIP dial peer 10:
c09.indd 360 9/21/11 11:25:39 AM
Router(config-dial-peer)#fax rate 7200 voice
Router#
fax-relay The fax-relay command is used to enable/disable two distinct fax features.
The command can be congured globally or on an individual dial peer.
One task for the fax-relay command is to disable error correction mode (ECM) using the
ecm disable keywords. This is commonly done on congested WAN links. By default, ECM
is enabled. The following example shows how to disable fax-relay ECM globally:
Router(config-voi-serv)#fax-relay ecm disable
Router#
When Error Correction Goes Wrong
Tiffany was in the process of converting multiple remote sites, that used analog lines for
fax connections to the corporate ofce, to use the IP WAN connections instead. The rst
three site conversions went smoothly, but she began struggling with fax transmission
failures at the fourth remote site.
After some investigation, it turned out that while the IP WAN connection was in no way
overutilized, it had a history of dropped packets. In fact, her tests showed almost a
3 percent packet loss from the remote site to the corporate ofce. The fax machines
being used utilized fax-relay packet loss concealment features. These are fax machines
that scan the entire image and store it in local memory. Once a page has been scanned,
the sending fax transmits the data in a series of frames. The receiving fax machine then
receives the frames and checks for any errors using ECM.
Tiffany found that ECM is enabled by default on her T.38 fax-relay voice gateway
conguration. In addition, Cisco recommends that ECM be disabled on networks with
a packet loss of 2 percent or more, because ECM requires a 100 percent error-free fax
transmission. If its not completely error free, the transmission fails.
After disabling ECM on her voice gateways, Tiffany was able to successfully send and receive
faxes between the remote and corporate ofces. This bought her enough time to investigate
the root cause of why her IP WAN was experiencing a high rate of dropped packets.
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A second task is to congure tone suppression. Tone suppression is a way to block fax
tone transmissions so that they can be transported at a lower rate. You might want to
do this if you have an overutilized link and want to lower the amount of bandwidth
a fax transmission requires. There are two tone-suppression methods. The fax-relay
ans-disable command is used to suppress answer (ANS) tones from the sending SG3-
compatible fax machine. This allows SG3 fax machine negotiation of lower, group 3
(G3) connection speeds. Super Group 3 (SG3) is a standard fax transmission method
that supports speeds up to 33.6 Kbps. Sometimes this speed is not reliable, or too much
bandwidth is consumed. That is why it is sometimes necessary to force the downgrade to
lower G3 speeds, which can be more reliable and consume less bandwidth.
The fax-relay sg3-to-g3 command essentially does the same thing as the ans-disable
keyword, except that the two voice gateways simply negotiate slower G3 speeds as opposed
to suppressing ANS tones.
Configuring T.38 Fax Relay with MGCP
Similar to conguring DTMF relay on MGCP gateways, negotiation of T.38 fax relay can
be controlled by either a call-processing agent or the voice gateway itself. The call agent
(CA) method is enabled by default, and therefore the actual conguration is performed on
the CUCM and is outside the scope of this study guide. Therefore, we will rst look at how
to congure gateway (GW)-controlled MGCP T.38 fax relay. Additionally, we will congure
MGCP T.38 fax relay to interoperate with SIP- and H.323-congured T.38 gateways.
The commands needed to congure T.38 fax relay using MGCP are performed in
global conguration mode. Two commands are required to enable this properly on a
voice gateway. The rst command is mgcp fax t38. There are several optional but useful
keywords that can follow the mgcp fax t38 command, described here:
ecm Enables error correction mode (ECM) for the gateway, which better ensures the
proper receipt of all packets. Keep in mind, however, that network congestion caused by
ECM can also cause more failures on slow or unreliable networks.
gateway force Forces the gateway-controlled fax relay service to use Ciscos
proprietary NSEs.
ls_redundancy Stands for low-speed redundancy and species the number of redundant
T.38 packets to be sent using the low-speed V.21-based T.30 protocol. Its range can be set
between 0 and 5; the default number of redundant packets is 0.
hs_redundancy Stands for high-speed redundancy and species the number of redundant
T.38 packets to be sent using the high-speed V.17, V.27, V.29, T.4, or T.6 protocol. Its
range can be set between 0 and 2; the default number of redundant packets is 0.
nsf Overrides non-standard facilities (NSF) code with unique code that depends on
the two-digit hexadecimal code entered to specify the fax-machine vendor. NSFs are
proprietary capability codes that fax-machine makers may build into their equipment.
By default, the NSF code is not overridden. You need to look up the code for the specic
vendor that you are using in order for this to work.
inhibit This command disables T.38 fax relay on MGCP. By default, T.38 fax relay is enabled.
c09.indd 362 9/21/11 11:25:40 AM
The second required command is mgcp tse payload, followed by a unique number to
specify the telephony service event (TSE) payload size. TSEs are special messages that can
provide a way to communicate telephony events between MGCP gateways. The valid TSE
range is 98 to 119. The default is 100; but this command is disabled by default, so it must be
enabled in order for the voice gateway to enable in-band signaling events. Also keep in mind
that the TSE payload sizes must be the same on both the source and destination voice gateways.
The following shows an example of how to congure MGCP T.38 signaling to use Cisco
NSEs and to enable error checking. The TSE payload size is enabled and set to 105:
Router(config)#mgcp fax t38 gateway force
Router(config)#mgcp tse payload 105
Router(config)#end
Router#
Understanding and Configuring Optional Fax MGCP Settings
You should be familiar with two optional MGCP fax conguration commands. The rst
adjusts the transmission rate at which a fax is sent. The other sets the amount of time that
a voice gateway waits to receive an NSC response packet from a peer. Lets briey look at
how to congure both.
mgcp fax rate This command statically sets the fax transmission rate. The default rate
depends on the codec method in use and is always set to the maximum. The value can be
lowered so that less bandwidth is used, but the transmission will take longer. The voice
keyword will reset any statically-assigned rate and revert to using the highest possible rate
for the codec being used. The command is congured while in global conguration mode.
Here is an example of how to congure a static transmission rate of 9600 bps:
Router(config)#mgcp fax rate 9600
Router(config)#end
Router#
mgcp timer nse-response t38 This command is used to change the default time that the
local voice gateway waits to receive a T.38 NSE response message from the remote voice
gateway. The possible range is 100 to 3000 ms, with the default being 200. It is sometimes
necessary to increase the wait time on low-bandwidth links. The following example
congures the T.38 NSE response wait time to 1000 ms:
Router(config)#mgcp timer nse-response t38 1000
Router(config)#end
Router#
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Understanding Fax Pass-through
Fax pass-through is the simplest yet least-reliable fax transport method. It can come in
handy, however, whenever you run across equipment that doesnt support any of the other
compatible fax methods. The pass-through method transports fax transmissions the same
way that voice calls are transmitted. The only difference is that when fax pass-through
is enabled, it ensures that fax transmissions are encoded using either G.711mu-law or
G.711a-law, which provides a high-quality digital representation of the original analog
source. Figure 9.3 shows how fax transmissions ow through the IP network.
Voice
gateway
V
Voice
gateway
V
Fax
Ext: 3000
Fax to
4000
T.30
VoIP
G.711a-law or
mu-law
enforced
Fax
Ext: 4000
FI GURE 9. 3 Fax pass-through
Using fax pass-through, even if voice calls are negotiated to a higher-compression
codec such as G.729a, the voice gateway detects fax transmissions locally by listening
for a 2100 Hz tone. As soon as this tone is detected, the local voice gateway sends an
NSE message in the RTP stream to inform the remote gateway that this stream is a fax
transmission and that the negotiated codec should be G.711.
Fax pass-through can operate with H.323, SIP, and MGCP protocols and is a fairly
simple process to congure on a voice gateway. The voice gateway can be congured
either globally while in conf-voi-serv conguration mode or on a single dial peer while
in config-dial-peer mode. Keep in mind that individual dial-peer congurations take
precedence over global congurations.
Configuring Fax Pass-through
Global conguration of fax pass-through is identical for H.323 and SIP. Here is an
example of how to congure fax pass-through globally:
Router(conf-voi-serv)#fax protocol pass-through g711ulaw
c09.indd 364 9/21/11 11:25:41 AM
Conguration of fax pass-through for MGCP is slightly different. There are three
required commands, which must be congured while in global conguration mode:
mgcp fax t38 inhibit This command is necessary to disable T.38 fax relay, which is
enabled by default on MGCP. If this command is not run, the voice gateway will attempt
to use T.38 fax relay with the peer voice gateway.
mgcp modem passthrough voip mode nse This command enables fax and modem pass-
through using NSE messages in a peer-to-peer fashion between two voice gateways.
mgcp package-capability rtp-package This command enables the RTP package for
MGCP so the two voice gateways can communicate with each other.
By default, the G.711 codec used is mu-law. This can easily be changed by issuing
the mgcp modem passthrough voip codec g711alaw command. The following example
shows how to congure fax pass-through on an MGCP voice gateway that uses
G.711a-law:
Router(config)#mgcp modem passthrough voip mode nse
Router(config)#mgcp modem passthrough voip codec g711alaw
Router(config)#mgcp package-capability rtp-package
Router(config)#mgcp fax t38 inhibit
Router(config)#end
Router#
Understanding T.37 Store-and-Forward Fax
T.37 store-and-forward fax is also commonly referred to as T.37 fax, because this is
the ITU-T standard that denes the fax-over-email method. Some fax machines natively
support T.37 and have the capability to send the contents of the fax transmission to
an email address instead of a telephone number. Alternatively, legacy fax machines
that do not support T.37 store-and-forward fax can be congured to send T.30 fax
transmissions to the destination fax as usual. The fax transmission passes through a voice
gateway congured for store-and-forward fax service. The voice gateway detects the fax
transmission and demodulates it. The demodulated transmission is then converted to an
image in a multipage tagged image le format (TIFF). The new le is then attached to an
email that is sent to a remote voice gateway, a T.37-compatible fax machine, or any device
that can receive email messages. If it is sent to the voice gateway, the gateway receives the
email and attached le. The TIFF image is converted back into a T.30 signal and sent to
the analog fax machine. Figure 9.4 shows an example of the store-and-forward method
between two phones that support only legacy T.30 transmissions.
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As you can see, the voice gateway that is responsible for taking the T.30 fax transmission
and converting it to a TIFF-attached email is known as the on-ramp gateway, and the voice
gateway that takes the TIFF-attached email and remodulates it back to a T.30 transmission
is called the off-ramp gateway. Obviously, a single gateway can be considered either the
on-ramp gateway or the off-ramp gateway depending on the direction in which the fax is
being sent on the network.
Because T.37 store-and-forward fax messages are sent in an email format,
they can easily be sent to multiple recipients in a single transfer.
The store-and-forward fax service is congured identically on voice gateways running
H.323 or SIP. MGCP does not support store-and-forward. The conguration of T.37 fax
is a fairly lengthy process. A separate interactive voice response (IVR) process is needed for
on-ramp and off-ramp gateway functionality. These processes are implemented on Cisco
voice gateways using Tool Command Language (TCL) scripts.
TCL scripts for on-ramp and off-ramp functionality can be found online
at Ciscos Software Support Center. The scripts are proprietary, and valid
CCO credentials are required in order to download them.
Of-ramp
voice gateway
V
On-ramp
voice gateway
V
Fax
Ext: 3000
Fax to
4000
T.30 T.30 Email
Conversion of
T.30 fax to
TIFF and sent as
email attachment
Conversion TIFF
to T.30 fax and
sent to Ext: 4000
PC- or T.37
capable
fax
SMTP
server
Fax
Ext: 4000
IP
Network
FI GURE 9. 4 T.37 store-and-forward fax
c09.indd 366 9/21/11 11:25:42 AM
Configuring Modem Support 367
In addition, because the process is email based, it requires an SMTP server to work.
SMTP is used to transport the message over the IP network between on- and off-ramp
gateways, and to send a conrmation of receipt back to the originating gateway. Because the
conguration process is complex, involving many components, and the conguration varies
greatly from one network to another, the CVOICE exam does not require you to know how
to congure store-and-forward fax, only that you understand how it works from a design
and signaling perspective.
Configuring Modem Support
Modem support on IP networks is similar to that for fax machines. The primary difference
between fax and modem transmissions is that data sent across a modem begins its life in a
digital format, while fax data is converted from paper to digital data using a scanning device.
A second difference is that with fax, the data transmission speeds are set prior to sending
of data, and these data rates are xed. Modem data rates, on the other hand, can uctuate
throughout a single connection. Modem trafc can be congured using either modem pass-
through or modem relay methods, which are discussed in the following sections.
Configuring Modem Pass-Through
Modem pass-through works similarly to fax pass-through, because modem signaling is
detected and then transported on a no-compression G.711 codec. This method can be
congured either globally, while in config-voi-serv conguration mode or on individual
dial peers. To enable modem pass-through, you use the modem passthrough command
followed by either the nse or system keyword. When conguring modem pass-through
globally, you have only the nse option, which stands for named service events. The system
option is available only when conguring pass-through on dial peers. This command
basically tells the voice gateway to use the options on the global conguration. There also
are other optional keywords that can be used to dene additional settings. All the modem
pass-through keyword options are listed here:
nse This keyword species that you use named service event packets for switching
the codec from a nonsupported codec to G.711. There also is an optional payload-type
keyword followed by a number specifying the payload for NSE packets. The default
payload type is 100, and the range is between 96 and 119 on most hardware platforms.
system This keyword is available only on dial-peer congurations and is used to tell the
voice gateway to use the globally congured modem pass-though conguration.
codec This keyword species which G.711 codec should be used. Your options are
g711ulaw and g711alaw. An optional redundancy keyword enables a single reception of
RFC 2198 packets to help ensure the receipt of the packets to improve reliability.
maximum-sessions This keyword sets a maximum number of simultaneous modem pass-
through sessions. The range of maximum-sessions varies by hardware platform.
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The following example shows how to congure modem pass-through on dial peer 20
that species use of the G.711-ulaw codec and the use of RFC 2198 redundancy packets:
Router(config-dial-peer)#modem passthrough nse codec g711ulaw redundancy
Router#
Configuring Modem Relay
Conguring modem relay is nearly identical to conguring modem pass-through, with just
a few exceptions. Modem relay can be congured either globally or at a dial-peer level.
To enable modem relay for H.323 or SIP, you use the modem relay command followed by
one or more options that are identical to the modem pass-through commands listed in the
previous section. There is one new keyword that can be used, however:
gw-controlled Species whether the voice gateway controls modem pass-through
transport parameters.
Here is an example of how to congure modem relay on a voice gateway using NSE and
gateway control functionality for SIP and H.323:
Router(config-voi-serv)#modem relay nse gw-controlled
Router#
For MGCP, you use the mgcp modem relay voip mode command with all of the
optional keywords available for H.323 and SIP. MGCP can only be congured globally,
but otherwise the commands are the same. Also remember that you need to enable the
necessary MGCP packages required for modem transport. The following example enables
modem relay on MGCP-operated voice gateways:
Router(config)#mgcp modem relay voip mode nse gw-controlled
Router(config)#mgcp package-capability dtmf-package
Router(config)#end
Router#
Configuring Voice Backup Paths
Part of a good voice network design is to provide backup mechanisms to help eliminate
single points of failure. The three different backup path designs we will focus on here are
c09.indd 368 9/21/11 11:25:44 AM
a WAN-to-PSTN fallback trigger using the preference command, an MGCP-to-H.323
fallback, and a COR and SRST conguration using CUCM Express.
Configuring a WAN-to-PSTN Fallback
Many IP voice networks that have multiple sites interconnected by way of a WAN
connection often use the WAN to transport voice as well. This will help to save money on
long-distance charges. But in case of a WAN failure, it is often necessary to congure a
backup path through the PSTN. While the PSTN will likely cost more, in most instances
the added cost is preferable to a lack of communication. To demonstrate how to congure
WAN-to-PSTN fallback, we will use Figure 9.5 as our example network.
As you can see, we have two sites connected by an IP WAN connection. A secondary
path will go through the PSTN. We want our voice gateways to use the WAN connection
as the primary path and only use the backup PSTN path in case of a WAN failure. To
accomplish this goal, we will congure two different dial peers to the same remote
destination. To force the voice gateway to choose the WAN path, we will use the
preference command in each dial peer and give the WAN path dial peer a lower preference
number. The lower the preference number, the more preferred it is.
We will congure the Seattle router with two dial peers (101 and 102). The VoIP dial
peer that uses the WAN path will have a preference of 1, while the POTS dial peer will
have a preference of 2 and therefore will be used only when the WAN is unavailable. The
commands to implement this are as follows:
Seattle#configure terminal
Seattle(config)#dial-peer voice 101 voip
Seattle(config-dial-peer)#destination-pattern 4....
Seattle
V
Las Vegas
Switch
Extensions:
4XXX
V
Switch
Extensions:
5XXX
Preferred path
B
a
c
k
u
p

p
a
t
h
PSTN
S0/0/0:23
172.16.1.1
IP WAN
FI GURE 9. 5 WAN-to-PSTN fallback
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Seattle(config-dial-peer)#no digit-strip
Seattle(config-dial-peer)#preference 1
Seattle(config-dial-peer)#session-target ipv4:172.16.1.1
Seattle(config-dial-peer)#exit
Seattle(config)#dial-peer voice 102 pots
Seattle(config-dial-peer)#destination-pattern 4...
Seattle(config-dial-peer)#prefix 13125554
Seattle(config-dial-peer)#preference 2
Seattle(config-dial-peer)#port 0/0/0:23
Seattle(config-dial-peer)#end
Seattle#
Dont forget that you will need to manipulate the calling digits as necessary. For
example, on VoIP dial peer 101, we trigger on four-digit extensions beginning with
the number 4. When an extension to our Chicago site is triggered and the WAN connection
is available, we will simply forward the four digits to the remote voice gateway. The
forwarding of all digits is accomplished using the no digit-strip command. But if the
WAN connection is not available, the voice gateway will use the next preferred path,
which is POTS dial peer 102. In this situation, we need to prepend the country code,
area code, and central ofce code. Additionally, we simply prepend the digit 4 and will
pass the other three wildcard digits along, effectively forwarding an 11-digit number to
the PSTN.
Configuring MGCP-to-H.323 Fallback
Our second example is a voice backup path that can be used when MGCP is the voice
gateway signaling protocol. As youve seen, MGCP is different from SIP and H.323 in that
all the call-routing intelligence (dial-peer rules) is controlled at the call control agent layer
and not directly on the voice gateway. In the event of a loss of communication between our
MGCP voice gateway and the call control agent, it would be nice to provide at least basic
calling service to calls going through our voice gateway. This would eliminate a single point
of failure in the event that the connection between the MGCP voice gateway and CUCM
(or CUCM cluster) is lost.
MGCP does not use dial peers but instead relies on the call-processing agent to tell the
voice gateway where to route calls. That means that if the gateway cannot communicate
with the CUCM, it doesnt have the intelligence to route calls on its own. To overcome the
lack of dial peers on a MGCP gateway, you can congure H.323 fallback on the router. With
H.323 fallback congured, the router will begin using H.323 signaling (and corresponding
dial peers that are congured locally) to route calls going into the private network and out
to the PSTN. The best way to show you how to congure H.323 fallback on MGCP voice
gateways is to walk through an example where the solution can be used. Our example will
use the network setup shown in Figure 9.6.
c09.indd 370 9/21/11 11:25:45 AM
Conguring MGCP to H.323 fallback would require the following three conguration
steps:
1. The rst command we enter tells our voice gateway that we are enabling MGCP fall-
back in the case that communications to the CUCM are lost:
Router(config)#ccm-manager fallback-mgcp
Router(config)#end
Router#
2. Next we enable MGCP fallback to the default signaling protocol H.323. This is simply
a matter of conguring the following:
Router(config)#application
Router(config-app)#global
Router(config-app-global)#service alternate default
Router(config-app-global)#end
Router#
Connection
failure
CUCM
192.168.1.2
PSTN
IP WAN
H.323 fallback
using dial peer
to PSTN
V
Voice
gateway
V
Switch
M
M
G
C
P
FI GURE 9. 6 H.323 fallback
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3. Then we congure any necessary POTS dial peers that will be controlled by H.323 in
the event the voice gateway needs to fall back. For example, we congure POTS dial peer
909 to accept any incoming calls and route NANP long-distance calls out the T1 CAS:
Router(config-dial-peer)#application mgcpapp
Router(config-dial-peer)#incoming called-number .
Router(config-dial-peer)#destination-pattern 91[2-8].........
Router#
Once completed, our voice gateway will use MGCP and the call-routing decisions
coming from the CUCM. But in the event of a loss of connectivity to the CUCM,
H.323 will take over and our dial peer will allow inbound and outbound NANP calls
to the PSTN. A good command to check the status of your voice gateway is the show
ccm-manager fallback-mgcp command. Here is an example of our router when it has
connectivity to the CUCM:
Router#show ccm-manager fallback-mgcp
Current active Call Manager: 192.168.1.2
MGCP Fallback mode: Enabled/OFF
Last MGCP Fallback start time: None
Last MGCP Fallback end time: None
And when our voice gateway loses connectivity, the MGCP fallback becomes enabled, as
shown here:
Router#show ccm-manager fallback-mgcp
Current active Call Manager: None
MGCP Fallback mode: Enabled/ON
Last MGCP Fallback start time: 11:25:42 UTC Apr 16 2011
Last MGCP Fallback end time: 11:25:18 UTC Apr 16 2011
Understanding and Configuring COR and SRST
In our previous example of MGCP-to-H.323 fallback, we enabled the voice gateway to fall
back to H.323, which can then utilize dial peers that are congured on the voice gateway.
The problem with this setup, however, is that all the phones are treated identically,
meaning they all can dial NANP numbers and nothing else. In many situations, we want to
allow some phones to call NANP, allow others to call NANP and international numbers,
and allow still other phones only local calling on the PSTN. In order to be able to do this,
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we need to congure Class of Restriction (COR) lists and place telephone extensions
within them. COR is commonly used to enforce dialing privileges in large voice networks,
and typically the restrictions are handled by the CUCM. But in the case that the voice
gateway loses connectivity with the CUCM, we can congure survivable remote site
telephony (SRST) in conjunction with local COR proles to get the job done. We will use
Figure 9.7 as our example of COR and SRST conguration to implement.
Connection
failure
CUCM
192.168.1.2
0/0/0:23 192.168.1.1
3006 3001 3000
PSTN
IP WAN
V
SRST voice
gateway
V
Switch
M
FI GURE 9. 7 COR and SRST
Lets rst look at understanding COR on IOS gateways and then how to apply COR lists
to dial peers when SRST kicks in.
Configuring Class of Restriction
Class of Restriction is a method used to dene calling search spaces, which can be applied
to a group of IP phones based on their extension number. COR lists can be dened as
inbound or outbound. Inbound COR lists are applied to calls.
When a call is made that passes through the router, it is matched with dial peers, which
dictate the next hop to the destination phone. If COR lists have been applied to the matched
dial peers, the following must be determined:

If the COR congured on the inbound dial peer has access rules equal to or greater
than the outbound COR list, the call can proceed. If the inbound COR list is not equal
to the outbound COR list, the call cannot proceed.

Additionally, if an inbound or outbound COR list is not dened on a dial peer, then the
voice gateway will use the default inbound or outbound COR list congured. The default
COR lists use the lowest priority setting and therefore will allow the call to proceed.
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The conguration of this on IOS routers begins with dening the COR members.
These are simply labels that will dene specic telephone access areas. Then actual
COR lists are dened within the COR members. The lists dene what numbers can be
dialed when the COR list is applied to a dial peer. For example, lets rst congure four
commonly congured COR members. To do this we must rst enter into config-dp-cor
conguration mode by using the dial-peer cor custom command. Once we enter this
mode, we can create our COR containers:
Router(config)#dial-peer cor custom
Router(config-dp-cor)#name 911
Router(config-dp-cor)#name local
Router(config-dp-cor)#name ld
Router(config-dp-cor)#name int
Router(config-dp-cor)#end
Router#
Now that we have our COR members dened, we can create our COR lists for both
inbound and outbound calls. First, well create our outbound COR lists that apply to calls
going out of the voice gateway (typically to the PSTN). In our example, we will congure four
COR lists that each contain their respective single COR membership that we congured earlier:
Router(config)#dial-peer cor list 911out
Router(config-dp-corlist)#member 911
Router(config-dp-corlist)#exit
Router(config)#dial-peer cor list localout
Router(config-dp-corlist)#member local
Router(config)#dial-peer cor list ldout
Router(config-dp-corlist)#member ld
Router(config)#dial-peer cor list intout
Router(config-dp-corlist)#member int
Router(config-dp-corlist)#end
Router#
Next, well create our inbound COR lists. These lists are used to dene the calling
privileges of the internal phones. Depending on the COR list, one or more COR members
are applied, as shown here:
Router(config)#dial-peer cor list 911
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Router(config)#dial-peer cor list local
Router(config)#dial-peer cor list ld
Router(config)#dial-peer cor list int
Router(config-dp-corlist)#member int
Router(config-dp-corlist)#end
Router#
So now we can see that phones dened to be within the 911 COR list are only allowed
to make calls within the 911 COR membership. Alternatively, the COR list named int
allows the phones to call numbers dened within all four COR memberships.
Then we need to apply our newly created COR lists to outbound dial peers that they are
associated with. For example, we will congure POTS dial peer 9 and dene it for local
destination patterns, as shown here:
Router(config-dial-peer)#destination-pattern 9[2-9]......
Router(config-dial-peer)#corlist outgoing localout
All the other required dial peers for our example are shown here:
Router(config-dial-peer)#corlist outgoing 911out
Router(config-dial-peer)#corlist outgoing 911out
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Router(config-dial-peer)#destination-pattern 91[2-9]..[2-9]......
Router(config-dial-peer)#prefix 1
Router(config-dial-peer)#corlist outgoing ldout
Router(config-dial-peer)#destination-pattern 9011T
Router(config-dial-peer)#prefix 011
Router(config-dial-peer)#corlist outgoing intout
Router#
Now we have our outbound dial peers dened on our voice gateway. The nal step is to
enable SRST and dene which phones are allowed which dialing privileges in the event of a
WAN failure.
Configuring Survivable Remote Site Telephony
The conguration of SRST on a compatible IOS gateway version is very similar to
conguring CUCM Express. The voice gateway can connect to IP phones that use either
SCCP or SIP signaling. This study guide will show you the basic steps to congure SRST
for SCCP. First, we need to enter into config-cm-fallback conguration mode and dene
our source IP address and max ephones and ephone-DNs as shown here:
Router(config)#call-manager-fallback
Router(config-cm-fallback)#ip source-address 192.168.1.1
Router(config-cm-fallback)#max-ephones 10
Router(config-cm-fallback)#max-dn 20
Router(config-cm-fallback)#end
Router#
Next, we need to dene our ephone-DNs and ephone settings. In this example, we will
congure an ephone and apply the ephone-DN of 3000 to button 1:
Router(config-ephone)#mac-address 0014.1c4d.2589
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Router#
Now that we have an SRST phone congured, we can go ahead and apply a COR list to
it through SRST, which is explained next.
Configuring SRST to Define COR Dialing Access
When our remote voice gateway fails to communicate with the CUCM, SRST will be
used to route calls. We also would like to enforce COR while in SRST mode. To do that,
we must rst enter into config-cm-fallback conguration mode and then dene the
telephone extensions or range of extensions that are applied a COR list. You can apply up
to 20 COR lists in SRST mode, and you need to number each rule individually using 120.
For example, we will assign extensions 3000 to 3005 on list 1 to use the local COR list and
extension 3006 on list 2 to dial any PSTN number:
Router(config)#call-manager-fallback
Router(config-cm-fallback)#cor incoming local 1 3000 3005
Router(config-cm-fallback)#cor incoming int 2 3006
Router(config-cm-fallback)#end
Router#
So the ephone that we congured with an extension of 3000 will be able to call outbound
on the PSTN to local numbers as well as to emergency services using 911 or 9911.
Toll Bypass and TEHO
One of the primary attractions of implementing a VoIP network is the inherent cost savings
that can be achieved. Cost savings are found in several areas. By combining voice and data
into a single network, you eliminate the need to support separate voice and data networks.
Additionally, a substantial saving in long-distance charges can be found when transporting
trafc over IP WAN connections, effectively eliminating long-distance charges. The key
point is that as long as you have enough bandwidth on your WAN connections, you pay
for that bandwidth whether you use it or not. So you may as well transport voice using
available IP bandwidth and avoid long-distances charges in the process.
Understanding Tail End Hop Off
The process of conguring your IP network to utilize WAN connections for voice transport
is known as toll bypass, and you already know how to congure this as well as how to
congure the PSTN as a backup in case of a WAN failure. You can extend the functionality
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378 Chapter 9

of toll bypass, however, when you congure what is known as tail end hop off (TEHO).
Toll bypass works great when you are calling from site to site where you have a high-speed
WAN connection to utilize. No matter what business you are in, eventually you are going
to have to use the PSTN to make calls to people outside of your VoIP network. But if you
have a vast IP network that covers a wide range of geographical locations, it is possible to
utilize the IP WAN to get remote calls as close to the PSTN destination as possible. The
calls can then be dumped off onto the PSTN locally, and the call will look as if it a local
call as opposed to a long-distance one.
In some countries, toll bypass and TEHO are not legal, and the PSTN must
provide transport for the entire call. If you are planning to implement either
of these features, you need to understand the laws in the regions you will
be covering.
To give you a better idea, lets look at Figure 9.8 as an example.
SanFran
V
Miami
Extensions:
5XXX
V
Ext 4000 call to
305-558-8442
To area code 305
PSTN
192.168.10.1 192.168.10.2
IP WAN PSTN
Local hop-of to PSTN
PSTN
Area code
305
Area code
415
FI GURE 9. 8 A TEHO network
Here you can see that we have our SanFran site interconnected with our Miami site
through an IP WAN. An employee at the SanFran site needs to call someone located in
Miami, but the user is not part of the IP network. We can congure our network, however,
to use the IP WAN to transport the call from SanFran to Miami, and then the voice
gateway in Miami will route the call out their PSTN connection. The call therefore looks
to the PSTN as if it is a local call, and no long-distance charges will be incurred. Lets walk
through the process of conguring TEHO on our SanFran and Miami voice gateways.
Configuring Tail End Hop Off
The secret to conguring TEHO lies in dial-peer congurations and digit translations. Its
really just a matter of determining the following:

What area codes are considered local in Miami?

What area codes are considered local in SanFran?
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Then you need to do the following:

Congure digit manipulation for Miami to SanFran local PSTN calls so that the
remote ANI is displayed.

Congure digit manipulation for SanFran to Miami local PSTN calls so that the
remote ANI is displayed.

Congure an outbound VoIP dial peer on the SanFran voice gateway to transport
Miami PSTN calls across the IP WAN.

Congure an outbound VoIP dial peer on the Miami voice gateway to transport San-
Fran PSTN calls across the IP WAN.
Once these steps are completed, long-distance PSTN calls will be transported over the
IP WAN. Once they reach the remote voice gateway, the standard PSTN dial peers already
congured will properly route the call to the PSTN and complete the local PSTN call.
To make our example a little easier to understand, we are going to assume that there
is only one area code in Miami and one area code in SanFran that are considered local.
The Miami area code is 305 and the SanFran area code is 415. Therefore, if someone in
SanFran calls someone in the Miami area code, we will use TEHO to make the call and
save on long-distance charges. The same will be true for Miami employees calling the
SanFran 415 area code.
The next step is to create translation rules and proles so the correct ANI will be
registered at the opposite end. We will start with our SanFran router to create a translation
rule and prole to add the SanFran area code of 415 and CO code of 555:
SanFran#configure terminal
SanFran(config)#voice translation-rule 1
SanFran(cfg-translation-rule)#rule 1 /^4/ /1415555/
SanFran(cfg-translation-rule)#exit
SanFran(config)#voice translation-profile miami-teho-loc-code
SanFran(cfg-translation-profile)#translate calling 1
SanFran(cfg-translation-profile)#end
SanFran#
Next, we need to create our VoIP dial peer that routes calls to the Miami 305 area code
out the IP WAN. We will also apply our translation prole outbound so the proper ANI
number will be sent to the Miami voice gateway:
SanFran#configure terminal
SanFran(config)#dial-peer voice 305 voip
SanFran(config-dial-peer)#destination-pattern 91305.......
SanFran(config-dial-peer)#session-target ipv4:192.168.10.2
SanFran(config-dial-peer)#translation-profile outgoing miami-teho-loc-code
SanFran(config-dial-peer)#end
SanFran#
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At this point, calls originating from the VoIP network in SanFran destined to Miami
phones with area code 305 will be sent across the IP WAN to the Miami voice gateway.
The Miami voice gateway should then be set up with typical dial peers for local, long-
distance, and international calling. When the call comes in from the WAN, the Miami
voice gateway will match it with a dial peer out to the PSTN, and the local call will be
made. The Miami voice gateway will also forward the ANI information it received from
the SanFran voice gateway. To complete our conguration, we will now congure the
Miami voice gateway for local calling to the 415 area code:
Miami#configure terminal
Miami(config)#voice translation-rule 1
Miami(cfg-translation-rule)#rule 1 /^2/ /1305555/
Miami(cfg-translation-rule)#exit
Miami(config)#voice translation-profile sanfran-teho-loc-code
Miami(cfg-translation-profile)#translate calling 1
Miami(cfg-translation-profile)#exit
Miami(config)#dial-peer voice 305 voip
Miami(config-dial-peer)#destination-pattern 91305.......
Miami(config-dial-peer)#session-target ipv4:192.168.10.1
Miami(config-dial-peer)#translation-profile outgoing sanfran-teho-loc-code
Miami(config-dial-peer)#end
Miami#
Always be aware of the increased bandwidth this can require on your IP WAN link. If
you are planning to implement toll bypass and TEHO, you should closely monitor WAN
utilization. You should also seriously consider implementing QoS techniques on your IP
WAN. QoS is detailed in Chapters 11, Introduction to Quality of Service (QoS), and 12,
Conguring Quality of Service, of this study guide.
Configuring Call Blocking
In many business voice networks, there are telephone numbers that you simply dont want
anyone to call or, conversely, numbers that cannot call you. This usually includes premium-
rate numbers that are typically in the 900 and 976 area code ranges in North America. Call
blocking is typically performed at the call-processing agent level, either on CUCM, CUCMBE,
or CUCM Express. Inbound call blocking can also be performed on SIP and H.323 gateways
by using translation-reject rules and applying the translation prole to an incoming dial peer.
From a voice gateways perspective, incoming dial peers are either coming
from telephony equipment that is connected to a telephony interface (such
as a T1 from the PSTN) or coming from a VoIP peer device.
c09.indd 380 9/21/11 11:25:50 AM
Configuring Call Blocking 381
To demonstrate how call blocking works on voice gateways, we will use an example where
we want to block all incoming calls from our T1 PRI that begin with the number 123555.
The rst step to take is to create our voice translation rule (rule 123) that will be
triggered by that number and prevent the call from completing. Here is how we can
congure our specic rule:
Router(cfg-translation-rule)#rule 1 reject /^123555/
Router(cfg-translation-rule)#end
Router#
Now that we have our voice translation rule dened, we can make our voice translation
prole and apply our rule. To do this, we will congure a new voice translation prole
and name it block_900. Once we are in cfg-translation-profile mode, we apply
translation rule 123, as shown here:
Router(config)#voice translation-profile block_900
Router(cfg-translation-profile)#translate calling 123
Router(cfg-translation-profile)#end
Router#
Our nal step will be to apply the voice translation prole to our inbound POTS dial
peer from the PSTN. In our case, lets say we already have POTS dial peer 100 congured
as our inbound dial peer for our PSTN connection. Therefore, we simply apply the
following conguration:
Router (config)#dial-peer voice 100 pots
Router(config-dial-peer)#call-block translation-profile incoming block_900
Router#
Now, our voice gateway will block any number with a calling number of 123555. It will
return a no service disconnect message to the calling party.
If you want to change the disconnect message from the default, you can
do so by using the call-block disconnect-cause incoming command
followed by one of four different disconnect causes that are configured in
the dial peer. These causes are shown in the following output:
Router(config-dial-peer)#call-block disconnect-cause incoming ?
call-reject Call Reject
invalid-number Invalid Number
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unassigned-number Unassigned Number
user-busy User Busy
Router(config-dial-peer)#call-block disconnect-cause incoming
We can test our call-blocking conguration by using the test voice translation-rule
command and specifying translation rule 123 along with a number beginning with 123555,
as shown here:
Router#test voice translation-rule 123 1235556542
1235556542 blocked on rule 1
Router#
Summary
Voice gateways are a bit of a Swiss army knife in terms of what they can be
congured to do for voice services, reliability, and customization. In Chapter 9, we
moved beyond basic voice gateway conguration to congure various scenarios that you
will likely face in the real world. We explored ways to more effectively transport DTMF,
fax, and modem transmissions when using H.323, SIP, and MGCP signaling. In addition,
we looked at several voice backup design methods that help to provide the mission-critical
reliability that is required in most voice networks. Lastly, we looked at various ways we
can restrict, bypass, and block calls that should turn into PSTN cost savings if
implemented properly.
Exam Essentials
Know the four methods for configuring H.323 DTMF relay. H.323 DTMF relay can be
congured using Ciscos proprietary method, the H.245 alphanumeric method, the H.245
signal method, and the RFC 2883 RTP-NTE method.
Know the two methods for configuring SIP DTMF relay. SIP DTMF can be congured
using the RFC 2883 RTP-NTE method or the SIP notify method.
Know the two methods for configuring MGCP DTMF relay. SIP DTMF relay
can be congured using the CUCM call agent (CA) or locally using the voice gateway (GW)
method.
Know the three fax-relay methods. The three methods are fax relay, fax pass-through,
and T.37 store-and-forward fax.
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Understand the two types of fax relay. The Cisco fax relay method is proprietary and
uses Cisco voice gateways to demodulate and remodulate analog transmissions across an IP
network. The T.38 method is essentially the same, but it uses ITU-T open standards.
Understand how fax pass-through functions. Fax pass-through simply creates a voice
channel using a low-compression codec for the transport of voice. It is the simplest yet least
reliable method.
Understand the benefits of T.37 store-and-forward fax. The advantages are in its
reliability and ease in sending a single message to multiple recipients because of its use of
SMTP for transport.
Know the two types of modem support on Cisco voice gateways. Cisco supports modem
pass-through and modem relay.
Understand the primary differences between sending fax transmissions and sending modem
transmissions. With modem transmissions, the data starts out in a digital format. Additionally,
modem speeds can vary throughout a transmission, but a fax rate is statically set.
Know the key command used to create WAN-to-PSTN fallback configurations. The
preference command is used to create multiple dial peers to the same destination but assign
a priority to them. That way, the voice gateway will always choose the most preferred route
that is currently available.
Understand the importance of configuring MGCP fallback. MGCP fallback is important
in cases where your MGCP gateway cannot communicate with the call-processing agent.
Understand the purpose of COR. Class of Restriction is used to create a structured set
of dialing rules that can be applied to a phone or a group of phones. It is often used in
situations where some phones should only be allowed internal, local, long-distance, or
international dialing access.
Know how to configure basic SCCP SRST. SRST is congured similarly to the way
CUCM Express is congured. You create an IP source address for the phones so they know
where the call-processing agent (the SRST router) resides. Then its a matter of conguring
ephone-DNs and associating them to buttons on ephones.
Understand the purpose of toll bypass and TEHO. Toll bypass and TEHO are congured
on networks that span a large geographical region in order to save on PSTN long-distance
charges. TEHO takes toll bypass one step further by dropping off connections at the PSTN
of a remote site so the call is technically local, as opposed to long distance.
Know how to configure call blocking on a voice gateway. Call blocking can be congured
for inbound calls into a voice gateway. To do this, you can create a translation rule and
prole that reject a number or range of numbers that you choose to block. Once the rule is
created, you then apply it as an inbound rule on your dial peer.
Exam Essentials 383
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Written Lab 9.1
1. What is the command that congures H.323 DTMF relay for the method that uses
RFC 2883 and explicitly drops in-band tones from being sent?
2. What fax transport method uses the concept of virtual fax machines?
3. What config-dial-peer command is used to statically set the fax transmission speed
to 9600 bps?
4. What SIP fax-relay command can be used to help ensure transmissions succeed on a
link with more than 2 percent packet loss?
5. What are the names of the two MGCP fax-relay conguration methods?
6. What tone frequency does a voice gateway, congured for fax pass-through, listen for
to detect the transmission of a line?
7. You are conguring a voice gateway for redundant paths. What config-dial-peer
command will ensure that the dial peer will be used unless unavailable because of a
failure?
8. What command can be used to verify the status of MGCP-to-H.323 fallback?
9. What config-dial-peer command can be used to set an outbound COR list named
int-out?
10. When conguring, you should use voice translation rules and proles to send the
correct to the PSTN.
Hands-On Labs
To complete the labs in this section, you need two Cisco routers with a voice-capable IOS.
Each lab in this section builds upon the last and will follow the logical voice gateway design
SiteA
V
SiteB
Ext: 888-3434
V
PSTN
S1/0:23
10.1.1.1
IP WAN PSTN
Area code
555
Area code
123
Ext: 456-7890
FI GURE 9. 9 Toll bypass and TEHO labs
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This lab assumes that basic IP networking and the local voice network is operational.
We are only concerned with conguring SiteAs voice gateway.
Lab 9.1: Conguring Toll Bypass and PSTN Redundancy
Lab 9.2: Conguring TEHO
Hands-On Lab 9.1: Configuring Toll Bypass
and PSTN Redundancy
In this lab, were going to congure the SiteA voice gateway for toll bypass to SiteB. The SiteB
extensions all begin with a site code of 888 followed by a four-digit extension. The preferred
path to SiteB should be across the IP WAN with a backup path across the PSTN. Additionally,
make sure to congure an off-network dial peer using 9 as the off-network trigger digit. SiteA
phones should be able to access local, long-distance, and international phones.
2. We will rst create dial peer 888 to send calls to SiteB phones across the IP WAN. To
do this, enter conf-dial-peer mode by typing dial-peer voice 888 voip.
3. Congure a destination-pattern to match on site-code 888 followed by four wildcard
digits, by typing destination-pattern 888....
4. Congure the dial peer to forward all seven digits, by typing no digit-strip.
5. Congure the next-hop destination to send calls over the IP WAN to SiteB, by typing
session-target ipv4:10.1.1.1.
6. Congure the dial peer to be the preferred path, by typing preference 1.
8. Create dial peer 889 to send calls to SiteB phones across the PSTN if the IP WAN
is unavailable. To do this, enter into conf-dial-peer mode by typing dial-peer
voice 889 pots.
9. Congure a destination-pattern to match on site-code 888 followed by four wildcard
digits by typing destination-pattern 888....
10. Congure the dial peer to add the USA nation code and area code for SiteB so the
correct 11-digit number is sent to the PSTN, by typing prefix 1555.
11. Congure the dial peer to forward all seven of the entered digits, by typing no
digit-strip.
12. Congure the next-hop destination to send calls out the T1 and over to the PSTN, by
typing port 1/0:23.
13. Congure the dial peer to be the backup path, by typing preference 2.
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15. Create dial peer 9 to send all off-network calls (except for SiteB phones) out the PSTN.
To do this, enter into conf-dial-peer mode by typing dial-peer voice 9 pots.
16. Congure a destination pattern to match on all numbers starting with 9, by typing
destination-pattern 9T.
17. Congure the next-hop destination to send calls out the T1 and over to the PSTN, by
typing port 1/0:23.
Hands-On Lab 9.2: Configuring TEHO
In Lab 9.2 we will congure TEHO on the SiteA voice gateway for all calls destined for
the 555 area code. We also want to make sure the correct ANI is displayed when phones at
SiteA use TEHO to call off-network phones in the 555 area code.
2. Create a voice translation rule by typing voice translation-rule 1. You will be placed
into cfg-translation-rule mode.
3. Create a rule that appends 1123 to the beginning of the seven-digit ANI that is
congured on phones at SiteA by typing rule 1 /^2/ /1123/.
4. Exit cfg-translation-rule mode by typing exit.
5. Congure a voice translation prole (named siteb-teho-loc-code) by typing voice
translation-profile siteb-teho-loc-code. You will be placed into cfg-translation-
profile mode.
6. Add translation rule 1 that we just created as a calling match rule by typing translate
calling 1.
7. Exit cfg-translation-profile mode by typing exit.
8. Congure a new VoIP dial peer that matches destination telephone numbers that
start with 91555 followed by seven wildcard digits by typing destination-pattern
91555.......
9. Congure the next-hop destination to send calls over the IP WAN to SiteB by typing
session-target ipv4:10.1.1.1.
10. Apply our translation prole to append a country code and area code to the ANI of the
calling number by typing translation-profile outgoing siteb-teho-loc-code.
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Review Questions
1. You configure the following config-dial-peer DTMF relay command:
Router(config-dial-peer)#dtmf-relay h245-alphanumeric h245-signal rtp-nte
What order of preference will the voice gateway use for the signaling method?
A. 1. h245-alphanumeric
2. cisco-rtp
3. h245-signal
4. rtp-nte
B. 1. cisco-rtp
2. rtp-nte
3. h245-signal
C. 1. cisco-rtp
2. h245-signal
4. rtp-nte
D. 1. h245-alphanumeric
2. rtp-nte
3. cisco-rtp
4. h245-signal
2. If DTMF relay support is not configured, how are DTMF tones transported across IP networks?
A. In band
B. In the data channel
C. As a pulse
D. In the bearer channel
3. What voice gateway fax transport method transmits fax messages over an IP network the
same way that voice messages are transported?
A. T.30
B. T.37
C. Fax relay
D. Fax pass-through
4. What fax relay method uses the concept of virtual fax interfaces?
A. T.37
B. Cisco fax relay
C. Fax pass-through
D. Store-and-forward fax
E. T.38
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5. You are reviewing a SIP voice gateway configured with T.38 fax relay and see the following
configuration entry:
dial-peer voice 101 voip
session protocol sipv2
fax protocol t38 ls-redundancy 3
What does ls-redundancy 3 mean?
A. Will attempt to use the Cisco proprietary fax-relay method as a backup in case T.38 fails
B. Specifies the number of times a T.38 fax can fail before giving up on low-speed
connections
C. Uses the group 3 fax transmission method
D. Specifies sending three T.38 packets using the low-speed V.21-based T.30 protocol
6. What is the maximum fax transmission rate for SG3 fax machines?
A. 14,400 bps
B. 28,800 bps
C. 33,600 bps
D. 56,000 bps
7. What are the two fax tone transmissions that can be used on T.38 fax relay?
A. fax protocol t38 nse
B. fax protocol t38 tse
C. fax-relay sg3-to-g3
D. fax-relay ans-disable
8. You are reviewing an MGCP voice gateway T.38 fax-relay configuration and run across the
following command entry:
mgcp fax t38 ecm gateway force
What is the purpose of the gateway force keyword?
A. It overrides nonstandard facilities (NSF) code with a unique code that is dependent on
the two-digit hexadecimal code entered to specify the fax-machine vendor.
B. It disables T.38 fax relay on MGCP. By default, T.38 fax relay is enabled.
C. It requires the use of Cisco proprietary named service events (NSEs).
D. It enables error correction mode (ECM) for the gateway, which better ensures the
proper receipt of all packets.
9. How are faxes transmitted across the IP network using the T.37 store-and-forward method?
A. TIFF images attached in emails
B. TCL scripts attached in emails
C. In special RFC 2883 RTP packets
D. In special Cisco proprietary packets
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10. Which of the following external components is required to operate T.37 store-and-forward fax?
A. CUCM, CUCM BE, or CUCM Express
B. Unity or Unity Express
C. SMTP server
D. DHCP server
11. Which of the following is not a primary difference between analog fax and modem
transmissions?
A. Fax transmission is not supported on MGCP gateways.
B. Modem transmission is only supported using pass-through techniques.
C. Modem transmission is only supported on H.323 and SIP networks.
D. Modem transmission rates can vary.
12. Which of the following commands and command modes is used to configure locally
controlled modem relay using MGCP NSE?
A. Router(config)#mgcp modem relay voip mode nse gw-controlled
B. Router(config-dial-peer)#mgcp modem relay voip mode nse gw-controlled
C. Router(config)#mgcp modem relay voip mode nse ca-controlled
D. Router(config-dial-peer)#mgcp modem relay voip mode nse ca-controlled
13. What command is needed within the dial-peer configuration when configuring MGCP to
H.323 fallback?
A. ccm-manager fallback-mgcp
B. service alternate default
C. application mgcpapp
D. mgcp-to-h323
14. What show command can be used to see if MGCP to H.323 fallback has taken place?
A. show ccm-manager fallback-mgcp
B. show ccm-manager fallback-h323
C. show voice fallback mgcp
D. show voice fallback h323
15. When is it common to configure COR on a voice gateway?
A. On large networks
B. On networks that utilize MGCP
C. On networks that utilize H.323 or SIP
D. When voice gateways are configured for SRST
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16. Which of the following commands and configuration modes are used to begin the process
of creating COR members?
A. Router(config)#dial-peer cor custom
B. Router(config-voi-serv)#dial-peer cor custom
C. Router(config-voi-serv)#dial-peer cor list
D. Router(config)#dial-peer cor list
17. Which of the following will assign extensions 40004010 to COR list 3 using a privilege of
emergency?
A. cor incoming emergency 3 4000-4010
B. cor assign emergency 3 4000-4010
C. cor incoming emergency 3 4000 4010
D. cor assign emergency 3 4000 4010
18. You are reviewing a voice gateway configured for TEHO and come across the following
configuration commands in the running configuration:
dial-peer voice 555 voip
destination-pattern 91555.......
session-target ipv4:192.168.1.10
translation-profile outgoing remote-teho-loc-code
What is likely to be the purpose of the translation-profile outgoing
remote-teho-loc-code configuration entry?
A. To hide the calling number from the remote voice gateway
B. To block the calling number from the remote voice gateway
C. So that the full and proper CLID number will be sent to the remote voice gateway
D. So that the full and proper ANI number will be sent to the remote voice gateway
19. Which of the following answers correctly identifies the configuration commands shown here?
Router(cfg-translation-rule)#rule 1 reject /^773555/
A. A class-of-restriction rule to block numbers that are greater than 7735555
B. A class-of-restriction rule to block numbers that begin with 773555
C. A call-blocking rule to block numbers that begin with 773555
D. A class-of-restriction rule to block a specific area code and central office code
E. A call-blocking rule to block numbers greater than 7735555
20. Which of the following is not a call-block disconnect cause message?
A. Reorder-reject
B. Call-reject
C. Invalid-number
D. User-busy
E. Unassigned-number
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1. B. The order that the voice gateway will prefer, regardless of order entered into the IOS, is
as follows:
1. cisco-rtp
2. rtp-nte
3. h245-signal
2. A. DTMF relay transports signals out of band. If this is not congured, the tones are sent
in band along with the standard voice signals.
3. D. Fax pass-through transmits fax messages over IP networks using the same RTP packets.
The only difference is that it will negotiate so that the G.711 low-compression codec is used
to increase the chances that the fax is transmitted accurately.
4. B. The Cisco fax relay method uses the concept of virtual fax interfaces that terminate at
the voice gateway.
5. D. ls-redundancy will send multiple, redundant packets to the remote end to help to
ensure the receipt of those packets. In this case, three redundant packets will be sent.
6. C. The maximum fax transmission rate for Super Group 3 (SG3) class fax machines is
33,600 bps.
7. C, D. Both the fax-relay sg3-to-g3 and fax-relay ans-disable commands are used
to suppress answer (ANS) tones. This is done if you want to transport faxes at a lower rate.
8. C. The gateway force keyword forces the gateway-controlled fax relay service to use
Ciscos proprietary NSEs.
9. A. T.37 faxes use emails for transport and attach fax transmissions as TIFF image les.
10. C. An SMTP server is required for T.37 store-and-forward fax because the method uses
emails to send faxes in attached TIFF images.
11. D. One primary difference between how fax and modem transmissions are handled is that
analog fax transmission rates are static, while modem transmissions can vary throughout
the connection.
12. A. The correct command is mgcp modem relay voip mode nse gw-controlled, and this
command can only be entered while in global conguration mode.
13. C. When conguring MGCP to H.323 fallback, you must congure dial peers. Within
those dial peers, you must add the application mgcpapp command to designate these dial
peers as backups in case the voice gateway loses connectivity to the call-processing agent.
14. A. The show ccm-manager fallback-mgcp command will display if the voice gateway can
communicate to the CUCM or if it has fallen back to H.323.
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15. D. Typically, COR is congured on the CUCM. One exception is when a voice gateway is
also congured for SRST. When in SRST mode, the voice gateway can also enforce COR
rules.
16. A. The correct command is dial-peer cor custom, and it is set while in global
conguration mode.
17. C. The correct syntax is cor incoming emergency 3 4000 4010. This is congured
while in config-cm-fallback conguration mode.
18. D. When you congure TEHO, you should also congure a translation rule and prole so
that the ANI number of the local calling phone is correct before sending it to the remote
voice gateway.
19. C. This is a call-blocking rule that will nd and reject numbers beginning with 773555.
20. A. All of the messages are possible disconnect cause messages when conguring call-block
except reorder-reject.
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1. dtmf-relay rtp-nte digit-drop
2. Cisco fax relay
3. fax rate 9600 voice
4. fax-relay ecm disable
5. Call agent (CA) and gateway (GW)
6. 2100 Hz
7. preference 1
8. show ccm-manager fallback-mgcp
9. corlist outgoing int-out
10. Automatic Number Identication (ANI)
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c09.indd 394 9/21/11 11:25:56 AM
Configuring and
Managing CUBE and
H.323 Gateways
CHAPTER:
Implement Cisco Unified Border Element.
Describe the Cisco Unified Border Element features

and functionality.
Configure Cisco Unified Border Element to provide

address hiding.
Configure Cisco Unified Border Element to provide protocol

and media internetworking.
Configure Cisco Unified Border Element to provide call

admission control.
Verify Cisco Unified Border Element configuration

and operation.
Chapter
10
c10.indd 395 9/21/11 11:26:22 AM
In this chapter well look at equipment that is used to help
manage large voice networks. First, well examine the H.323
gatekeeper to see how it can be used to break networks into
zones and how to interact with multiple gatekeepers that control different zones within a
network. You might notice that gatekeepers arent part of the ofcial exam objectives, but
understand that they are a critical part of the 642-437 exam. Once we nish our coverage
of gatekeepers, well move on to look at the Cisco Unied Border Element (CUBE) to see
how it is different from a standard voice gateway and how it can connect two voice net-
works using a pure IP-to-IP solution when the networks are running either SIP or H.323.
What Is an H.323 Gatekeeper?
H.323 can function fairly well on its own just being congured on voice gateways, as you
learned in Chapter 7, Conguring Voice Gateway Signaling Protocols. When you begin
dealing with larger networks, H.323 simply doesnt scale well without the help of an
H.323 gatekeeper to manage your voice network, by breaking it up into multiple zones.
Your H.323 gateways will quickly become cluttered with multiple dial peers that often
cause confusion, and are a pain to maintain when you are dealing with multiple voice
gateways. A better solution is to install one or more gatekeepers into an H.323 network
to perform the following mandatory and optional functions, shown in Table 10.1.
TABLE 10.1 Mandatory and optional H.323 gatekeeper functions
Mandatory Optional
Zone management Call authorization
Address translation Call management
Call admission control (CAC) Bandwidth management
Bandwidth control
Lets break down each of these mandatory and optional H.323 functions to better
understand what the H.323 gatekeeper can provide.
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What Is an H.323 Gatekeeper? 397
H.323 Gatekeeper Mandatory Features
The primary responsibilities of an H.323 gatekeeper are to control call routing, call
permission, and call settings on the network. The H.323 mandatory features control each
of these functions.
Zone Management
Gatekeepers use the concept of logical zones to segment large networks into small and more
manageable chunks. A single zone may contain one or more voice gateways, multipoint
control units (MCUs), or H.323 endpoints. The H.323 gatekeepers responsibility is to
manage all registered devices within the zone and to provide information about how to route
calls between zones. Figure 10.1 shows an example of a gatekeeper managing two different
zones in a network.
Zone1 Zone2
Gatekeeper
V
Voice
Gateway1
V
Voice
Gateway2
V
FI GURE 10.1 A network controlled by a single H.323 gatekeeper
In our example, you see that we have two zones connected to our gatekeeper. The
gatekeeper that directly controls a zone considers them to be local zones, yet our gatekeeper
does not specically belong to a zone itself.
There can also be multiple gatekeepers congured that manage different zones, as shown
in Figure 10.2.
FI GURE 10. 2 A network controlled by multiple H.323 gatekeepers
Zone1 Zone2
Gatekeeper1 Gatekeeper2
V V
Voice
Gateway1
V
Voice
Gateway2
V
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398 Chapter 10
Configuring and Managing CUBE and H.323 Gateways

Now you see that Gatekeeper1 controls Zone1 and Gatekeeper2 controls Zone2. When
a device in Zone1 needs to contact a phone in Zone2, the H.323 endpoint or gateway
contacts its local gateway. The local gateway does not know about the zone, because it
does not directly control it. The two zones in the example therefore are considered to be
remote zones to each other, since they are not controlled by the same gatekeeper. In order
for remote gateway calls to work, the two gatekeepers communicate with each other and
Gatekeeper1 forwards the request to the unknown zone to Gatekeeper2. Gatekeeper2
knows about Zone2 because it is local to it, and the call can be completed.
Address Translation
The H.323 gatekeeper maintains a telephone-number-to-IP-address table. When an H.323
endpoint or gateway sends call information to the gatekeeper, it provides the destination
telephone address but does not know how to reach the remote endpoint. The gatekeepers
table maps telephone numbers to next-hop IP addresses. The IP address information is
passed back to the original requester so it can attempt to establish a call with the intended
remote device. The IP address that is given is where the H.225 setup packet should be sent
from the H.323 calling endpoint.
Call Admission Control
So now you know that H.323 gatekeepers control H.323 zones for management purposes
and possess routing information about where endpoints reside in the network. Given this
information, we can make calls across the entire network. But what if we want to prevent
some calls from being made between zones? Since the H.323 gatekeeper is at the center
of all the action, there is no better place to implement admission control. To accomplish
admission control between H.323 devices, gatekeepers use H.225 Registration Admission
and Status (RAS) messages. RAS messages are used for multiple H.323 services and are
explained in detail in the Understanding Gatekeeper Signaling section of this chapter.
Bandwidth Control
Because the gatekeeper is for providing call admission services using an H.323 gatekeeper,
the centralized location of a gatekeeper is ideal for controlling bandwidth usage between
endpoints. Bandwidth control uses RAS messages to negotiate codec rates and bandwidth
limits with endpoints.
H.323 Gatekeeper Optional Features
The optional H.323 gatekeeper functions revolve around the optimal management of the
network for operation of voice over an IP network. This includes functions such as call
authorization, call management, and bandwidth control; we will talk about each of these next.
Call Authorization
Situations arise where you want to restrict the access to endpoints or entire zones based on
various policies. These policies can be congured as either permit or deny rules, depending
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Understanding Gatekeeper Signaling 399
on the structure of the rule set. A common example of a call authorization would be to
deny calls to call-center endpoints based on the time of day.
Call Management
Call management deals with using in-progress call status information to better manage call
routing based on the information. For example, if the gatekeeper knows that a particular
H.323 endpoint is already in a connected call, and a second call comes into the gatekeeper,
it can respond to the calling party with a busy signal on behalf of the called party. Call
management can further be used for call-redirection purposes as well.
Bandwidth Management
When you think of bandwidth management, you should think of call admission control
(CAC), the technique by which bandwidth usage is tracked across the network by the
gatekeeper. When new calls come into the gatekeeper, it can make the decision to allow the
call to proceed because sufcient bandwidth is available. If, however, the gatekeeper nds
that there is not enough bandwidth available at the current time, it can reject new calls
from being made until sufcient bandwidth has been reclaimed. CAC is one of the most
common reasons for implementing a gatekeeper.
Understanding Gatekeeper Signaling
Gatekeepers communicate with other H.323-speaking devices, such as endpoints or H.323
voice gateways, by using signaling. RAS falls under the H.225 protocol within the overall
H.323 umbrella. Signaling messages are sent between devices using the User Datagram
Protocol (UDP). When a gatekeeper is involved in the call-setup process on an H.323
network, RAS messages are sent between the party requesting the call and the gatekeeper.
If no gatekeeper is used, these messages are sent directly between the registered endpoints
involved in the call.
There are a number of RAS messages used within the H.323 protocol suite for various
communication messages between gatekeepers and endpoints/voice gateways. This section
will detail the most commonly used messages in a Cisco environment.
RAS Gatekeeper Discovery Messages
H.323 voice gateways and endpoints need to be able to nd the gatekeeper. Following are
the three gatekeeper discovery message types:

Gatekeeper Request (GRQ)

Gatekeeper Conrm (GCF)

Gatekeeper Reject (GRJ)
On voice gateways, there are two slightly different methods of discovering a gatekeeper.
The rst method is to precongure your H.323 voice gateways with the IP address of your
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400 Chapter 10

gatekeeper. Because the gateway is precongured with an IP address, it can send a unicast
GRQ message and wait to receive either a GCF, which means the gatekeeper is available, or
a GRJ if for some reason the gatekeeper cannot allow endpoints to register.
If the voice gateway (or any other H.323 endpoint) is not precongured with the
gatekeepers IP address, it will send the same GRQ message but this time as a multicast
message instead of a unicast message. This is known as dynamic gatekeeper discovery, and
the message is sent to the multicast address of 224.0.1.41.
RAS Gateway Registration Messages
If the discovery process resulted in a gatekeeper conrm (GCF) message, the next step is
registration. The following messages are used in the registration process:

Registration Request (RRQ)

Registration Conrm (RCF)

Registration Reject (RRJ)
The endpoint or voice gateway initiates this process by sending an RRQ message to
the gatekeeper that was previously discovered. This is essentially a permission message,
and the gatekeeper can either accept the registration using the RCF message or reject the
registration using an RRJ response message.
If an RCF message is sent back, that endpoint or voice gateway is now considered to
be registered to the gatekeeper. Thats not the end of the registration messages, however.
H.323 also uses these messages as keepalives to ensure connectivity. H.323 version 1
devices will send all of the original message information contained in a standard RRQ
message. These messages are sent every 30 seconds. This information obviously isnt
required if the registration process has been completed, and it leads to wasted bandwidth.
Fortunately, version 2 of the H.323 protocol suite sends what is known as lightweight
registration messages. If this version is used, it means that RRQ messages contain only
basic information and consume far less bandwidth. The lightweight registration messages
specify a time to live (TTL) either in the endpoint/voice gateway RRQ or the gatekeeper
response RCF. Each time registration messages are sent, the TTL is decreased and the
keepalive eld is set to true until the TTL expires. When the TTL expires, the full RRQ
message is sent to verify that no changes have been made.
RAS Call Admission Messages
After the H.323 endpoints or voice gateways have received a registration conrm (RCF)
message from the gatekeeper, calls can be attempted through the gatekeeper. Notice that
the word attempted is used here. What happens is that when an endpoint wants to make
a call, it sends admission messages between itself and the gatekeeper. Here are all of the
admission message types:

Admission Request (ARQ)

Admission Conrmation (ACF)

Admission Reject (ARJ)
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The H.323 Gatekeeper Discovery, Registration, and Admission Process 401
When the H.323 gatekeeper receives an ARQ message from a registered device, it makes
the following two decisions:
1. Is the call permitted to go through? Or more specically, is there enough bandwidth
available over the link to support the call at the current time? If the call is denied, an
ARJ message is sent back to the requesting device.
2. If the call is permitted, how should it be routed? This is where the gatekeeper does a
table lookup to determine where the next-hop IP address is for this particular E.164
telephone number. Once the routing information is known, the gatekeeper sends back
an ACF message that both permits the call to be made and provides the IP location
where the calling device can nd the called device.
The H.323 Gatekeeper Discovery,
Registration, and Admission Process
Lets visualize the three H.323 gatekeeper RAS processes youve learned up to this point.
Well use Figure 10.3 as our example network.
Zone1 Zone2
Gatekeeper
H.225
H.245
RTP
RTP
R
A
S
R
A
S
V
Voice_GW_1
4444 5555
V
Voice_GW_2
V
FI GURE 10. 3 Gatekeeper RAS discovery, registration, and admission
In our example network, we have two H.323 voice gateways that require the services
of an H.323 gatekeeper. All of our RAS communication will take place between the voice
gateways and the gatekeeper, while the H.225, H.245, and RTP streams occur directly
between the two voice gateways. Lets step through an example of all the messaging that
occurs to complete a call, using Figure 10.3.
Voice_GW_1 has the IP address manually congured and therefore sends a unicast GRQ
to the gatekeeper. Voice_GW_2, on the other hand, does not have the IP of the gatekeeper
and must therefore send a unicast message across the WAN. The gatekeeper receives both
messages and is available for registration, so a GCF is sent back to the voice gateways.
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Receiving the GCF, the voice gateways attempt to register to the gatekeeper by sending
an RRQ. The messages are received and the gatekeeper permits both devices to register. A
RCF message is sent back to both voice gateways to conrm this. At this point, our two
voice gateways are registered and ready to eld calls to the gatekeeper.
Now lets say that our phone at extension 4444 attempts to call extension 5555. The
call is handled by Voice_GW_1, which in turn sends an ARQ to its registered gatekeeper.
The ARQ contains E.164 numbers of the calling and called parties. The gatekeeper ensures
that the call has enough bandwidth to be made and nds the next-hop IP address where the
called party phone is located. This information is packaged and sent back to Voice_GW_1
in the form of an ACF message.
Now that Voice_GW_1 knows the location of the remote phone, it sends an H.225
call-setup message to the remote voice gateway, which happens to be Voice_GW_2 in this
example. Now that our remote voice gateway is involved in a new call, it too must send
an ARQ message and wait for an ACF message response before the call can be permitted.
Once the ACF message is received by both voice gateways, the H.225 message-exchange
process can be completed as usual. As soon as this process is completed, the two voice
gateways handle the H.245 exchange messages, and two RTP sessions are set up between
the endpoints when the call is established.
RAS Location Messages
If your H.323 network uses multiple gatekeepers, the gatekeepers will use interzone
messages among one another to exchange information regarding zones that each of them
are responsible for. The following location request messages are used:

Location Request (LRQ)

Location Conrm (LCF)

Location Reject (LRJ)
Using Figure 10.4 as our example, lets go through the location request message process.
Zone1 Zone2
Voice_GW_1
7777 8888
V
Voice_GW_2
V
RAS
location
messages
Gatekeeper1 Gatekeeper2
V V
FI GURE 10. 4 RAS location messages
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The H.323 Gatekeeper Discovery, Registration, and Admission Process 403
Lets say that our voice gateways are properly registered to their respective gatekeepers.
A call is made from extension 7777 to extension 8888. The Voice_GW_1 sends an
ARQ to Gatekeeper_1. This gatekeeper does not have any information about extension
8888, because it is in a zone managed by Gatekeeper_2. Gatekeeper_1 sends an LRQ
to its neighbor, Gatekeeper_2, which is responsible for the zone that extension 8888
resides in. Gatekeeper_2 veries permissions and looks up the necessary next-hop IP
address information and sends it back to Gatekeeper_1 in the form of an LCF message.
Gatekeeper_1 then takes the newly acquired information, places it into an ACF message,
and sends it back to Voice_GW_1, and the connection can be established.
If you have more than two gatekeepers, location messages can be sent either sequentially
to individual gatekeepers or in a blast, where all gatekeepers are sent LRQ messages
at one time. If you have gatekeepers congured with identical zones, you will want to
use sequential forwarding, which happens to be the default setting. With sequential
forwarding, you can specify which gatekeepers should be sent LRQ messages over others.
If you dont have any overlapping zones, however, you may want to consider the blast
method, because it provides faster response times. Figure 10.5 shows the sequential method
of location message forwarding.
Zone_B
Zone_C
Zone_D
Zone_A
Gatekeeper2
V
Gatekeeper3
V
Gatekeeper1
ARQ
LRQ #1
LRJ
LRJ
LCF
LRQ #2
LRQ #3
V
Gatekeeper4
V
FI GURE 10. 5 Sequential location message forwarding
As you can see, Gatekeeper1 receives an ARQ from Zone_A for a phone in Zone_C.
Gatekeeper1 is set up to sequentially send LRQ messages to remote gatekeepers, so it rst
sends a request to Gatekeeper2. Gatekeeper1 receives an LRJ message from Gatekeeper2
that indicates this gatekeeper has no knowledge of Zone_C. Gatekeeper1 sends a second
LRQ message to Gatekeeper3 and again receives a reject message. Finally, Gatekeeper1
sends a third LRQ message to Gatekeeper4, which has knowledge of Zone_C and therefore
sends an LCF message to Gatekeeper1.
Finally, the blast method of location message forwarding is shown in Figure 10.6.
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404 Chapter 10

Here you see that LRQs are sent to all gatekeepers at once. The rst gatekeeper to
respond with an LCF is the one that will be used.
RAS Resource Availability Messages
Special resource availability messages sent from voice gateways to gatekeepers can be
used to update information about current call capacity and resource availability. This
information is then used to permit calls and set bandwidth limitations on H.323 calls.
The following messages are used:

Resource Availability Indicator (RAI)

Resource Availability Conrmation (RAC)

Resource in Progress (RIP)
The H.323 voice gateway sends RAI messages that contain resource availabilities such
as available bandwidth. If the gatekeeper successfully receives the RAI, it will process
the information and return an RAC to the voice gateway. If there is a problem with
resource availability, such as an RAS message timing out, the gatekeeper will send out a
RIP message to the voice gateway to wait additional time for the gatekeeper conrmation
message before the call can be attempted.
RAS Bandwidth Messages
After H.323 calls are established, it is possible that an endpoint can request that the
bandwidth for a particular call be adjusted. The following bandwidth request messages are
used between endpoints/voice gateways and gatekeepers:

Bandwidth Request (BRQ)

Bandwidth Conrm (BCF)

Bandwidth Reject (BRJ)
Zone_B
Zone_C
Zone_D
Zone_A
Gatekeeper2
V
Gatekeeper3
V
Gatekeeper1
ARQ
LCF
LRQ Blast
V
Gatekeeper4
V
FI GURE 10. 6 Blast location message forwarding
c10.indd 404 9/21/11 11:26:28 AM
The endpoint sends a BRQ message to the gatekeeper that can accept the request and
make bandwidth adjustments to the stream. An acceptance of the BRQ requires that the
gatekeeper send a BCF back to the requestor. If, however, the bandwidth request is denied,
the BRJ message is sent back and bandwidth settings remain the same.
Configuring an H.323 Gatekeeper
Now that you have a solid understanding of what H.323 gatekeepers offer and how they
communicate, we will explore how to congure the following gatekeeper functions:

Local zones

Remote zones

Zone prexes

Technology prexes
Well go over each of these conguration steps in the following few sections. Figure 10.7
shows the network with dual gatekeepers that will be used to show how to congure H.323
gatekeeper and gateway interoperation.
Zone: Miami
Miami
4XXX
S0/0
V
Zone: Boston
Boston
Gatekeeper1
Domain: example.com
Gatekeeper2
3XXX
S0/0 S0/0
V
V
Zone: LA
IP WAN
5XXX
Tech prex: 1# 99#
10.99.99.1 10.5.5.1
V
LA
V
FI GURE 10. 7 H.323 multi-gatekeeper network
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406 Chapter 10

Configuring Local Zones
Local zones are those zones that are managed by the local gatekeeper. For example,
Gatekeeper1 has two local zones, Boston and Miami. H.323-speaking endpoints and voice
gateways register directly to this gatekeeper. To congure our local zones, we must rst
enter into config-gk mode by issuing the gatekeeper command. Next, we enter the zone
local command, followed by the zone name, the domain name, and the RAS IP address.
The zone name is the name of a local zone. The domain name is the fully qualied domain
name (FQDN) of the gatekeeper and is used when DNS names are entered instead of IP
addresses. Note that the domain-name information is required even if DNS services are not
used. Finally, the RAS IP address is the IP address of the local gatekeeper that will be the
source IP used for sending and receiving RAS messages. Because there can be only one IP
address used for RAS communication on a gatekeeper, this command can be specied only
once in a local zone conguration. Once the IP address is congured for one local zone,
this address is used for all other congured local zones.
Here is an example of how to congure Gatekeeper1 with local zones:
Gatekeeper1#configure terminal
Gatekeeper1(config)#gatekeeper
Gatekeeper1(config-gk)#zone local Boston example.com 10.99.99.1
Gatekeeper1(config-gk)#zone local Miami example.com
Gatekeeper1(config-gk)#no shutdown
Gatekeeper1(config-gk)#end
Gatekeeper1#
Dont forget to issue the no shutdown command, which enables the gatekeeper service
on your router.
The RAS IP address is often a loopback interface in situations where
redundant connections exist. Since a virtual interface generally does not
go down like a physical interface, it is considered to be more reliable.
Configuring Remote Zones
Remote zones are the zones that are not congured locally and are handled by an external
gatekeeper. As discussed previously, gatekeepers that are responsible for different zones
communicate with each other using location RAS messages. In order for the RAS messages
to be sent between gatekeepers, you must congure a remote zone and specify the FQDN
and IP address of the remote gateway. In our example, we will congure a remote zone
for our LA zone and specify that calls that are not congured locally should send an
LRQ message to Gatekeeper2 to see if it has routing information for the unknown E.164
address. Here is an example of the remote zone conguration:
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Gatekeeper1(config-gk)#zone remote LA la.example.com 10.5.5.1
Gatekeeper1#
Configuring Zone Prefixes
Gatekeepers keep track of zones by using a unique zone prex. A zone prex uses E.164
numbers to dene each zone. If a call made by an H.323 device reaches the gatekeeper,
it checks the dialed number to see if it matches a specic prex for a known local zone.
If a match is made, the gatekeeper routes that call to the zone that is mapped to the zone
prex. To congure zone prexes, we use the zone prefix command followed by the local
zone name and a range of E.164 numbers that represent numbers within that zone. In our
example, we will congure Gatekeeper_1 with two zone prex commands for the Boston
and Miami local zones:
Gatekeeper1(config-gk)#zone prefix Boston 3...
Gatekeeper1(config-gk)#zone prefix Miami 4...
Gatekeeper1#
Now weve successfully mapped all 3XXX calls to the Boston zone and all 4XXX
calls to Miami. Since weve already congured the next-hop IP address in our local zone
conguration commands, the gatekeeper will perform an E.164-number-to-IP-address
lookup and return the IP address of the correct gateway to the sending voice gateway.
Gateway Redundancy
Anthony was developing a highly redundant H.323 gatekeeper and needed to congure
gateway redundancy, in which multiple voice gateways are responsible for the same
zone and thus the same phone groupings. In Anthonys case, he was creating zone
redundancy within the Atlanta_gw1 and Atlanta_gw2 gateways that covered phones in
the 5XXX extension range. Anthony has already registered both voice gateways with the
gatekeeper, but now he needs to congure the gatekeeper to send all calls to Atlanta_
gw1unless it is unreachable, at which point the gatekeeper would use the Atlanta_gw2
route. To do this, Anthony discovered the gw-priority keyword within the zone prefix
command. Anthony simply creates two zone prefix commands for extensions 5XXX
in the Atlanta zone for both Atlanta_gw1 and Atlanta_gw2 voice gateways. But at the
end of each of these commands, Anthony added gw-priority followed by a priority
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408 Chapter 10

Configuring Technology Prefixes
On large networks, you might have voice gatekeepers that handle additional H.323 tasks. If
this is true, you will need to congure technology prexes, which are special numbers that
when dialed connect to the appropriate voice gateway for the service that is needed. Your users
will need to know the prex numbers in order to include them as a prex when the destination
number is dialed. In addition, note that the technology prex takes precedence over any
zone prexes. So even though the number dialed matches a zone prex, the technology
prex number will be matched rst and sent to the proper voice gateway. Also note that the
technology prex will be stripped off prior to forwarding the call to the destination voice
gateway. By default, no technology prexes are dened, so therefore LRQ messages will be
sent to all gatekeepers either sequentially or simultaneously if the blast method is used.
In our example, we will congure two technology prex commands on Gatekeeper1. To
congure a technology prex, we use the gw-type-prefix command, followed by the prex
extension and additional conguration keywords, as shown in our conguration:
Gatekeeper1(config-gk)#gw-type-prefix 99# gw ipaddr 10.99.99.1
Gatekeeper1(config-gk)#gw-type-prefix 1# default-technology
Gatekeeper1#
number and the gateway alias name. The higher the priority number, the more preferred
the destination is. The range is 0 to 10. A priority of 0 means that the gatekeeper will
never use the route. An example of using 0 as a priority would be when you want to
specically exclude a gateway from a gateway pool, because that pool would require
that the gateway incur an expensive long-distance charge if it was allowed through. The
default priority is 5. Therefore, Anthony chose to use a priority of 10 for Atlanta_gw1 and
a priority of 5 for Atlanta_gw2, as shown here:
Gatekeeper#configure terminal
Gatekeeper(config)#gatekeeper
Gatekeeper(config-gk)#zone prefix Atlanta 5... gw-priority 1 Atlanta_gw1
Gatekeeper(config-gk)#zone prefix Atlanta 5... gw-priority 2 Atlanta_gw2
Gatekeeper(config-gk)#end
Gatekeeper#
Once this is congured, the gatekeeper will tell other gateways to use the Atlanta_gw1
path to the Atlanta zone. If it ever goes down, the gatekeeper will use the second most
preferred route, which is the Atlanta_gw2 voice gateway.
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Voice Gateway Interoperation with Gatekeepers 409
You will notice that our examples use the # sign as part of our technology
prefix. This convention is commonly used in the real world to easily dif-
ferentiate technology prefixes from other extensions. A technology prefix
does not need to have any special characters, however, and can range
from 1 to 11 digits in length.
In this scenario, weve just mapped the prex number 99# to our Boston remote voice
gateway. In addition, the second technology prex that uses the default-technology
keyword species that 1# be used as the default technology prex for this gatekeeper. Thus,
all gateways that register with Gatekeeper_1 and use the 1# prex option are used as the
default for routing any addresses that cannot be resolved. Keep in mind that your second
gatekeeper should have technology prex congurations identical to the rst, to ensure
proper interoperation of H.323 services.
Voice Gateway Interoperation
with Gatekeepers
Now that youve learned how to congure H.323 gatekeepers on an IP network, we need to
shift our focus and explore how to congure H.323 voice gateways to interoperate with them.
The following conguration steps are required to register a voice gateway to a gatekeeper:
1. Congure the necessary H.323 commands on your designated H.323 signaling
interface.
2. Congure one or more dial peers that point to the local gatekeeper.
3. Enable the H.323 process on your voice gateway.
Lets go through how to congure each of these required steps on the Boston router
using the multi-gatekeeper diagram from Figure 10.7.
Configuring H.323 Interface Commands
As we congured for the H.323 gatekeeper, the voice gateway needs a designated interface
that will always be used when communicating with the gatekeeper using RAS messages.
First, we will enter into interface conguration mode for an interface that has an IP address
congured. Next, we will use the h323-gateway voip command followed by various
keywords to set up our interface for gatekeeper interoperation. The primary h323-gateway
voip keywords are described next.
interface This keyword marks the interface as being a voice gateway interface.
bind srcaddr The bind srcaddr command is followed by the IP address of the interface
you are using for H.323 functions.
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410 Chapter 10

id This keyword is used to specify the local zone the voice gateway operates in and the IP
address of the local gatekeeper. This command is optional and allows the voice gateway
to contact the gatekeeper using a unicast GRQ RAS message. If this command is not
congured, the voice gateway will send the GRQ RAS message using multicast.
h323-id The h323-id keyword followed by the local zone name is another optional
command that is used to specify what zone the voice gateway should register under.
tech-prefix This keyword is followed by a prex number and is used to specify that our
voice gateway wants to register with technology prex services.
Using our network diagram in Figure 10.7, we will congure our Boston voice
gateway to interoperate with Gatekeeper_1 on our serial 1/0 interface using the following
commands:
Boston#configure terminal
Boston(config)#interface serial 0/0
Boston(config-if)#h323-gateway voip interface
Boston(config-if)#h323-gateway voip id Boston ipaddr 10.99.99.1
Boston(config-if)#h323-gateway voip h323-id Boston
Boston(config-if)#h323-gateway voip tech-prefix 1#
Boston(config-if)#h323-gateway voip tech-prefix 99#
Boston(config)#end
Boston#
Configuring Dial Peers for Gatekeeper Interoperation
When using an H.323 gatekeeper for routing information, you need to enter the VoIP dial
peer session target ras command. This tells the router to request routing information
from its locally congured H.323 gatekeeper using RAS messaging. The gatekeeper will
determine where the call should be routed (if possible) and send that information back to
our Boston voice gateway. You should also specify the tech-prefix digits that you plan to
use on your dial peers. The remainder of the dial-peer conguration statements should look
familiar to you by now. Here is an example of how to congure a dial peer on our Boston
voice gateway for the Miami and LA extensions:
Boston(config)#dial-peer voice 1000 voip
Boston(config-dial-peer)#destination-pattern ....
Boston(config-dial-peer)#tech-prefix 1#
Boston(config-dial-peer)#tech-prefix 99#
Boston(config-dial-peer)#session target ras
Boston(config-dial-peer)#end
Boston#
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Configuring Call Admission Control on H.323 Gatekeepers 411
Notice that our VoIP dial peer 1000 has a four-digit wildcard. This is a catch-all for
our voice network that will forward any four-digit extensions to the gatekeeper. Our local
gatekeeper will know about the locally congured 3XXX and 4XXX extensions but nothing
else. If another extension is dialed (such as the LA 5XXX extensions), the caller will be
forwarded to Gatekeeper_2 to see if it has any knowledge of the location of the extensions.
Enabling the H.323 Service on a Voice Gateway
Once we have all of our congurations set on the voice gateway, we must enable the H.323
gateway-to-gatekeeper service, by issuing the gateway global conguration command. Note
that we also include a no shutdown command while in config-gk conguration mode to
bring up the service, as shown here:
Boston(config)#gateway
Boston(config-gk)#no shutdown
Configuring Call Admission Control
on H.323 Gatekeepers
One of the true strengths of implementing an H.323 gatekeeper is the ability to manage
your H.323 devices and voice gateways by zones. When you have the ability to segment a
network into distinct zones, you can easily congure and control the amount of bandwidth
used when calls are placed between those zones. Bandwidth control is an important part
of a voice network, especially when low-speed WAN connections are being used with a
remote site. By limiting the number of calls that can be made at one time, this technique
ensures that the calls in progress have sufcient bandwidth.
This section shows how to congure call-admission control to limit bandwidth used.
But before we congure bandwidth control, we must rst discuss how the gatekeeper keeps
track of the bandwidth being used for voice calls at any given time.
Understanding the CAC Bandwidth Control
on H.323 Gatekeepers
An H.323 gatekeeper can control CAC between itself and the following voice components:

H.323-enabled voice gateways

CUCM

CUCM Business Edition

CUCM Express
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412 Chapter 10

The policies that are congured on the gatekeeper are static, so it is important to
understand how voice call bandwidth is calculated so that the bandwidth settings you
implement are ideal for your network.
The formula for determining the current amount of voice bandwidth being utilized
between H.323 zones couldnt be simpler. Here is the equation:
Zone_Bandwidth = Number_of _Current_Calls Codec_Payload_Bandwidth 2
Notice that we multiply the calls and bandwidth by 2. We do so because that part of the
equation calculates only the codec bandwidth and nothing else. Because the gatekeeper has
no knowledge of the network topology, it simply doubles the codec bandwidth to dene a
static number that should take care of any overhead. For example, lets say that we have
ve concurrent calls in place between zone_A and zone_B. Three of the calls are using the
G.711 codec and the other two are using the G.729 codec. You should know that the payload
bandwidth for G.711 is 64 Kbps and the payload bandwidth for G.729 is 8 Kbps. Therefore,
our H.323 gatekeeper with CAC enabled will calculate the current bandwidth as the following:
Zone_Bandwidth = 3 64 2 = 384 Kbps
Zone_Bandwidth = 2 8 2 = 32 Kbps
The gatekeeper will add 384 to 32 to get a total of 416 Kbps current bandwidth between
zone_A and zone_B. Note again that the simplicity of this calculation at no time takes into
account compression techniques (such as cRTP) or LAN/WAN header size differences.
Now that you understand CAC interoperation and zone bandwidth calculations, lets see
how to congure bandwidth limitations between H.323 zones on a gatekeeper.
Configuring CAC Bandwidth Control on H.323 Gatekeepers
To congure bandwidth control on an H.323 gatekeeper, you must rst enter into
config-gk mode and use the bandwidth command followed by one of these keywords:
session This keyword denes the maximum amount of bandwidth permitted for a single
H.323 stream on a zone.
interzone This keyword denes the maximum amount of bandwidth allowed between
different zones.
total This keyword denes the total amount of voice bandwidth permitted (both inter-
zone and intrazone) on a zone.
remote This keyword denes the total amount of voice bandwidth permitted between
gatekeepers in a multi-gatekeeper environment.
Once you decide whether you want to control bandwidth at the session, interzone, or
total level, you can further rene bandwidth control based on the following:
default bandwidth-amount This keyword sets the default maximum bandwidth for all zones.
The default is overridden with more specic bandwidth congurations that use the zone
keyword, described next. If no specic bandwidth congurations exist, this default value is used.
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Configuring Call Admission Control on H.323 Gatekeepers 413
zone zone-name bandwidth-amount The zone keyword, followed by a previously congured
zone name and then a bandwidth amount in Kbps, sets the maximum amount of voice
bandwidth for a specic zone.
If you are configuring either the interzone, total, or remote bandwidth
control amounts, you can specify a bandwidth (in Kbps) between 1 and
10,000,000. If you are configuring bandwidth control at the session level,
the range is 1 to 5,000.
To demonstrate how to congure zone bandwidth control using the bandwidth command,
we will use the gatekeeper-controlled network with three zones depicted in Figure 10.8.
Gatekeeper
V
zone_C
V
zone_A
V
zone_B
V
IP WAN
FI GURE 10. 8 An H.323 gatekeeper bandwidth-controlled network
On our gatekeeper, we want to set the following bandwidth limitations:

Interzone default: four G.711 calls

Interzone for zone_A: six G.711 calls

Total bandwidth for each zone: eight G.711 calls
Gatekeeper#configure terminal
Gatekeeper(config)#gatekeeper
Gatekeeper(config-gk)#bandwidth interzone zone default 512
Gatekeeper(config-gk)#bandwidth interzone zone zone_A 768
Gatekeeper(config-gk)#bandwidth total default 1024
Gatekeeper(config-gk)#end
Gatekeeper#
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414 Chapter 10

That should give you a solid understanding of H.323 gatekeepers and how to congure
them. Next, well look at how to verify and troubleshoot H.323 gatekeepers in a network.
Gatekeeper Verification and
Troubleshooting Commands
There are several show and debug commands that will be useful when verifying gatekeeper
conguration and for troubleshooting purposes. These commands are to be used on the
gatekeeper itself to verify current calls, endpoints, zones, and RAS communications.
show gatekeeper status The output of this command shows the status of the gatekeeper
service. It is a great way to verify that the gatekeeper is up and operational and which local
zones are congured, as shown in this example output:
Gatekeeper#show gatekeeper status
Gatekeeper State: UP
Load Balancing: DISABLED
Flow Control: DISABLED
Zone Name: Zone1
Zone Name: Zone2
Accounting: DISABLED
Endpoint Throttling: DISABLED
Security: DISABLED
Maximum Remote Bandwidth: unlimited
Current Remote Bandwidth: 0 kbps
Current Remote Bandwidth (w/ Alt GKs): 0 kbps
From the output you can see that the gatekeepers state is UP and we have two zones
(Zone1 and Zone2) that are congured locally.
show gatekeeper calls This command shows a real-time snapshot of all the current
calls on the voice network that utilize the local gatekeeper. Heres an example of its output:
Gatekeeper#show gatekeeper calls
Total number of active calls = 1.
GATEKEEPER CALL INFO
====================
LocalCallID Age(secs) BW
9-34668 124 16(Kbps)
Endpt(s): Alias E.164Addr
src EP: voice1@example.com 3001
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Gatekeeper Verification and Troubleshooting Commands 415
CallSignalAddr Port RASSignalAddr Port
10.1.1.100 1720 10.1.1.100 55136
Endpt(s): Alias E.164Addr
dst EP: voice2 4001
CallSignalAddr Port RASSignalAddr Port
10.2.1.101 1720 10.2.1.101 51329
In this example, you can see that there is one active call that the gatekeeper is aware
of. You can see the source and destination IP addresses, port numbers, and aliases for the
endpoints currently communicating. Also, you see that this call is using a bandwidth of
16 Kbps, which means that the codec being used is probably G.729. This is because G.729
uses 8 Kbps of bandwidth, and we need two RTP streams for our outgoing and incoming
voice for a single call.
show gatekeeper endpoints This command displays all of the currently known H.323
end devices and voice gateways from the local gateways perspective. Here is an example of
the output from this command:
Gatekeeper#show gatekeeper endpoints
GATEKEEPER ENDPOINT REGISTRATION
================================
CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags
-------------- ----- -------------- ---- -------- ---- -----
10.1.1.2 1720 10.1.1.2 1719 Zone1 VOIP-GW S
H323-ID: gway1 (static)
10.2.2.2 1720 10.2.2.2 1719 Zone2 VOIP-GW S
H323-ID: gway2 (static)
Total number of active registrations = 2
You can see that our gatekeeper currently has two endpoints associated with the gatekeeper,
and they both are voice gateways (Type: VOIP-GW). You can also verify the IP address of
the endpoints and the ports they used to communicate signaling with.
debug ras The debug ras command is great for troubleshooting RAS communication
problems in real time or simply to understand how the different RAS message types operate
for various gatekeeper functions. In the example output, we have enabled RAS debugging
on our gatekeeper. A voice gateway is registering with the gatekeeper, and we are watching
the RAS messages in the process:
Gatekeeper#debug ras
RASLib::RASRecvData: successfully rcvd message of length 34 from
10.1.1.2:24999
RASLib::RASRecvData: GRQ rcvd from [10.1.1.2:24999] on sock[5C8D28]
RASlib::ras_sendto: msg length 45 sent to 192.168.1.100
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416 Chapter 10

RASLib::RASSendGCF: GCF sent to 192.168.1.100
RASLib::RASRecvData: successfully rcvd message of length 76 from
10.1.1.2:24999
RASLib::RASRecvData: RRQ rcvd from [10.1.1.2:24999] on sock [0x5C8D28]
RASlib::ras_sendto: msg length 81 sent to 192.168.1.100
RASLib::RASSendRCF: RCF sent to 192.168.1.100
The output shows a Gatekeeper Request (GRQ) by our voice gateway. The gatekeeper
accepts the request and sends back a Gatekeeper Conrm (GCF). Next, the voice gateway
sends a Registration Request (RRQ), and the gatekeeper registers the voice gateway and
sends back an acknowledgement in the form of a Registration Conrm (RCF).
Introducing the Cisco Unified
Border Element
Dont be concerned about the name Cisco Unied Border Element (CUBE), because it is
simply a Cisco marketing term for a voice gateway that uses only IP-based connections instead
of traditional PSTN analog and digital lines. It used to be that the CUBE was called an IP-
to-IP gateway, and that is still a good way to describe the duties of a CUBE. But you should
know that a CUBE is commonly deployed at the border or edge of the network and is used
to connect either to other networks managed by the organization or to an Internet Telephony
Service Provider (ITSP). There are some instances, however, where a CUBE is deployed within
a large enterprise voice environment that nds a need to translate between legacy H.323
equipment and newer hardware that may only operate with SIP signaling. A CUBE is also
deployed internally to provide CAC support between CUCMs in a clustered environment.
The CUBE is responsible for terminating and establishing call legs to external VoIP
networks. The voice signaling protocols must be terminated and reestablished at the CUBE.
Media sessions, however, can be congured to terminate at the CUBE or to ow around
the CUBE.
As we know, voice networks can run on various voice signaling protocols. Fortunately,
the CUBE can interconnect VoIP networks using the following voice signaling protocol
scenarios:

SIP-to-SIP

H.323-to-H.323

SIP-to-H.323

H.323-to-SIP
This is different from traditional voice gateways, which typically take an IP call on the
internal network and translate it for transport on a PSTN circuit such as a T1 PRI. VoIP
dial peers would be used for internal call routing and POTS dial peers used for external
call routing. But with the CUBE, both internal and external call routing is performed using
physical VoIP dial peers and logical call legs, as shown in Figure 10.9.
c10.indd 416 9/21/11 11:26:34 AM
CUBE Features
As noted earlier, a CUBE and a voice gateway are very similar. They operate on the same
Cisco router hardware, and many functions are the same. The primary difference is that a
CUBE runs a specialized version of IOS software to achieve the protocol internetworking
function as well as some of the other more commonly implemented features. Heres a
summary of the primary CUBE gateway services:
Protocol Internetworking This is the ability to terminate and reinitiate IP voice sessions
between devices that run H.323, SIP, or H.323-to-SIP.
Call Admission Control A CUBE provides dynamic CAC either statically or dynamically
in the form of the Resource Reservation Protocol (RSVP).
Secure Deployment A CUBE can be deployed on the DMZ arm of a rewall to provide
voice/video services to external (and therefore untrusted) networks.
IP Address Hiding Because the CUBE can terminate and reinitiate VoIP sessions, it can be
used to either replace or hide the true IP address of endpoint devices. This can add an addi-
tional layer of security if needed.
Codec Negotiation Because a CUBE sits on the border of a network, it is an ideal device
to provide codec negotiation between signaling protocols. You can congure a CUBE to
take an interest in codec negotiation, meaning that the two endpoint devices and the CUBE
must all agree on the codec, or the CUBE can be set to transparent mode where the codec
negotiation between endpoints is ignored.
Next, well go through the CUBE essentials and congure a CUBE device to bridge two
voice networks together.
CUBE Media Flow Options
When calls that are destined for external voice networks pass through a CUBE, the internal
voice signaling protocol is terminated and reestablished. This type of behavior is known
as a proxy. While the voice signaling protocol must be terminated, the voice/video media
streams may or may not be proxied as well. These two approaches, known as media
ow-through and media ow-around, are detailed next.
CUBE
IP voice
network 2
IP voice
network 1
VoIP call leg/
dial peer
VoIP call leg/
dial peer
FI GURE 10. 9 CUBE VoIP-to-VoIP dial peers
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418 Chapter 10

Media Flow-Through
In media ow-through, voice/video streams come into and are proxied by the CUBE. Because
the connection is proxied, the CUBE will replace the source IP address of the actual device
with its own IP address. While this is done primarily for routing purposes, it also provides
the following two benets:

IP address hiding for added security

Prevention of duplicate network address spaces between separate networks
Media ow-through is the default transport method on the CUBE and is the only option
when terminating two different signaling protocols such as SIP-to-H.323. Figure 10.10
shows an example of using media ow-through between two different VoIP protocols.
While the added benets of IP address hiding and duplicate address protection are
useful, the ow-through method also has drawbacks:

Increased CPU and bandwidth load on the CUBE and its connected network

The possibility of suboptimal paths that can introduce unnecessary latency for calls.
Because these two drawbacks might become an issue on some networks, Cisco has a
second option, called media ow-around, that xes these problems.
Media Flow-Around
Media ow-around does not act as a proxy for voice/video transmissions such as RTP.
Instead, the media streams ow freely between the two networks and nd their own path
to the destination. This solves the CUBE load and suboptimal-path problems inherent in
the media ow-through method but at the cost of giving up IP address hiding and duplicate
IP network protection. Figure 10.11 shows an example of using media ow-around
between two different VoIP networks.
CUBE
IP WAN
R
T
P
R
T
P
H
.
2
2
5
/
H
.
2
4
5
H
.
2
2
5
/
H
.
2
4
5
V V
FI GURE 10.10 Media flow-through
c10.indd 418 9/21/11 11:26:35 AM
CUBE
RTP
IP WAN
H
.
2
2
5
/
H
.
2
4
5
H
.
2
2
5
/
H
.
2
4
5
V V
FI GURE 10.11 Media flow-around
It is important to keep in mind that you must verify that the two networks you are connecting
with your CUBE using the media ow-around method do not have overlapping IP address space.
If that is the case, youll need to revert to the default media ow-through method.
CUBE Signaling Protocol Interoperation
Now that weve determined how we can manipulate the ow of media streams, we must
next look at the voice signaling protocols that can be implemented on a CUBE to provide
interoperating functions. Specically, the CUBE provides interoperation using either SIP or
H.323. In Chapter 7 you learned that H.323 can be congured with either fast- or slow-
start initiation and SIP can be congured with early or delayed offer. With a CUBE, these
initiation methods may or may not be available, depending on the signaling types being
used. The following methods are supported according to Table 10.2.
TABLE 10. 2 CUBE signaling interoperation
H.323 fast
H.323-to-SIP
H.323-to-H.323
SIP-to-SIP
H.323 slow H.323 fast
H.323 slow
SIP early
SIP delayed SIP early
SIP delayed
H.323 fast
SIP delayed SIP early
H.323 slow
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420 Chapter 10

CUBE RSVP-CAC
Earlier in this chapter, you learned how to congure CAC on H.323 gatekeepers. While
this static method of bandwidth control does work, its not very exible. Resource
Reservation Protocol (RSVP)based CAC, on the other hand, can be congured between
two CUBE routers to provide a much more intelligent method of bandwidth control
between two Cisco Unied Communications (CUCM) systems or voice gateways. RSVP
is a transport-layer protocol that is designed to reserve bandwidth resources dynamically
across an IP network. RSVP-CAC is initiated by the calling-side network. As soon as the
call-setup message is received by the local CUBE, the path and reservation messages are
sent to the remote CUBE. It determines whether there is enough bandwidth and either
accepts or denies the RSVP request. As soon as an RSVP conrm message is returned
to the local CUBE, the call is considered to be admitted, and the H.225 call-setup
process continues.
When running RSVP-CAC on your network, you must make sure that your
CUBE routers are configured for media flow-through, because media flow-
around is not supported.
Figure 10.12 shows the signaling steps and responsibilities with two H.323 voice
gateways and two RSVP-CAC CUBE routers sitting in between.
CUBE 1
Gateway1
5. H.225 call setup
2. RSVP request
1. H.225 call
setup
3. RSVP conrm
Called party
phone
Called party
phone
Gateway2
CUBE 2
V V
4. H.225 call setup
FI GURE 10.12 RSVP-CAC signaling
c10.indd 420 9/21/11 11:26:36 AM
CUBE Call Flow Differences
Call signaling with a CUBE depends on not only the voice-signaling type used but also the
type of hardware that is connected. In the following sections we will focus on two CUBE
network scenarios.
Communication with a SIP ITSP
Our rst example is interconnecting our CUCM Express, running the default H.323
protocol, to a SIP ITSP voice gateway through a CUBE. Figure 10.13 shows the call
signaling ow from end to end.
CUBE
SIP
H.225/H.245
CUCM
Express
V
PSTN
ITSP
SCCP
M
Internet
FI GURE 10.13 CUBE network call flow to a SIP ITSP
Dont forget that if the SIP ITSP uses only the early-offer method, your H.323 session
must be congured for fast start. And if SIP late-offer is congured between the SIP
ITSP and CUBE, then H.323 slow-start initiation must be used. Also remember that
media ow-through must be used because we are connecting calls that use different
signaling protocols.
Communication through a Gatekeeper to a SIP ITSP
In our next example, you see CUCM Express and an H.323 gatekeeper on one managed IP
network. Then, on the opposite network, we have a remote gatekeeper and a CUBE,
as depicted in Figure 10.14.
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422 Chapter 10

You can see that CUCM Express and the remote SIP ITSP network both utilize an H.323
gatekeeper to provide telephone-number-to-IP-address lookups. Remote zone RAS
signaling can be congured between the gatekeepers to exchange call-routing information.
Once RAS is completed and the call is permitted, the CUCME works directly with the
CUBE, as does the CUBE with the SIP voice gateway.
Next you will learn how to congure a CUBE in various network topologies.
Configuring the CUBE
In this section, youll see how to congure a CUBE to operate in VoIP networks that run
SIP, H.323, or both.
Configuring Protocol Interoperation
To enable VoIP-to-VoIP interoperation, you need to rst get into conf-voi-serv conguration
mode by issuing the voice service voip command. Once there, you use the allow-
connections command, followed by the protocol interoperation you want to use on your
CUBE. This is accomplished by rst choosing the from protocol, which is the protocol used
by the originating endpoint. Then you issue the to command followed by the protocol used by
the terminating endpoint. Here are the possible protocol interoperation options:
allow-connections sip to sip
allow-connections h323 to h323
allow-connections sip to h323
allow-connections h323 to sip
CUBE
SIP
H.225/H.245
CUCM
Express
V
V V
PSTN
ITSP
Remote-GK
Remote zone RAS
Admission
RAS
Local-GK
SCCP
M
Internet
FI GURE 10.14 CUBE network call flow through a gatekeeper
c10.indd 422 9/21/11 11:26:37 AM
Configuring the CUBE 423
Always remember when working to interconnect SIP and H.323 networks that
the allow-connections command is unidirectional. If you want your CUBE to work
bidirectionally (that is, able to make outbound calls and receive inbound calls), you must
congure two commands to allow SIP-to-H.323 and H.323-to-SIP. Figure 10.15 shows
CUBE interoperation between H.323 and SIP networks.
CUBE
SIP
ntetwork
Bidirectional communication
H.323
network
FI GURE 10.15 An H.323-to-SIP network
We want either network to be able to initiate calls that terminate on the other side, so we
must congure two allow-connections statements as shown here:
CUBE#configure terminal
CUBE(config)#voice service voip
CUBE(conf-voi-serv)#allow-connections h323 to sip
CUBE(conf-voi-serv)#allow-connections sip to h323
CUBE(conf-voi-serv)#end
CUBE#
Configuring Media Flow-Around
You can modify the default media ow-through behavior, which forces all media ows to
be proxied at the CUBE. As long as you are running the same gateway protocol (either
SIP-to-SIP or H.323-to-H.323) and are not using RSVP-CAC, you can modify the media
ow behavior to ow-around. Lets say you have a SIP-to-SIP network interconnected with
a CUBE. You want to modify the media ow to use the ow-around method. There are
three ways of conguring this:

At a global level

At a voice-class level

At a dial peer level
In this example, we will congure media flow-around on VoIP dial peer 111:
CUBE(config)#dial-peer voice 111 voip
CUBE(config-dial-peer)#media flow-around
CUBE(config-dial-peer)#end
CUBE#
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424 Chapter 10

Configuring Codec Transparency
Earlier in the chapter, we discussed how the CUBE can take an interest and help negotiate
which codec is used between endpoints. This is the default behavior. If you dont want the
CUBE to interfere with codec negotiation, the default behavior can be changed using the
codec transparent command; you dont need to specify particular codecs that can be used
between connected endpoints. Here is an example of how to congure codec transparency
on VoIP dial peer 111:
CUBE(config)#dial-peer voice 111 voip
CUBE(config-dial-peer)#codec transparent
CUBE(config-dial-peer)#end
CUBE#
Now the CUBE will stay out of the codec negotiation process. However, it should be
mentioned that even though codec transparency is enabled, the CUBE still looks to see
what codec is being attempted by the endpoints. If the codec is not known to the CUBE,
the call cannot be completed.
Configuring H.323 Fast-to-Slow-Start Signaling
When conguring an H.323-to-H.323 network, you need to decide how to handle the H.323
initiation process. This behavior can be modied while in conf-serv-h323 configuration
mode by using the call start command, followed by one of these keywords:
fast This command forces all H.323 dial peers to use H.323v2 fast-start initiation. This
is the default CUBE setting for H.323.
slow This command forces all H.323 dial peers to use H.323v1 slow-start initiation.
interwork This command is used where there is either a fast-start-to-slow-start or
slow-start-to-fast-start interoperation. Note that this option disables slow-to-slow or
fast-to-fast call-matching setups. As an example of how to congure this, Figure 10.16
illustrates a network that requires an H.323 fast-to-slow-start setup.
CUBE
H.323
network
Fast-start
H.323
network
Slow-start
FI GURE 10.16 H.323 fast-to-slow start
Focusing on CUBE1, well rst congure H.323-to-H.323 protocol interoperation and
then enable fast-to-slow-start signaling, as shown here:
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CUBE Verification and Troubleshooting Commands 425
CUBE1#configure terminal
CUBE1(config)#voice service voip
CUBE1(config-voi-serv)#allow-connections h323 to h323
CUBE1(config-voi-serv)#exit
CUBE1(config)#h323
CUBE1(conf-serv-h323)#call start interwork
CUBE1(conf-serv-h323)#end
CUBE1#
Once this is completed, you simply need to congure the proper VoIP dial peers for
routing calls and youre all set.
Configuring SIP Delayed-to-Early-Offer Signaling
Like H.323, SIP can be modied to use either early- or delayed-offer signaling when
operating a SIP-to-SIP network. You can congure a delayed-to-early-offer SIP network
either globally for all SIP dial peers while in config-voi-serv mode or individually while
in config-dial-peer mode. In our example, we will rst allow SIP-to-SIP interoperation
and then enable SIP delayed-offer-to-early-offer by using the command early-offer
forced, as shown in this example:
CUBE1#configure terminal
CUBE1(config)#voice service voip
CUBE1(config-voi-serv)#allow-connections sip to sip
CUBE1(config-voi-serv)#early-offer forced
CUBE1(config-voi-serv)#exit
Now all SIP dial peers that are congured on this CUBE will participate in early- to
late-offer early-media negotiations.
CUBE Verification and Troubleshooting
Commands
To end this chapter well cover some of the show and debug commands that are especially
useful when verifying and troubleshooting a CUBE-supported voice network.
show call active voice brief To display the number of currently active call legs and
signaling type that are traversing a CUBE, you can use the show call active voice brief
command. A portion of the output of this command is shown here:
CUBE#show call active voice brief
-output cut
Telephony call-legs: 0
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426 Chapter 10

SIP call-legs: 1
H323 call-legs: 1
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
-output cut
CUBE#
In the output of this example, we have one call with two total call legs, and the CUCM
is providing voice network interoperation between a SIP and an H.323 network.
show call history voice brief You can also take a look at the past history of call
leg connections using the show call history voice command. The output is nearly
identical to that of the show active call brief command, but this command takes the
accumulated calls in the CUBE routers memory. The history can be cleared out by issuing
the clear call history voice command.
show voip rtp connections If you are running your CUBE as a media ow-through
device, RTP sessions will be terminated and proxied. You can view the RTP sessions by
issuing the show voip rtp connections command, as shown here:
CUBE#show voip rtp connections
Load for five secs: 1%/0%; one minute: 1%; five minutes: 1%
Time source is NTP, 19:18:43.542 CST Mon May 19 2011
VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 3001 4001 18546 17402 10.10.10.100 10.10.3.50
2 4001 3001 17778 19596 10.10.10.100 10.10.4.50
Found 2 active RTP connections
CUBE#
The output shows a single voice call with a transmitting and receiving RTP stream.
Notice how the LocalIP address for both RTP sessions is the same. The 10.10.10.100 IP
address is the address of the CUBE because it is acting as a proxy for the RTP stream. This
allows the CUBE to hide the actual endpoint IP addresses.
debug voip ipipgw The debug voip ipipgw command is useful to see the CUBE pro-
cesses it is responsible for when connecting separate voice networks together. Youll nd
information such as H.323 or SIP initiation processes, RTP and RTCP port information,
and media ow settings. In this example, you see output showing that the H.323 incoming
call leg is set to use the ow-through media stream method:
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Exam Essentials 427
CUBE#debug voip ipipgw
-output cut
May 19 20:27:53.430 CST: cch323_media_flow_mode: IPIPGW(3001):Flow Mode=1
May 19 20:27:53.430 CST: cch323_set_h245_state_mc_mode_outgoing:call_spi_
mode = 1
You can see that the call originated from extension 3001. If the CUBE device is set for the
ow-through method, the debug output will show Flow Mode=1.
Summary
This chapter introduced the H.323 gatekeeper and CUBE devices. You learned how these
two devices interact with voice end devices and voice gateways and how to congure each
of these devices and verify their operational status. In the next two chapters, well take
what youve learned from the entire book and add one nal layer of serviceability to IP
voice networks in the form of Quality of Service (QoS). It is the last step that smoothens
out the rough spots in terms of the quality of voice calls on a packet-switched network.
Exam Essentials
Know the mandatory and optional H.323 gatekeeper features. Mandatory features
include zone management, address translation, CAC, and bandwidth control. Optional
features include call authorization, call management, and bandwidth management.
Understand the purpose of RAS messages. RAS messages are used between H.323
endpoints and the gatekeeper to register to the gateway, perform call lookups and
admissions, and provide information about where calls to remote zones should be routed.
Understand how endpoints discover the H.323 gatekeeper. Endpoints can either be
statically congured and sent as a unicast GRQ RAS message, or if not statically
congured, that same GRQ is sent as a multicast message.
Understand the purpose of RAS location messages. These messages are used to request
information from one gatekeeper to another in a multi-gatekeeper environment.
Know where local zones reside in an H.323 environment. Local zones are the zones
congured between a voice gateway and their local gatekeeper.
Know where remote zones reside in an H.323 environment. Remote zones are the zones
congured on a gatekeeper other than the local gatekeeper.
Know the difference between zone prefixes and technology prefixes. Zone prexes are
E.164 numbers used to represent an H.323 zone that includes endpoints and voice
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428 Chapter 10

gateways. Technology prexes are special E.164 numbers that, if dialed, direct the calling
device to a location where special H.323 functions reside.
Know how to configure a voice gateway dial peer to use a gatekeeper. To direct a VoIP
dial peer to use a gatekeeper, you use the session target ras command while in config-
dial-peer mode.
Understand how an H.323 gatekeeper calculates zone bandwidth for CAC services. The
equation is Zone_Bandwidth = Number_of _Current_Calls Codec_Payload_Bandwidth 2.
Understand the primary purpose of a CUBE. The CUBE is primarily used for IP-to-IP
gateway connections.
Know the four possible CUBE voice gateway signaling scenarios. They are: SIP-to-SIP,
H.323-to-H.323, SIP-to-H.323, and H.323-to-SIP.
Understand how a CUBE can provide address hiding. The CUBE offers address hiding by
acting as a proxy and terminating and reinitiating signaling and media ows.
Know the pros and cons of media flow-through. Media ow-through offers IP address
hiding and prevention of duplicate network address schemes. The downside is that all
media ows must terminate at the CUBE and therefore increase CPU and bandwidth load.
Know the pros and cons of media flow-around. Media ow-around allows the media ow
to move around the network as opposed to forcing it through the CUBE. The downside is
that it does not provide IP address hiding or prevention of duplicate IP network schemes.
Understand the concept of RSVP. RSVP is a transport-layer protocol that is designed to
dynamically reserve bandwidth resources across an IP network.
Know which command is useful when you want to see CUBE setup messages in real time. The
debug voip ipipgw command is ideal when troubleshooting CUBE connection problems.
Written Lab 10.1
1. What config-gk conguration mode command assigns zoneA in domain example.com
as a local zone?
2. What config-gk conguration mode command assigns zoneEXT in domain example.
com as a remote zone controlled by a gatekeeper with the IP address of 192.168.9.101?
3. What config-gk mode command maps zoneA with E.164 numbers that range between
5500000 and 5599999?
4. What config-if command is used on a voice gateway to identify it as the interface
used for interoperation with the H.323 gatekeeper?
5. When conguring dial peers for voice gateways in a gatekeeper-controlled network,
how do you congure the dial peer to nd the next-hop IP address from the gatekeeper
when in config-dial-peer conguration mode?
6. What gatekeeper verication command lets an administrator view communication
messages between itself and other H.323 components in real time?
c10.indd 428 9/21/11 11:26:40 AM
Hands-On Labs 429
7. You are conguring a CUBE for bidirectional communication between SIP and H.323
networks. You have already congured H.323-to-SIP communications. What conf-
voi-serv command is used to allow SIP-to-H.323 communications?
8. You are conguring a CUBE VoIP dial peer and dont want the media ow to be prox-
ied. What config-dial-peer command is used to do this?
9. You have a CUBE congured for H.323 to SIP functionality. What command is used
verify that you have an H.323 and a SIP call leg for a call that is currently going on?
10. You are running your CUBE as a media ow-through proxy. What command can be
used to view active RTP sessions?
Hands-On Labs
To complete the labs in this section, you need two routers to act as voice gateways and
one router as a gatekeeper. There is a second gatekeeper (Dub_Gatekeeper_1) and third
voice gateway (Dublin_gw1), which act as our remote gatekeeper zone, but we will only
congure our local GB_Gatekeeper_1 and two local zone voice gateways in the labs. The
labs will follow the logical network design shown in Figure 10.17.
Zone: Glasgow
Glasgow_gw1
5554XXX
S0/0
V
Zone: London
London_gw1
GB_Gatekeeper_1
Domain: example.com
Dub_Gatekeeper_1
5553XXX
S0/0
V
V
Zone: Dublin
IP WAN
5555XXX
Default
tech prex:
1#
10.88.88.1
10.77.77.1
V
Dublin_gw1
V
FI GURE 10.17 H.323 Gatekeeper lab diagram
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430 Chapter 10

These labs build on each other, so it is best to perform them in the order listed:
Lab 10.1: Conguring GB_Gatekeeper_1
Lab 10.2: Conguring London_gw1 and Glasgow_gw1
Hands-On Lab 10.1: Configuring GB_Gatekeeper_1
In this lab, we assume that GB_Gatekeeper_1 is precongured on the WAN except for the
gatekeeper-specic settings. It is our responsibility to congure local zones, remote zones,
and zone prexes according to Figure 10.17. In addition, the gatekeeper will be used as a
default technology prex for extension 1#.
1. Log into GB_Gatekeeper_1 and go into privileged exec mode by typing enable.
3. Enter into config-gk conguration mode by typing gatekeeper.
4. Congure the London zone as a local zone and specify the local IP address as the
source IP for RAS messages by typing zone local London example.com 10.88.88.1.
5. Congure the Glasgow zone as a local zone by typing zone local Glasgow example.com.
6. Congure the Dublin zone as a remote zone by typing zone remote Dublin example.com
10.77.77.1.
7. Congure the local zone prex for the London zone to be 5553 by typing zone
prefix London 5553...
8. Congure the local zone prex for the Glasgow zone to be 5554 by typing zone
prefix London 5554...
9. Congure the gatekeeper to be the default technology prex when users key in
extension 1# by typing gw-type-prefix 1# default-technology.
10. Enable gatekeeper services by typing no shutdown.
11. Exit config-gk configuration mode by typing end.
Hands-On Lab 10.2: Configuring London_gw1
and Glasgow_gw1
In this lab, we assume that both the London_gw1 and Glasgow_gw1 gateways are
precongured on the WAN except for the gatekeeper-specic settings and dial peers. We
must congure the voice gateway to communicate with a gatekeeper, congure the VoIP
dial peer pointing to the gatekeeper, and set the default technology prex we have set up on
our gatekeeper.
1. Log into London_gw1 and go into privileged exec mode by typing enable.
c10.indd 430 9/21/11 11:26:41 AM
Hands-On Labs 431
3. Enter into interface serial 0/0 mode by typing interface serial 0/0.
4. Enable gateway-to-gatekeeper operation on this interface by typing h323-gateway voip
interface.
5. Set the zone name to London by typing h323-gateway voip h323-id London.
6. Set the gateway to use the 1# default technology prex by typing h323-gateway voip
tech prefix 1#.
7. Exit config-if mode by typing exit.
8. Enable this router as an H.323 gateway by typing gateway.
9. Congure a VoIP dial peer (dial peer 555) by typing dial-peer voice 555 voip.
10. Congure a destination pattern to match all seven-digit numbers beginning with 555
by typing destination pattern 555...
11. Congure the dial peer to use the default technology prex that our gatekeeper is con-
gured for by typing tech-prefix 1#.
12. Congure the dial peer to look to the gatekeeper for next-hop call-routing information
by typing session target ras.
13. Exit config-dial-peer conguration mode by typing end.
14. Repeat steps 1 to 13 on the Glasgow_gw1 voice gateway to complete the end-to-end
setup between the two networks.
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Review Questions
1. Which of the following is not a mandatory H.323 gatekeeper feature?
A. Zone management
B. Address translation
C. Call authorization
D. Admission control
2. What does H.323 address translation accomplish?
A. Translates IP addresses into physical interface ports
B. Translates IP addresses into physical MAC addresses
C. Translates E.164 numbers into interface ports
D. Translates E.164 numbers into MAC addresses
E. Translates E.164 numbers into IP addresses
3. If a voice gateway sends a RAS gatekeeper discovery message and the gatekeeper
determines that the gateway can register, what RAS message type is returned to the
voice gateway?
A. GRJ
B. GCF
C. RCF
D. RRJ
4. What are the two options that can be used with voice gateways to discover a local H.323
gatekeeper?
A. Using a broadcast message
B. Using a static IP address
C. Using a multicast message
D. Using a MAC address message
5. RAS location messages are sent and received between what two devices?
A. A gatekeeper and an MCU
B. A gateway and a gatekeeper
C. A gatekeeper and any H.323 compatible endpoint
D. Between two gatekeepers
E. Between two MCUs
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6. When a gatekeeper determines that there is a resource problem on the H.323 network,
what type of message does it send to the calling endpoint to inform it that it must wait
before the call setup process can begin?
A. RIP
B. RRJ
C. BCF
D. RAI
7. What IOS command mode must an administrator be in to configure H.323 zones on
a gatekeeper?
A. config-if
B. config-voi-serv
C. config-gk
D. config-h323-gk
8. You are reviewing an H.323 gatekeeper configuration and see the following command:
zone local zoneA example.com 10.101.13.99
What does the 10.101.13.99 represent?
A. The IP address of an endpoint in zoneA that is used as the source IP for RAS messages
B. The IP address of the local gatekeeper that is used as the source IP for RAS messages
C. The IP address of a voice gateway in zoneA that is used as the source IP for
RAS messages
D. The IP address of a remote gatekeeper that is used as the source IP for RAS messages
9. Which of the following is the correct IOS configuration mode and syntax used to configure
a remote zone?
A. Gatekeeper(config-gk)#remote zone zoneA example.com 192.168.1.1
B. Gatekeeper(conf-voi-serv)#remote zone zoneA example.com 192.168.1.1
C. Gatekeeper(config-gk)#zone remote zoneA example.com 192.168.1.1
D. Gatekeeper(conf-voi-serv)#zone remote zoneA example.com 192.168.1.1
10. You are reviewing an H.323 gatekeeper configuration and see the following commands:
zone prefix Denver 50...
zone prefix Seattle 51...
Based on this information, which of the following statements is true?
A. Denver and Seattle are remote zones.
B. Denver and Seattle are local zones.
C. The Denver and Seattle zones have their own H.323 gatekeeper.
D. Voice gateways within the Denver and Seattle zones do not use VoIP dial peers.
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11. You are configuring an H.323 gatekeeper that has two paths to the Dallas zone. Which of
the following is the correct command syntax used to ensure there is a backup path to the
secondary voice gateway in the event that the primary path fails?
A. zone prefix Dallas 4.. gw-priority 4 Dallas_gw1
zone prefix Dallas 5.. gw-priority 5 Dallas_gw2
B. zone prefix Dallas 4.. gw-priority 4 Dallas_gw1
C. zone prefix Dallas 4.. gw-priority 4 Dallas_gw1
zone prefix Dallas2 4.. gw-priority 5 Dallas_gw2
D. zone prefix Dallas 4.. gw-priority 4 Dallas_gw1
12. A user dials a unique E.164 prefix extension to connect to a gatekeeper-controlled device
that provides unique services. What is this called?
A. Technology gateway
B. Prefix service
C. Call admission control (CAC)
D. Technology prefix
13. Which of the following commands and configuration modes will enable the H.323
gateway-to-gatekeeper service on a voice gateway?
A. Gateway(config-gw)#gatekeeper
B. Gateway(config)#gatekeeper
C. Gateway(config-gw)#gateway
D. Gateway(config)#gateway
14. Your network has 5 G.711 and 3 G.729 calls operating on a gatekeeper controlled network
between two zones. According to the gatekeeper, how much bandwidth is being utilized?
A. 688 Kbps
B. 640 Kbps
C. 384 Kbps
D. 344 Kbps
15. You are reviewing an H.323 gatekeeper configuration and come across the following command:
bandwidth interzone zone default 1024
Which of the following statements is correct?
A. This command is used to create static CAC.
B. This command is used to create static RSVP-CAC.
C. This command is used to create dynamic CAC.
D. This command is used to create dynamic RSVP-CAC.
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16. What is the primary difference between a CUBE configured for media flow-through as
opposed to media flow-around?
A. Media flow-through does not proxy media streams on the CUBE.
B. Media flow-through proxies media streams on the CUBE.
C. Media flow-through allows the RTP sessions to find the optimal path from end-to-end
on an IP network.
D. Media flow-through does not prevent overlapping IP address space.
17. A CUBE is configured for RSVP-CAC. When are path reservation messages exchanged?
A. Before the call setup message is received
B. After the call setup message is received
C. Before the H.323 endpoint capabilities message is received
D. After the H.323 endpoint capabilities message is received
18. A CUBE is providing interoperation between a SIP and an H.323 network. Which of the
following call-initiation types can a CUBE be configured for? (Choose all that apply.)
A. Early offer to fast start
B. Early offer to slow start
C. Delayed offer to fast start
D. Delayed offer to slow start
19. Which of the following configuration examples correctly configures a CUBE for
bidirectional SIP-to-H.323 interoperation?
A. CUBE(config)#voice service voip
CUBE(config-voice-serv)#allow-connections h323 to sip
B. CUBE(config)#voice service cube
C. CUBE(config)#voice service voip
D. CUBE(config)#voice service cube
20. You enable CUBE debugging by issuing the debug voip ipipgw command and see the
following:
May 19 20:27:53.430 CST: cch323_media_flow_mode: IPIPGW(3001):Flow Mode=1
May 19 20:27:53.430 CST: cch323_set_h245_state_mc_mode_outgoing:
call_spi_mode = 1
What does Flow Mode=1 mean?
A. The CUBE currently has one H.323 call leg.
B. The CUBE is configured for media flow-through.
C. The CUBE is configured for media flow-around.
D. The CUBE currently has one SIP call leg.
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1. C. Call authorization is an H.323 feature that can be optionally congured, while the
other three features are mandatory when conguring H.323 gatekeepers.
2. E. An H.323 gatekeeper maintains a table of E.164 numbers to next-hop IP addresses of
the local zones it controls.
3. C. A gatekeeper RAS registration RCF message is returned to the voice gateway when the
gatekeeper decides it can register to it.
4. B, C. H.323 voice gateways can discover their local gatekeeper by either statically conguring
the IP address of the gatekeeper and sending a unicast RAS or by sending a multicast message.
5. D. RAS location messages are exchanged between two gatekeepers to send query messages
about remote zones.
6. A. The Resource in Progress (RIP) RAS message is used by the gatekeeper to inform the
H.323 endpoint that a resource constraint has been discovered and to allow for more time
to begin the call setup process.
7. C. H.323 zones are congured while in config-gk mode on a gatekeeper.
8. B. 10.101.13.99 is the IP address of the gatekeeper you are currently conguring. It signies
that this is the IP that will be used to source RAS messages. This command can only be entered
on a single zone-conguration command, but it is then used for all congured zones.
9. C. Remote zones are congured while in config-gk conguration mode, and the proper
syntax is zone remote zoneA example.com 192.168.1.1.
10. B. Given the conguration information in the question, the gatekeeper manages the Denver
and Seattle zones locally.
11. D. When you are conguring zone redundancy using priority commands, the E.164 numbers
and zone names must match. The priority numbers are used to determine the primary path
and therefore one should be more preferred (a higher number).
12. D. Technology prex numbers are special E.164 prexes that users can dial to access
special gatekeeper-controlled resources.
13. D. The gateway-to-gatekeeper interoperation must be enabled on a voice gateway by
issuing the gateway command while in global conguration mode.
14. A. The equation the gatekeeper uses is Zone_Bandwidth = Number_of _Current_Calls
Codec_Payload_Bandwidth 2.
15. A. CAC on gatekeepers is static in nature. This command is used to limit the maximum
bandwidth for H.323 trafc to 1024 Kbps.
16. B. Media ow-through acts as a proxy for media streams such as RTP for voice transport.
17. B. RSVP-CAC messages are sent as soon as the call setup message is received by the local
CUBE.
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18. A, D. A CUBE can be congured to interoperate between a SIP early offer to H.323 fast-
start initiation process or a SIP delayed offer to H.323 slow-start process only.
19. C. To access conf-voi-serv conguration mode, you must use the voice service
voip command. Then two allow-connections commands must be entered for bidirectional
communication between the SIP and H.323 networks.
20. B. The Flow Mode=1 output from the debug voip ipipgw command means that the voice
gateway is processing H.323 media that are congured for the ow-through method.
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1. zone local zoneA example.com
2. zone remote zoneEXT example.com 192.168.9.101
3. zone prefix zone1 55.....
4. h323-gateway voip interface
5. session target ras
6. debug ras
7. allow-connections sip to h323
8. media flow-around
9. show call active voice brief
10. show voip rtp connections
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Introduction to
Quality of Service
CHAPTER:
Describe the need to implement QoS for voice and video.
Describe the causes of voice and video quality issues.
Describe how to resolve voice and video quality issues.
Describe QoS requirements for voice and video traffic.

Describe and configure the DiffServ QoS model.
Describe the DiffServ QoS model.
Describe marking based on CoS, DSCP, and IP Precedence.
Describe trust boundaries.
Describe the operations of the QoS classifications and

marking mechanisms.
Describe Low Latency Queuing.
Describe the operations of the QoS WAN link efficiency

mechanisms.
Chapter
11
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As soon as IP networks were designed and implemented
with sufcient redundancy mechanisms in place to rival
traditional voice systems in stability, it was only a matter
of time before voice made the transition to IP. During this early transition period, early
adopters began noticing that for voice trafc to function as well on a packet network as it
did on traditional circuit-switched networks, the transport method used by IP networks
needed some additional policies and compression techniques in place. Thus began the rise
of Quality of Service (QoS), the collective term for queuing techniques devised to help
eliminate bottlenecked areas on a network.
This chapter covers the who, what, when, where, and why of QoS on IP networks.
Newly added voice trafc began creating bottlenecks, and these bottlenecks led to the
need to create a way to prioritize and queue these packets that are highly sensitive to drops
and latency. Specically, you will learn what it is that causes IP networks to falter when
running real-time streaming voice and video and how QoS and compression techniques can
be used to eliminate each of those problems.
In Chapter 12, Conguring Quality of Service, well move on to the how of QoS on
IP networks as we explore conguring various QoS scenarios.
Problems with Voice/Video
on IP Networks
To understand what QoS does, you need to understand the problems it was introduced
to solve. Before the convergence of time-sensitive transport such as voice and video, IP
networks dealt with applications and data that had the following characteristics:

Large packet payloads

Bursty transport ow

Time-exible transmissions

No one application or data ow with higher priority than another on shared links

The ability to recover in the event of packet drops
As you can see, most data trafc before voice and video were added was inherently
robust. It didnt really matter how long it took for data to get from point A to point B,
as long as it was transported without errors. Thus you see that most data applications
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were built using TCP, which has built-in CRC checks and retransmission of lost or
damaged packets.
Todays modern IP networks that carry voice and video have very different transport
needs outside of the standard data ows just described. Now a network must also provide
mechanisms to carry trafc with these characteristics:

Small packet payloads

Continuous transport ow

Time-sensitive payloads

A way to dene some data ows as higher priority than others on shared links

High sensitivity to packet drops
Because of these new requirements, network administrators must focus on four primary
modications to ensure that voice/video trafc does not suffer on an IP network. Well look
at those factors in the next section.
Mitigating IP Network Voice Issues
Now that converged voice/video and data networks are here to stay, network designers and
administrators must educate themselves about addressing IP network issues so that time-
sensitive data can properly be transported in a reliable and efcient manner. There are four
primary issues to address:

Providing sufcient bandwidth for a converged network

Reducing end-to-end delay

Reducing jitter

Eliminating packet loss
Lets break down each of these issues to see how they can be resolved on a network. You
will then learn how to implement QoS congurations to mitigate the issues in Chapter 12.
Providing Sufficient Bandwidth for
a Newly Converged Network
When planning for a converged voice/video and data network over IP, you must carefully
consider how to allow for the increase in bandwidth usage. There are several considerations
when determining how much bandwidth will increase when adding IP voice to the mix.
These include things such as:

Number of users

Internal versus external calling

Remote site bottlenecks
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442 Chapter 11
Introduction to Quality of Service

Codec choice

Required voice features and services

Future growth estimates
Keep in mind that there are other reasons for determining how much bandwidth
is required, but these are the primary ones to focus on. See Chapter 5, VoIP Design
Options, if you need to revisit bandwidth calculations.
Reduce End-to-End Delay
Chapter 5 discussed xed and variable delay as aspects of the quality of voice calls on
IP networks. It should be stated again, however, that while a certain amount of delay is
necessary and acceptable, it is the responsibility of the network administrator to limit
variable delay whenever possible. You can use three basic techniques to reduce variable
delay: eliminate bottlenecks, add compression, and prioritize time-sensitive trafc.
Table 11.1 describes the advantages of each. Again, youll learn how to implement these
techniques in Chapter 12.
TABLE 11.1 Variable-delay reduction techniques
Technique Description
Eliminate
bottlenecks
Bottlenecks not only drop packets, but they can also force packets
into queues until the network router/switch can process them. These
queuing delays can substantially add to the delay of a packet.
Add compression Compression reduces the packet size and therefore reduces the
amount of overall bandwidth consumed on a link.
Prioritize time-
sensitive traffic
Not all IP packets require low delay times. You can pinpoint voice/
video traffic to give it priority when it enters a queue. By moving
time-sensitive traffic to the front of the queue, you can reduce
variable delay for the data streams that absolutely require it.
Reduce Jitter
Network jitter and variable delay often go hand in hand. While delay attempts to reduce
the time it takes for packets to be transported from one end to the other, jitter attempts to
stabilize the time in between the receipt of packets at the destination. If you dont keep jitter
within a specied range (30 ms for voice), the audio stream at the destination will end up
sounding distorted and garbled. The same techniques used to eliminate variable delay can
also be used to reduce jitter.
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Eliminate Packet Loss
When I think about network bandwidth and packet loss, Im often reminded of a quote by
Abraham Lincoln: You cannot escape the responsibility of tomorrow by evading it today.
I am reminded of that quote because most networks today have more applications utilizing
more and more bandwidth. It is a major responsibility of the network administrator to
constantly monitor end-to-end link utilization on a network. Even though your network
utilization may be ne today, you need to have plans in place for the time when you are
approaching the point where your network becomes overutilized. When a link becomes
overutilized, packet loss often occurs in the form of interface output queue drops
(sometimes called tail drops). When an interface becomes overwhelmed with trafc, it
begins placing packets into an output queue buffer in the hope that trafc will eventually
die down and the network device can catch back up. If the trafc does not die down,
however, the queue lls up. Those packets that cannot be placed into the already full
output queue are dropped. Additional but less-frequent reasons for packet loss due to
bottlenecks include these:

Input queue drops

Overruns

Ignored packets

Frame errors including CRC, runts, and giants
The primary areas of concern are at the network bottlenecks, as Chapter 5 briey
described. Network bottlenecks can cause a network interface to become overwhelmed.
And when the interface cannot handle any more data, some of it is dropped. Because voice
and video streams are highly sensitive to dropped packets, it is important to be able to
identify various bottlenecks on a network. Figure 11.1 shows the network location where a
bottleneck is most likely to occur between two IP phones.
FI GURE 11.1 Network bottleneck
1000 Mbps 1000 Mbps
1000 Mbps 1000 Mbps
Possible
bottleneck
45 Mbps
V V
IP WAN
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By identifying possible bottleneck interfaces before problems occur, administrators can
closely monitor the increase in utilization that will occur as network usage grows. When
utilization begins reaching capacity, its time to consider either implementing compression
techniques or increasing bandwidth. Ideally, increasing bandwidth is the way to go in most
situations, but compression can be used when upgrades are not possible. Compression
techniques include compressing the IP header information and compressing the IP payload.
Both of these techniques are described in more detail later in this chapter.
Putting the Pieces Together
Now that weve identied problems with voice on IP networks and looked at some of the
solutions, there are three primary steps that we can take so that voice/video can operate
well on a converged network. The rst step is to add bandwidth wherever it is needed. This
is a simple yet highly effective solution. Unfortunately, it can also be expensive. The other
two steps involve careful planning and conguration to accomplish, and these are what the
remainder of the book will cover. First, we have Quality of Service, which is used to give
time-sensitive trafc priority on the network to limit delay, jitter, and packet loss. Next, we
can congure link efciency and compression techniques to lower our bandwidth utilization
footprint. While link efciency and compression isnt technically QoS, it is good to combine
the two methods because they can help signicantly reduce bandwidth utilization and
ultimately move trafc across a network with less latency and packet drops. Well begin by
covering QoS.
The Three-Step QoS Process
So the goal for us is to implement QoS in order to provide a much more consistent and
steady transport mechanism for voice, video, and other time-sensitive data ows. While our
best-effort design may work well for data, voice trafc requires a bit more care to function
optimally. Now that you know what were trying to accomplish with QoS, lets turn our
attention to how it works.
The QoS function has three stages, which well look at each in turn:
1. Trafc classication
2. Trafc marking
3. Trafc queuing
Traffic Classification
Trafc classication is the process of identifying trafc based on different characteristics
in order to group the same trafc types together for QoS. The identication process must
be performed rst because the equipment must be able to clearly identify certain trafc.
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QoS Policy Considerations 445
Creating voice VLANs makes it easy to identify voice trafc, because we can assume that
any packets on the voice VLAN should be classied as such.
Traffic Marking
Trafc marking is the process of agging critical packets so that the rest of the network can
properly identify them and give them priority over all other trafc. Cisco phones have the
ability to mark voice packets with a Class of Service (CoS) and Differentiated Services Code
Point (DSCP) value. The CoS is a eld within the Layer 2 Ethernet frame header that marks
trafc as being one of eight (0 to 7) classes. The higher the CoS value, the more priority is
given. By default, voice trafc is marked with a classication of 5. If data is not marked with
a CoS, it is given a value of 0. The CoS is used by Layer 2 switches for proper queuing.
The Cisco phone also marks the IP packet with a DSCP value at Layer 3. By marking
the ToS/DS eld, DSCP essentially does the same thing as CoS but is intended to be used
by Layer 3 devices such as routers and switches. Also keep in mind that the Layer 2 headers
change at each hop, while Layer 3 header information always remains until it reaches its
destination.
Traffic Queuing
Trafc queuing is the process of ordering certain types of trafc for transport over LAN/WAN
interfaces. Queues are logical storage devices that can be used for egress interface trafc.
Egress basically means that the trafc is exiting the interface as opposed to coming into it.
Queuing for ingress trafc is not possible, because no queues are available. There are several
queuing techniques, discussed under Congestion Management, later in this chapter.
QoS Policy Considerations
With every managed network, there are network providers and network customers or
users. When conguring QoS on an IP network, it is important to create a written policy
that details what kind of service end users should expect depending on the trafc type or
application used.
The Three-Step QoS Policy Development Process
The construction of a QoS policy consists of the following three steps:
1. Consider the trafc types on your network and determine their network delay, jitter,
and packet loss requirements. The ITU-T G.114 recommendation states that a one-way
delay should not exceed 150 ms for voice. Additionally, Cisco recommends that jitter
stay under a 30 ms average and that packet loss should be held under 1 percent.
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2. Put your various trafc types into categories based on network requirements. The
more sensitive the trafc is to latency, jitter, and packet loss, the higher the priority.
For example, voice and video would be placed into the high-priority category, while
FTP would be considered low priority. Other applications that are not necessarily time
sensitive but are important to the business may also be higher on the priority list.
While you might think it to be a good idea to have dozens of different QoS
priorities so that your policy is highly granular, having categories over the Cisco
recommended maximum of 11 adds very little additional value.
3. Document your QoS policy to show users where their application trafc ts into the
QoS policy structure. Additionally, explain why some trafc is given priority over
others. In this way, your network becomes highly transparent to end users, so they
understand why some trafc is given a higher priority on the IP network.
Methods of Configuring QoS Policies
As QoS has evolved over the years, so too have the methods for conguring QoS policies
on Cisco hardware. Following are the three primary methods of conguring QoS policies on
QoS-aware Cisco equipment such as routers and switches.
Command Line Interface
The command line was at one time the only way to congure QoS on Cisco equipment.
As it is with all Cisco command-line interface (CLI) methods, its highly robust in the fact that
everything that you can do and modify for QoS, you can do with the CLI. Unfortunately, the
major drawbacks are the fact that conguring QoS using the command line requires many
steps on multiple interfaces of your hardware. This often led to misconguration errors on
the equipment, which in turn often led network administrators to scrap QoS congurations
altogether. But if you know what you are doing and you like the exibility, the CLI is
certainly an option for QoS.
AutoQoS
To help simplify the QoS conguration process as well as help eliminate
miscongurations, Cisco developed AutoQoS, which is essentially a CLI script that can
be run on QoS-capable Cisco interfaces. This script has only a couple of conguration
options to choose from, depending on your network type and the device(s) connected to
the interface being congured. The standard Cisco AutoQoS for VoIP is used to congure
Cisco routers and switches within a LAN. The AutoQoS for the Enterprise feature, on the
other hand, is used at the WAN edge for remote-site QoS conguration across common
WAN interfaces such as serial, Frame Relay, and ATM circuits. We will use AutoQoS for
basic QoS conguration in Chapter 12.
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Modular QoS CLI
The modular QoS CLI (MQC) method strikes a happy medium between the CLI and
AutoQoS methods. Using MQC-specic CLI commands, a network administrator can
construct a single QoS module on the IOS router or switch. Once that module has been
congured, it can then be applied to any interface on that hardware. This gives us a
highly exible and robust QoS conguration system that eases conguration and
management issues.
MQC has a modular three-step hierarchical structure when conguring a module:
1. Congure a trafc class that is used to identify a priority of network trafc such as voice.
2. Create a trafc policy that denes the amount of network resources that should be
reserved. The trafc class is assigned to a trafc policy.
3. Assign the trafc policy to the appropriate network interface.
Chapter 12 also demonstrates using MQC for class-based QoS conguration.
QoS Classification Models
We can categorize all QoS functionality within three distinct QoS feature models:

Best-effort

IntServ

DiffServ
The next three sections will cover what each of these models provides in regard to
service of trafc on an IP network.
The Best-Effort Model
The Best-effort QoS model is really no model at all. When IP networks run without QoS, all
trafc is considered to be best-effort, meaning that there is no guarantee that the packets will
be delivered. Additionally, all trafc is treated identically. Thus an email message would be
treated by the network the same way that an IP voice call would be. This is how the Internet
currently works, as well as any private network that does not have QoS implemented.
The IntServ Model
Integrated services, or IntServ, is the only model that guarantees the quality of service for
specic types of trafc from end to end. IntServ provides these guarantees by reserving
a dedicated amount of bandwidth to specic trafc. Once that trafc has been reserved,
it is set aside to be used only by the intended trafc regardless of all other trafc. This
bandwidth guarantee is why the IntServ model is often referred to as hard QoS. In a way,
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IntServ carves out its own connection for specied trafc, similar to a PSTN circuit.
When a PSTN circuit is not in use, it sits idle. In the same sense, bandwidth that has
been reserved by IntServ might sit idle as well while the rest of the bandwidth becomes
overutilized.
So what type of IP trafc might be congured with IntServ? The classic example would
be a dedicated video-streaming application that uses a well-dened amount of continuous
bandwidth. It is important to point out that IntServ is inherently granular, because one
specic ow type must have its own bandwidth reservation. That is why only the most
critical applications are congured with IntServ. Considering that you may have hundreds
or even thousands of different data ows, it would be impossible to congure IntServ for
each of them.
IntServ is built upon the Resource Reservation Protocol. RSVP is used for admission
control and instructs the QoS device as to what classication the packets should be given
along the path. This classication is then used along the entire path of the trafc stream to
reserve a set amount of bandwidth.
While it is true that IntServ provides an absolute guarantee of bandwidth for specic
applications, there are some major drawbacks. For one, IntServ must be congured at every
Layer 3 device along the path of the trafc ow. Because of this requirement, IntServ does
not scale well. Second, when bandwidth is reserved and not in use, no other trafc can use
that bandwidth even when it might be needed. The hard reservation of bandwidth is often
wasteful and can lead to utilization problems for other types of trafc. Think of IntServ as
an overprotective mother. She wont let her children play football because of the off chance
that they might get hurt. But since this scenario rarely happens, the child never gets to play
the game. Similarly, RSVP reserves bandwidth because of the fear the link will be over
utilized. But if a network is properly designed, overutilization is probably rare and thus
reserve bandwidth is wasted. Because of these drawbacks, IntServ is rarely used and the
DiffServ model is used instead.
The DiffServ Model
The differentiated services or DiffServ model is sometimes referred to as soft QoS. DiffServ
classies (differentiates) IP trafc ows and marks them for use on other QoS-aware devices
along the trafc ow path. This is similar to the IntServ model in terms of classication.
DiffServ can group together multiple data-ow types into a single group, however. This helps
tremendously with conguration scalability.
Another difference from IntServ is that DiffServ does not make an explicit reservation
along the path for classied trafc. Instead, the DiffServ marking is used along each hop
of the path that trafc takes. Each hop could potentially give the trafc a different level of
service, and therefore the quality cant be considered guaranteed. But if a network is
managed by one administrative source, DiffServ could be congured so that service
is nearly guaranteed. Additionally, bandwidth is never set aside for one specic type of
bandwidth, so a network can be more cost-effective.
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Comparing the QoS Models
Before we look in more detail at the widely used DiffServ model, it will be useful to
summarize and compare the three QoS models: Best-effort, IntServ, and DiffServ. Each has
its benets and drawbacks. It is important that CVOICE candidates understand when one
model is better than another in any given situation. Therefore, Table 11.2 lists the pros and
cons for each model.
TABLE 11. 2 The three QoS models compared
Model Pros Cons
Best-effort Highly scalable No traffic differentiation
No configuration required No service guarantee
IntServ Absolute service guarantee Not scalable because of configuration
complexities
Complete bandwidth control Wasted bandwidth when services not in use
Highly granular Continuous signaling, which wastes a small
amount of bandwidth
DiffServ Highly scalable Attempted service guarantee but not
absolute
Highly granular Mildly complex to configure
Understanding the DiffServ ToS/DS Byte
DiffServ uses a trafc-marking mechanism based on either the Type of Service (ToS) byte
or the Differentiated Services (DS) byte contained in every IP header. In reality, the ToS
and DS byte are one and the same. ToS (originally dened in IETF RFC 791) was used to
assign packet priorities using IP Precedence. The ToS byte was later renamed the DS byte
when it became obvious that IP Precedence was not granular enough for many networks.
Instead of IP Precedence, the DSCP marking method was used to create a more granular
marking structure as well as provide congesting markings. Lets take a look at both IP
Precedence and DSCP to compare the two marking mechanisms.
IP Precedence
This marking method uses 3 of the 8 bits of the ToS byte. The remaining 5 bits are unused.
Specically, IP Precedence uses the 3 leftmost bits of the ToS, as shown in Figure 11.2.
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FI GURE 11. 2 IP Precedence and the ToS byte
ToS byte
0 1 2
IP precedence
bits
Unused
3 4 5 6 7
Since IP Precedence uses 3 binary bits, that means that there are eight possible IP
Precedencemarking values, which are numbered 0 to 7. However, the two highest numbers
(6 and 7) are reserved for network control trafc such as routing protocols. That means
there are six categories that a network administrator can prioritize trafc into. The higher
the IP Precedence, the more preferred the trafc is. That is why it is most common to mark
voice and video with an IP precedence of 5. Table 11.3 lists the eight possible IP Precedence
values and their descriptions according to the RFC.
TABLE 11. 3 IP Precedence priorities
3-Bit Binary Decimal Description
000 0 Routine
001 1 Priority
010 2 Immediate
011 3 Flash
100 4 Flash override
101 5 CRITICAL/ECP
110 6 Internetwork control
111 7 Network control
While the IP Precedence method of marking packets is simple to understand, the limitation
that packets could only be assigned six different priorities clearly was a drawback, because it
did not provide enough markings to classify the dozens or hundreds of different trafc ows
and applications used on todays networks. Add to that the fact that IP Precedence did not
utilize all of the bits within the ToS byte, leaving room to expand the number of priorities,
and that is precisely why DSCP was created. You will learn about it next.
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DSCP
The Differentiated Services Code Point (DSCP) method effectively replaces IP Precedence and
is dened in RFC 2474. It uses a 6-bit eld in the newly renamed Differentiated Services (DS)
byte (previously known as the ToS byte). DSCP is far more granular, because 6 bits are used
to prioritize packets instead of only 3 as with IP Precedence. A network administrator can
congure similar behavior ows to operate inside various classes. These similar ows that are
traveling in the same direction on a network device are called a behavior aggregate (BA).
You are probably curious what the other 2 bits of the DS byte are used for. When DSCP
was rst developed, the last 2 bits were unused and served no purpose. In 2001, RFC 3168
was introduced and there was nally a role for the last 2 bits. According to RFC 3168,
the 2 rightmost bits are for explicit congestion notication (ECN). Layer 3 devices can
be used to monitor congestion and mark the ECN bits when congestion is detected on an
interface. The ECN can then be read by other ECN-aware network devices to reduce their
transmission rates. ECN is often used on network equipment that uses Weighted Random
Early Detection (WRED) congestion management, which is a more robust congestion
tool for voice/video than packet drops. ECM and queuing mechanisms will be discussed
later in this chapter. Figure 11.3 shows the DS byte, with its 6 leftmost bits used for DSCP
markings and its 2 rightmost bits used for ECN.
FI GURE 11. 3 DSCP and the DS byte
DS byte
0 1 2
DSCP bits ECN bits
3 4 5 6 7
Now network administrators could theoretically have up to 64 different priority
markings if they choose. This made DSCP almost too exible, so some guidelines were
needed so that DSCP values can be uniform when crossing into a network managed by
a different administration group. The IETF has created four structured DSCP per-hop
behaviors (PHB):

Default PHB

Expedited Forwarding (EF) PHB

Assured Forwarding (AF) PHB

Class Selector (CS) PHB
These DSCP subsets of the 64 possible DSCP markings are dened next.
Default PHB The default PHB is for all IP data that requires only a best-effort level of
service. This would likely be used on the majority of your network data including FTP,
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HTTP, and other noncritical and non-delay-sensitive data ows. The default PHB is dened
in DSCP as all 0s. Because DSCP uses 6 bits, a default PHB is 000000.
Expedited Forwarding PHB The Expedited Forwarding (EF) PHB is dened in RFC
3246 and is used for IP data ows that require low latency, packet loss, and jitter. These
characteristics are ideal for real-time trafc such as voice and video, and therefore most
voice trafc is tagged with EF PHB. Using the 6 bits of the DSCP eld, EF is 46 in decimal
or 101110 in binary format.
Assured Forwarding PHB The Assured Forwarding (AF) PHB is dened in RFC 2597
and 3260 and has 12 different priority classes within the group. The priorities are broken
up into four classes each containing three drop probabilities. The priorities within each
class are divided into low, medium, and high drop probabilities, as shown in Table 11.4.
TABLE 11. 4 AF PHB classes and drop priorities
Drop Probability Class 1 Class 2 Class 3 Class 4
Low drop AF11 (DCSP 10) AF21 (DCSP 18) AF31 (DCSP 26) AF41 (DCSP 34)
Med drop AF12 (DCSP 12) AF22 (DCSP 20) AF32 (DCSP 28) AF42 (DCSP 36)
High drop AF13 (DCSP 14) AF23 (DCSP 22) AF33 (DCSP 30) AF43 (DCSP 38)
The higher the AF class number, the more preferred the packet will be on a network.
But instead of using strict priority queuing between classes, a fair queuing algorithm is
commonly used so that lower-class packets are not choked off completely. Additionally,
packets within a class have a drop precedence applied to them. If congestion occurs within
a single class, the packets marked with a higher drop precedence are dropped before ones
marked with medium and low precedence. Drop precedence is handled using traffic-policing
mechanisms, which are used to drop excess packets that venture above a defined rate limit.
If you look at the binary conversion of the 12 AF PHB classes, you can better see the
structure and backward compatibility inherent in AF PHBs, as shown in Table 11.5.
TABLE 11. 5 AF PHB binary values
Class 1 Class 2 Class 3 Class 4
Low drop 001 010 010 010 011 010 100 010
Med drop 001 100 010 100 011 100 100 100
High drop 001 110 010 110 011 110 100 110
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Recall that IP Precedence uses only the 3 leftmost bits in the field (highlighted in gray in
Table 11.5). Therefore all AF class 1 values would be treated as having an IP Precedence of 1.
Class 2 AFs would have an IP Precedence of 2, and so on. Alternatively, you will notice
that all of the drop precedence bit values (in white) are identical for the low (binary 010 or
decimal 2), medium (binary 100 or decimal 4), and high (binary 110 or 6) drops.
One final thing to keep in mind is that the highest number here does not represent the
highest priority packet. The leftmost 3 bits that represent the AF classes are based on
the strategy that a higher number is better, but the drop preference bits use a lower-number
strategy. Therefore, a class 2, low-drop packet (binary 010010 or decimal 18) is less likely
to be dropped than a class 2, high-drop packet (binary 010110 or decimal 22).
Class Selector PHB The Class Selector (CS) PHB is dened in RFC 2474 and is the DSCP
subset that most closely follows IP Precedence values. This is because CS PHBs technically
use only the 3 leftmost bits, and the 3 rightmost bits are all 0s. So when a CS value is
110000 (or a decimal value of 40), devices that are compatible only with IP Precedence
would read 110, or a decimal IP Precedence of 5, which means this packet would be treated
as a voice/video real-time priority packet.
DiffServ Service Quality Features
Once IP packets are placed in classes and properly marked, a number of quality tools can
be implemented on a network to make the network play nice for time- and drop-sensitive
packets. This includes features such as the following:

Congestion management

Congestion avoidance

Trafc policing and shaping

Link efciency
In the next few sections, we will detail what each of these features can do for priority
trafc ows.
Congestion Management
Congestion management uses logical queues within network hardware interfaces to store
packets that are waiting to be transmitted on a congested link. Different queuing mechanisms
can be used to determine which packets leave the queue rst and which ones have to stay a
longer period of time. This is where classication and marking of packets comes into play.
Once you can classify trafc ows and mark them so that network equipment can differentiate
between packets of different classes, queue emptying strategies can be implemented.
Cisco currently supports these queuing mechanisms:
First-In First-Out The rst-in rst-out (FIFO) queuing mechanism does not place any
emphasis on packet priorities. Instead, the rst packet to be placed in the queue is the rst
one to come out. This is the default queuing method on Cisco hardware for any interface
above an E1 speed.
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Priority Queuing Priority queuing (PQ) is a strict queuing mechanism that is used to give
explicit priority to certain trafc types. These trafc types can be divided into categories
such as protocol type, source/destination IP, packet size, and incoming interface. PQ
can place trafc into one of four different queueslabeled high, medium, normal, and
lowbased on the priority assignment by the network administrator. Packets that are
not prioritized are placed into the normal queue. Packets in the higher queues are given
absolute preferential treatment and allowed out of the queue. Therefore, PQ suffers from
queue starvation for trafc at lower preferred levels. Priority queuing is also known as
strict priority queuing because of its strict nature.
Custom Queuing The custom queuing (CQ) mechanism divides the total number of
queue slots into different classes. Each class gets a certain number of queue spaces; this
value can be congured by the network administrator. The more preferred a class is, the
more queue slots it is given. The queued trafc is then performed in a round-robin fashion,
where the classes with more queue allocation will have the opportunity to transmit packets
out of the queue more frequently. CQ was designed to let administrators allocate more
resources for trafc with minimum bandwidth and low-latency requirements such as
voice. There are 17 total queues, but queue 0 is designated for network signaling messages.
Queues 1 through 16 are handled in the round-robin fashion, so the complete queue
starvation that was found in PQ is eliminated. One drawback is that applications with
larger packet sizes inadvertently receive more bandwidth than others because the round-
robin mechanism sends the complete packet. Therefore, smaller voice packets can suffer
under this method.
Weighted Fair Queuing As youve learned, custom queuing (CQ) has a downside when
it comes to the fairness of handling large versus small packets. The larger packets get
preferential treatment because CQ transmits the entire packet from a queue regardless
of size. Weighted fair queuing (WFQ), on the other hand, is much fairer to small-packet
transmissions because it transmits data from queues at the byte level as opposed to the
packet level. Therefore, if queue 1 has 10 packets that are 200 bytes each and queue 2 has
20 packets that are 100 bytes each, WFQ will transmit 1 packet in queue 1 and 2 packets
in queue 2. Weighted fair queuing also has the ability to prioritize trafc based on data
ows. It does this by matching frame header information such as source/destination IP or
MAC address, protocol and port numbers, and IP Precedence in the ToS eld. WFQ then
places the categorized trafc into either high- or low-bandwidth trafc. Low-bandwidth
trafc ows are given higher priority and receive preferential queuing treatment. The WFQ
mechanism provides a much more consistent one-way delay and reduces jitter. That is why
WFQ is great on low-speed WAN links and is the default queuing method for interfaces at
E1 speeds or below.
Class-Based Weighted Fair Queuing An extension of standard WFQ, class-based
weighted fair queuing (CBWFQ), lets an administrator create classes and places
categorized trafc into multiple classes. Each class can then be given a minimum
bandwidth requirement level, which CBWFQ will attempt to meet for the entire class. Note
that the minimum bandwidth level is not at a ow level but based on all the ows within a
c11.indd 454 9/21/11 11:27:17 AM
class. CBWFQ is very fair, but this fairness can sometimes prevent sensitive voice or video
trafc from getting out of queues because of the inability to use strict queuing methods.
Low Latency Queuing Low Latency Queuing (LLQ) is a priority queue (PQ) mechanism
with CBWFQ classes built in. Essentially, LLQ is two different queuing methods in one.
This mechanism provides a very fair level of service and is commonly used on high-speed
LAN interfaces that transport voice and video. LLQ allows for priority queuing based
on administrator-dened class ows instead of specic trafc types. LLQ is perfect for
voice and video because it allows an administrator to congure voice and video ows in a
high-priority class. This class can then be congured for a strict queuing priority to ensure
preferential treatment. Other trafc that does not need strict priority queuing can use WFQ
or CBWFQ queuing instead.
As you can see, there are many queuing mechanisms to choose from. When QoS was
new, there were only a few queuing strategies, such as PQ and CQ, and they solved very
specic problems while introducing others. Later on, more general-purpose queuing
methods such as WFQ and CBWFQ came around and were very popular. Finally, LLQ was
developed, which is a combination of simple PQ and CBWFQ and is the preferred queuing
mechanism that Cisco recommends for voice and video networks because it was built
specically to service real-time trafc.
Congestion Avoidance
Sometimes an interfaces queue gets lled up to the maximum. When this happens without
intervention, new packets that have no place to go are dropped. For applications that are
not time sensitive and are running TCP, tail-dropped packets are not a big deal because
the packets can simply be retransmitted. But time-sensitive trafc that uses UDP suffers
greatly from dropped packets. Therefore, it is often wise to implement congestion-avoidance
features such as Weighted Random Early Detection (WRED). WRED is a congestion-
avoidance tool that uses the original Random Early Detection (RED) tool and combines
it with IP Precedence intelligence to drop packets based on priority levels. RED uses TCPs
built-in capability to retransmit packets that are dropped. If the sending device begins to
see that it has to continuously retransmit dropped packets, it will automatically reduce its
transmission speed to help ease any possible congestion.
WRED adds an extra layer of intelligence that will discard packets on a congested interface
that have a lower IP Precedence rst. This works out well because voice and video (which
often operate on UDP and thus cannot retransmit packets) will be at a higher precedence,
while more robust data transactions using TCP will have lower IP Precedence and their
packets will be dropped rst, triggering the host to temporarily slow down its transmissions.
WRED is ideal in bottleneck situations and is commonly congured on WAN interfaces,
which are likely to be slower than LAN interfaces.
Traffic Policing and Shaping
Trafc policing and shaping is all about setting maximum limits for classes of trafc.
This approach is in contrast to setting a minimum value. Trafc shaping and policing is
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also known as trafc conditioning, because you condition your trafc not to overreach its
bandwidth boundaries.
Traffic Policing Trafc policing is the more hard core of the two trafc-conditioning
methods. The technique does not rely on interface queues and can therefore be applied
either inbound or outbound on an interface. When data attempts to come into or out of
an interface and it exceeds a congured policing maximum level, the packets are dropped
or marked, based on conguration details. Policing does not buffer packets. Because no
buffering occurs, you can see that bursty data is simply cut off at the tips when graphed, as
FI GURE 11. 4 Traffic policing
Time
Dropped trafc
Maximum
rate
T
r
a
f
c
Notice that when traffic bursts above the maximum rate, traffic policing allows only the
maximum traffic rate to be sent. This is how policing works; the interface is configured to
send only a specific amount of traffic over a set period of time. Because of this characteristic,
you commonly see a sawtooth diagram of peaks and valleys. Traffic policing is also
commonly implemented in network bottleneck situations and usually found configured on
WAN interfaces. The two possible traffic-policing methods that can be configured on Cisco
hardware are these:

Committed Access Rate (CAR): A legacy trafc-policing method that limits trafc
rates based on criteria such as IP Precedence, MAC address, or IP address.

Class-based policing: A newer policing method that uses a more advanced
algorithm and can match trafc on information such as DSCP values, class maps,
and Layer 2 CoS as well as the same criteria that CAR matches on.
Traffic Shaping Trafc shaping buffers data into queues. Because queuing techniques
are only used outbound on interfaces, trafc shaping can only be applied outbound.
Also, because queues are used with trafc shaping, those packets that are queued should
eventually be transmitted. Therefore, if we graph the same interface data as we did with the
trafc policing, instead of the sawtooth graph, we should see a much smoother trafc rate,
as shown in Figure 11.5.
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FI GURE 11. 5 Traffic shaping
Time
Maximum
rate
T
r
a
f
c
The primary differences between policing and shaping are that policing will cause more
drops and therefore more TCP retransmissions, while shaping will add additional variable
delay because packets have to be placed into queues and wait to be transmitted on the wire.
Link Efficiency
In addition to conguring QoS on your network for voice support, you can use two other
link-efciency techniques to help with the consistent transport of VoIP. These techniques are
compression and link fragmentation and interleaving (LFI). Again, keep in mind that link-
efciency techniques should only be applied to low-speed WAN interfaces below 768 Kbps.
Compression Techniques
There are two primary types of frame-compression techniques. The rst is payload
compression, in which compression techniques are used to reduce the data payload size.
Obviously, the smaller the frame size, the more frames can be transmitted over a
Policing vs. Shaping on Low-Speed WAN Connections
Richard was responsible for conguring IP voice services at a remote site that utilized
a shared voice/data WAN connection. At times the WAN became overutilized. Richard
decided to implement trafc policing in an attempt to proactively avoid congestion. After
trafc policing was implemented, Richard noticed that it had little effect on the quality and
reliability of calls during periods of high bandwidth utilization on the WAN. So he scrapped
the trafc-policing conguration and instead congured trafc shaping on his outbound
interfaces across the WAN. This time, there was a noticeable difference in the quality and
reliability of calls. Richard discovered that since voice packets are transported using UDP,
when trafc policing happened to drop those packets, UDP could not retransmit them.
This caused calls to stutter or fail altogether if congestion was high. Alternatively, with
trafc shaping, the voice UDP packets were queued rather than completely dropped. In this
situation, the voice packets might have been delayed, but they eventually made it out of the
queue, and call clarity improved and there were fewer dropped calls.
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xed-bandwidth link. Cisco router hardware commonly uses two forms of payload
compression. The rst method is called stacker compression and uses a special encoded
dictionary that both routers possess. The router replaces streaming data with much smaller
codes found in the dictionary. The data is then sent to the receiving router interface, where
the symbols are looked up and converted back into the original data stream. The predictor
compression method, on the other hand, attempts to predict the next sequence of characters
in a data stream by using an index in the compression dictionary. Predictor looks at a
portion of the data stream and looks it up in the compression dictionary. If a sequence match
is found, that data is replaced with the sequence that was previously looked up.
The other type of frame compression is header compression, such as cRTP, which
compresses the standard 40-byte headers into either 2- or 4-byte sizes depending on CRC
settings. Keep in mind that cRTP can only be congured on serial interfaces, as shown in
Figure 11.6.
FI GURE 11. 6 An example of compression
Router-B Router-A
T1 PPP
S1/0
S1/0
cRTP compression
In general, compression should be used only on low-speed WAN links where potential
bottlenecks exist. Otherwise, the benet of compression simply isnt worth the increased
memory and CPU utilization that compression processes consume. Also remember that
these techniques affect only a single hop along the entire path. When a voice packet must
traverse multiple low-speed WAN links, each of them needs to be congured separately.
Link Fragmentation Interleaving
A second link-efciency technique that is commonly used on low-speed PPP multilink
circuits is called link fragmentation interleaving (LFI). This process takes large data
frames and fragments them into smaller, more manageable sizes. If we dont break down
large frames into smaller ones, our smaller voice packets that are waiting behind the large
data packets can experience serialization delay, which can seriously hurt voice quality.
Serialization delay is the amount of time it takes the router to place a packet onto the
outbound interface. The amount of time depends on the packet size and interface link
speed. The formula for calculating serialization delay is:
Serialization_delay (packet_byte_size 8) / link_bps_speed
For example, lets say we have a 512 Kbps WAN link and a packet that is 1024 bytes.
Therefore we calculate serialization delay as:
Serialization_delay (1024 8) / 512000
Serialization_delay 8192 / 512000
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Layer 2 Class of Service and QoS Trust Boundaries 459
Serialization_delay 0.016 seconds
Serialization_delay 16 milliseconds
If you play around with larger packet sizes across WAN links with low bandwidths,
youll begin to see the serious impact that serialization delay can have on time-sensitive
trafc. This is why LFI is so important on low-speed links. LFI breaks up large, Layer 2
frames into much smaller sizes. It then is able to interleave voice frames between the newly
fragmented data frames. This process ensures that voice packets have a more consistent
variable delay and signicantly cuts down on voice jitter. In Chapter 12 we will congure
the two most commonly implemented LFI mechanisms, MLP LFI and FRF.12, for Frame
Relay connections.
Layer 2 Class of Service
and QoS Trust Boundaries
This section explains how we can design our IP network to best handle QoS for voice and
video. First, we will look at a way to enable Layer 2 devices to prioritize frames using a
eld in an 802.1Q frame. This allows us to mark data either at Layer 2 or Layer 3 in the
network. Next, we will look at the different locations in a network where classication and
marking can and ideally should occur.
Layer 2 Classification and Marking with CoS
So far weve discussed in detail how to mark packets with either IP Precedence or DSCP.
Unfortunately, these two marking methods are Layer 3 mechanisms, and therefore Layer 2
devices (such as Layer 2 switches) cannot perform any QoS marking. Fortunately, Class of
Service (CoS) allows us to congure Layer 2 switches to classify and mark Ethernet frames
that pass through a CoS-capable Layer 2 switch. The CoS consists of 3 bits that are found
inside a eld within the 2-byte 802.1Q header Tag Control Information (TCI). The 3-bit CoS
markings follow the exact same structure as IP Precedence, where there are eight possible
values (07), and priorities 6 and 7 are recommended for network information usage only.
The CoS bits within an 802.1Q frame are shown in Figure 11.7.
FI GURE 11. 7 CoS bits in an 802.1Q frame header
Preamble DA SA
3 Bits of the TCI
are for CoS
TCI FCS PT Data
Start frame
delimiter
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If you are using CoS at the Access layer, voice frames from Cisco IP phones will already
be marked with both CoS (in the 802.1Q frame header) and DSCP (in the IP DS eld)
priorities. Remember that Cisco IP phones transport data in 802.1Q frames, so they can
place voice trafc in one VLAN and (if a computer is connected to the phones Ethernet
port) data in a separate VLAN, as shown in Figure 11.8. The switch can either trust the
CoS markings contained within the 802.1Q header or rewrite them. Native VLAN frames
that come into the switch untagged are assigned an administrator-denable CoS, which is 0
by default.
FI GURE 11. 8 IP phone 802.1Q tags
Switch
Fa0/5 Trunk link
Cisco phone
PC
Voice VLAN
Data VLAN
Once the switch either trusts or re-marks frames, they are sent out the egress port to the
next hop along the network path to their destination. By default, the egress port will send
all trafc to a single queue. Alternatively, an egress port can be congured to place trafc
into one of four queues where frames are placed into a queue. The de-queuing mechanism
can be congured to use either strict priority scheduling or weighted round-robin (WRR)
scheduling.
Identifying QoS Trust Boundaries
You can classify, mark, and begin enforcing queuing strategies for IP trafc
at several points along a network. But where should this process begin? The simplest
answer is to push your trust boundary out as far to the endpoint as possible. But,
depending on the type of network, you may have to pull the boundary in a bit based on
how much you trust markings from end devices (thats why its called a trust boundary!).
If you have full control of endpoints, then you control the CoS and ToS/DS markings that
are generated, and you can push the trust boundary out to the phone and even PC levels.
If you do not have as much control over your network, it might be better to begin marking
(and possibly overwriting) CoS/ToS values as soon as the trafc hits your switch. Also,
you may run into a situation where your access switches are not capable of QoS. Because
of this, you have no choice but to congure the trust boundary at the Distribution layer.
Figure 11.9 illustrates where trust boundaries can be implemented within a
typical network.
c11.indd 460 9/21/11 11:27:21 AM
FI GURE 11. 9 Trust boundaries
Si
Trust at
Distribution layer
Trust at
Access layer
Trust cisco
phones
Trust any
endpoint
Most organizations will trust CoS/ToS/DS markings from Cisco IP phones but will not
trust the markings from devices attached to the phone, such as a PC. When network data
from the PC reaches the Cisco phone, the switch will ignore the CoS/ToS/DS markings and
consider all data packets to have a value of 0 by default.
QoS Baseline Models
When you begin designing QoS policies for your network, it can be a little overwhelming.
Every network is different and runs very different applications. Additionally, data that may
be a priority for some networks may be farther down the list for others. While prioritizing
your network data is completely up to your discretion, there are several available QoS
baseline models that give you a classication framework to organize your data more easily
and with a sense of consistency.
Comparing the Cisco QoS Baseline Model
While there are multiple QoS baseline models, the Cisco QoS baseline model is the one that
we will focus on. The Cisco QoS baseline model consists of 11 different classes that you
can place your applications/data into. Figure 11.10 shows the Cisco QoS baseline model
beside common 8-class and 5-class models so you can compare and contrast how each class
is broken down.
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FI GURE 11.10 The Cisco QoS baseline model compared with 5- and 8-class models
Voice
Cisco QoS
baseline
8-Class
model
5-Class
model
Video conf
Call signaling
Video stream
Routing
Network mgmt
Critical
Transactional
Bulk data
Best-efort
Scavenger Scavenger Scavenger
Voice
Real-Time
Call signaling Call signaling
Network
control
Critical data
Critical data
Bulk data
Best-efort Best-efort
Network
control
Recommended Cisco Baseline Classification Markings
Since we have 11 different classes within our Cisco QoS baseline model, we should mark
our packets with DSCP, PHB, or CoS markings according to the Cisco recommended
method, as shown in Table 11.6.
TABLE 11. 6 Cisco QoS baseline recommended markings
Cisco QoS Baseline PHB DSCP CoS
Routing CS6 46 6
Voice EF 46 5
Video conf AF41 34 4
Video stream CS4 32 4
Critical AF31 26 3
Call signaling CS3 24 3
Transactional AF21 18 2
Network mgmt CS2 16 2
Bulk data AF11 10 1
Scavenger CS1 8 1
Best-effort 0 0 0
c11.indd 462 9/21/11 11:27:22 AM
The beauty of using a QoS model such as the Cisco baseline is the fact that you can more
easily provide an end-to-end classication and marking model. Once you have a consistent
model in place, it is much easier to treat your marked data with the same level of service
throughout the entire network.
Recommended Cisco Baseline Congestion-Management
and -Avoidance Tools
So far, the Cisco QoS baseline model has given us a structured model for classication and
marking. The nal piece of the puzzle is to implement consistent congestion management in
the form of queuing and congestion avoidance using RED and WRED. Table 11.7 lists each
of the 11 Cisco baseline classes, in order from highest to lowest priority, and their Cisco
recommended congestion-management and -avoidance congurations.
TABLE 11. 7 Cisco QoS baseline recommendations for congestion management
and avoidance, from highest to lowest priority
Cisco QoS Base Recommended QoS
Routing CBWFQ + RED
Voice RSVP + LLQ
Video conf RSVP + CBWFQ + DSCP-WRED
Video stream RSVP + CBWFQ + RED
Critical CBWFQ + DSCP-WRED
Call signaling CBWFQ + RED
Transactional CBWFQ + DSCP-WRED
Network mgmt CBWFQ + RED
Bulk data CBWFQ + DSCP-WRED
Best-effort BW Guarantee CBWFQ + RED
Scavenger No BW Guarantee + RED
Notice that some of these recommended congurations, such as voice, include call
admission control. CAC is used to ensure that only a specied number of simultaneous calls
can be made. This is typically done on WAN interfaces that may be a possible bottleneck.
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Summary
This chapter covered the terminology, methods, and models used to implement QoS on a
network for voice and video. This included the three-step QoS process of trafc classication,
marking, and queuing. You also learned two different congestion-avoidance techniques in the
form of compression and LFI. In Chapter 12, we will put all this knowledge to use when we
set about actually conguring QoS on Cisco routers and switches.
Exam Essentials
Understand how to resolve potential problems with voice/video on IP networks. Time-
sensitive trafc such as voice and video can suffer from delay, jitter, and packet loss. To
resolve these problems, network administrators should provide sufcient bandwidth,
eliminate bottlenecks, use QoS to prioritize trafc, and use link-efciency techniques to
reduce bandwidth requirements.
Understand the three-step QoS process. QoS can be broken down into three steps. First
is the classication of data. Next is the marking of classied trafc. And the third is using
interface queuing techniques on marked trafc.
Understand the purpose of a QoS policy. A QoS policy should be created so network
users understand what levels of service to expect for various networked applications.
Know the three QoS IOS configuration methods. The three methods that can be used to
congure QoS on Cisco routers are command-line interface (CLI), AutoQoS, and Modular
QoS CLI (MQC).
Know the three QoS classification models. There are three different QoS classication
models with which to categorize trafc. The models are Best-effort, IntServ, and DiffServ.
Understand the DiffServ ToS/DS byte. The ToS or DS is a byte within every IP header.
This byte is used for the marking of packets so they can be placed into different classes.
It is called the ToS byte when IP Precedence is being used and called the DS byte when DSCP
is being used.
Know the IP Precedence values. IP Precedence is a marking system that uses 3 bits. There
are eight possible values. The higher the value, the more preferred the packet is.
Know the DSCP values. IP Precedence is a marking system that uses 6 bits. There are 64
possible values. DSCP can be used on its own, or per-hop behavior models can be used to
classify trafc.
Know the four IETF PHB systems. The four structured per-hop behaviors (PHB) are Default
PHB, Expedited Forwarding (EF), Assured Forwarding (AF), and Class Selector (CS).
c11.indd 464 9/21/11 11:27:23 AM
Know the four DiffServ quality features. DiffServ can be congured for the following
quality features: congestion management, congestion avoidance, trafc policing/shaping,
and link efciency techniques.
Understand CoS markings. Class of Service markings are 3 bits contained with Layer 2
Ethernet frames that can be used for marking trafc that can be understood by
Layer 2switch hardware.
Know the possible trust boundary locations. Trust boundaries can be located at
endpoints, IP phones, access switch ports, and the Distribution layer.
Understand the Cisco QoS baseline model. Cisco recommends that trafc be classied
into 11 specic groups based on trafc type and sensitivity to delay.
Written Lab 11.1
1. Of all the possible methods used to prevent delay problems on time-sensitive IP trafc,
which method is preferred if possible?
2. What are the three steps of the QoS process?
3. What QoS method uses a three-stage hierarchical conguration structure?
4. Which QoS classication model is also called soft QoS?
5. When using IP Precedence, what is voice trafc commonly marked as?
6. When using DSCP, what is voice trafc commonly marked as using both the DSCP
decimal and PHB formats?
7. How are the 12 PHB AF classes categorized?
8. LLQ is a combination of what two queuing mechanisms?
9. The primary difference between trafc policing and trafc shaping is that trafc
policing drops packets outright instead of using what?
10. A Cisco Catalyst switch operating at Layer 2 can understand which type of QoS
markings?
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466 Chapter 11

Review Questions
1. Which of the following is not a characteristic of voice traffic on an IP network?
A. Small packet payloads
B. Time sensitive
C. Sensitive to compression
D. Sensitive to packet loss
2. What type of delay should a network administrator focus most on reducing using QoS
mechanisms?
A. Fixed delay
B. Variable delay
C. Compression delay
D. Signaling delay
3. What is another name for an occurrence when an output interface queue fills up and begins
discarding packets?
A. Delay
B. Jitter
C. Frame overflow
D. Tail drop
4. Which of the following is not a reason for packet loss on a router interface that has limited
bandwidth?
A. Input queue drops
B. Tail drops
C. DSCP marking of 0
D. CRC errors
5. Which QoS classification model uses either the ToS or DS byte to mark packets?
A. DiffServ
B. IntServ
C. Best-effort
D. Both IntServ and DiffServ
6. IP Precedence marking uses which bits of the ToS IP header field?
A. The 6 leftmost bits
B. The 4 leftmost bits
C. The 3 leftmost bits
D. The 4 rightmost bits
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7. According to best-practice strategies, network administrators can use IP Precedence to
classify traffic into how many different groups?
A. 6
B. 7
C. 8
D. 46
8. What is the decimal equivalent to the AF binary value11?
A. 8
B. 10
C. 36
D. 46
9. Voice traffic is commonly marked with what DSCP value? (Choose two.)
A. 64
B. AF42
C. EF
D. 46
E. CS4
10. You are installing a T1 circuit for an IP WAN to a remote site. What queuing mechanism
will be enabled by default?
A. CQ
B. FIFO
C. None
D. WFQ
11. What QoS queuing mechanism can use strict priority queuing for voice traffic and WFQ or
CBWFQ for other types of traffic?
A. CQ
B. PQ
C. LLQ
D. Traffic policing
E. Traffic shaping
12. Which congestion-avoidance technique drops packets based on IP Precedence values before
buffers begin to get congested?
A. WRED
B. Traffic shaping
C. RED
D. Traffic policing
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13. What are the two traffic-policing methods on Cisco equipment?
A. WFQ
B. Class-based
C. RED
D. CAR
E. WRED
14. Why is traffic shaping more suited for voice and video traffic than policing?
A. Traffic shaping prioritizes voice and video traffic more aggressively.
B. Traffic policing will not queue any packets, including voice and video.
C. Traffic policing will put voice/video packets into queues, which causes delay and jitter.
D. Neither traffic policing nor traffic shaping is recommended when operating voice/video
over IP.
15. We will configure LFI on a 256 Kbps WAN interface but first need to know what
serialization delay to expect if we break up packets into 128-byte sizes. Based on this
information, what is the approximate serialization delay for this interface?
A. 8 ms
B. 16 ms
C. 4 ms
D. 24 ms
16. Which of the following is not a negative aspect of IntServ?
A. No service guarantee
B. Potential for wasted bandwidth
C. Large amount of signaling traffic
D. Not scalable
17. Cisco best-practice methodologies recommend that network administrators separate traffic
into or fewer classes for QoS purposes.
A. 64
B. 32
C. 11
D. 12
18. Using the Cisco QoS baseline model, voice traffic will be marked with which values for
PHB, DSCP, and CoS?
A. EF, 42, and 1
B. AF41, 46, and 7
C. EF, 46, and 5
D. AF41, 42, and 5
E. AF41, 46, and 1
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19. The Cisco recommended congestion-management tools for voice include which two of the
following?
A. CBWFQ
B. LLQ
C. RSVP
D. DSCP-WRED
E. RED
20. Which classification of traffic is given QoS tools that technically give it a lower priority
than a best-effort level of traffic?
A. Bulk data
B. Transactional
C. Scavenger
D. Call signaling
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1. C. Voice trafc is not sensitive to compression, which in many instances is implemented to
provide better voice services.
2. B. Using QoS mechanisms, a network administrator should focus on reducing variable
delay on IP networks for time-sensitive trafc.
3. D. Interface output drops are also known as tail drops.
4. C. Having a DSCP marking of 0 will not cause the packet to be dropped on a bottlenecked
interface unless it is explicitly congured.
5. A. Of the three QoS classication models, only DiffServ uses the ToS or DS byte to mark
packets for prioritization purposes.
6. C. IP Precedence uses the 3 leftmost bits of the ToS byte for marking purposes.
7. A. IP Precedence has eight total markings but best-practice documentation states that
classes 6 and 7 should be reserved for network control trafc and therefore should not be
used. Therefore, there are six available classes to group trafc under.
8. B. AF11 is equivalent to 10 in decimal.
9. C, D. Voice trafc is commonly classied with a DSCP PHB of hexadecimal EF, which is
46 in decimal format.
10. D. WFQ is enabled on interfaces that are E1 speeds or lower.
11. C. LLQ uses a strict priority-queuing technique, which is ideal for voice and/or video
trafc. All other trafc is queued using either WFQ or CBWFQ.
12. A. Weighted Random Early Detection (WRED) is a mechanism that randomly drops
packets before buffer queues get completely full. WRED uses IP Precedence and drops
packets with lower values more often.
13. B, D. There are two methods of implementing trafc policing. CAR is the older method,
used for legacy trafc, and class-based is the newer method that is commonly implemented
in new installations.
14. B. Trafc policing will drop packets outright, whereas trafc shaping places them into
queues. While voice or video packets may be delayed, its better than not getting the packet
at all. Trafc shapers have a queue associated in order to get the best probability of packet
transmission.
15. C.
Serialization_delay (128 8) / 256000
Serialization_delay 1024 / 256000
Serialization_delay 0.004 4 ms
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16. A. One positive aspect of IntServ is its ability to offer absolute service guarantees.
17. C. Ciscos QoS baseline model recommends you segment trafc into one of the 11 different
class categories. Depending on your network size and trafc types, not all classes need to
be used.
18. C. According to the Cisco QoS baseline model, voice trafc should be marked with an EF
PHB, DSCP 46, or CoS value of 5.
19. B, C. The QoS tools recommended for voice are RSVP and LLQ.
20. C. Scavenger trafc is recommended to have no bandwidth guarantee at all.
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1. Increasing bandwidth
2. Trafc classication, trafc marking, and trafc queuing
3. Modular QoS CLI (MQC)
4. DiffServ
5. 5
6. 46 or PHB EF
7. Four classes with three drop priorities
8. PQ and CBWFQ
9. Queue buffers
10. CoS
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Configuring Quality
of Service
CHAPTER:
Describe and configure the DiffServ QoS model.
Configure Layer 2 to Layer 3 QoS mapping.
Configure trust boundary on Cisco switches.
Describe the operations of the QoS classifications and

marking mechanisms.
Describe Low Latency Queuing.
Describe the operations of the QoS WAN link efficiency

mechanisms.
Enable QoS mechanisms on switches using AutoQoS.
Configure Low Latency Queuing.
Chapter
12
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Chapter 12 is where the QoS rubber meets the network
road. Chapter 11, Introduction to Quality of Service,
covered the basics of QoS, and now it is time to apply what
you learned to the routers and switches. You will learn how to implement QoS policies
using the AutoQoS methods at Layer 2 and Layer 3, as well as the three-tiered MQC
mechanism, where we mark trafc ow, set policies, and apply them to interfaces.
In addition, we will look at how to congure class-based link efciency techniques, trafc
policing and shaping, trust boundaries, and Layer 2 to Layer 3 mapping modications.
At the end of this chapter, you should have a solid understanding of how to congure key
QoS components as well as how to verify their operation.
Configuring QoS Policies
Using AutoQoS
If you quickly want to get a uniform QoS implementation up and operational on a network,
AutoQoS is the way to go. Essentially, AutoQoS is a built-in script where the router
automatically evaluates a network and then applies QoS settings based on the scripts best
guess at a policy that will work in a particular infrastructure environment. The evaluation
includes verication of interface types and link speeds. The AutoQoS conguration method
is by far the easiest method to implement because little knowledge is required of you in
order to implement it. QoS deployment times are greatly reduced, and the best-practice
congurations are uniform on all network equipment.
AutoQoS can be congured on both routers and switches, although their congurations
and operations vary. Only certain routers are capable of using AutoQoS. The current
generation of ISR and ISR G2 routers supports AutoQoS. From a router perspective,
AutoQoS is commonly congured on WAN interfaces that may be bottlenecks at some
point along a path. AutoQoS may congure the following features on WAN interfaces:

Automatic classication of RTP, cRTP, and voice gateway signaling protocols (SCCP,
H.323, SIP, MGCP)

Automatic building of service policies for priority trafc

LLQ implementation for high-priority trafc

Trafc shaping where appropriate

Link fragmentation where appropriate

cRTP compression where appropriate
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The automatic classification function within AutoQoS uses the Network-
Based Application Recognition (NBAR) feature to identify and classify
different application and data types based on Layer 4 UDP and TCP port
numbers. In order for NBAR to work on a router interface, Cisco Express
Forwarding (CEF) must be enabled first. CEF has been enabled by default,
beginning at IOS 12.2, so if you are running an earlier version, you must
make sure to manually enable it.
On the LAN side of the network, any Cisco Catalyst switch can have AutoQoS for VoIP
congured on its access ports and trunk ports. The following QoS features can be enabled
on switchports using AutoQoS for VoIP:

Set the trust boundary at the Cisco IP phone

Set the trust boundary at the access port or trunk-port level

Automatically enable PQ and WRR queuing when appropriate

Automatically add or modify CoS markings where appropriate

Automatically adjust queue sizes and weights where appropriate

Perform CoS-to-DSCP or IP precedenceto-DSCP mappings
You must choose from two AutoQoS implementation methods when conguring
AutoQoS on a router:

AutoQoS for VoIP

AutoQoS for the Enterprise
The AutoQoS for VoIP is the least complex AutoQoS method, and it primarily
focuses on prioritizing trafc for voice. It can be congured on either routers or switches.
Larger networks with a substantial number of remote site WAN connections may benet
from additional prioritization for trafc types other than voice (such as video and other
streaming applications), and the more complex AutoQoS for the Enterprise is likely to be
a better t. Note that QoS for the Enterprise can be congured only on routers and not
switches. Well start by going through the AutoQoS for VoIP conguration for both routers
and switches followed by conguring AutoQoS for the Enterprise, while pointing out
differences between the two implementation methods along the way.
Configuring AutoQoS for VoIP on a Router
Conguring AutoQoS on routers is truly a magical thing to see. It seems magical because
AutoQoS intelligently recognizes a network setup and appropriately congures multiple
QoS settings. And when I say multiple, I mean it. The AutoQoS for VoIP command is
entered while in config-if mode on any router interface you choose. The command to kick
off the AutoQoS for VoIP process is auto qos voip. There is one optional keyword that
can follow this command, trust. This keyword tells the router to trust the DSCP values
that have been already marked on incoming packets. If the trust keyword is not used, the
router uses NBAR and marks (or re-marks) packets. If you trust your endpoints and their
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476 Chapter 12
Configuring Quality of Service

QoS markings, then you should use trust. If not, then dont include it and instead rely on
NBAR. To show you how to congure AutoQoS for VoIP on a router, we will use
Figure 12.1 as our example network.
FI GURE 12.1 AutoQoS for VoIP on a router
CUCM
Express
S0/0
256 Kbps
Switch Switch
Router-A Router-B
S0/0
We will congure the interface serial 0/0 of Router-A shown in the diagram. To show
you what AutoQoS for VoIP will congure for our particular router, here is the current
conguration on interface s0/0:
Router-A#show run int s0/0
!
interface Serial0/0
bandwidth 256
ip address 192.168.1.1 255.255.255.0
encapsulation ppp
clock rate 2000000
You can see that we have an enabled serial interface that has a set bandwidth of 256
Kbps. Additionally, the interface has an IP address and is set to use PPP as the transport
mechanism.
Now we can congure AutoQoS for VoIP. In our example, we will let the router use
NBAR to mark packets with a DSCP value; we do that by not using the trust keyword in
the auto qos voip command, as shown here:
Router-A#configure terminal
Router-A(config)#interface serial 0/0
Router-A(config-if)#auto qos voip
Router-A(config-if)#end
Router-A#
As soon as this command is entered on the serial 0/0 interface, the router kicks off a
script to determine the best possible QoS conguration for this interface. The number of
actual congurations made is fairly staggering. But dont take my word for it: you can use
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the show auto qos command on Router-A to see exactly what the router chose to congure
as our QoS policy, as shown here:
Router-A#show auto qos
!
policy-map AutoQoS-Policy-UnTrust
class AutoQoS-VoIP-RTP-UnTrust
priority percent 70
set dscp ef
class AutoQoS-VoIP-Control-UnTrust
bandwidth percent 5
set dscp af31
class AutoQoS-VoIP-Remark
set dscp default
class class-default
fair-queue
!
class-map match-any AutoQoS-VoIP-Remark
match ip dscp ef
match ip dscp cs3
match ip dscp af31
!
class-map match-any AutoQoS-VoIP-Control-UnTrust
match access-group name AutoQoS-VoIP-Control
!
class-map match-any AutoQoS-VoIP-RTP-UnTrust
match protocol rtp audio
match access-group name AutoQoS-VoIP-RTCP
!
ip access-list extended AutoQoS-VoIP-RTCP
permit udp any any range 16384 32767
!
ip access-list extended AutoQoS-VoIP-Control
permit tcp any any eq 1720
permit tcp any any range 11000 11999
permit udp any any eq 2427
permit tcp any any eq 2428
permit tcp any any range 2000 2002
!
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478 Chapter 12

rmon event 33333 log trap AutoQoS description AutoQoS SNMP traps for Voice
Drops owner AutoQoS
rmon alarm 33333 cbQosCMDropBitRate.146.12938593 30 absolute rising-threshold
1 33333 falling-threshold 0 owner AutoQoS
Serial0/0 -
!
interface Serial0/0
no ip address
encapsulation ppp
no fair-queue
ppp multilink
ppp multilink group 2001100115
!
interface Multilink2001100115
bandwidth 256
ip address 192.168.1.1 255.255.255.0
ppp multilink
ppp multilink interleave
ppp multilink group 2001100115
ppp multilink fragment delay 10
service-policy output AutoQoS-Policy-UnTrust
ip rtp header-compression iphc-format
Router-A#
Our QoS conguration on Router-A now consists of the following:

Policy maps and class maps to classify and mark our voice and voice-related trafc

Access lists for VoIP RTCP and control packets

RMON settings to trigger alerts via SNMP based on thresholds

A PPP multilink conguration on the serial 0/0 interface along with LFI congurations

Compressed RTP conguration on the serial 0/0 interface
The PPP multilink LFI configuration in the previous example was configured
because the router discovered that serial 0/0 had a bandwidth below 768
Kbps. As mentioned in Chapter 11, any WAN interface that is below 768 Kbps
should be configured with link-efficiency techniques such as LFI and cRTP.
We will discuss PPP multilink in more detail later in this chapter.
You can see that AutoQoS for VoIP congured four different classes with which to mark
packets and enforce policy. Specically, our voice (RTP) trafc will be tagged as DSCP
EF and will have a strict priority queue of 70 percent of the total bandwidth of our 256
Kbps serial interface.
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If we went ahead and performed AutoQoS on Router-B, the same QoS settings would
be implemented there as well, and we would have a completely uniform QoS structure on
either side of our WAN connection.
Now that youve seen how AutoQoS for VoIP works on routers, lets take a look at how
to congure switches.
Configuring AutoQoS for VoIP on a Switch
There are only three options for conguring AutoQoS on a switch interface using the auto
qos voip command. Once you understand these three options, conguring QoS on your
Cisco Catalyst will be a snap! Here is the output of the switch when conguring AutoQoS:
Switch(config-if)#auto qos voip ?
cisco-phone Trust the QoS marking of Cisco IP Phone
cisco-softphone Trust the QoS marking of Cisco IP SoftPhone
trust Trust the DSCP/CoS marking
Lets look at the options to understand when each one should be used:
cisco-phone You should use this option when you want to trust the QoS markings from
your Cisco phone. Note that I said Cisco phone and not IP phone. Cisco uses CDP
between the switch and phone to ensure that the device is indeed a phone and not some
other device attempting to get a better classication for its trafc. Because CDP is Cisco
proprietary, it works only when Cisco IP phones are connected or when other companies
license CDP technology (such as Mitel IP Phones). You must also make sure that CDP is
enabled both globally and on the access port your Cisco phone is connected to.
cisco-softphone This option is very similar to the cisco-phone option, except that
it trusts the CoS/ToS markings on PCs that are running the Cisco IP Communicator
software. The IP Communicator software runs CDP once again to ensure that the device is
properly identied as a Cisco phone.
trust The trust option basically means that the switch will trust any CoS/ToS value
received and treat the trafc accordingly. Be cautious when conguring this on access
ports, because people in the know can manipulate the classication markings of data
trafc on their PCs and have their data sent as priority trafc when it should be treated as
normal trafc. But where the trust option should absolutely be used is between all of the
switch and router interfaces that interconnect your network equipment. As soon as you set
a location for your trust boundary, all other devices within that boundary can safely trust
the CoS/ToS markings they receive.
Thats all there is to AutoQoS for VoIP on a switchport. Lets use Figure 12.2 as our
network example for conguring QoS on a production network. Assume that the Sales,
Marketing, Management, and Voice VLANs are precongured on the network. Switchport
Fa0/5 is congured to use VLAN 10 for data and VLAN 100 for voice trafc. Also assume
that 802.1Q trunking is congured between the switch and the CUCM Express router.
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480 Chapter 12

FI GURE 12. 2 AutoQoS for VoIP on a switch
CUCM
Express
Fa0/1
802.1Q
Trunk
Trust boundary
Fa0/1 Fa0/5 Fa0/5
Switch-A Switch-B
We need to set our trust boundary. Lets assume that well trust the Cisco phones but
not trust ordinary PCs. Therefore, our trust boundary is set at the phone, using the auto
qos voip cicso-phone command. But rst, lets view the current conguration of our
interface using the show run interface fastEthernet 0/5 command, as shown here:
Switch-A#sh run int fa0/5
Building configuration...
Current configuration : 487 bytes
!
interface FastEthernet0/5
switchport access vlan 10
switchport mode dynamic desirable
switchport voice vlan 100
Now we will go into interface-conguration mode and congure AutoQoS to trust
Cisco phones attached to Fa0/5.
Switch-A(config)#interface fastEthernet 0/5
Switch-A(config-if)#auto qos voip cisco-phone
Lets see exactly what AutoQoS has congured on our port, using the show run interface
fastEthernet 0/5 command a second time:
Switch-A#sh run int fa0/5
!
switchport access vlan 10
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switchport mode dynamic desirable
switchport voice vlan 100
mls qos trust device cisco-phone
mls qos trust cos
auto qos voip cisco-phone
wrr-queue bandwidth 10 20 70 1
wrr-queue min-reserve 1 5
wrr-queue cos-map 1 0 1
wrr-queue cos-map 3 3 6 7
wrr-queue cos-map 4 5
priority-queue out
spanning-tree portfast
You can see that the auto qos voip command actually congured all kinds of things on
the interface, including trust settings, weighted round-robin (WRR) queuing policies, and
priority queuing specically for VoIP trafc. The important thing you need to identify is
that we are trusting the Cisco phone with the auto qos voip cisco-phone entry.
Once the trust boundary is set, we know that the interfaces connecting our Layer 2
switch to the CME router should be congured using the auto qos voip trust command.
To show you what is actually congured using the AutoQoS command, we will do a show
run interface fa0/1 command to view the initial conguration settings:
Switch-A#show run interface fa0/1
!
switchport trunk encapsulation dot1q
switchport trunk allowed vlan 10,20,100
switchport mode trunk
Now we will congure AutoQoS to trust markings passing through Fa0/1:
Switch-A(config)#interface fastEthernet 0/1
Switch-A(config-if)#auto qos voip trust
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You can now view our new running conguration for our switch uplink to see the
differences between the auto qos voip trust congurations and the auto qos voip
cisco-phone output:
Switch-A#show run interface fa0/1
!
switchport trunk encapsulation dot1q
switchport trunk allowed vlan 10,20,100
switchport mode trunk
mls qos trust cos
auto qos voip trust
wrr-queue bandwidth 10 20 70 1
wrr-queue cos-map 3 3 6 7
wrr-queue cos-map 4 5
priority-queue out
Notice that from a QoS conguration standpoint, the only difference between the trust
and cisco-phone conguration is the auto qos voip trust command.
After conguring AutoQoS for VoIP on the switch, you will also nd a couple of global
congurations in the running conguration shown here:
Switch-A#show run | include mls qos
mls qos map cos-dscp 0 8 16 26 32 46 48 56
mls qos
The mls qos command is what enables QoS on our switch. The mls qos map cos-dscp
command followed by DSCP numbers instructs the switch to map Layer 2 CoS values that
are read by the switch coming from the Cisco phone. The switch will adjust DSCP markings
according to the CoS values. Remember that CoS uses eight different markings. A CoS of
0 will be mapped to a DSCP value of 0, while a CoS of 5 (which is what voice frames are
tagged with) will use a DSCP value of 46, which corresponds to an AF PHB class of EF.
We will discuss CoS-to-DSCP mappings and how to modify them later in this chapter.
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The conguration of the opposite-end switch in our example diagram should be
identical. Once youve completed conguring all the interfaces, congratulations; youve
successfully implemented QoS for voice on your switch-based LAN.
Configuring AutoQoS for the Enterprise
on a Router
If your network requires a more generalized QoS approach that classies and marks
trafc other than simply voice, the AutoQoS for the Enterprise method might be the
right choice for you. In addition to identifying, classifying, and queuing voice trafc with
LLQ, AutoQoS for the Enterprise classies other forms of application trafc into possibly
10 different queues. You might remember from Chapter 11 that the Cisco QoS baseline
model has 11 different classes. So why does AutoQoS for the Enterprise have only up to 10
queues? The critical classication is the missing class that AutoQoS for the Enterprise
does not attempt to use. This is because the tool has no way of knowing what your
particular business or organization deems as critical application ows. If you wish to use
the critical classication and markings, youll have to go back and congure it manually
by specifying what trafc you deem to be critical. Keep in mind that AutoQoS for the
Enterprise will not use this class by default. The other 10 classes are available and may or
may not be used depending on what type of trafc the router sees and classies according
to Cisco best-practice policies.
AutoQoS for the Enterprise will also identify low-speed WAN links and congure
compression and link-efciency techniques on interfaces that are less than 768 Kbps
(just like AutoQoS for VoIP).
Unlike the AutoQoS for VoIP method, AutoQoS for the Enterprise is a two-phased
approach:
1. The AutoQoS autodiscovery phase is started when a router monitors trafc passing
through a specic interface. The router monitors its local interfaces and collects
baseline information about the data ows it sees, and attempts to classify them into
one of 10 possible classes. Either discovery can either be made using NBAR, or the
router can be set to trust the DSCP markings of packets and classify them based on the
markings that packets currently have.
2. Once the AutoQoS autodiscovery phase has had sufcient time to collect data and
classify it, AutoQoS for the Enterprise automatically creates QoS templates for
classication, marking, queuing, and link efciency. A network administrator should
review the templates, and enable them during the AutoQos installation phase.
Lets rst go through the AutoQoS autodiscovery phase and then see how we can verify
and implement the recommended settings in the AutoQoS installation phase. We will use
the network shown in Figure 12.3 as our example network and congure Router-A with
AutoQoS for the Enterprise.
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FI GURE 12. 3 AutoQoS for the Enterprise
IP phones
IP phones
Switch Switch
Router-A
S0/0
S0/0
1540
Kbps
Router-B
CUCM
Desktop
computers
Desktop
computers
Application
servers
Configuring the AutoQoS Discovery Phase
As you can see from Figure 12.3, we have a fairly large network with multiple application
servers that we would like to be able to categorize, mark, and queue accordingly. We will
focus on the Router-A conguration, but keep in mind that Router-B must go through the
same process.
In the AutoQoS discovery phase we will enable the discovery of trafc on our serial
0/0 interface by using the auto discovery qos command. This command will use NBAR
to discover and classify trafc. Any trafc that NBAR does not have in its database will
be placed into the best-effort queue. If we wanted to use and trust DSCP markings that
packets may already be congured with, we would use the auto discovery qos trust
command. This would disable NBAR discovery and solely rely on DSCP marking for
classication. In our example, we will use NBAR for classication, as shown here:
Router-A(config-if)#auto discovery qos
Router-A#
When you turn on autodiscovery, it runs in the background while the interface operates
normally, so you should not be concerned about it interfering with trafc ows. Youll
notice in the interface conguration that there is an entry that has auto discovery qos
enabled, as shown here:
Router-A#show run interface s0/0
interface Serial0/0
bandwidth 1540
ip address 192.168.1.1 255.255.255.0
encapsulation ppp
auto discovery qos
clock rate 2000000
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The duration of the AutoQoS autodiscovery phase is completely up to your discretion,
although it is highly recommended that you run it for several days. In many situations, a
complete weeks worth of data discovery is highly recommended in case you have some
applications that only run during specic times of the day or days of the week.
While the autodiscovery phase is going on, you can use the show auto discovery qos
command while in privileged EXEC mode to see what trafc the router has identied and
the suggested conguration policy commands according to Cisco best practices, as shown
in this example:
Router-A#show auto discovery qos
Serial0/0
AutoQoS Discovery enabled for applications
Discovery up time: 1 hours, 12 minutes
AutoQoS Class information:
Class Voice:
Recommended Minimum Bandwidth: 519 Kbps/52% (PeakRate).
Detected applications and data:
Application/ AverageRate PeakRate Total
Protocol (kbps/%) (kbps/%) (bytes)
----------- ----------- -------- ------------
rtp audio 3/<1 517/52 703323
Class Interactive Video:
No data found.
Class Control:
Recommended Minimum Bandwidth: 0 Kbps/0% (AverageRate).
----------- ----------- -------- ------------
h323 0/0 75/7 30212
rtcp 0/0 7/<1 1540
Class Streaming Video:
No data found.
Class Transactional:
No data found.
Class Bulk:
----------- ----------- -------- ------------
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ftp 0/0 330/31 74480
Class Scavenger:
No data found.
Class Management:
----------- ----------- -------- ------------
dhcp 0/0 84/8 115543
ldap 0/0 169/16 55434
Class Routing:
----------- ----------- -------- ------------
icmp 0/0 2/<1 300
Class Best Effort:
Current Bandwidth Estimation: 355 Kbps/34% (AverageRate).
----------- ----------- -------- ------------
unknowns 336/32 99650/97 949276
http 14/1 15557/15 41545
Suggested AutoQoS Policy based on a discovery uptime of 1 hours, 12 minutes:
!
class-map match-any AutoQoS-Voice
match protocol rtp audio
!
class-map match-any AutoQoS-Signaling
match protocol sip
match protocol rtcp
!
class-map match-any AutoQoS-Bulk
match protocol exchange
policy-map AutoQoS-Policy
class AutoQoS-Voice
priority percent 1
set dscp ef
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class AutoQoS-Signaling
bandwidth remaining percent 1
set dscp cs3
class AutoQoS-Bulk
bandwidth remaining percent 1
random-detect dscp-based
set dscp af11
class class-default
fair-queue
Router-A#
From this output, you can see that the discovery process has identied several
different data ows using NBAR and classied them into one of the 10 possible classes
according to Cisco best practices. Notice that not all 10 classes are used in the example.
Your specic network may use all or only a few of the possible classes. Additionally,
at the end of the command, you can see the recommended conguration settings the router
will implement during phase 2 of the AutoQoS for the Enterprise process. This gives
you an opportunity to review the settings to make sure they perform the QoS functions
you desire.
Dont Forget to Rediscover Your Network
Jeremy is a network consultant working on a WAN project to upgrade a companys
fractional 512 Kbps PRI data connections to full 1.536 Kbps PRI speeds. The process
would help alleviate some of the congestion that the 512 Kbps circuits were previously
experiencing.
The upgrade to the full PRI speeds went smoothly, but Jeremy noticed that the AutoQoS
policies did not automatically update. All of the cRTP compression and LFI congurations
remained on the conguration. This was an important lesson for Jeremy to learn. Now he
knows that when bandwidth upgrades occur, the AutoQoS for the Enterprise discovery
process must be completely redone so the process can suggest the optimal QoS
conguration policies for the newly upgraded network. After running the AutoQoS process
for a week as recommended, Jeremy found that the AutoQoS process correctly saw the
WAN interface as a full PRI and removed the cRTP and LFI settings, which are recommended
only for slower links.
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Configuring AutoQoS Implementation Phase
After you have completed the discovery phase and reviewed the conguration that
AutoQoS for the Enterprise has recommended specically for our network, its time to
apply those changes. In our example, it is simply a matter of going back into the serial 0/0
interface and issuing the auto qos command, as shown here:
Router-A(config-if)#auto qos
Router-A#
This command applies all of the QoS classication, marking, and queuing
congurations. In addition, it will congure any link-efciency techniques for low-speed
WAN interfaces when necessary.
After you complete the implementation phase, you should go ahead and
disable the AutoQoS discovery process on any interfaces it is currently enabled
on. To do this, simply go into config-if configuration mode and issue the no
auto discovery qos command. This will end the data-collection process and
delete all data-collection information on the router for that interface.
To review the complete list of QoS settings implemented on a router, you can issue the
same show auto qos command as we did with AutoQoS for VoIP. This command again
shows all of the class maps, policy maps, service policies, and link-efciency settings on the
local router. If you want to go in and manually modify or add any QoS policies, you can do
so using the standard CLI method.
In the next section, we will step away from the fully automated QoS policy-creation
method and look at a more robust, yet fully structured conguration tool called Modular
QoS CLI (MQC).
Configuring QoS Policies Using MQC
Youve seen the power of AutoQoS and how it can provide sound QoS policies for most
networks. But sometimes network administrators want to control the conguration
process completely. If this is the case, they typically will choose to use the Modular QoS
CLI (MQC) method. This is also known as class-based (CB) QoS. Although there are
more conguration steps involved, MQC lets administrators decide for themselves how
data ows should be classied and marked, and what queuing policies to use. MQC uses a
structured three-step command process, as shown here:
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class-map Class maps are used to congure different trafc class types. This is your
matching function. Keep in mind that a packet will be matched by only one class.
The matching order is set by the appearance of classes in the policy map. Therefore, it is
important to congure your most specic matching statements rst.
policy-map Policy maps are used to associate a trafc class type with one or more QoS
operations. This is your set or do function.
service-policy A service policy is used to apply a policy to router interfaces, including
subinterfaces and virtual circuits. The service policies can be applied inbound or outbound
depending on the need.
The beauty of MQC lies in its highly scalable and modular nature. Classications can be
used within multiple policies and applied to multiple interfaces. Figure 12.4 shows how one
or more class maps can be placed in a policy map. Then that policy map is applied to an
inbound or outbound (or both) interface using the service-policy command.
FI GURE 12. 4 MQC structure
Class 1
Class 2
Class 3
class-map policy-map
Policy
service-policy
Interface
Class-map match statements can also match against other class-maps.
That is, you can have nested class maps. Nested class maps and policy
maps are fairly common and are configured similarly. An example of a
nested policy map can be found later in this chapter.
Now that you have a solid foundation in the structure of MQC, we will try out the
three-step conguration process, using Figure 12.5 as our example network.
FI GURE 12. 5 MQC network example
IP WAN
Router-A
S0/0
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Configuring Class Maps
The rst step of any QoS policy is to segment various types of trafc into classes. Using
MQC, this process is kicked off by using the class-map command while in global
conguration mode. You are also required to assign the new class a unique name to
identify it. Additionally, two optional keywords can be entered when creating a new
class map:
match-any (default) This command states that any single match statement contained
within the class map can be used to match a packet to that class. If you have multiple
match statements congured within the class, your packet needs to meet only one of the
requirements.
match-all This command states that a packet must adhere to all match statements
contained within the class map. Therefore if you have three match statements in a
match-all class, your packet must meet all three distinctions to be part of that class.
Again, note that the match-any and match-all keywords are optional. If the class-map
command simply has a unique identier in the statement and no keyword, that class map is
a match-any class map.
In our rst example, we will congure a class map named class-1, where we want to
trigger on any one match statement:
Router(config)#class-map class-1
Router(config-cmap)#
Notice that we did not include a match-any or match-all keyword. This means that this
class map is a match-any map. Also note that we are placed into config-cmap mode. This is
where we can congure our match statements.
The process of conguring match statements is straightforward. You use the match
command followed by any of the following match options listed in Table 12.1. Keep in
mind that the available options vary depending on the hardware type and version of IOS
you are running:
TABLE 12.1 Class-map match statement options
Option Description
access-group Match against an access control list (ACL) configured on
the local router.
any Match any packet.
class-map Match against another class map.
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Option Description
cos Match against the packets COS value.
destination-address mac Match against the packets destination
MAC address.
discard-class Match against the packets discard class (within DSCP).
dscp Match against the packets DSCP value for both IPv4 and
IPv6 packets using either DSCP values (1-63) or PHB values.
flow Match against the packets flow QoS parameters.
fr-de Match against the Frame Relay discard eligibility (DE) bit.
fr-dlci Match against the Frame Relay DLCI identifier.
input-interface Match against the input interface the packet came into the
router on.
ip Match against specific IP values such as RTP, IPv4 DSCP,
and IPv4 precedence.
mpls Match against specific multi-protocol label-switching
values within the packet.
not Negate a match statement.
packet Match against packet size (for v4 or v6).
precedence Match against the packets IP Precedence value for both
IPv4 and IPv6 packets.
protocol Match against the packets protocol with NBAR.
qos-group Match against a preconfigured QoS group.
source-address mac Match against the packets source MAC address.
vlan Match against the VLAN assignment.
In our rst example, we will create a class map named voice and use it to classify any
trafc marked with either an IP Precedence of 5 or a DSCP of 46:
Router(config)#class-map voice
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Router(config-cmap)#match precedence 5
Router(config-cmap)#match dscp 46
Router(config-cmap)#end
Router#
As a result, any packet that crosses an interface where this class map is applied will be
put into the voice class.
Lets congure a second class map that uses an access-list as well as DSCP
PHB values CS6 and CS2 for our network control ows. First well congure an ACL
to dene what IP addresses to match against. Then we will create our new match-all
class map named network and use the appropriate match statements to classify
our trafc:
Router(config)#access-list 10 permit 10.0.1.0 0.0.0.255
Router(config)#class-map match-all network
Router(config-cmap)#match access-group 10
Router(config-cmap)#match dscp cs2 cs6
Router#
Notice that you can match multiple values for DSCP (up to eight) on a
single line. Similarly, you can use the precedence command to match one
of eight IP precedence values on a single match statement.
Lastly, you will see how to use class maps to classify trafc using NBAR. In this
example, we will create a new match-any class map named web and use NBAR to identify
HTTP and HTTPS trafc:
Router(config)#class-map web
Router(config-cmap)#match protocol http
Router(config-cmap)#match protocol secure-http
Router#
So as you can see, there are multiple ways to classify trafc, using all sorts of
differentiation techniques, including ACLs, IP precedence/DSCP values, and NBAR.
Now that we have classied some trafc, lets see how we can apply QoS policies using the
policy-map command.
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Configuring Policy Maps
To apply various QoS policies to classes we have created, we use the policy-map global
conguration command followed by a unique name for the policy. We will then be placed
into config-pmap conguration mode. Here, we can identify a class map by its unique name.
We will then be put into config-pmap-c conguration mode. It is within this conguration
mode that QoS policies are applied. Table 12.2 lists all of the QoS policy options available,
with their descriptions. Again, the options will vary based on the hardware and IOS version
you are running.
TABLE 12. 2 QoS policy-map options
Option Description
bandwidth Sets the CBWFQ bandwidth by Kbps, percent of link, or percent of
remaining bandwidth on the link.
compression Enables compression on all packets.
drop Drops all packets.
log Logs IPv4 and ARP packets.
netflow-sampler Performs NetFlow actions.
no Negates an option.
police Enables traffic-policing policies.
priority Sets strict priority queuing either by Kbps or bandwidth percent of
the link.
queue-limit Set the maximum tail-drop threshold.
random-detect Enables Random Early Detection (RED) or Weighted Random Early
Detection (WRED) as the drop policy.
service-policy Used to configure nested policy maps.
set Sets QoS values such as IP Precedence, COS and DSCP values, DE
bits, and discard class values.
shape Enables traffic-shaping policies.
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We will use our three previously congured class maps and apply them to a single policy
map in our example. We will call the policy policyWAN and apply specic trafc condition
settings for each class, as shown here:
Router(config)#policy-map policyWAN
Router(config-pmap)#class voice
Router(config-pmap-c)#priority percent 70
Router(config-pmap-c)#exit
Router(config-pmap)#class network
Router(config-pmap-c)#bandwidth percent 5
Router(config-pmap-c)#random-detect dscp-based
Router(config-pmap)#class web
Router(config-pmap-c)#set dscp 0
Router(config-pmap-c)#end
Router#
In this example you can see that we gave our voice trafc a strict-priority queue of 70
percent of the total link bandwidth. Next, we chose to use CBWFQ (5 percent of the total
bandwidth), and we also turned on WRED. Finally, our web trafc is simply handed a
DSCP value of 0. Next, you will learn how to apply our policies to interfaces using the
service-policy interface conguration command.
The Parent and Child Relationship
Nathan was working on conguring classes and policies using MQC. He wanted an
elegant way to apply CBWFQ on a wide range of trafc (a class named big) and
enable trafc policing on a subset of that class (a class named small). After looking over the
MQC policy-map options, Nathan found the service-policy command option with
the policy-map that allows for nested classes. Therefore, Nathan could create two policy
maps: one parent map that could set the priority for all trafc and a child map that would
use the parent priority settings but also enable trafc policing, as shown in this example:
Router(config)#policy-map child
Router(config-pmap)#class big
Router(config-pmap-c)#priority 30
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Applying Policy Maps to Interfaces with a Service Policy
The nal step in the MQC process, after we have dened trafc in classes and congured
QoS policies, is to apply those policies to an interface. To do this, we navigate to config-if
conguration mode of the interface we choose to apply our policy on, and then issue the
service-policy command. We must then choose to specify whether we want the policy to
apply for trafc coming into or out of the interface, using the input or output keywords.
Keep in mind that an interface can have policies applied both inbound and outbound if you
choose, but anything that involves the use of queues can only be applied outbound. In our
example network we apply our service policy outbound on interface s0/0:
Router(config)#interface s0/0
Router(config-if)#service-policy output policyWAN
Router#
Thats all there is to MQC! To recap, you must perform the following MQC steps:
1. Congure a class map.
2. Congure a policy map.
3. Apply a policy map to an interface (input or output) using a service policy.
Next youll learn how to verify our QoS congurations and policy mechanisms using
various show commands.
MQC QoS Configuration Show Commands
In this section, well cover three useful show commands to verify a class-based QoS
policy that was congured using MQC. The last of these commands, show policy-map
interface, can be used to verify packet matching and queue usage on interfaces congured
with QoS policies.
Router(config-pmap)#exit
Router(config)#policy-map parent
Router(config-pmap)#class small
Router(config-pmap-c)#police 256000 conform-action transmit exceed-action drop
Router(config-pmap-c-police)#exit
Router(config-pmap-c)#service-policy child
Router#
Now Nathan has a single policy map named parent that polices all trafc and at the same
time applies PQ to a subset policy map named child.
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show class-map Use this command to quickly view all of the class maps congured on
your local router. Notice that the DSP entries list both the PHB values and DSCP values
together, as shown in this example:
Router#show class-map
Class Map match-any class-default (id 0)
Match any
Class Map match-all web (id 3)
Match protocol http
Match protocol secure-http
Class Map match-all voice (id 1)
Match precedence 5
Match dscp ef (46)
Class Map match-all network (id 2)
Match access-group 10
Match dscp cs2 (16) cs6 (48)
show policy-map This command is useful when you want to review all policy-map
congurations on a router, as shown here:
Router#show policy-map
Policy Map policyWAN
Class voice
Strict Priority
Bandwidth 70 (%)
Class network
Bandwidth 5 (%) Max Threshold 64 (packets)
Class web
Set dscp default
show policy-map interface Lastly, the show policy-map interface command, followed
by an interface that has a service policy applied to it, will show which policy and class
maps are in force, along with packet-matching statistics for trafc that has passed through
the interface. This is a very useful command to verify that your policies are operating
correctly, as demonstrated in the following example output:
Router#show policy-map interface s0/0
Serial0/0
Service-policy output: policyWAN
Class-map: voice (match-all)
0 packets, 0 bytes
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5 minute offered rate 0 bps, drop rate 0 bps
Match: precedence 5
Match: dscp ef (46)
Queueing
Strict Priority
Output Queue: Conversation 264
Bandwidth 70 (%)
Bandwidth 31500 (kbps) Burst 787500 (Bytes)
(pkts matched/bytes matched) 0/0
(total drops/bytes drops) 0/0
Class-map: network (match-all)
0 packets, 0 bytes
Match: access-group 10
Match: dscp cs2 (16) cs6 (48)
Queueing
Output Queue: Conversation 265
Bandwidth 5 (%)
Bandwidth 2250 (kbps)Max Threshold 64 (packets)
(pkts matched/bytes matched) 0/0
(depth/total drops/no-buffer drops) 0/0/0
Class-map: web (match-all)
0 packets, 0 bytes
5 minute offered rate 0 bps
Match: protocol http
Match: protocol secure-http
QoS Set
dscp default
Packets marked 0
Class-map: class-default (match-any)
25 packets, 600 bytes
Match: any
Router#
Notice that at the end of this command is a map identied as class-default. While we did
not congure this class map, it is always included anytime you apply a policy to an interface.
This class map is used for all other trafc that does not match a congured class map.
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Now that you understand how to congure QoS using MQC, we will take a look at how
to congure RED and WRED, which as you learned in Chapter 11 are congestion-avoidance
techniques for bottlenecked links.
Configuring Congestion-Avoidance
Techniques
Random Early Detect (RED) is a congestion-avoidance technique that is set into motion
when queues begin to ll up, and packets need to be discarded on bottleneck interfaces. Before
the queue buffer completely lls and sensitive trafc begins to suffer from TCP synchronization,
you can use RED to drop random packets in an attempt to have TCP applications back off
their current TCP window rates and send packets to their destination more slowly. TCP
synchronization often occurs when queue buffers ll up and begin dropping packets, because
the multiple TCP sessions will all back off their TCP window size at the same time.
When this happens, youll see a seesaw effect in the interface trafc. All of the TCP
transmissions slow down simultaneously and then begin speeding up simultaneously, and
ultimately buffers overow over and over again. To avoid this, RED will randomly drop
packets in an attempt to have only some TCP sessions adjust their windowing downward.
When it does this, youll see a much more even transport ow across an interface.
Weighted RED, or simply WRED, is Ciscos proprietary advancement of RED to make
the dropping of packets a little less random by selecting packets that are marked lower
than others. WRED can be congured directly on the interface, but we will use class-based
WRED (CB-WRED) for our examples. Also keep in mind that CBWFQ must be congured
in conjunction with WRED within a policy map.
We can congure WRED to look for either IP Precedence or DSCP markings, and we can
determine the probability of dropped packets based on these values that we can classify.
To do this, we must have our class maps dened and create a policy map. Within the policy
map we reference the class map of trafc we wish to apply WRED on. We can then issue the
random-detect command. From here we have some decisions to make. First, do we want
WRED to used IP Precedence or DSCP markings? If we want to use IP Precedence, we enter
the prec-based keyword. If we wish to use DSCP values, we use the dscp-based keyword.
For example, lets assume weve created a class map named tcp-traffic that classies a
group of TCP data packets owing through an interface. We will create a policy map for
this class and give it a CBWFQ percent of 10. Additionally, we will congure WRED to use
DSCP markings, as shown here:
Router(config-pmap)#class tcp-traffic
Router(config-pmap-c)#random-detect dscp-based
Router(config-pmap-c)#
At this point, we are ready to tell WRED manually how we want our packets to be
dropped. To do this, we again use the random-detect command followed by dscp
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Configuring Congestion-Avoidance Techniques 499
(or precedence if using IP Precedence). We then need to congure the following four
WRED settings:
DSCP or IP Precedence Value This is the marking that you want to specify for WRED.
Minimum Threshold (Number Of Packets) This sets the number of packets that must
be in the queue before the router starts to discard packets with a specic DSCP or IP
Precedence marking.
Maximum Threshold (Number Of Packets) This sets the maximum number of packets
that can be in the queue before all other packets are dropped.
Mark Probability Denominator This is a proportion of the number of packets that are
dropped when the queue maximum threshold is reached (but not yet surpassed).
This can be really confusing, so we will use a visual diagram depicting the different
WRED thresholds, shown in Figure 12.6.
FI GURE 12. 6 WRED packet drop procedure
100% packet drop
Packets go from a
portion being dropped to
100% dropped after the
maximum threshold is
reached.
Mark probability
denominator
0% packet drop
Queue size
Minimum
threshold
Maximum
threshold
P
a
c
k
e
t

d
r
o
p

p
r
o
b
a
b
i
l
i
t
y
As you can see from the diagram, all packets are sent to a queue until the minimum
threshold is reached. When the number of packets in a queue is between the minimum and
maximum, some packets are queued and others are dropped. The number of dropped
versus queued packets increases until it reaches the maximum threshold. Once the number
of packets exceeds the maximum threshold, you see that the portion of dropped packets
becomes 100 percent, and no packets are queued until the number of queued packets drops
below the maximum threshold.
So to show you an example of this, we will complete our conguration process by dening
minimum, maximum, and mark-probability numbers for the three DSCP values shown here:
Router(config-pmap-c)#random-detect dscp af13 25 100 4
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500 Chapter 12

Looking at DSCP value AF11, you see that we have a minimum threshold of 35 packets.
Once a queue reaches 25 packets, WRED will begin dropping packets marked with a value
of AF13. WRED will continue dropping packets more proportionally up until the queue
reaches 100 packets. Right at this point, when there are 100 packets waiting in the queue,
the probability of an AF13 packet being dropped is 1 in 4, because the mark probability
indicator is set to 4. That means that 25 percent of AF11 packets will be dropped when
there are 100 packets waiting in the queue. If the queue were to exceed 100 packets,
WRED would begin dropping all packets. The same is true for the congurations for AF12
and AF11 as well, although their minimum thresholds are different.
Next, lets look at how we can congure class-based trafc policing and shaping to help
condition and control trafc rates.
Configuring Class-Based Traffic
Policing and Shaping
QoS techniques such as priority queuing (PQ) and class-based weighted fair queuing
(CBWFQ) focus on setting minimum-bandwidth requirements for certain types of trafc.
While this is important, sometimes it is just as important to set maximum-bandwidth
levels for trafc ows as well. This is especially critical when dealing with trafc that can
handle dropped packets and still function. On Cisco routers, there are two methods for
conguring maximum levels of trafc, known as trafc policing and trafc shaping. We
covered the basics of these two methods and their differences in Chapter 11.
In this section we will rst examine the token bucket mechanism that both trafc
policing and shaping use. Next well show how to congure trafc policing, which is the
stricter of the two options. Last, we will go on to congure trafc shaping, which utilizes
queues to slow down TCP ows.
Understanding Token Buckets
Trafc policing and shaping use the concept of token buckets to manage trafc. The token
bucket is used to regulate data in a ow to manage overall bandwidth of an interface. The
token bucket mean rate equation denes the average (mean) rate of transfer for trafc on a
specic interface. The three components for determining token bucket rates are as follows:
Committed Information Rate (CIR) This is the average amount (in bps) of data that can
be forwarded out of an interface. The mean rate for policing and shaping is commonly
represented in bps.
Burst Conforming Size (Bc) This is a number in bits that tells us how much trafc can be
sent on a token within a specic time interval without disrupting other trafc.
Time Interval (Tc) This number tells us the time in seconds allowed for a burst of trafc.
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The equation to determine the mean rate is:
CIR Bc / Tc
Alternatively, this equation could be reworked to find either Tc or Bc if the CIR
is known. For example, to find the time interval, we would use the following
equation:
Tc Bc / CIR
For example, lets say we have 16000 bits worth of tokens of burst trafc (Bc) moving in
and out of the token bucket every 250 ms. We would see a CIR of the following:
CIR 16000 / 0.25
CIR 64000 bps or 64 Kbps
Using a trafc-policing example, the maximum trafc rate for this interface cannot go
over 64 Kbps or it will either be dropped or discarded. Policed trafc that is at or below
this rate is our burst-conforming (Bc) trafc. Any data ows that go above the CIR are
called burst-exceeding (Be) trafc.
Lets take a closer look at the token bucket itself. The bucket always has a xed capacity
size. If the bucket completely lls up, all new tokens are dropped. Tokens are metaphors
used to describe a routers permission for a source device to send a specied number of
bits out to the network. There must be enough tokens to send a complete IP packet. These
tokens move in and out of the token bucket at a rate specied by the router. If the bucket
does not contain enough tokens to send a packet, one of two things happens, depending on
whether you are using trafc policing or trafc shaping:
Not Enough Tokens: Traffic Policing When there are not enough tokens to send a packet
with trafc policing, the packet is either dropped or marked with a new DSCP or IP
Precedence value and then forwarded on.
Not Enough Tokens: Traffic Shaping When there are not enough tokens to send a packet
with trafc shaping, the packet is placed into a queue and waits until enough free tokens
are available.
Lets next examine the different trafc-policing bucket types and how to congure them.
Understanding Traffic-Policing Token Buckets
Trafc policing can use a single bucket, dual bucket with a single rate, or dual bucket with
dual rates.
The single bucket has only the Bc bucket to work with. Figure 12.7 shows the single-
bucket method.
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502 Chapter 12

FI GURE 12. 7 A single token bucket
Exceed-action
performed on
these tokens
Conform-action
performed on
these tokens
Bc
Exceeding
the CIR
Token
ow (CIR)
As you can see from Figure 12.7, there are two possible outcomes for single-bucket trafc:
Conform-action If the number of bytes in the packet is equal to or less than the number
of token bytes available in the bucket, the trafc conforms to the rules. The bucket
tokens are used to remove the conforming data from the bucket, and specic actions are
performed on that data that meet the conform rules. One of these actions is commonly a
forward action, so trafc can move to the next hop to their destination.
Exceed-action If the number of bytes in the packet is greater than the number of token bytes
available in the bucket, the trafc exceeds the rule limits. The exceeding data remains in the
token bucket, and specic actions are performed on the data that exceed conform rules.
In a dual-bucket, single-rate model, there is a burst conform (Bc) bucket and a burst
exceed (Be) bucket, as shown in Figure 12.8. The Bc bucket is used rst, and any exceeding
trafc that ts into the Be bucket will be categorized and handled differently. If the
exceeding trafc does not t into the Be bucket, it is considered to be in violation.
FI GURE 12. 8 Dual-bucket, single-rate model
Conform-action
performed on
these tokens
Exceed-action
performed on
these tokens
Bc
Be
Exceeding
the CIR
Token
ow (CIR)
Violate-action
performed on
these tokens
c12.indd 502 9/21/11 11:28:06 AM
Here are the three specic outcomes of trafc using the dual-bucket, single-rate method:
Conform-action Just as in the single-bucket method, if the number of bytes in the packet
is equal to or less than the number of token bytes available in the Bc bucket, the trafc
conforms to the rules and the conformed trafc is handled accordingly.
Exceed-action If the packet is higher than the CIR and cannot t into the rst Bc bucket,
it exceeds the limit. However, there is a second bucket that it can possibly t into. This is
the Be bucket. If the CIR overow packet data can t inside this bucket, it is classied as an
exceed-action trafc type and specic actions can be performed on this data.
Violate-action If the number of overow bytes in the packet is greater than the number
of token bytes available in the Be bucket, the trafc is considered to be in violation of the
policing rules. The data remains in the bucket and specic actions can be performed on this
data. The default action here is to drop, and please note that not all hardware can modify
the violate-action.
Finally, we have a dual-bucket, dual-rate method, as shown in Figure 12.9.
FI GURE 12. 9 Dual-bucket, dual-rate method
Violate-action
performed on
these tokens
Exceeding
the PIR
Token
ow (CIR)
Token
ow (PIR)
Conform-action
performed on
these tokens
Exceed-action
performed on
these tokens
Exceeding
the CIR
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504 Chapter 12

Youll notice that in this situation, there are two different trafc rates. We have our
standard CIR and a second rate called the peak information rate (PIR). When a packet arrives
at an interface, the PIR bucket is checked rst to see if there are enough tokens available for
the packet. If there are not enough tokens, the packets fall into the violate-action category.
If there are enough PIR tokens available, that is good, but it does not mean the trafc
conforms to the rules yet. The packet is then checked against the CIR bucket to see if there
are enough tokens in this bucket to transmit the trafc. If so, the trafc does indeed meet
the conform-action rules; if not, the trafc is considered to exceed the CIR limits and must
abide by the exceed-action rules.
Here are the possible token outcomes for the dual-bucket, dual-rate method:
Conform-action If the number of bytes in the packet is equal to or less than the number
of token bytes available in the Bc bucket, the trafc conforms to the rules and the
conformed trafc is handled accordingly.
Exceed-action If the packet is higher than the CIR and cannot t into the rst Bc bucket,
it exceeds the limit. However, if there are enough tokens in the PIR bucket, that packet is
considered to have exceeded the conform limit and abides by the exceed-action rules.
Violate-action If there are not enough tokens to handle the packet in the conform or exceed
buckets, that trafc is considered to be in violation and abides by the violate-action rules.
Having two different trafc rates allows for the following benets:

Improved bandwidth manageability

Sustained (non-bursty) excess trafc rates

Preferred rate limiting on network edges for packet conforming and marking
Configuring Class-Based Traffic Policing
With class-based trafc policing, we can manipulate packets and data ows in one of the
two following ways:

Setting a rate that limits (or drops completely) the data ow transmission rate on
packets coming into or out of an interface.

Marking packets using CoS, IP Precedence, and/or DSCP.
Class-based trafc policing conguration occurs during the policy-map stage, after we
have dened a class of trafc we wish to police. We will use the police command followed
by one or several keywords. The best way to understand how to congure trafc policing is
to use an example and then break down each command to see exactly whats going on. In
our rst example, we want to use the following rates and burst sizes:

CIR = 8000

Bc = 2000

Be = 4000
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For any trafc that conforms to the CIR, we will transmit the data. For trafc that falls
into the exceed-action category, we will re-mark it with a DSCP value of 0. Finally, any trafc
that falls into the violate-action category is immediately dropped. Because we are using the
violate-action keyword, we know we are conguring a dual-bucket policing strategy.
For our policing and shaping conguration examples, we will use Figure 12.10 as our
sample network.
FI GURE 12.10 A traffic-policing network example
LAN LAN
Router-A Router-B
S0/0
256 Kbps
We have already dened a class map named police-me on Router-A. Lets create our
policy map named policyPOLICE and police our class map as shown here:
Router-A(config)#policy-map policyPOLICE
Router-A(config-pmap-c)#class police-me
Router-A(config-pmap-c)#police 8000 2000 4000 conform-action transmit exceed-
action set-dscp-transmit 0 violate-action drop
Router-A(config-pmap-c-police)#end
Router-A#
To make this a single-bucket structure, we would simply remove the
violate-action command.
Now that we have our policy map created, we need to apply it to interface s0/0
according to gure 12.10. With trafc policing, we have the ability to apply the policy
for either inbound or outbound trafc. In our particular case, we will choose to apply the
policy to outbound trafc, as shown here:
Router-A(config)#interface s0/0
Router-A(config-if)#service-policy output policyPOLICE
Router-A#
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506 Chapter 12

Next, youll learn how to congure the other rate-limiting technique using trafc-shaping
methods.
Configuring Class-Based Traffic Shaping
Conguring class-based trafc shaping is done in a similar manner as trafc policing on
policy-map congurations of a specic class of trafc. The command used to congure
trafc shaping is shape, followed by one of these keywords:
Router(config-pmap-c)#shape ?
adaptive Enable Traffic Shaping adaptation to BECN
average configure token bucket: CIR (bps) [Bc (bits) [Be (bits)]],
send out Bc only per interval
fecn-adapt Enable Traffic Shaping reflection of FECN as BECN
fr-voice-adapt Enable rate adjustment depending on voice presence
max-buffers Set Maximum Buffer Limit
peak configure token bucket: CIR (bps) [Bc (bits) [Be (bits)]],
send out Bc+Be per interval
While there are several keywords available to choose from, we want to congure generic
trafc shaping (GTS). For GTS, the two commands we are interested in are average and
peak. Notice that the keyword descriptions are using token bucket terms you are already
familiar with, including CIR, Bc, and Be. If we congure the average rate, this limits trafc
to the committed burst (Bc). The tokens are emptied, and a period of inactivity occurs.
Right after that point, Bc + Be can be sent. Peak rate shaping, on the other hand, allows the
router to send trafc up to the committed burst (Bc) as well as up to the excess burst (Be)
rate at every time interval and not only after periods of inactivity.
The trade-off between average and peak is that peak will squeeze a bit more bandwidth
out of the link but at the cost of the possibility of more dropped packets. The equation to
calculate the peak rate in bps is:
Peak_rate CIR [1 (Be / Bc)]
Because of the possibility of dropped packets, conguring trafc shaping using an
average rate is preferred and implemented in most situations.
When conguring using the shape command followed either by average or peak, we will
congure only the average/peak bps for the classied trafc. Optionally, we can manually
set the Bc and Be rates. If we do not manually congure these, the router will set the values
automatically. The router will always choose to set the Bc and Be rates to be the same rate.
For example, if our CIR is 16 Kbps, the router will set Bc and Be to 8 Kbps. Using our
equation we will get:
Peak_rate 16000 [1 (8000 / 8000)]
Peak_rate 16000 [1 1]
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Peak_rate 16000 2
Peak_rate 32000 or 32 Kbps
Lets look at a trafc-shaping conguration example again using Figure 12.10 as our
network.
We have two classes already congured on Router-A. One class is named sensitive and
the other class is named tolerant. The sensitive class will use an average shape of 128
Kbps, while the tolerant class will use a peak shape of 64 Kbps, as shown here:
Router-A(config)#policy-map policySandT
Router-A(config-pmap)#class sensitive
Router-A(config-pmap-c)#shape average 128000
Router-A(config-pmap-c)#exit
Router-A(config-pmap)#class tolerant
Router-A(config-pmap-c)#shape peak 64000
Router-A(config-pmap-c)#exit
Router-A(config-pmap)#exit
Router-A(config)#interface s0/0
Router-A(config-if)#service-policy output policySandT
Router-A#
From this conguration, trafc dened in the sensitive class will transmit to the CIR
at a rate of 128 Kbps, and the tolerant class will peak to a rate of 128 Kbps. How did
we come up with the peak rate number? We let the router determine the Bc and Be rates.
We can use the show policy-map interface interface-name interface-number output
command to see what the router set the Bc and Be rates to:
Router#show policy-map interface serial 0/0 output
Serial0/0
Service-policy output: policySandT
Class-map: sensitive (match-all)
0 packets, 0 bytes
Match: dscp ef (46)
Traffic Shaping
Target/Average Byte Sustain Excess Interval Increment
Rate Limit bits/int bits/int (ms) (bytes)
128000/128000 1984 7936 7936 62 992
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508 Chapter 12

Adapt Queue Packets Bytes Packets Bytes Shaping
Active Depth Delayed Delayed Active
- 0 0 0 0 0 no
Class-map: tolerant (match-all)
0 packets, 0 bytes
Match: dscp default (0)
Traffic Shaping
Target/Average Byte Sustain Excess Interval Increment
Rate Limit bits/int bits/int (ms) (bytes)
128000/64000 2000 8000 8000 125 2000
Adapt Queue Packets Bytes Packets Bytes Shaping
Active Depth Delayed Delayed Active
- 0 0 0 0 0 no
Class-map: class-default (match-any)
0 packets, 0 bytes
Match: any
Router#
You can see here that the average shaping policy has a target and average rate of
128000. And our peak shaping policy has a target of 128000 and an average of 64000.
How did the router arrive at the 128000 peak rate number? We can use our equation again
to calculate the peak rate. In the output the Sustain bits/int of 8000 is the Bc and the
Excess bits/int is the Be. Therefore we can calculate the following:
Peak_rate 64000 [1 (8000 / 8000)]
Peak_rate 64000 [1 1]
Peak_rate 64000 2
Peak_rate 128000 or 128 Kbps
Configuring Link Efficiency
Techniques
Sometimes QoS just isnt enough to provide efcient transport for sensitive trafc. This is
especially true with low-speed WAN connections. Fortunately, there are several different
techniques to manipulate frames and packets so that the transport of sensitive data is
c12.indd 508 9/21/11 11:28:09 AM
more efcient. In this section Ill show how to use LFI on both MLP and Frame Relay
connections as well as how to congure class-based header compression.
Configuring Link Fragmentation and Interleaving for
MLP and Frame Relay
In Chapter 11, you learned that link fragmentation is a technique used to accomplish
two goals:

Break up large frames into smaller frame chunks

Intermix the smaller frames with frames from other trafc ows
LFI allows us to have a more steady serialization delay on an interface, which helps
control jitter that can cripple time-sensitive voice and video trafc. LFI is recommended on
links at or below 768 Kbps and is an optional conguration on WAN links with bandwidth
speeds between 768 Kbps and 2048 Kbps. Any circuit that is higher than an E1 (2048
Kbps) is not recommended because the trade-off between fragmenting frames and higher
CPU and memory utilization is not favorable.
LFI is commonly implemented on multilink PPP (MLP) and Frame Relay circuits, and
this guide will focus on how to congure these two scenarios.
Configuring LFI for Multilink PPP
Multilink PPP is a Layer 2 transport mechanism dened in RFC 1990 that encapsulates
Layer 3 trafc over point-to-point links including ISDN. While the protocol does have
built-in abilities to load-balance trafc from multiple links into a single connection, keep in
mind that MLP can be used for a single P2P connection as well.
Ciscos LFI feature uses the MLP protocol for transport because the protocol natively
allows frames to be fragmented and passed across a WAN connection, where the frame is
then put back together. Cisco then adds the interleaving feature, which is used to provide
a special transmit queue for time-sensitive frames such as voice and video. This special
queue is given priority over other trafc so it can avoid serialization delays. The queuing
mechanism used to differentiate between time-sensitive and regular trafc is WFQ, which
can differentiate trafc ows at Layers 3 and 4.
To congure MLP, you must rst create an MLP virtual interface by using the
interface multilink command followed by a unique virtual interface identier. The
virtual interface identier will later be used to map the physical interface to the multilink
interface. In our example, we will use the number 1 as our MLP virtual interface identier:
Router(config)#interface multilink 1
Router(config-if)#
At this point, we are in config-if conguration mode, and we should rst congure
an IP address on it. Note that the virtual interface contains the IP address and not the
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510 Chapter 12

physical interface that we will apply the virtual interface to. If you have an IP address already
applied to the physical interface, it needs to be removed for the multilink connection to work.
Next, we will use the ppp multilink command to enable MLP on our interface. Then we
need to turn on the LFI feature by issuing the ppp multilink interleave command. As an
optional conguration we will use the ppp multilink fragment-delay command, followed
by a delay time in milliseconds. This command will alter the LFI fragmentation techniques
to abide by a specic delay timer. This is very useful for voice trafc, which is delay sensitive.
So, for example, if we were to set the fragment delay to 20 ms, MLP would fragment frames
in a way that the delay would be 20 ms or less on the interface. Here is the complete virtual
interface conguration:
Router(config-if)#ppp multilink
Router(config-if)#ppp multilink interleave
Router(config-if)#ppp fragment-delay 20
Router(config-if)#exit
Router(config)#
Lastly, we can apply our multilink virtual interface to a physical interface. In our example,
we will congure LFI on interface serial 0/1. We must rst change the default encapsulation
method of HDLC to PPP, by issuing the encapsulation ppp command. Then we must
enable multilink by issuing the ppp multilink command. Last, we can reference our virtual-
multilink interface by issuing the ppp multilink-group 1 command, as shown here:
Router(config)#int s0/1
Router(config-if)#encapsulation ppp
Router(config-if)#ppp multilink
Router(config-if)#ppp multilink-group 1
Router(config-if)#no shutdown
Router#
At this point, MLP is up and running on your interface. Make sure that you also congure
the opposite-end router with a multilink interface and LFI that is identical to this one.
Configuring LFI for Frame Relay
Frame Relay is one of those legacy technologies that seem to stick around year after year.
Many remote sites are still connected to their primary site through the use of Frame Relay
circuits. But because the technology is dated, the bandwidth speeds are less than ideal. When
you attempt to run time-sensitive trafc such as voice over these circuits, you can experience
congestion problems quickly. In this section, you will learn how to congure FRF.12, which
is a specication for fragmenting large Frame Relay frames into smaller chunks.
FRF.12 does not have Layer 3 or 4 intelligence and therefore cannot distinguish between
a voice packet and an HTTP packet, for example, as MLP can. Instead, FRF.12 simply
c12.indd 510 9/21/11 11:28:10 AM
fragments any frame larger than the congured fragment size. Because of this, you must
be cautious in setting your fragmentation size so you dont accidentally fragment time-
sensitive voice frames, which can cause added delay. Make sure you properly calculate your
voice frame size and set the fragmentation size to something higher. Frame Relay trafc
shaping must also be congured on the Frame Relay map and physical interface in order for
fragmentation to work.
The map-class frame-relay command followed by a unique name identier is used
to congure FRF.12 fragmentation. Once we have created a Frame Relay map class, we
must use the frame-relay fragment command, followed by the maximum fragment size in
bytes. We can use the following equation to gure out the optimal maximum fragment size
for our circuit:
Max_fragment_size (bandwidth / 8) target_delay
In order to use this calculation, you must know the bandwidth of the Frame Relay
connection (in bps) and your target serialization delay time. For this example, we will
assume a bandwidth of 256 Kbps and a target serialization delay of 10 ms (0.01 seconds).
The equation is:
Max_fragment_size (256000 / 8) 0.01
Max_fragment_size 32000 0.01
Max_fragment_size 320 bytes
We can then congure our Frame Relay trafc-shaping conguration as shown in this
example, where we congure a Frame Relay class named Frag-Me:
Router(config)#map-class frame-relay Frag-Me
Router(config-map-class)#frame-relay fragment 320
Router(config-map-class)#frame-relay cir 128000
Router(config-map-class)#frame-relay fair-queue
Router(config-map-class)#exit
Router(config)#
Now its time to apply our Frame Relay fragment and shaping class to the Frame Relay
interface with a congured DLCI. Similar to MLP, we must modify our Layer 2 protocol
from HDLC to Frame Relay. Also remember that we must enable Frame Relay trafc
shaping on the physical interface.
We can then congure DLCI 100 on the subinterface s0/0.1 point-to-point circuit. This
is where we need to apply our Frame Relay class map and set the bandwidth for the circuit,
as shown here:
Router(config-if)#frame-relay traffic-shaping
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512 Chapter 12

Router(config-if)#exit
Router(config)#int s0/0.1 point-to-point
Router(config-subif)#frame-relay interface-dlci 100
Router(config-fr-dlci)#class Frag-Me
Router(config-fr-dlci)#end
Router#
To verify the result, we can use the show frame-relay fragment command to display
the maximum fragment size as well as the total number of fragmented frames, as shown in
this example:
Router#show frame-relay fragment
interface dlci frag-type size in-frag out-frag dropped-frag
Serial0/0.1 100 end-to-end 320 643 790 0
Router#
Now that you understand how to congure LFI for MLP and Frame Relay connections,
lets move on to learn how to congure class-based header compression.
Configuring Class-Based Header Compression
Class-based (CB) header compression can perform either RTP, TCP, or both TCP and
RTP compression on packets that are dened within a policy map. Using the layered MQC
conguration method, we can congure trafc for compression as well as multiple other
QoS policies such as PQ and CBWFQ.
The command used to congure CB header compression while within config-pmap-c
conguration mode is compression header ip. If you simply enter this command, it will
enable both cRTP and TCP compression. If you want to enable only one or the other, you
can use the optional rtp and tcp keywords.
In our example conguration, we want to create a class map named voice. This class
will group voice packets by matching on the DSCP EF markings. We will then create a new
policy map named policyRTPcompress. This policy will give a strict policy of 50 percent of
the link bandwidth to voice trafc and will enable cRTP compression:
Router(config-cmap)#match dscp ef
Router(config-cmap)#exit
Router(config)#policy-map policyRTPcompress
Router(config-pmap)#class voice
Router(config-pmap-c)#compression header ip rtp
Router#
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Configuring Trust Boundaries 513
Now its all a matter of applying our policy map to an interface using the service-policy
command, and were now using a strict priority queue and cRTP on voice packets with the
DSCP EF markings congured on them. Here is an example of how to apply our policy map
to interface serial 0/0:
Router(config-if)#service-policy output policyRTPcompress
Router#
Configuring Trust Boundaries
Chapter 11 discussed Layer 2/Layer 3 markings and trust boundary locations on Catalyst
switches. Recall that the best-practice decision about where to set a trust boundary depends
on the level of trust from endpoint devices and the capabilities of Access layer switches. If
your access switches are QoS capable, you can trust markings from all endpoints or only
the Cisco IP phones. Most organizations choose to trust the Cisco phones, and any other
markings from PCs are rewritten regardless of what they come in as. In fact, by default, if a
PC is connected to a Cisco phone, the phone will rewrite any frame coming from the PC with
a CoS value of 0. Keep in mind, however, that the DSCP markings will not be rewritten.
To congure trust boundaries on an Access layer switch, we must be in config-if
conguration mode of the switchport that connects to the device we want to create a
boundary for. In Figure 12.11 we will use interface gi0/5 as our example interface to apply
a trust boundary on.
FI GURE 12.11 Access switch trust boundary configuration
Switch
Gi0/5
VLAN 10
Trusted markings
Untrusted markings
VLAN 99
The mls qos trust command is used to set what type of trafc is to be trusted that
comes inbound on the interface. We can then set the boundary to trust one of the two
markings:
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cos Trusts all Layer 2 CoS markings and rewrites the DSCP markings using the default
CoS-to-DSCP map.
dscp Trusts the DSCP markings and rewrites the CoS markings using the default
DSCP-to-CoS map.
The default CoS-to-DSCP and DSCP-to-CoS mappings are listed here:
CoS-to-DSCP mappings
CoS markings 0 1 2 3 4 5 6 7
DSCP markings 0 8 16 24 32 40 48 56
DSCP-to-CoS mappings
DSCP markings 0 8, 10 16, 18 24, 26 32, 24 40, 46 48 56
CoS markings 0 1 2 3 4 5 6 7
The default CoS-to-DSCP and DSCP-to-CoS mappings can be modified,
and you will learn how this is done in the next section.
The preferred method for a Cisco IP phone network is to use the Layer 2 CoS values on
lower-end Catalyst switches. One optional command that can be added when conguring
trust boundaries to use CoS is mls qos trust cos pass-through dscp. The pass-through
dscp keywords state that the DSCP markings are not to be overwritten with the CoS-to-
DSCP map.
Another popular conguration command that can be used when a network has Cisco IP
phones is mls qos trust device cisco-phone. This command works with the Cisco
Discovery Protocol (CDP), which both the Cisco phone and switchport (if enabled) understand.
This effectively pushes our trust boundary out past the access switch to Cisco IP phones. But
keep in mind that only Cisco IP phones are trusted and no other end-device markings.
The following example shows how to congure an Access layer switchport with both a
voice (10) and data (99) VLAN. In addition, the switchport will be congured to trust CoS
markings from Cisco IP phones:
Switch(config)#interface gi0/5
Switch(config-if)#switchport mode access
Switch(config-if)#switchport voice vlan 10
Switch(config-if)#switchport access vlan 99
Switch(config-if)#mls qos trust cos
Switch(config-if)#mls qos trust device cisco-phone
Switch#
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Configuring CoS-to-DSCP Mappings 515
One nal switch interface conguration command that you should be familiar with is
mls qos cos, followed by a CoS value. This command statically assigns a CoS value to
trafc that comes into the switchport untagged and thus does not have a CoS value assigned.
The optional override keyword can be used so the static CoS value used in this command
will rewrite the CoS value of frames that already have CoS markings. In the following
example, we will congure our switch interface gi0/5 to assign frames with a CoS of 1 if
they come in untagged:
Switch(config)#interface gi0/5
Switch(config-if)#mls qos cos 1
Switch#
Next well discuss the need for CoS-to-DSCP mappings (and vice versa) and how to
modify the default settings for better end-to-end QoS.
Configuring CoS-to-DSCP Mappings
As you know, trust boundaries are congured to trust (or not to trust) QoS markings
that come into a port. When a switch is congured to trust the CoS values of a frame,
the switch will rewrite the DSCP value based on the default CoS-to-DSCP values. This
remapping of DSCP numbers is a critical step and allows us to have a truly end-to-end
QoS policy.
Unfortunately, the default CoS-to-DSCP mappings (Figure 12.12) built into the Catalyst
switch IOS are not ideal for all situations. For example, a Cisco IP phone marks trafc
coming from it with a CoS of 5 and a DSCP of 46. But if you look at Figure 12.12, we have
a switch congured to trust the CoS values and rewrite the DSCP value using the default
CoS-to-DSCP mappings.
FI GURE 12.12 CoS-to-DSCP default mapping
Router
trust DSCP CoS = 5
DSCP = 40
Cos markings 0 1 2 3 4 5 6 7
0 8 16 24 32 40 48 56 DSCP markings
CoS = 5
DSCP = 46
Switch
trust CoS
CoS-to-DSCP lookup
and remapping
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Youll notice that when the switch remaps the DSCP value, it changes from 46 to 40.
Best-practice documentation states that RTP trafc should be classied as DSCP 46 (PHB
EF), and therefore we have our voice operating with suboptimal QoS levels, which can
degrade the Quality of Service applied to it. To x this problem, we can adjust the default
CoS-to-DSCP mappings to make them conform to best-practice policy.
To modify the default mapping settings, we can issue the mls qos map cos-dscp
command, followed by eight DSCP values that we want to map to CoS values 0 to 7. The
remapping is congured globally on the switch. This example shows a commonly remapped
CoS-to-DSCP mapping:
Switch(config)#mls qos map cos-dscp 0 10 18 26 34 46 48 56
Switch(config)#end
Switch#
When our Catalyst switch makes queuing decisions that come from the DSCP-operated
network, it performs a DSCP-to-CoS remapping, again in an attempt to use a single QoS
mark from end to end. If the default DSCP-to-CoS mappings are not optimal for your
network, they too can be modied. We use the same mls qos map command, but this
time we specify dscp-cos. But because there are multiple DSCP markings and only eight
possible CoS values, we have the ability to congure up to 13 DSCP markings to a single
CoS number. Each CoS value is congured using a single CLI command. For example, we
will congure DSCP markings 0, 8, and 10 to a CoS of 0. Then, on the next line, we will
map DSCP 16, 18, 24, and 26 to a CoS value of 1:
Switch(config)#mls qos map dscp-cos 0 8 10 to 0
Switch(config)#mls qos map dscp-cos 16 18 24 26 to 1
Switch(config)#end
Switch#
To review the current mappings of a switch we can use the show mls qos maps
command followed by the type of map we want to see. Here we will use the show command
to view our CoS-to-DSCP mappings:
Switch#show mls qos maps cos-dscp
cos-dscp map:
cos: 0 1 2 3 4 5 6 7
--------------------------------
dscp: 0 10 18 26 34 46 48 56
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Exam Essentials 517
Summary
Quality of Service is no longer a nice to have feature on networks. When networks
begin intermixing standard TCP data transmissions along with time-sensitive UDP voice
and video transmissions, you must be able to properly mark sensitive trafc so it gets
the interface priority that it requires. The CVOICE version 8.0 exam now requires that
certication candidates know the two primary QoS conguration methods, AutoQoS and
MQC. Without this knowledge, youll quickly nd that many IP networks will not meet the
strict bandwidth, latency, and jitter requirements, and voice quality will suffer.
Exam Essentials
Know the two different AutoQoS types. They are AutoQoS for VoIP and AutoQoS for
the Enterprise.
Understand when AutoQoS for the Enterprise is recommended. AutoQoS for the Enterprise
is used on larger networks with a signicant number of remote sites interconnected with
WAN connections.
Know what type of AutoQoS can be configured on Cisco Catalyst switches. Cisco
Catalyst switches can only be congured for AutoQoS for VoIP.
Know the two phases of AutoQoS for the Enterprise. The rst phase is called the AutoQoS
autodiscovery phase and is used to discover interfaces and data ows on a network. The
second phase creates QoS templates based on the ndings from the autodiscovery phase.
Know the three primary commands used to configure QoS using MQC. The three
commands are class-map, policy-map, and service-policy.
Understand the difference between class-map match-any and match-all statements. The
match-any statement triggers when any single match statement is met. The match-all
statement is triggered only when all match statements are met.
Understand the difference between RED and WRED. RED is used to drop packets
randomly before queue buffers ll up. WRED uses classication markings to drop less-
important packets rst.
Understand the concept of a token bucket. A token bucket is used to regulate data in a
ow to manage the overall bandwidth of an interface.
Understand the three different types of class-based header compression. CB header
compression can perform either RTP, TCP, or RTP and TCP compression on packets
matched in a policy map.
Understand the most common need to modify CoS-to-DSCP mappings on a VoIP
network. The default CoS-to-DSCP mappings will mark voice trafc with a lower DSCP
value that is recommended by Cisco.
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Written Lab 12.1
1. What Cisco equipment can Auto-QoS for VoIP be congured on?
2. AutoQoS for the Enterprise is recommended on large networks with multiple
connections.
3. What Cisco switch AutoQoS command is used to congure an interface to trust DSCP
and/or CoS markings?
4. What router interface command is used to congure AutoQoS for the Enterprise to
monitor and classify trafc using NBAR?
5. What class-map command will congure a class named voice and set it to classify
trafc when all match statements are met?
6. What class-map match statement will match packets that have DSCP values of 48 or 56?
7. What policy-map statement will congure WRED?
8. What is the switch-interface command used to rewrite the CoS to be a value of 2?
9. What is the switch-interface command used to map the DSCP values of 0 and 8 to a
CoS value of 2?
10. What switch command can be used to verify CoS to DSCP mappings?
Hands-On Labs
To complete the labs in this section, you need a Cisco Catalyst switch for Layer 2 QoS
congurations. In addition, you will need a router with a voice-capable IOS, one serial
interface, and one Ethernet interface. Each lab in this section builds upon the last and will
follow the logical network design as shown in Figure 12.13.
FI GURE 12.13 QoS lab diagram
Phone
(VLAN 5)
PC
(VLAN 101)
Switch
Fa0/5
Fa0/0
S0/0
Fa0/0
Router
Remote site
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Hands-On Labs 519
Lab 12.1: Conguring a Switchport to Trust Cisco IP Phone QoS Markings
Lab 12.2: Modifying CoS-to-DSCP Mappings
Lab 12.3: Conguring a Router for QoS Using MQC
Hands-On Lab 12.1: Configuring a Switchport to Trust
Cisco IP Phone QoS Markings
In this lab, were going to congure our access port on fa0/5 to place trafc coming from
the phone on VLAN 5 and from the PC on VLAN 101. Additionally, we will congure the
interface to trust QoS markings that come from the Cisco IP phone but not the connected PC.
1. Log into the switch and go into conguration mode by typing enable and then
configure terminal.
2. We can then begin to congure the Cisco IP phone that is attached to fa0/5 by rst
entering into interface conguration mode by typing interface fa 0/5.
3. We will congure this port as an Access layer port by typing switchport mode access.
4. Now we will congure the switchport to place voice trafc on VLAN 5 by typing
switchport voice vlan 5.
5. Next, we will congure all other trafc coming from the PC to use VLAN 101 for
transport by typing switchport access vlan 101.
6. Our rst QoS command is to congure the port to trust CoS markings by typing mls
qos trust cos.
7. Lastly, we will congure the port to trust CoS markings only from the Cisco phone by
typing mls qos trust device cisco-phone.
8. Exit config-interface mode by typing end.
Hands-On Lab 12.2: Modifying CoS-to-DSCP Mappings
From our conguration in lab 12.1, we are now trusting CoS markings that come from the
Cisco IP phone connected to fa0/5. However, we must now modify the default CoS-to-DSCP
mappings so our voice trafc will be marked with a DSCP value of 46 (EF). Remember
that if we use AutoQoS for VoIP on our switchport, the AutoQoS script will modify the
CoS-to-DSCP mappings for us. But since we are manually conguring QoS on our switch,
we need to congure mapping rules so that our voice trafc will receive the proper QoS on
the network. We will use the following CoS-to-DSCP mappings according to Table 12.3.
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TABLE 12. 3 CoS-to-DSCP mappings
CoS Markings DSCP Markings
0 0
1 10
2 18
3 26
4 34
5 46
6 48
7 56
configure terminal.
2. Congure the CoS-to-DSCP mappings by typing mls qos map cos-dscp 0 10 18 26 34
46 48 56.
3. Exit global conguration mode by typing exit.
Hands-On Lab 12.3: Configuring a Router
for QoS Using MQC
We will now shift our attention from the Cisco switch QoS conguration to the router QoS
conguration. In this lab we will use MQC to congure QoS to match our voice trafc
coming from our switch. We will then apply QoS policies and apply them to the outbound
serial 0/0 interface. We will rst congure a class map named voice-traffic. Within this
class map, we will match packets that have a DSCP value of 46. Remember that in our
previous two labs, we congured our switch to trust markings from the Cisco IP phone and
modied the CoS to DSCP mappings so voice trafc will be marked with a DSCP value of 46.
1. Log into the switch and go into conguration mode by typing enable and then
configure terminal.
2. Congure and name our class map by typing class-map voice-traffic.
3. Differentiate between voice and non-voice packets by typing match dscp 46.
4. Exit out of class-map conguration mode by typing exit.
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Hands-On Labs 521
Now we will create our policy map named voice-policy, which will tell our router how
voice trafc that is matched against the class map rules should be handled. In our example,
we will give our voice trafc a strict priority queuing capability up to 60 percent of the
total link bandwidth.
5. Congure and name our policy map by typing policy-map voice-policy.
6. Specify the voice-trafc class map by typing class voice-policy.
7. Apply the QoS strict-priority queuing mechanism for this trafc by typing priority
percent 60.
8. Exit out of policy-map conguration mode by typing exit.
Finally, we will apply the policy to our serial interface 0/0 so the voice trafc can have
plenty of bandwidth across the WAN to the remote network.
9. Enter into interface conguration mode for our serial interface by typing interface
serial 0/0.
10. Apply the policy map outbound by typing service-policy output voice-policy.
11. Exit interface conguration mode by typing end.
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Review Questions
1. AutoQoS works in conjunction with which two features?
A. CDP
B. SIP
C. CEF
D. NBAR
E. H.323
2. Which of the following are not AutoQoS features that can be enabled on Cisco Catalyst
switches?
A. Setting the trust boundary at the Cisco IP phone
B. Setting the trust boundary at the access or trunk port
C. Automatic classification of RTP, cRTP, and voice gateway signaling protocols
D. Automatic enabling of PQ and WRR when appropriate
3. When configuring AutoQoS for VoIP on a Cisco router, what optional keyword(s) instructs
the router to believe DSCP markings from incoming packets?
A. trust interface
B. trust cisco-phone
C. trust dscp
D. trust
4. You are configuring a Cisco switch interface and issue the following command:
auto qos voip
Which of the following is not a keyword that can be used to complete this command?
A. cisco-phone
B. dscp
C. trust
D. cisco-softphone
5. What two global configuration commands will be found after enabling AutoQoS on one or
more interfaces on a Cisco switch?
A. mls qos
B. mls qos trust cos
C. mls qos trust dscp
D. mls qos map cos-dscp 0 8 16 26 32 46 48 56
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6. What command can be used to verify that the AutoQoS for the Enterprise autodiscovery
phase is identifying and classifying traffic?
A. show qos auto discovery
B. show auto discovery qos
C. show mls qos auto discovery
D. show auto discovery mls qos
7. Which of the following switch-interface commands will trust DSCP markings coming from
an upstream router?
A. auto qos voip trust
B. qos voip trust
C. qos voip trust dscp
D. mls qos trust dscp
8. A QoS class map is always configured by default when a class map is configured on Cisco
hardware. What is the name of this class map?
A. class-voip
B. class-best-effort
C. class-default
D. class-network
9. How can the MQC service policy be applied on a Cisco router?
A. Inbound on an interface
B. Outbound on an interface
C. Inbound and/or outbound on an interface
D. Inbound or outbound on an interface but not both inbound and outbound
10. You are configuring an MQC policy for voice traffic and enter the following commands:
Router(config-cmap)#match precedence 5
Router(config-cmap)#match dscp 46
In order for a packet to trigger on this class map, what must be true?
A. The packet must have an IP Precedence value of 5 or a DSCP value of 46.
B. The packet must have an IP Precedence value of 5 and a DSCP value of 46 or higher.
C. The packet must have an IP Precedence value of 5 and a DSCP value of 46.
D. The packet must have an IP Precedence value of 5 or higher or a DSCP value of 46 or
higher.
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11. You have created the following access list:
access-list 5 permit 192.168.1.0 0.0.0.255
access-list 5 permit 192.168.2.0 0.0.0.255
You want to use this access list in a class map. Which of the following commands properly
does this?
A. Router(config)#class-map match-all access-group 5
B. Router(config-cmap)#match access-list 5
C. Router(config-cmap)#match access-group 5
D. Router(config)#class-map match-any access-group 5
12. Which of the following is the proper way to create a policy-map named mypolicy for a class
map named ipt? You want to apply a strict PQ of 60 percent of the overall link bandwidth.
A. Router(config)#policy-map mypolicy
Router(config-pmap)#class ipt
B. Router(config)#policy-map mypolicy
Router(config-pmap)#class ipt
C. Router(config)#policy-map mypolicy
Router(config-pmap)#class ipt bandwidth percent 60
D. Router(config)#policy-map mypolicy
Router(config-pmap)#class ipt priority percent 60
13. Which of the following policy-map keywords sets a CBWFQ value?
A. priority
B. police
C. shape
D. bandwidth
14. Which of the following show commands can be issued on a router to view the policy map(s)
that has been applied on interface serial 0/1?
A. show policy-map interface serial 0/1
B. show service policy serial 0/1
C. show policy-map serial 0/1
D. show service-policy interface serial 0/1
15. What is a Cisco proprietary congestion-avoidance technique that intelligently drops packets
based on QoS markings?
A. RED
B. WRED
C. CQ
D. CBWFQ
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16. If there are 56,000 bits worth of tokens of burst traffic (Bc) and these bits are moved in and
out of the token bucket every 250 ms, what is our token bucket CIR?
A. 14,000 bps
B. 140,000
C. 24,400 bps
D. 224,000 bps
17. When traffic in a token bucket goes above the CIR, what is the traffic called?
A. Burst conforming (Bc)
B. Burst violated (Bv)
C. Burst overflow (Bo)
D. Burst exceeded (Be)
18. In a dual-bucket, dual-rate token model, which of the following is not a possible token
outcome?
A. Overflow-action
B. Conform-action
C. Violate-action
D. Exceed-action
19. Tokens within a traffic-policing mechanism using a single token bucket can have which two
possible outcomes?
A. Conform-action
B. Violate-action
C. Exceed-action
D. Shaping-action
E. Policing-action
20. You are reviewing a router policy map configuration as shown here:
policy-map policyONE
police 8000 2000 4000 conform-action transmit exceed-action set-dscp-
transmit 0
What token bucket structure is used?
A. Single-bucket
B. Single-bucket with dual-rates
C. Dual-bucket with single-rates
D. Dual-bucket with dual-rates
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1. C, D. The automatic classication function of AutoQoS uses NBAR to identify and classify
trafc at Layer 4. NBAR requires that CEF be enabled.
2. C. The automatic classication of RTP, cRTP, and voice-signaling protocols cannot be
performed on Catalyst switches.
3. C. The optional trust keyword indicates that inbound packets that are already marked
with DSCP values should be trusted.
4. B. The three possible keyword options are cisco-phone, trust, and cisco-softphone.
5. A, D. The mls qos command enables QoS on the switch, and the mls qos map cos-dscp
0 8 16 26 32 46 48 56 command sets the new set of CoS-to-DSCP mappings.
6. B. The show auto discovery qos command displays information about how long the
discovery process has been running, and the types of trafc found and classied while
monitoring. Additionally, the command output shows suggested QoS conguration
commands based on the trafc-discovery process.
7. D. The mls qos trust dscp interface command will trust DSCP markings coming
inbound on the interface.
8. C. The class-default class is included when a new policy is created and applied to an
interface.
9. C. A service policy can be applied to trafc coming into or out of an interface.
10. A. The command class-map voice does not include the keywords match-any or
match-all. Because of this, the router will use the default match-any statement.
Therefore, a packet that has an IP Precedence of 5 or a DSCP of 46 will be classied into
this class.
11. C. The match access-group class-map command followed by the access-list number
should be congured while in config-cmap mode.
12. B. You must rst specify the class map while in config-pmap mode. Then you use the
priority percent command followed by a percentage value to set the strict priority queue
for this trafc.
13. D. The bandwidth command sets the CBWFQ bandwidth.
14. A. The show policy-map interface command followed by the interface type and
number is used to view the input and/or output policy maps applied to that specic
interface.
15. B. WRED is a Cisco proprietary mechanism that uses the same dropping techniques of
RED but drops packets with lower IP Precedence or DSCP markings before those of a
higher priority.
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16. D.
CIR = Bc / Tc
CIR = 56,000 / 0.25
CIR = 224,000 bps or 224 Kbps
17. D. Trafc that exceeds the CIR is considered to be Be trafc and is placed into the Be token
bucket.
18. A. All of these token outcomes for tokens are possible except for the overow-action.
19. A, C. The two possible outcomes of a single token bucket-policing mechanism are
conform-action or exceed-action.
20. A. Because the police command only has conform-action and exceed-action settings, this
is a single-bucket structure.
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528 Chapter 12

1. Routers and Catalyst switches
2. WAN (or remote site)
3. auto qos voip trust
4. auto discovery qos
5. class-map match-all voice
6. match dscp 48 56
7. random-detect dscp-based
8. mls qos cos 2
9. mls qos map dscp-cos 0 8 to 2
10. show mls qos maps cos-dscp
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About the
Companion CD
IN THIS APPENDIX:
What youll find on the CD.
System requirements.
Using the CD.
Troubleshooting.
Appendix
bapp.indd 529 9/20/11 1:34:45 PM
What Youll Find on the CD
The following sections are arranged by category and
summarize the software and other goodies youll nd on the CD. If you need help with
installing the items provided on the CD, refer to the installation instructions in the Using
the CD section of this appendix.
Sybex Test Engine
The CD contains the Sybex test engine, which includes two bonus practice exams for
Exam 642-437.
Electronic Flashcards
These handy electronic ashcards are just what they sound like. One side contains a
question and the other side shows the answer.
PDF of the Glossary
We have included an electronic version of the Glossary in PDF format. You can view the
electronic version of the book with Adobe Reader.
Adobe Reader
Weve also included a copy of Adobe Reader so you can view PDF les that accompany the
books content. For more information on Adobe Reader or to check for a newer version,
visit Adobes website at www.adobe.com/products/reader/.
bapp.indd 530 9/20/11 1:34:46 PM
Troubleshooting 531
System Requirements
Make sure your computer meets the minimum system requirements shown in the following
list. If your computer doesnt match up to most of these requirements, you may have
problems using the software and les on the companion CD. For the latest and greatest
information, please refer to the ReadMe le located at the root of the CD-ROM.
A PC running Microsoft Windows 98, Windows 2000, Windows NT4 (with SP4 or
later), Windows Me, Windows XP, Windows Vista, or Windows 7
An Internet connection
A CD-ROM drive
Using the CD
To install the items from the CD to your hard drive, follow these steps:
1. Insert the CD into your computers CD-ROM drive. The license agreement appears.
Windows users: The interface wont launch if you have autorun disabled.
In that case, click Start Run (for Windows Vista or Windows 7, Start
All Programs Accessories Run). In the dialog box that appears, type
D:\Start.exe. (Replace D with the proper letter if your CD drive uses a
different letter. If you dont know the letter, see how your CD drive is listed
under My Computer.) Click OK.
2. Read the license agreement, and then click the Accept button if you want to use the CD.
The CD interface appears. The interface allows you to access the content with just one or
two clicks.
Troubleshooting
Wiley has attempted to provide programs that work on most computers with the minimum
system requirements. Alas, your computer may differ, and some programs may not work
properly for some reason.
The two likeliest problems are that you dont have enough memory (RAM) for
the programs you want to use or you have other programs running that are affecting
installation or running of a program. If you get an error message such as Not enough
bapp.indd 531 9/20/11 1:34:46 PM
532 Appendix
About the Companion CD

memory or Setup cannot continue, try one or more of the following suggestions and
then try using the software again:
Turn off any antivirus software running on your computer. Installation programs
sometimes mimic virus activity and may make your computer incorrectly believe that
its being infected by a virus.
Close all running programs. The more programs you have running, the less memory is
available to other programs. Installation programs typically update les and programs;
so if you keep other programs running, installation may not work properly.
Have your local computer store add more RAM to your computer. This is, admittedly,
a drastic and somewhat expensive step. However, adding more memory can really help
the speed of your computer and allow more programs to run at the same time.
Customer Care
If you have trouble with the books companion CD-ROM, please call the Wiley Product
Technical Support phone number at (800) 762-2974.
bapp.indd 532 9/20/11 1:34:46 PM
Index
Note to the Reader: Throughout this index boldfaced page numbers indicate primary
discussions of a topic. Italicized page numbers indicate illustrations.
A
A-law algorithm, 55
access codes for private plans, 117
access-group option, 490, 492
access-list option, 492
ACF (Admission Confirmation) message,
400401
Adaptive Differential Pulse Code Modulation
(ADPCM), 155
Adaptive MultiRate Wideband
(AMR-WB), 155
address signaling, 3537, 3637
address translation, 398
Admission Confirmation (ACF) message,
400401
Admission Reject (ARJ) message, 400401
Admission Request (ARQ) message,
400401
ADPCM (Adaptive Differential Pulse Code
Modulation), 155
AF (Assured Forwarding) PHB, 452453
agents
call processing, 1517, 17
fax relay, 362
SIP, 86, 240
AIM (AOL Instant Messenger), 156
ALERTING ephone extension state, 336, 338
aliases, directory, 299
allow-connections command, 422423
allow-connections sip to sip
command, 306
alternate mark inversion (AMI), 59, 59
AMI option, 63, 192
ampersands (&) in regular expressions, 128
AMR-WB (Adaptive MultiRate
Wideband), 155
analog telephones, 3
analog telephony adapters (ATAs), 15
analog-to-digital conversion
compression process, 5455
encoding process, 5354, 54
overview, 5152
quantization, 53, 53
signal sampling, 5253, 52
analog-to-IP adapters, 15
analog voice, 34
conversion to digital. See analog-to-digital
conversion
exam essentials, 6667
ports, 3435
FXO outbound, 184187, 185
FXS and FXO PLAR OPX,
180184, 180
FXS basic, 4750
FXS/DID inbound, 184187, 185
review questions, 6974
signaling, 35
address, 3537, 3637
E&M, 4146, 4446
ground-start, 4041, 41
informational, 3738
supervisory, 3841, 39, 41
summary, 66
T1 CAS to analog cross-connect, 191195,
191
written lab, 6768, 75
ANI (Automatic Number Identification), 112,
188189
ani mapping command, 213
answer-address command, 112
answer (ANS) tones, 362
AOL Instant Messenger (AIM), 156
application mgcpapp command, 259
application-specific packets, 82
application-specific routing (ASR), 91
applications, 15
area codes, 8, 115, 190
ARJ (Admission Reject) message, 400401
ARQ (Admission Request) message, 400401
ASR (application-specific routing), 91
associate application SCCP command, 199
associate ccm command, 201
associate profile command, 201
Assured Forwarding (AF) PHB, 452453
asterisks (*) in regular expressions, 128
bindex.indd 533 9/20/11 1:38:36 PM
ATA 180 series phones, 15
ATAs (analog telephony adapters), 15
AUCX (audit connection) command, 256
audible rings, 310
audio fidelity, 148, 148
AUEP (audit endpoint) command, 256
Authentication and Message Integrity packets, 83
authentication username command, 269
authentication username password command,
240
authorization, call, 398399
auto command, 285
auto assign command, 310
auto discovery qos command, 484, 488
auto discovery qos trust command, 484
auto qos command, 488
auto qos cisco-phone command, 480481
auto qos voip command, 475476, 481
auto qos voip cisco-phone command, 482
auto qos voip trust command, 481482
autodiscovery phase in AutoQoS, 483487
Automatic Number Identification (ANI), 112,
188189
AutoQoS, 446, 474475
autodiscovery phase, 483487
enterprises, 475, 483, 484
implementation phase, 488
installation phase, 483
VoIP, 475
on routers, 475479, 476
on switches, 479483, 480
average keyword, 506
B
B (bearer) channels, 61
b (silent with beep) ephone button
separator, 318
B8ZS (Bipolar 8-bit Zero Substitution),
59, 60, 192
B8ZS option, 63
BA (behavior aggregate) in DSCP, 451
Baby Bell companies, 8
background noise, 149151, 150
backhauled connections, 7
backslashes (\) in regular expressions, 128130
backup paths, voice, 368369
COR, 372377, 373
MGCP-to-H.323 fallback, 370372, 371
SRST, 376377
WAN-to-PSTN fallback, 369370, 369
backward compatibility of inline power switches,
284285
bandwidth
H.323 gatekeepers, 398399
CAC control, 411414, 413
RAS messages, 404405
IP voice, 164
calculations, 165, 167169, 169
codec bit rate, 166167
packet and frame size information,
165166, 166
providing, 441442
bandwidth command
H.323 gatekeepers, 412413
QoS policy maps, 493
Bandwidth Confirm (BCF) message, 404405
Bandwidth Reject (BRJ) message, 404405
Bandwidth Request (BRQ) message, 404405
baseline models in QoS, 461463, 462
Basic Rate Interface (BRI), 56, 61
Bc (Burst Conforming) size for token buckets,
500501, 506
BCF (Bandwidth Confirm) message, 404405
Be (burst-exceeding) traffic for token buckets,
501502, 506
bearer (B) channels, 61
behavior aggregate (BA) in DSCP, 451
Bell, Alexander Graham, 2
Best-effort QoS model, 447, 449
bind command, 247
bind all source-interface command, 269
bind control source-interface
command, 306
bind interface command, 201
bind srcaddr command, 409
binding
SIP sources to IP addresses, 247
virtual H.323 gateway addresses,
233234, 233
Bipolar 8-bit Zero Substitution (B8ZS),
59, 60, 192
bipolar variations, 59
bit rate for codecs, 166167
blast method for location message
forwarding, 403
bottlenecks
end-to-end delays, 442
packet loss from, 443444, 443
as quality issue, 151152
boundaries, trust
configuring, 513515, 513
identifying, 460461, 461
534 ATA 180 series phones boundaries, trust
bindex.indd 534 9/20/11 1:38:37 PM
BRI (Basic Rate Interface), 56, 61
BRJ (Bandwidth Reject) message, 404405
BRQ (Bandwidth Request) message, 404405
buffer packets in traffic policing, 456
buffering delay, 152
Burst Conforming (Bc) size for token buckets,
500501, 506
burst-exceeding (Be) traffic for token buckets,
501502, 506
Busy informational signals, 38
button command
ephone, 318
SCCP, 344
button separator options for ephone, 318319
expansion line, 324
monitor line, 319320, 320
overlay line, 320323
overlay with call waiting, 323324
watch phone, 320
Bye retry type, 246
bytes setting, 230
C
c (overlay with call waiting) ephone button
separator, 319, 323324
C549 DSP chipset (PVDM), 157158
C5510 DSP chips (PVDM2), 157, 159160
CA (call agent) fax relay method, 362
call-admission control (CAC), 9091, 93, 398,
411414, 413, 417
call admission messages, 400401
call agent (CA) fax relay method, 362
call authorization, 398399
call-block disconnect-cause incoming
command, 381
call blocking, 380382
call clarity, codecs for, 164
call control devices, 87
call flow differences, 421422, 421422
call management in H.323, 399
call processing agents, 1517, 17
call-processing clusters, 90, 90
call progress (CP) tones, 38
call routing, 108
call legs, 110111, 111
POTS dial peers, 108109
VoIP dial peers, 109110, 110
call signaling, 17, 17
call start config-serv-h232
command, 228
call waiting, 323324
Call waiting informational signals, 38
called type in translation profiles, 131
caller-ID blocking, 249
calling privileges, 90, 104
calling type in translation profiles, 131
CAMA (Centralized Automatic Messaging
Accounting) trunks, 188191, 189, 213
Cancel retry type, 246
CAR (Committed Access Rate), 456
carets (^) in regular expressions, 128130
CAS (Channel Associated Signaling), 56, 6061
categories of traffic, 446
CB (class-based) header compression,
512513
CB (class-based) QoS, 488489
CB-WRED (class-based WRED), 498
CBWFQ (class-based weighted fair queuing),
454455
CC (Country Code) in E.164, 113114
ccm-manager mgcp command, 257
CCME licenses, 297
CCS (Common Channel Signaling), 56, 6162
CDP (Cisco Discovery Protocol), 285, 514
CEF (Cisco Express Forwarding), 475
central office (CO)
overview, 45, 4
trunks, 7, 7
central office code in NANP, 115
Centralized Automatic Messaging Accounting
(CAMA) trunks,
188191, 189, 213
centralized call-control systems, 93
centralized services deployment model,
20, 21
cfg-translation-profile mode, 131, 381
CH1 licenses, 297
Channel Associated Signaling (CAS),
56, 6061
child maps, 494
CIR (Committed Information Rate), 500501
Cisco Cius Tablet, 14, 14
Cisco Discovery Protocol (CDP), 285, 514
Cisco Express Forwarding (CEF), 475
Cisco fax relay, 357359, 358
Cisco IP Communicator, 1314
cisco option, 360
cisco-phone option, 479
Cisco products overview. See Unified
Communications Model overview
cisco-rtp option, 354355
cisco-softphone option, 479
BRI (Basic Rate Interface) cisco-softphone option 535
bindex.indd 535 9/20/11 1:38:38 PM
Cisco Unified Border Element (CUBE), 396,
421422, 421422
codec transparency, 424
debug voip ipipgw command, 426427
features, 417
H.323 fast-to-slow signaling, 424425
hands-on labs, 429431, 429
media flow-around, 418419, 419, 423
media flow-through, 418, 418
overview, 416, 417
protocol interoperation, 422423, 423
RSVP-CAC, 420, 420
show call active voice brief command,
425426
show call history voice brief command,
426
show voip rtp connections
command, 426
signaling protocol interoperation,
419, 419
SIP delayed-to-early-offer signaling, 425
summary, 427
Cisco Unified Communications Manager
(CUCM), 16, 420
gatekeepers, 8990
IP soft phones, 1314
RTP, 79
Cisco Unified Communications Manager Business
Edition (CUCMBE), 16
Cisco Unified Communications Manager Express
(CUCME), 16, 294295
Cisco Unified Mobile Communicator, 1314
Cisco Unified Personal Communicator, 13
Cisco Video Advantage product, 14
Cius tablet, 14, 14
clarity, voice codecs for, 160163
class-based (CB) header compression,
512513
class-based (CB) QoS, 488489
class-based traffic policing
description, 456
token buckets, 500504, 502503
class-based traffic shaping, 500
configuring, 506508
token buckets, 500501
class-based weighted fair queuing (CBWFQ),
454455
class-based WRED (CB-WRED), 498
class-default command, 497
class-map command, 489490
class-map voice-traffic command, 520
class maps, 489492
Class of Restriction (COR), 372377, 373
Class of Service (CoS), 445, 459460, 459460
Class Selector (CS) PHB, 453
class voice-policy command, 521
classification, traffic, 444445
classification markings, baseline, 462463
classification models in QoS
Best-effort, 447, 449
DiffServ. See DiffServ QoS model
IntServ, 447449
clear call history voice command, 426
clear h323 gateway h225 command, 235
clear mgcp statistics command, 264
CLI (command line interface) in QoS, 446
clid command, 127
clid strip pi-restrict command, 249
clid substitute name command, 248
clock source command, 6364
clock source line command, 213
clock timezone command, 293
clocking, 6365, 192
CO (central office)
overview, 45, 4
trunks, 7, 7
codec command
ephones, 309
modem pass-through, 367
SCCP, 344
codec complexity command, 158159
codec preference command, 229230
codec transparent command, 424
codecs
bit rate, 166167
choosing, 163164
complexity, 156160
CUBE negotiation, 417
H.323 preference, 229231
transparency, 424
types, 153156
cold spares, 297
colons (:) ephone button separator, 318
comfort-noise command, 151
comfort noise synthesis, 151
command line interface (CLI) in QoS, 446
commas (,) for prefix adding, 125
Committed Access Rate (CAR), 456
committed burst (Bc), 506
536 Cisco Unified Border Element (CUBE) committed burst (Bc)
bindex.indd 536 9/20/11 1:38:38 PM
Committed Information Rate (CIR),
500501
Common Channel Signaling (CCS), 56, 6162
companding, 55
compatibility
codecs, 158159
ephone-DN line, 163
inline power switches, 284285
complexity of codecs, 156160, 164
compressed RTP (cRTP)
header compression, 166
overhead, 8283
compression
analog-to-digital conversion, 5455
CB header, 512513
link efficiency, 457458, 458
packets, 493
compression command, 493
compression header ip command, 512
conf-dial-peer mode, 359360
conf-serv-sip mode, 306
conf-voi-serv mode
CUBE, 422
fax pass-through, 364
fax relay, 360
H.323, 227, 229
SIP, 248
conference calling, 147, 147
config-class mode, 232
config-cm-fallback mode, 376377
config-cmap mode, 490
config-controller mode
T1 CAS, 192
T1 PRI, 196
config-dial-peer mode, 354
fax pass-through, 364
H.323, 354
outbound dial peers, 214
SIP, 242243, 425
config-dp-cor mode, 374
config-dsp-farm-profile mode, 199
config-ephone mode, 330, 332
config-ephone-dn mode, 308
config-ephone-type mode, 309
config-gk mode, 406, 411412
config-if mode
AutoQoS, 475, 488
MLP, 509
QoS, 495
trust boundaries, 513
config-pmap mode, 493
config-pmap-c mode, 493, 512
config-register-global structure, 306
config-sccp-ccm mode, 201
config-serv-h323 mode, 231
config-serv-sip mode, 242, 247
config-sip-ua mode, 240241, 243247
config-telephony mode, 300, 304305, 310,
325, 329
config-voi-serv mode, 242, 359, 425
config-voicecard mode, 158
config-voiceport mode, 180181, 207
configure terminal command, 4750
conflicts, IP address, 293
conform-actions with token buckets,
502504, 502503
Confucius, 170
Congestion informational signals, 38
congestion management
avoidance techniques, 498500, 499
baseline congestion, 463
DiffServ, 453455
connect timer, 245
connect voice-port command, 194
CONNECTED ephone extension state, 336, 338339
connection plar command, 183
Contributing Source (CSRC) field in RTP
headers, 81
controller command, 63
conversion, analog-to-digital. See analog-to-
digital conversion
COR (Class of Restriction),
372377, 373
CoS (Class of Service), 445, 459460, 459460
cos option
class maps, 491
CoS-to-DSCP mappings, 514516, 515, 519520
country codes
E.164 telephone numbers, 113114
FXS ports, 4748
progress tones, 181182
CP (call progress) tones, 38
cp-tone command, 47
CPE (customer premise equipment), 35
cptone command, 181182
CQ (custom queuing) mechanism, 454
CRC (cyclic redundancy check) feature, 58
CRCX (create connection) command, 256
create cnf-files command, 305
create profile command, 307
cRTP (compressed RTP)
Committed Information Rate (CIR) cRTP (compressed RTP) 537
bindex.indd 537 9/20/11 1:38:38 PM
cRTP (compressed RTP) (continued)
overhead, 8283
CS (Class Selector) PHB, 453
csim start command, 209
CSRC (Contributing Source) field in RTP
headers, 81
CSRC counter field in RTP headers, 80
CUBE. See Cisco Unified Border Element (CUBE)
CUCM (Cisco Unified Communications
Manager), 16, 420
gatekeepers, 8990
RTP, 79
CUCM Express, 281282
capabilities, 294295
date and time format, 328
default phone configuration file, 304305
ephones
button options. See button separator
options for ephone
extension states, 336339
registration states, 335336
state, 334335
firmware load files, 303304, 304
hardware requirements, 295
initial configuration, 297298
IP phone
keepalive timer, 329
restart vs. reset, 329332
overview, 293294
SCCP
ephone configuration, 308310, 314317
ephone directory number, 308
ephone-DN line configuration,
311313, 313314
individual lines, 317318
phone operation, 343344
signaling, 300305, 301304
SIP
signaling, 305307
voice register DNs, 310311
software licensing, 296297
summary, 339
as TFTP server, 298300, 342343
troubleshooting
DHCP, 332333
phone registrations, 332
TFTP, 333334
user locale and network locale, 325328
voice network infrastructure
power options for IP phones, 282286
VLANs, 286290, 287, 289
VoIP support, 290293
CUCMBE (Cisco Unified Communications
Manager Business Edition), 16
CUCME (Cisco Unified Communications
Manager Express), 16, 294295
custom queuing (CQ) mechanism, 454
customer premise equipment (CPE), 35
cyclic redundancy check (CRC) feature, 58
D
D (delta) channel, 61
date-format command, 328
date format in CUCM Express, 328
de-jitter buffer delay, 152
debug dialpeer command, 209210
debug ip dhcp server events command, 332
debug ras command, 415416
debug tftp events command, 333
debug voice translation command, 134
debug voip dialpeer command, 134
debug voip ipipgw command,
426427
DECEASED ephone registration state,
335337
default command, 412
default-technology keyword, 409
defaults
DHCP routers, 291292
dial peer 0, 112
PHB, 451
phone configuration files, 304305
SIP retry settings, 246
zone bandwidth, 412
delay, network, 151152
delay-dial command, 50
delay-dial E&M signaling, 46, 46
delayed offer in SIP, 238, 238
delayed-to-early-offer signaling, 425
delta (D) channel, 61
demarcation point (demarc) points, 5, 5
deployment models, 20
centralized services, 20, 21
distributed services, 21, 21
geographical diversity, 2223, 23
inter-networking of services, 22, 22
538 cRTP (compressed RTP) deployment models
bindex.indd 538 9/20/11 1:38:39 PM
deregulation, 8
destination-address mac option, 491
destination-pattern command
dial-plan digit manipulation, 123
DNIS, 112
FSX ports, 184
outbound dial peers, 112113, 214
POTS dial peer, 109
SIP, 270
tool bypass, 385386
VoIP dial peers, 109
wildcards, 117120, 119, 121
detector state in voice port tests, 207
dhcp-config mode, 291
DHCP relay command, 292
DHCP (Dynamic Host Control Protocol) services
monitoring and troubleshooting,
292293, 332333
for voice, 290292
dial peer 0, 112
dial-peer command
FSX ports, 184
H.323, 228
dial-peer cor custom command, 374
dial-peer voice command, 214
H.323, 227
POTS, 109
SIP, 270
tool bypass, 385386
VoIP, 109
dial-peer voice 911 pots command, 214
dial peers
gatekeeper interoperations, 410411
inbound rules, 111112
outbound rules, 112113
POTS, 372, 381
T1 CAS, 194
wildcards, 117121, 119, 121
dial-plan digit manipulation, 104, 123
digit stripping, 123124
forwarding last X digits, 124125
number substitution, 126127
prefixes, 125
translation rules and profiles, 127132
verifying, 132134
dial plan path-selection process, 104
call routing, 108111, 110111
International Numbering Plan,
113114, 114
NANP, 114116, 115
private plans, 116117
PSTN, 113
site-code dialing, 122123, 122
strategies, 111113
voice call types, 104108, 105108
wildcards, 117121, 119, 121
Dial tone informational signals, 38
dial-type command, 49, 183
Dialed Number Identification Service (DNIS)
interfaces, 111112
DID (direct inward dial)
inbound ports, 184187, 185
PSTN, 116
ranges and extensions, 122
Differentiated Services (DS) byte,
449450, 450
Differentiated Services Code Point (DSCP), 445,
451453, 451, 514
DiffServ QoS model, 448449, 453
congestion avoidance, 455
congestion management, 453455
link-efficiency techniques, 457459, 458
traffic policing and shaping, 455457,
456457
DiffServ ToS/DS byte, 449450, 450
digit-collection methods, IP phones,
305306
digit-drop keyword, 355356
digit manipulation. See dial-plan digit
manipulation
digit stripping, 123124
digital ports, 56
configuring, 6365
CUCM, 201202, 202
DSP profiles, 199200
SCCP, 200201
T1 CAS to analog cross-connect,
191195, 191
T1 PRI, 195198, 195
digital signal processors (DSPs), 18, 146, 156159
chipsets, 157160
CUCM configuration, 201202, 202
delay from, 152
farms, 198199, 198
profiles, 199200
SCCP communications, 200201
status, 205206
summary, 210
voice gateway functions, 146147, 147
written lab, 211
digital telephones, 3
digital trunks, 54
digital voice, 51
deregulation digital voice 539
bindex.indd 539 9/20/11 1:38:39 PM
digital voice (continued)
analog-to-digital conversion, 5155, 5254
framing, 5758, 58
multiplexing, 5657, 57
physical transport, 59, 5960
port configuration, 6365
port types, 56
signaling, 6063
summary, 66
dir flash command, 298299, 343
direct inward dial (DID)
inbound ports, 184187, 185
PSTN, 116
ranges and extensions, 122
direct-inward-dial command, 185186
directory aliases, 299
discard-class command, 491
disconnect timer, 245
discovery messages in H.323, 399400
discrete signals, 52
distributed call-control systems, 93
distributed services deployment model, 21, 21
DLCX (delete connection) command, 256
DNIS (Dialed Number Identification Service)
interfaces, 111112
DNs
configuring, 301302
SCCP, 308
SIP voice register, 307, 310311
DNS servers, 291292
dollar signs ($) in regular expressions,
128129
domain names, 87, 291292
DOWN ephone extension state, 336337
drop option, 493
drop precedence, 452
dropped calls, 159
ds0-group command
MGCP, 259
T1 CAS circuits, 64, 193194
DSCP (Differentiated Services Code Point), 445,
451453, 451, 514
dscp command
class maps, 491
dscp-based command, 498
DSCP or IP Precedence Value setting, 499
DSCP-to-CoS mappings, 514
dspfarm_assist keyword, 309
DSPs. See digital signal processors
(DSPs)
DTMF (dual-tone multi-frequency),
36, 37
dtmf command, 49
dtmf-package packages, 258259, 356
dtmf-relay command, 354, 356
DTMF relay support, 354
H.323, 354355
MGCP, 356
SIP, 355356
dual-bucket traffic, 502504, 502503
dual-line phones, 315317
dual-tone multi-frequency (DTMF), 36, 37
dynamic auto trunk mode, 288
dynamic desirable trunk mode, 288
dynamic gatekeeper discovery, 400
Dynamic Host Control Protocol (DHCP)
services
292293, 332333
for voice, 290292
E
E.164 standard, 113114, 114
E wire in E&M signaling, 42
E1 ports, 56
E&M ports and signaling, 35, 4142
configuring, 50
line-seizure, 4346, 4446
physical wiring types, 4243
trunks, 187188, 187
E911 calling
ports, 213
trunks, 188189, 189
X11 services, 116
Early Media in H.323, 226, 226
early-offer forced command, 425
early offer in SIP, 237, 237
earth wire in E&M signaling, 4243
echo and echo cancellation, 148149
echo-cancel command, 149
echo return loss (ERL) levels,
207, 209
ECM (error correction mode), 361362
ecm command, 362
ecm disable command, 361
ECN (explicit congestion notification), 451
540 digital voice ECN (explicit congestion notification)
bindex.indd 540 9/20/11 1:38:40 PM
edge devices, 3
EF (Expedited Forwarding) PHB, 452
egress interface traffic, 445
802.1Q trunk links, 289
802.3af standard, 284285
email, TIFF-attached, 366
emergency calls
ports, 213
trunks, 188189, 189
X11 services, 116
en-bloc digit collection, 305306
encapsulation, trunk, 288
encapsulation ppp command, 510
encoding process in analog-to-digital conversion,
5354, 54
encryption
payloads, 83
SIP passwords, 240
end command, 213
endpoints, 11
analog-to-IP adapters, 15
codec issues, 164
gatekeepers for, 90
private plans, 117
SIP
availability, 239
capabilities, 237239, 237238
locations, 237
video phones and tablets, 14, 14
wired IP phones, 1213
enterprises
AutoQoS for, 483, 484
EPCF (endpoint configuration) command, 256
ephone-config mode, 318
ephone-dn command, 344
ephone-DNs
dual- and octo-lines, 315317
SCCP line configuration, 311313,
313314
two with one number, 314315
ephone-type command, 309
ephones, 307
button options. See button separator options
for ephone
directory number, 308
extension states, 336339
registration states, 335336
SCCP configuration, 308310
state, 334335, 344345
ERL (echo return loss) levels, 207, 209
error correction mode (ECM), 361362
ESF (Extended Super Frame), 58, 58
exceed-action for token buckets, 502505,
502503
excess burst (Be) rate, 506
exit command, 213
expansion line (x) ephone button separator,
319, 324
Expedited Forwarding (EF) PHB, 452
expires timer, 245
explicit congestion notification (ECN), 451
Extended Super Frame (ESF), 58, 58
extension field in RTP headers, 80
extension states for ephones, 336339
extensions in PBX systems, 10, 317
F
f (feature ring) ephone button separator, 318
f8 cipher mode, 83
fallback, 87
MGCP-to-H.323, 370372, 371
WAN-to-PSTN, 369370, 369
fallback keyword, 360
farms, DSP, 198199, 198
fast command, 424
fast start in H.323, 226, 228229
fast-to-slow signaling, 424425
fax protocol t38 command, 359
fax rate command, 360
fax-relay command, 361
fax-relay ans-disable command, 362
fax-relay sg3-to-g3 command, 362
fax transmission package, 257
faxes
fax relay, 357359, 358
MGCP settings, 364, 364
pass-through, 364365
SIP and H.323 settings, 360
T.37 store-and-forward, 365367, 366
T.38, 362363
feature licenses for CUCM Express, 296
feature ring (f) ephone button separator, 318
FIFO (first-in first-out) queuing
mechanism, 453
filters, low-pass, 52
edge devices filters, low-pass 541
bindex.indd 541 9/20/11 1:38:41 PM
firewalls in VoIP, 80
firmware load command, 307
firmware load files, 303304, 304
first-in first-out (FIFO) queuing
mechanism, 453
fixed-bytes setting, 230
fixed delay, 151152
flex option, 160
flow option, 491
Foreign Exchange Office (FXO) ports, 35, 40
configuring, 4950, 180184, 180
outbound, 184187, 185
Foreign Exchange Station (FXS) ports, 34
configuring, 4749, 180184, 180
inbound, 184187, 185
forward-digits command, 124125, 214
forward-digits all command, 214
forward-digits shutdown command, 191
FQDNs (fully qualified domain
names), 406
fr-de option, 491
fr-dlci option, 491
Frame Relay, 510512
frame-relay fragment command, 511
frames
compression techniques, 457458, 458
digital voice, 5758, 58
frame errors and packet loss, 443
size information, 165166, 166
framing command, 63, 192
framing esf command, 213
FRF.12, 510511
fully qualified domain names (FQDNs), 406
FXO (Foreign Exchange Office) ports, 35, 40
configuring, 4950, 180184, 180
outbound, 184187, 185
fxr-package package, 257
FXS (Foreign Exchange Station) ports, 34
configuring, 4749, 180184, 180
inbound, 184187, 185
G
G.711 codec, 5455, 154
G.711mu-law and G.711a-law, 364
G.722 codec, 155
G.723.1 codec, 154155
G.726 codec, 155
G.728 codec, 155
G.729 codec, 155156
gatekeeper command, 406
Gatekeeper Confirm (GCF) message, 399400
Gatekeeper Reject (GRJ) message, 399400
Gatekeeper Request (GRQ) message,
399400
gatekeepers. See H.323 gatekeepers
gateway command, 411
gateway force option, 362
gateways. See voice gateways
GCF (Gatekeeper Confirm) message,
399400
generic traffic shaping (GTS), 506
geographical diversity deployment model,
2223, 23
glare, 40
gm-package packages, 258259
Goodbye RTCP packets, 82
GRJ (Gatekeeper Reject) message, 399400
ground-start signaling, 4041, 41
ground wire in supervisory signaling, 3839
groundstart signaling, 181
group 3 fax machines, 357
growth, dialing plans for, 113
GRQ (Gatekeeper Request) message,
399400
GSM Full Rate (GSMFR) codec, 156
GTS (generic traffic shaping), 506
gw-controlled command, 368
gw-priority command, 407
gw-type-prefix command, 408, 430
H
H.225 protocol
description, 85
settings, 231232
timers, 232233
H.235 protocol, 85
H.245 protocol, 85
H.323 gatekeepers, 8991, 8990, 396
address translation, 398
bandwidth control, 398
bandwidth management, 399
call admission control, 398, 411414, 413
call authorization, 398399
call management, 399
configuration overview, 405, 405
debug ras command, 415416
dial peers, 410411
542 firewalls in VoIP H.323 gatekeepers
bindex.indd 542 9/20/11 1:38:42 PM
enabling, 411
interface commands, 409410
overview, 396
RAS messages, 399400
bandwidth, 404405
call admission, 400401
location, 402404, 402404
registration, 400
resource availability, 404
sample network, 401402, 401
show gatekeeper calls command, 414415
show gatekeeper endpoints
command, 415
show gatekeeper status command, 414
signaling, 399401
summary, 427
technology prefixes, 408409
zones
local, 406
managing, 397398, 397
prefixes, 407
remote, 406407
H.323 protocol
codec preference, 229231
DTMF relay, 354355
fast and slow start connections, 228229
fast-to-slow signaling, 424425
fax settings, 360
gatekeepers. See H.323 gatekeepers
gateway configuration, 227228, 228
H.225 settings, 231233
MCU, 9192
overview, 8485, 224226, 224226
proxy servers, 91
session transport mode, 231
show gateway command, 234
show h323 gateway h225 command, 234236
T.38 fax relay with, 359360
virtual gateway addresses, 233234, 233
H.450 protocol, 85
h225 timeout setup command, 233
h225 timeout tcp call-idle command, 232
h225 timeout tcp establish command, 232
h245-alphanumeric method, 355
h245-signal method, 355
h323-gateway voip command, 409
h323-gateway voip interface
command, 431
h323-id command, 410
hardware
codec compatibility, 163
CUCM Express, 295
voice gateways, 1819, 19
header fields in RTP, 8081
high-complexity codec calls, 157160
high-speed fax relay redundancy, 359
house wiring, 5
hs-redundancy option, 359360
hs_redundancy option, 362
huntstop command, 315316
huntstop channel command, 316
hybrid systems, 323
hyphens (-) in dates, 328
I
id keyword, 410
identifier setting, 200
IDLE ephone extension state, 336337
IEEE 802.3af standard, 284285
IETF (Internet Engineering Task Force)
protocols, 85
IFP (Internet fax packets), 358
ignored packets, 443
iLBC (Internet Low Bit Rate Codec), 156
ILP (inline power) functionality, 283286
immediate signaling type, 50
immediate-start E&M signaling, 44, 44
impedance
FXS ports, 48
mismatches, 149
impedance command, 207
implementation phase in AutoQoS, 488
inbound calls, translation profiles for, 132
inbound dial-peer rules, 111112
inbound SIP transport protocols, 243244
incoming called-number command,
111112, 185
informational signaling, 35, 3738
inhibit command, 362
inject-tone state, 207
inline power (ILP) functionality, 283286
input-interface option, 491
H.323 protocol input-interface option 543
bindex.indd 543 9/20/11 1:38:42 PM
input queue drops, 443
installation phase in AutoQoS, 483
Integrated Services Digital Network (ISDN)
BRI, 56, 61
with H.323, 224, 224
with MGCP, 253254, 254
PRI, 62, 6465, 195198, 195
with SIP, 236, 236, 247249, 247
Intelligent Power Management (IPM), 285286
inter-networking of services deployment model,
22, 22
interactive voice response (IVR) services, 354
intercluster trunk calls, 108, 108
interdigit timeout, 118
interexchange networks, 9
interface binding, 233234, 233
interface command for voice gateways, 409
interface commands for H.323 gatekeepers,
409410
interface multilink command, 509
interface serial command, 213, 521
internal T1 CAS option, 63
international calling, 8, 9
international networks, 9
International Numbering Plan,
113114, 114
Internet Engineering Task Force (IETF) protocols,
85
Internet fax packets (IFP), 358
Internet Low Bit Rate Codec (iLBC), 156
Internet Protocol Telephony (IPT), 11
Internet Speech Audio Codec (iSAC), 156
Internet Telephony Service Providers
(ITSPs), 12, 154, 416, 421422,
421422
interoffice trunks, 78, 8
interwork command, 424
interzone keyword command, 412
IntServ QoS model, 447449
Invite SIP retry type, 246
IOS licenses, 296
IP addresses
binding SIP sources to, 247
conflicts, 293
CUBE, 417
CUCM Express, 300301, 301
DHCP
292293, 332333
for voice, 290292
ip command for class maps, 491
ip dhcp excluded address command, 291
ip dhcp pool command, 291
ip-helper address command, 292
IP networks
bandwidth, 441442
jitter, 442
packet loss, 443444, 443
SIP configuration, 239241, 240
voice issues, 441444, 443
voice/video on, 440441
IP phones
digit-collection methods, 305306
keepalive timers, 329
power options, 282286
restart vs. reset, 329332
soft phones, 1314
VLAN configuration, 288290, 289
wired, 1213
wireless, 13
IP Precedence in ToS byte, 449450, 450
ip source-address command, 307, 343
IP-to-IP gateways, 416
IP to PSTN translations, 87
IP/UDP header size, 165
IP voice bandwidth consumption, 164
codec bit rate, 166167
packet and frame size information,
165166, 166
providing, 441442
IPM (Intelligent Power Management), 285286
IPSec, 166, 166
IPT (Internet Protocol Telephony), 11
iSAC (Internet Speech Audio
Codec), 156
ISDN. See Integrated Services Digital Network
(ISDN)
isdn incoming-voice command, 197
isdn incoming-voice voice command, 213
isdn supp-service name calling
command, 248
isdn switch-type command, 64, 196
isdn switch-type primary-ni
command, 212
ISR router series, 19
ITSPs (Internet Telephony Service Providers), 12,
154, 416, 421422, 421422
ITU-T T.38 fax relay, 358359, 358
IVR (interactive voice response)
services, 354
544 input queue drops IVR (interactive voice response)services
bindex.indd 544 9/20/11 1:38:43 PM
J
jitter, network, 152153, 152, 442
K
keepalive command, 329
keepalive timers, phone, 329
key systems, 1011, 311312
L
languages in CUCM Express, 325328
LATAs (Local Access and Transport Areas), 8
Layer 2 CoS, 459460, 459460
Layer 2 header size, 165
LCF (Location Confirm) message,
402404, 403
LdapDirectories.xml files, 334
leases, DHCP, 291292
legacy PBX, 187188, 187
LFI (link fragmentation and interleaving),
457459, 509512
licensing CUCM Express, 296297
lightweight registration messages, 400
Lincoln, Abraham, 443
line command, 63
line-package packages, 256, 258
line seizure, 40
E&M signaling, 4346, 4446
ephone extension state, 336338
linecode command, 63, 192
linecode b8zs command, 213
link efficiency, 457, 508509
compression, 457458, 458, 512513
LFI, 509512
link fragmentation and interleaving (LFI),
457459, 509512
listener echo, 149
lists in regular expressions, 128
LLQ (Low Latency Queuing), 455
load files in CUCM Express, 303304, 304
Local Access and Transport Areas (LATAs), 8
local calls, 105, 105
local loop, 5, 5
local name segment in MGCP, 87
local zones, 406
locales in CUCM Express, 325328
Location Confirm (LCF) message,
402404, 403
location messages in RAS, 402404, 402404
Location Reject (LRJ) message, 402403, 403
Location Request (LRQ) message,
402404, 403
log option, 493
loop-start signaling, 3940, 39, 181
loopback interface, 233234, 233
loopback state in voice port tests, 207
loopstart signaling, 181
low-bit-rate keyword, 356
low-complexity codec calls, 157160
Low Latency Queuing (LLQ), 455
low-pass filters, 52
low-speed fax relay redundancy, 359
low-speed WAN connections, 457
LRJ (Location Reject) message, 402403, 403
LRQ (Location Request) message,
402404, 403
ls-redundancy option, 359360
ls_redundancy option, 362
M
m (monitor line) ephone button separator,
318320, 320
M wire in E&M signaling, 42
mac-address command, 310, 344
MAC addresses for ephones, 310
magnet wire in E&M signaling, 4243
management, dialing plans for, 113
map-class frame-relay command, 511
maps
class, 490492
CoS-to-DSCP, 514516, 515, 519520
policy, 493495
Mark Probability Denominator setting, 499
marker field in RTP headers, 81
marking traffic, 445
match-all command, 490, 492
match-any command, 490, 492
max-dn command, 302303, 343
max-ephones command, 302, 343
max-forwards command, SIP, 246
max-pool command, 307
maximum sessions command, 199200, 367
maximum sessions hardware command, 200
Maximum Threshold (Number Of Packets)
setting, 499
MC (Multipoint Controller), 92
MCU (Multipoint Control Unit), 9192
MDCX (modify connection) command, 256
jitter, network MDCX (modify connection) command 545
bindex.indd 545 9/20/11 1:38:44 PM
Mean Opinion Score (MOS) test, 161162
media flow-around, 418419, 419, 423
media flow-around command, 423
media flow-through, 418, 418
Media Gateway Control Protocol
(MGCP), 8788
DTMF relay, 356
fallback, 87, 370372, 371
fax relay with, 362363
fax settings, 364, 364
overview, 253254, 254
residential gateways, 254255, 255,
257259, 258
show ccm-manager command, 264265
show mgcp command, 261263
show mgcp profile command, 260261
show mgcp statistics command, 263264
trunking gateways, 255257, 255,
259260, 260
media termination point (MTP), 146147
medium-complexity codec calls, 157160
mgcp call-agent command, 257, 259
mgcp dtmf-relay voip codec
command, 356
mgcp fax rate command, 363
mgcp fax t38 command, 362
mgcp fax t38 inhibit command, 365
mgcp modem passthrough voip mode nse
command, 365
mgcp modem relay voip mode command, 368
mgcp package-capability command, 257258
mgcp package-capability rtp-package
command, 365
mgcp timer nse-response t38
command, 363
mgcp tse payload command, 363
Minimum Threshold (Number Of Packets)
setting, 499
minus signs (-) in regular expressions, 128
MLP (multilink PPP), 509510
mls qos command, 482
mls qos map command, 516
mls qos trust command, 513514
mls qos trust cos command, 519
mode cme command, 306
modem relay command, 368
modems, 367
pass-through, 367368
relay, 368
modular QoS CLI (MQC) method, 447
policies using, 488498, 489,
520521
show class-map command, 496
show policy-map command, 496
show policy-map interface command,
496497
MOH (music on hold), 150
monitor line (m) ephone button separator,
318320, 320
monitoring DHCP service, 292293
MOS (Mean Opinion Score) test,
161162
MP (Multipoint Processor), 92
mpls option, class maps, 491
MQC. See modular QoS CLI (MQC) method
MTP (media termination point), 146147
mu-law codecs, 365
multilink PPP (MLP), 509510
multiplexing, 54, 5657, 57
Multipoint Control Unit (MCU), 9192
Multipoint Controller (MC), 92
Multipoint Processor (MP), 92
music on hold (MOH), 150
N
named service events (NSE), 359
NANP (North American Numbering Plan),
114116, 115, 190
narrowband communication, codecs for, 154
narrowband sampling, 148, 148
national calling PSTN, 8, 9
National Destination Code (NDC), 113114
National Institute of Standards and Technology
(NIST), 293
NBAR (Network-Based Application Recognition)
feature, 475
NDC (National Destination Code), 113114
netflow-sampler option, 493
Network-Based Application Recognition (NBAR)
feature, 475
network capacity in codec selection, 164
network delay, 151152
network infrastructure, 20
network jitter, 152153, 152
network-locale command, 326327
network locales in CUCM Express, 325328
network-number command, 127
Network Time Protocol (NTP), 290, 293
never command in PoE, 285
9951 IP video phone, 14
NIST (National Institute of Standards and
Technology), 293
546 Mean Opinion Score (MOS) test NIST
bindex.indd 546 9/20/11 1:38:45 PM
no command in QoS policy maps, 493
no digit-strip command
CAMA, 191
forwarding digits, 124125
tool bypass, 385
WAN-to-PSTN fallback, 369
no huntstop command, 315, 321
no maximum sessions command, 200
no shutdown command
CAMA, 190191, 213
DSP profiles, 199
FSX ports, 182
H.323 service, 411
local zones, 406
voice port tests, 207
VoIP service, 227
noise, background, 149151, 150
non-standard facilities (NSF) code, 362
none keyword in SIP, 248
normal ring (:) ephone button separator, 318
North American Numbering Plan (NANP),
114116, 115, 190
not option for class maps, 491
NPD (Numbering Dialing Plan) in emergency
calls, 190
NSE (named service events), 359
nse keyword
T.38 fax relay, 359
NSF (non-standard facilities) code, 362
nsf option, 362
NTFY (notify) command, 256
NTP (Network Time Protocol), 290, 293
ntp server command, 293
NULL ciphers, 83
NULL rules, 129
num-exp command, 126
number command, 310311
Number not in service informational
signals, 38
Numbering Dialing Plan (NPD) in emergency
calls, 190
Nyquist, Harry, 52
Nyquist Sampling Theorem, 52, 52
O
o (overlay line) ephone button separator,
319323
octo-line phones, 315317
off-hook state
pulse dialing, 36
supervisory signaling, 3839, 39
off-net calls, 106, 106
off-premises extension (OPX), 180
off-ramp gateways, 366367, 366
on-hook state
pulse dialing, 36
on-net calls, 105, 105
on-net-to-off-net calls, 106, 107
on-ramp gateways, 366367, 366
one-stage dialing, 186
operation command, 50
option command in DHCP, 291
OPX (off-premises extension), 180
out-of-band signaling, 61
outbound calls, translation profiles for, 132
outbound dial peers
to PSAP, 214
to PSTN, 214
rules, 112113
outbound SIP transport protocols, 243244
overlay line (o) ephone button separator,
319323
overlay with call waiting (c) ephone button
separator, 319, 323324
override keyword in trust boundaries, 515
overruns, packet loss from, 443
P
packages, 256259
packet loss concealment (PLC)
methods, 153
packet option for class maps, 491
packetization delay, 152
packets
information for, 165166, 166
loss of, 153, 443444, 443
packets per second (PPS), 165, 167168
padding field in RTP headers, 80
PAM (pulse-amplitude modulation), 51
parent maps, 494
parentheses ()
destination pattern wildcards, 120
in regular expressions, 128130
pass-through
fax, 364365
modems, 367368
pass-through keyword, 360
no command in QoS policy maps pass-through keyword 547
bindex.indd 547 9/20/11 1:38:45 PM
pass-through dscp keywords, 514
passwords in SIP, 240
Payload Encryption packets, 83
payload type field in RTP headers, 81
PBX (private branch exchange) systems, 10
E&M trunks with, 187188, 187
extensions, 317
PBX-to-PBX calls, 107, 107
PBX-to-PBX switch connections, 40
PBX-to-PSTN switch connections, 40
PCM (pulse-code modulation),
53, 53, 55, 154
peak information rate (PIR), 504
peak keyword, 506
peer-to-peer architecture, 8485
peer-to-peer protocols
H.323, 224
SIP, 236
UDP, 86
per-hop behaviors (PHB), 451453
percent signs (%) in destination pattern
wildcards, 120
Perceptual Evaluation of Speech Quality (PESQ)
measure, 163
Perceptual Objective Listening Quality Analysis
(POLQA), 163
Perceptual Speech Quality Measure (PSQM),
162163
periods (.)
destination pattern wildcards,
118119, 119
PESQ (Perceptual Evaluation of Speech Quality)
measure, 163
phantom power, 283
PHB (per-hop behaviors), 451453
phone configuration files, 304305
phone registrations, 332
phone switches, 34
phones, IP. See IP phones
physical transport in digital voice, 59, 5960
physical wiring in E&M signaling, 4243
PIR (peak information rate), 504
PLAR (Private Line Automatic Ringdown),
180184
PLC (packet loss concealment) methods, 153
plus signs (+)
PoE (Power over Ethernet) switches, 283285
police command, 493, 504
policy-map command, 489, 492493
policy-map voice-policy command, 521
policy maps, 493495
QoS, 489
service policies, 495
POLQA (Perceptual Objective Listening Quality
Analysis), 163
pools
DHCP, 291
voice register, 311
port command
POTS dial peer, 109
ports
analog voice
configuring, 4750
types, 3435
digital. See digital voice
inbound dial-peer rules, 112
POTS dial peers, 108109, 372, 381
power
IP phones, 282286
power bricks, 282283
power injectors, 283
power inline command, 285
Power over Ethernet (PoE) switches, 283285
powered patch panels, 283
ppp multilink command, 510
ppp multilink fragment-delay
command, 510
ppp multilink-group command, 510
ppp multilink interleave command, 510
PPP multilink LFI configuration, 478
PPS (packets per second), 165, 167168
PQ (priority queuing), 454455
prec-based keyword, 498
precedence option, 491
predictor compression method, 458
preference command
ephone DNs, 314315
overlay lines, 321
tool bypass, 385
WAN-to-PSTN fallback, 369
prefix adding, 125
prefix command, 125
CAMA, 191
tool bypass, 385
prefixes
technology, 408409
zone, 407
PRI (Primary Rate Interface), 62, 6465,
195198, 195
pri-group timeslots command, 65, 213
primary-5ess command, 65
548 pass-through dscp keywords primary-5ess command
bindex.indd 548 9/20/11 1:38:46 PM
primary-qsig command, 65
Primary Rate Interface (PRI), 62, 6465,
195198, 195
priorities
call-processing units, 201
gateways, 408
IP Precedence, 450
traffic marking, 445
priority percent command, 521
priority queuing (PQ), 454455
priority setting
SCCP, 200
private branch exchange (PBX) systems, 10
E&M trunks with, 187188, 187
extensions, 317
Private Line Automatic Ringdown (PLAR),
180184
private numbering plans, 116117
private switches, 4
private telephone systems, 2, 910
profiles
dial-plan digit manipulation, 127132
DSP, 199200
translation, 131132
propagation delay, 151152
protocol internetworking, 417
protocol interoperation, 422423, 423
protocol option for class maps, 491
proxies, CUBE, 417
proxy servers
H.323, 91
SIP, 8687, 246247
PSAPs (Public Safety Answering Points)
CAMA trunks, 189
outbound dial peers to, 214
PSQM (Perceptual Speech Quality Measure),
162163
public switched telephone network
(PSTN), 3, 89, 9
central office, 45, 4
local loop, 5, 5
with MGCP, 253254, 254
numbering plan, 113
outbound dial peers to, 214
redundancy, 385386
with SIP, 247248, 247
termination, 146
public telephone systems, 2
pulse-amplitude modulation (PAM), 51
pulse-code modulation (PCM),
53, 53, 55, 154
pulse command, 49
pulse dialing, 36, 36
pulse interval command, 182
PVDM (C549 DSP chipset), 157158
PVDM2 (C5510 DSP chips), 157,
159160
PVDM3 (C5510 DSP chips), 157, 159
Q
Q.921 signaling, 253
Q.931 signaling, 62, 253
Q signaling (QSIG), 6263
QoS. See Quality of Service (QoS)
qos-group option, 491
QSIG (Q signaling), 6263
quality considerations, 147148
background noise, 149151, 150
echo and echo cancellation, 148149
network delay, 151152
network jitter, 152153, 152
packet loss, 153
Quality of Service (QoS), 439440
AutoQoS. See AutoQoS
baseline models, 461463, 462
Best-effort model, 447, 449
class-based, 488489
class maps, 490492
command line interface, 446
congestion management, 453455, 463,
498500, 499
CoS-to-DSCP mappings, 514516, 515
DiffServ model, 448, 453459
DiffServ ToS/DS byte, 449450, 450
DSCP method, 451453, 451
exam essentials, 464465, 517
IntServ model, 447448
Layer 2 classification, 459460, 459460
link efficiency techniques, 457458, 458,
508513
mitigating IP network voice issues,
441444, 443
models comparison, 448
policies
primary-qsig command Quality of Service (QoS) 549
bindex.indd 549 9/20/11 1:38:47 PM
Quality of Service (QoS) (continued)
considerations, 445447
using MQC, 488498, 489
policy maps, 493495
review questions, 466471, 522527
settings, 153
496497
summary, 464, 517
three-step process, 444445
traffic policing and shaping, 455457,
456457, 500508, 502503, 505
trust boundaries
voice/video on IP networks, 440441
written lab, 465, 472, 518, 528
quantization in analog-to-digital
conversion, 53, 53
question marks (?)
queue-limit option, 493
queuing
delay, 152
traffic, 445, 456
R
R wire in E&M signaling, 42
R1 wire in E&M signaling, 42
RAC (Resource Availability
Confirmation), 404
RAI (Resource Availability Indicator), 404
random-detect command, 493, 498
Random Early Detection (RED) tool,
455, 498
RAS. See Registration Admission and Status
(RAS) messages
RBS (robbed-bit signaling), 60
RCF (Registration Confirm) message, 400
Real-time Transport Control Protocol (RTCP),
8182
Real-time Transport Protocol (RTP),
17, 7881
header fields, 8081
header size, 165
Receiver off-hook informational signals, 38
Receiver Report packets, 81
RED (Random Early Detection) tool,
455, 498
redirect-called type, 131
redirect servers
maximum, 246247
SIP with ISDN, 247249, 247
redundancy
gateways, 407408
PSTN, 385386
redundancy keyword, 367
register device-name command, 201
register servers, 87
REGISTERED ephone registration state,
335339
registrar server command, 306
Registration Admission and Status (RAS)
messages, 398
bandwidth, 404405
call admission, 400401
location, 402404, 402404
registration, 400
resource availability, 404
Registration Confirm (RCF) message, 400
registration messages, 400
Registration Reject (RRJ) message, 400
Registration Rejected message, 302, 303
Registration Request (RRQ) message, 400
registration states for ephones, 335336
registrations, phone, 332
regular expressions, 128
relay
fax, 357359, 358, 362363
modem, 368
relay state in voice port tests, 207
remote keyword, 412
remote zones in H.323, 406407
Reorder informational signals, 38
repeaters for analog signal, 51
Replay Protection packets, 83
reset, IP phone, 330332
reset command, 330
reset all command, 330
residential gateways, 254255, 255, 257259, 258
Resource Availability Confirmation
(RAC), 404
Resource Availability Indicator (RAI), 404
resource availability messages, 404
Resource in Progress (RIP), 404
Resource Reservation Protocol (RSVP)
CAC, 420, 420
QoS, 448
550 Quality of Service (QoS) Resource Reservation Protocol (RSVP)
bindex.indd 550 9/20/11 1:38:48 PM
Response retry type, 246
restart, IP phone, 329330
restart command, 330
restart all command, 329
retries, SIP, 245246, 270271
retry invite command, 271
retry response command, 271
right-to-use licenses, 297
Ring-back informational signals, 38
ring cadence command, 182
ring frequency command, 182
ring number command, 183
RINGING ephone extension state, 336, 338
ringing time, 149
RIP (Resource in Progress), 404
robbed-bit signaling (RBS), 60
rotary dialing, 36, 36
routers
DHCP, 291292
Enterprise, 483, 484
VoIP on, 475479, 476
RQNT (request for notification) command, 256
RRJ (Registration Reject) message, 400
RRQ (Registration Request) message, 400
RSIP (restart in progress) command, 256
RSVP (Resource Reservation Protocol)
CAC, 420, 420
QoS, 448
RTCP (Real-time Transport Control Protocol),
8182
RTP (Real-time Transport Protocol), 17, 7881
header fields, 8081
header size, 165
rtp-nte method, 355356
rtp-package packages, 258259
S
s (silent ring) ephone button separator, 318
sampling
analog-to-digital conversion, 5253, 52
SB (signal battery) wire, 4243
SB-ADPCM (Sub-Band Adaptive Differential
Pulse Code Modulation), 155
SBCS (Smart Business Communications System)
suite, 12
SC (Subscriber Code), 113115
SCCP. See Skinny Client Control Protocol (SCCP)
sccp cucm group command, 201
sccp local command, 200
SDP (Session Description Protocol)
with MGCP, 254
with SIP, 237238, 238
second-number strip command, 127
secure codec option, 160
secure RTP (sRTP), 83, 242243
secure RTP packages, 257
security
CUBE deployment, 417
SIP, 241243
voice packets, 166, 166
Segmented Integer Counter Mode cipher mode, 83
SEIZE ephone extension state, 336338
seizure, line, 40
E&M signaling, 4346, 4446
ephone extension state, 336338
Sender Report packets, 81
separators, button. See button separator options
for ephone
sequence field in RTP headers, 81
serialization delay, 152, 458
service policies
policy maps, 495
QoS, 489
service-policy command, 489, 493495, 513
service-policy output voice-policy command,
521
service-type mgcp command, 257
Session Description Protocol (SDP)
with MGCP, 254
with SIP, 237238, 238
Session Initiation Protocol (SIP), 17, 8587
basic configuration, 269270
call signaling, 12
CUCM signaling, 305307
delayed-to-early-offer signaling, 425
DTMF relay, 355356
endpoints
availability, 239
capabilities, 237239, 237238
locations, 237
ephone-DN line compatibility, 312
fax relay with, 359360
fax settings, 360
IP voice gateways, 239241, 240
overview, 236237
register servers, 87
secure communications, 241243
sessions, 239
show sip-ua calls command, 252253
show sip-ua retry command, 252
Response retry type Session Initiation Protocol (SIP) 551
bindex.indd 551 9/20/11 1:38:48 PM
Session Initiation Protocol (SIP) (continued)
show sip-ua statistics command,
249251
show sip-ua status command, 251
show sip-ua timers command, 252
timers and retries, 270271
voice gateway settings, 243
binding sources to IP addresses, 247
inbound and outbound transport
protocols, 243244
ISDN call-ID blocking, 249
ISDN interoperation settings, 247248, 247
proxy and redirect servers, 246247
signaling retries, 245246
signaling timers, 244245
voice register pools, 311
session keyword, gatekeepers, 412
session protocol sipv2 command,
241, 270, 355
session target command
SIP, 270
tool bypass, 385
VoIP dial peer, 109
session target ipv4 command, 227, 241
session target ras command, 410
session target sip-server command, 241
session transport mode in H.323, 231
session transport tcp calls-per-connection
command, 231
session transport udp command, 231
sessions, SIP, 239
set option in QoS policy maps, 493
7921G wireless IP phone, 13
7925G and 7925G-EX wireless IP
phones, 13
7985G IP video phone, 14
SF (Super Frame), 5758, 58
SG (signal ground) wire, 4243
SG3 (Super Group 3) fax transmissions, 362
shape command, 493, 506
shared lines with ephone-DN, 312313, 313314
show auto discovery qos command, 485487
show auto qos command, 477478
show call active voice brief command,
425426
show call history voice brief command, 426
show ccm-manager command, 264265
show ccm-manager fallback-mgcp command, 372
show connection all command, 195
show controller command, 205
show dial-peer command, 132133
show dialplan number command, 133134
show ephone command, 321323, 335336,
338, 344
show frame-relay fragment command, 512
show gatekeeper calls command,
414415
show gatekeeper endpoints command, 415
show gatekeeper status command, 414
show gateway command, 227, 234
show h323 gateway h225 command,
234236
show ip dhcp binding command, 292293
show ip dhcp conflict command, 293
show mgcp command, 261263
show mgcp profile command, 260261
show mgcp statistics command, 263264
show mls qos maps command, 516
495497, 507508
show power inline command, 286
show run interface fa0/1
command, 481
show run interface fastEthernet 0/5
command, 480
show sip-ua calls command,
252253
show sip-ua retry command, 252, 271
show sip-ua statistics command,
249251
show sip-ua status command, 251
show sip-ua timers command,
252, 271
show telephony-service tftp-bindings
command, 334
show vlan brief command, 290
show voice dsp command, 160,
205206
show voice port command, 203205
show voice port summary command,
196197
show voip rtp connections command, 426
shutdown command
CAMA, 190191, 213
voice port tests, 207
shutdown forced command, 227
side-car modules, 309
signal battery (SB) wire, 4243
signal command for FSX ports, 181
signal cama command, 213
signal ground (SG) wire, 4243
552 Session Initiation Protocol (SIP) signal ground (SG) wire
bindex.indd 552 9/20/11 1:38:49 PM
signal groundstart command, 49
signal loopstart command, 47
signaling
analog voice, 35
address, 3537, 3637
E&M, 4146, 4446
ground-start, 4041, 41
informational, 3738
supervisory, 3841, 39, 41
call, 17, 17
digital voice, 6063
gatekeeper, 399401
retries, 245246
timers, 244245
signaling forward command, 248
signaling protocols
with CUBE, 419, 419
voice gateway. See voice gateway signaling
protocols
silent ring (s) ephone button separator, 318
silent rings, 310
silent with beep (b) ephone button
separator, 318
simplicity, dialing plans for, 113
single-bucket traffic, 502503,
502503
SIP. See Session Initiation Protocol (SIP)
SIP ITSP, 421422, 421422
sip-notify method, 356
SIP secure (SIPS) mechanism, 241
sip-server command, 241
sip-ua command, SIP, 271
SIPS (SIP secure) mechanism, 241
site-code dialing, 122123, 122
sites in private plans, 117
size of frames, 165166, 166
Skinny Client Control Protocol (SCCP),
17, 88
configuring, 200201
CUCM Express, 300305, 301304
with DSP farms, 198, 198
ephone configuration, 308310
ephone directory number, 308
ephone-DN line configuration, 311313,
313314
individual lines, 317318
phone operation, 343344
slashes (/)
in dates, 328
slow command, 424
slow start connections in H.323, 228229
slow start initiation mode in H.323,
225, 225
small businesses, wired IP phones for, 12
Smart Business Communications System (SBCS)
suite, 12
Smart Phone Control Protocol (SPCP), 12
soft phones, 1314
software-activated voice licensing, 297
software licensing for CUCM Express, 296297
source-address command, 307
source-address mac command, 491
source-bind feature, 247
Source Description packets, 82
source IP addresses in CUCM Express,
300301, 301
SPA 300 and 500 series IP phones, 12
spare phones, 297
SPCP (Smart Phone Control Protocol), 12
square brackets ([])
destination pattern wildcards, 118120, 120
SRST (Survivable Remote Site Telephony), 21
configuring, 376377
with COR, 373, 373
with MGCP, 254
sRTP (secure RTP), 83, 242243
srtp command, 242243
srtp fallback command, 242243
srtp-package package, 257
SSDC5, 43
SSRC (Synchronization Source Identifier) field in
RTP headers, 81
stacker compression method, 458
standards for dialing plans, 113
states, ephones, 344345
extensions, 336339
registration, 335336
static command, 285
station-id command, 182
store-and-forward fax, 365367, 366
stripping, digit, 123124
Sub-Band Adaptive Differential Pulse Code
Modulation (SB-ADPCM), 155
Subscriber Code (SC), 113115
substitution, number, 126127
Super Frame (SF), 5758, 58
Super Group 3 (SG3) fax transmissions, 362
supervisory signaling, 35, 3841, 39, 41
Survivable Remote Site Telephony (SRST), 21
configuring, 376377
with COR, 373, 373
with MGCP, 254
signal groundstart command Survivable Remote Site Telephony (SRST) 553
bindex.indd 553 9/20/11 1:38:50 PM
switches and switchports, 34, 519
inline power, 283286
QoS markings, 519
in voice port tests, 207
VoIP on, 479483, 480
switchport access vlan command, 519
switchport mode access command, 519
switchport mode trunk command, 288
switchport trunk encapsulation
command, 288
switchport voice vlan command, 289, 519
synchronization
CAS, 60
CCS, 61
ISDN, 62
TCP, 498
time, 290, 293
Synchronization Source Identifier (SSRC) field in
RTP headers, 81
system keyword, 367
T
T.30 fax machines, 357358
T.37 fax, 365367, 366
T.38 fax relay
with H.323 and SIP, 359360
with MGCP, 362363
T wildcard, 118, 121, 121
T wire in E&M signaling, 42
T1 circuits
CAS configuration, 6364, 191195, 191
ports, 5657, 57
PRI configurations, 6465, 195198, 195,
212213
T1 wire in E&M signaling, 42
tablets, 14, 14
Tag Control Information (TCI), 459
tagged image file format (TIFF),
365366
tail end hop off (TEHO), 377380,
378, 386
talker echo, 149
Tc (Time Interval) in token buckets, 500
TCI (Tag Control Information), 459
TCL (Tool Command Language) scripts, 366
TCP (Transmission Control Protocol)
RTP with, 79
with SIP, 86
synchronization, 498
tcp-traffic keyword, 498
TDM (time-division multiplexing),
5657, 57
tech-prefix keyword, 410
technology prefixes, 408409
TEHO (tail end hop off), 377380,
378, 386
telephony. See traditional telephony
telephony-service command, 343
telephony service event (TSE) payload
size, 363
termination, PSTN, 146
test voice port command, 206209
test voice translation-rule command,
130, 382
tftp-server flash command, 299300
TFTP servers
CUCM Express as, 298300, 342343
DHCP, 291292
troubleshooting, 333334
tie trunks, 67, 6
TIFF (tagged image file format), 365366
time
CUCM Express format, 328
synchronizing, 290, 293
time-division multiplexing (TDM), 5657, 57
Time Interval (Tc) in token buckets, 500
time-sensitive traffic in end-to-end delays, 442
time to live (TTL) in registration
messages, 400
time zones, 293
timeouts, interdigit, 118
timers
H.225, 232233
keepalive, 329
SIP, 244245, 270271
timers command, 245
timers connect command, 271
timers disconnect command, 271
timers trying command, 271
timeslots command, 259
timestamp field in RTP headers, 81
tip wire in supervisory signaling, 3839
TLS (Transport Layer Security),
86, 241
toll bypass, 377380, 378, 385386
tone suppression for faxes, 362
Tool Command Language (TCL)
scripts, 366
ToS (Type of Service) byte, 449450, 450
total keyword, 412
554 switches and switchports total keyword
bindex.indd 554 9/20/11 1:38:51 PM
touch-tone pads, 36, 37
traditional telephony, 23
edge devices, 3
exam essentials, 24
local loop, 5, 5
phone switches, 34
private systems, 910
PSTN, 89, 9
summary, 2324
trunks, 68, 68
written lab, 25, 32
traffic classification in QoS, 444445
traffic marking in QoS, 445
traffic policing and shaping
class-based, 504508
DiffServ features, 455457, 456457
traffic queuing in QoS, 445
transcoding, 146
translate command, 131
translation-profile command, 132, 386
translation profiles, 131132
translation rules in dial-plan digit manipulation,
127132
Transmission Control Protocol (TCP)
RTP with, 79
with SIP, 86
synchronization, 498
transmission rate for faxes, 360
transparent setting for H.323 codecs, 230
transport
digital voice, 59, 5960
SIP protocols, 243244
Transport Layer Security (TLS) protocol,
86, 241
transport tcp command, 243
triggers for FSX ports, 184
troubleshooting
Cisco phone registrations, 332
DHCP, 292293, 332333
TFTP, 333334
trunk encapsulation command, 288
trunk-package packages, 256, 258259
trunking gateways
overview, 255257, 255
trunks, 6
CAMA, 188191, 189, 213
digital, 54
E&M, 187188, 187
intercluster, 108, 108
interoffice, 78, 8
tie, 67, 6
VLANs, 286288, 287
trust boundaries
trust keyword in AutoQoS, 475476, 479
trying timer, 245
TSE (telephony service event) payload
size, 363
TTL (time to live) in registration messages, 400
tunnels for voice packets, 166, 166
2900 and 3900 series ISR routers, 19
two-stage dialing, 186
type command for ephones, 309
Type of Service (ToS) byte, 449450, 450
U
u-law algorithm, 55
UAC (user agent clients), 86
UAS (user agent servers), 86
UAs (user agents) in SIP, 86, 240
UC500 series products, 12
UDP (User Datagram Protocol), 7980, 86
unconditional keyword, 248
Unified Communications Model overview, 11
applications, 15
call processing agents, 1517, 17
endpoints, 1115, 14
exam essentials, 24
network infrastructure, 2023, 2123
summary, 2324
voice gateways, 1819, 19
written lab, 25, 32
Uniform Resource Locators (URLs), 86
Unity Express voicemail module, 294
UNREGISTERED ephone registration state,
335337
upgrades of key systems, 10
url sips command, 242
URLs (Uniform Resource Locators), 86
User agent clients (UAC), 86
User agent servers (UAS), 86
user agents (UAs) in SIP, 86, 240
User Datagram Protocol (UDP),
7980, 86
touch-tone pads User Datagram Protocol (UDP) 555
bindex.indd 555 9/20/11 1:38:51 PM
user licenses for Cisco phone, 297
user-locale command, 325326
user locales in CUCM Express, 325328
usernames in SIP, 240
V
VAD (Voice Activity Detection), 150151,
150, 156
variable delay, 152
version field in RTP headers, 80
version setting in SCCP, 200
VG200 series appliances, 15
video phones, 14, 14
violate-action command, 505
violate-actions for token buckets, 503505
virtual dial peers in CUCM, 294
virtual gateway addresses, 233234, 233
vlan option for class maps, 491
VLANs
configuring and verifying, 288290, 289
trunks, 286288, 287
voice
analog. See analog voice
clipping, 150
digital. See digital voice
IOS licenses, 296
IP network issues, 441444, 443
Voice Activity Detection (VAD), 150151, 150, 156
voice backup paths, 368369
COR, 372377, 373
SRST, 376377
voice call types
intercluster trunk, 108, 108
local, 105, 105
off-net, 106, 106
on-net, 105, 105
on-net-to-off-net, 106, 107
PBX-to-PBX, 107, 107
voice class codec command, 230
voice class h323 command, 232233
voice-class sip url sips command, 242
Voice Codec Bandwidth Calculator,
169, 169
voice codecs, 153
clarity, 160163
complexity, 156160
types, 153156
voice gateway ports, 179180
CAMA trunks, 188191, 189
csim start command, 209
CUCM, 201202, 202
debug dialpeer command, 209210
DSP profiles, 199200
E&M trunks, 187188, 187
FXS and FXO PLAR OPX, 180184, 180
FXS/DID inbound and FXO outbound,
184187, 185
SCCP, 200201
show controller command, 205
show voice dsp command, 205206
show voice port command, 203205
summary, 210
T1 CAS to analog cross-connect,
191195, 191
T1 PRI, 195198, 195
test voice port command, 206209
voice gateway signaling protocols, 8384, 84,
223224
comparisons, 88
H.323. See H.323 protocol
MGCP. See Media Gateway Control Protocol
(MGCP)
SCCP, 88
selecting, 93
SIP. See Session Initiation Protocol (SIP)
summary, 265
voice gateways, 1819, 19, 353354
call blocking, 380382
dial peers, 410411
DSP farms on, 198199, 198
DSP functions, 146147, 147
DTMF relay support, 354356
fax. See faxes
H.323 interface commands, 409410
H.323 service on, 411
modems, 367368
redundancy, 407408
556 user licenses for Cisco phone voice gateways
bindex.indd 556 9/20/11 1:38:52 PM
SIP settings, 243249, 247
summary, 382
toll bypass and TEHO, 377380, 378
voice backup paths. See voice backup paths
voice media transmission protocols, 78
cRTP, 8283
RTCP, 8182
RTP, 7881
sRTP, 83
voice network infrastructure considerations
power options for IP phones,
282286
VLANs, 286290, 287, 289
VoIP support, 290293
voice payload size, 165
voice-port command, 4750, 63
voice register dn command, 310
voice register global command, 306
voice register pools, 307, 311
voice service voip command, 227,
229, 269, 422
voice translation-profile command,
131, 386
voice translation-rule command,
129, 386
voice transport, 17, 17
voice/video hardware protocols, 93
voice/video on IP networks, 440441
voice VLANs
configuring and verifying, 288290, 289
trunks, 286288, 287
VoIP AutoQoS policies
on routers, 475479, 476
on switches, 479483, 480
VoIP design, 145146
codecs
clarity concerns, 160163
selecting, 163164
types, 153160
IP voice bandwidth consumption, 164169,
166, 169
quality considerations, 147153, 148,
150, 152
summary, 170
voice gateway DSP functions,
146147, 147
VoIP network infrastructure support, 290293
VoIP operation, 77
firewalls, 80
gatekeepers, 8991, 8990
H.323
MCU, 9192
proxy servers, 91
summary, 94
voice gateway signaling protocols,
8388, 84, 93
voice media transmission protocols,
7883, 79
VoIP path-selection process, 103
dial peers, 109110, 110
dial-plan digit manipulation. See dial-plan
digit manipulation
dial plans. See dial plan path-selection
process
summary, 135
W
w (watch phone) ephone button separator,
318, 320
WAN connections, 457
watch phone (w) ephone button separator,
318, 320
weighted fair queuing (WFQ), 454
Weighted Random Early Detection (WRED), 451,
455, 498500, 499
white noise, 151
wideband communication, codecs for, 154155
wideband sampling frequencies, 148, 148
wildcards
dial-peer configurations, 117121, 119, 121
in digit stripping, 124
in prefix adding, 125
wink-start command, 50
wink-start dialing, 185
wink-start E&M signaling, 4445, 45
voice media transmission protocols wink-start E&M signaling 557
bindex.indd 557 9/20/11 1:38:53 PM
wireless IP phones, 13
WRED (Weighted Random Early Detection), 451,
455, 498500, 499
X
x (expansion line) ephone button separator,
319, 324
X11 services, 116
Z
zone keyword, 413
zone local command, 406, 430
zone prefix command, 407, 430
zones
H.323 gatekeepers, 397398, 397
local, 406
prefixes, 407
remote, 406407
558 wired IP phones zones
bindex.indd 558 9/20/11 1:38:54 PM
Glossary
bgloss.indd 1 8/19/11 12:40:26 PM
2 Glossary
802.3af An IETF standard PoE method for powering networked devices.
A
address signaling The transmission of telephone digits from the calling-party phone to
the called-party phone. A unique sequence of digits identifies each individual phone on the
network so the call reaches the correct destination.
alternate mark inversion (AMI) An older digital circuit method for dictating how binary
is sent and interpreted on the wire.
analog telephone An edge device that sends and receives voice using two wires. The voice
signal is sent and received in analog waveforms.
application-specific routing (ASR) An H.323 feature that allows streams to be routed
based on the application being used.
assured forwarding (AF) PHB A PHB classification system that has 12 priority classes,
which are segmented into four classes, each with three drop priorities: low, medium, and
high.
Automatic Number Identification (ANI) The source telephone number of the calling
party. Also known as caller ID.
AutoQoS A Cisco QoS configuration method that automatically determines the best-
practice configurations for an interface and applies them for you.
AutoQoS autodiscovery phase The first step on the two-step AutoQoS for the
Enterprise configuration process. The router monitors interfaces and collects information
about the data flows it sees and attempts to classify them into one of 10 possible classes.
AutoQoS for the Enterprise An automated QoS feature that configures large-scale
networks for voice transport based on Cisco best-practice methodologies.
AutoQoS for VoIP An automated QoS feature that configures small to medium-size
networks for voice transport based on Cisco best-practice methodologies.
AutoQoS installation phase The second step on the two-step AutoQoS for the Enterprise
configuration process. The router uses the information collected during the AutoQoS
autodiscovery phase to configure classes and policies, and then applies them to the
appropriate interfaces.
B
backhaul Trunks used to transport multiple voice calls between the private site and the
service provider core network.
Basic Rate Interface (BRI) See ISDN Basic Rate Interface (BRI).
bgloss.indd 2 8/19/11 12:40:28 PM
Glossary 3
behavior aggregate (BA) A QoS term used to describe similar traffic flows that are
traveling in the same direction on a network device. Typically you want to classify traffic
into groups that have a similar BA.
best-effort QoS model The model in which a network device treats all traffic the same
and does not guarantee the delivery of traffic.
Bipolar 8-bit Zero Substitution (B8ZS) A newer digital circuit method for dictating how
binary is sent and interpreted over the wire. It solves the AMI 8-zeroes-in-a-row problem by
sending a distinct pattern that can be interpreted as such.
bottleneck The part of a network between two points where bandwidth is at its lowest.
This is the area where congestion is most likely.
C
call admission control (CAC) A voice protection feature that monitors the amount of
bandwidth on a path, and either permits or denies a call from being established based on
the amount of bandwidth available.
call leg A one-way logical connection of a call setup between two voice gateways.
call-processing agent Hardware and software responsible for call-processing and call-
control functions on an IPT network. From a Cisco perspective, call-processing agents are
any of the three Cisco Unified Communications Managers.
call waiting The ability of a phone to receive two or more simultaneous calls.
caller ID See: Automatic Number Identification (ANI).
central office (CO) A PSTN switch equipment office that is geographically dispersed to
handle the need of users based on population density and telephone usage.
central office (CO) trunk Circuits that connect a private business PBX to the PSTN.
Centralized Automatic Messaging Accounting (CAMA) A specialized trunk
configuration often used in North America for connecting to emergency services (E911).
Channel Associated Signaling (CAS) A digital signaling method that allows for up to 24
simultaneous calls at one time. In order to be able to squeeze 24 calls into an SF or ESF
frame, CAS uses robbed-bit signaling.
Cisco Discovery Protocol A Cisco proprietary Layer 2 messaging protocol that
is commonly used between Cisco devices to determine neighboring devices and their
capabilities.
Cisco fax relay Ciscos proprietary fax relay method, which uses special RTP packets to
transport the communication stream.
bgloss.indd 3 8/19/11 12:40:28 PM
4 Glossary
Cisco phone user license A license for each individual phone endpoint.
Cisco power injector A midspan device that provides power to a single phone endpoint.
Cisco Unified Border Element (CUBE) A specialized voice gateway IOS that can perform
IP-to-IP gateway functionality.
Cisco Unified Communications Manager (CUCM) A hardware appliance that runs on a
hardened Linux operating system. Each server appliance is capable of handling up to 7,500
endpoints and can be clustered to support up to 30,000 endpoints.
Cisco Unified Communications Manager Business Edition (CUCMBE) A hardware
appliance that runs on a hardened Linux operating system. The appliance is used in
medium-size businesses and can handle up to 500 endpoints.
Cisco Unified Communications Manager Express (CUCME) Specialized IOS software
that runs on Cisco routers. The voice hardware and software are commonly used in small
business environments and supports up to 250 endpoints.
class maps The first tier of a class-based QoS policy that defines a specific subset of traffic.
Class of Restriction (COR) Within the voice gateway, a method that allows you
to configure calling privileges and assign them to dial peers and telephone extensions
configured on the voice gateway or CUCM Express.
Class of Service (CoS) A field within the Layer 2 Ethernet frame header that marks
traffic as being one of eight (0 to 7) classes for QoS prioritization purposes.
Class Selector (CS) PHB A PHB classification system that uses only the three leftmost
bits. The other three bits are always 0s. The CS was created for backward compatibility
with IP Precedence values.
class-based (CB) QoS A term used to describe the three-tiered MQC configuration
process for QoS. The three tiers include class maps, policy maps, and applying the policy
maps using a service policy.
class-based weighted fair queuing (CBWFQ) A queuing mechanism that is an extension
of WFQ and also can be used to classify and prioritize traffic based on flow types.
codec An algorithm that converts analog waves into a digital format that may or may not
include compression. There are multiple codecs that use different fidelities, sampling rates,
and packet payload sizes. The word codec is short for coder/decoder.
comfort noise Artificial white noise created locally and played to let the user receive
audio feedback that a call is still in progress and has not been terminated.
Common Channel Signaling (CCS) A digital signaling method that uses in-band
signaling by taking an entire channel out of the TDM structure to use exclusively
for signaling.
bgloss.indd 4 8/19/11 12:40:28 PM
Glossary 5
companding A bandwidth-saving technique used to reduce the total number of bits that
are required for the digital circuit to be encoded and transported.
compressed RTP (cRTP) A technique used to shrink the size of the IP/UDP/RTP header
from 40 bytes to 25 bytes by not passing static information in every packet of an RTP
stream.
compression A method of reducing bandwidth by eliminating redundant 8-bit binary
samples on the receiving end. This is done by using a known sample or group of samples
and sending a signal to represent the known samples.
congestion avoidance A QoS method used to drop packets when congestion is detected
on an interface.
congestion management The use of logical queues within network hardware interfaces
to store packets that are waiting to be transmitted on a congested link.
CUCM Express feature license A license that determines how many phones you can run
on the CUCM.
custom queuing (CQ) A queuing mechanism that divides the total number of queue slots
into different classes. Each class gets a certain amount of queue spaces that is configurable
by the network administrator. The more preferred a class is, the more queue slots it is given.
customer premise equipment (CPE) Telephone equipment owned by a private party that
connects to the PSTN network.
D
default PHB A PHB classification describing traffic that requires only best-effort QoS.
The default PHB has a binary value of 000000.
default phone configuration file An XML configuration file that provides a Cisco IP
phone with all the general information it needs to communicate with the CUCM Express
system.
demarc The termination point that separates cabling responsibility between the
customers house wiring and the PSTN local-loop wiring.
DHCP relay A configuration setting that relays DHCP messages between requesting
endpoints and a DHCP server that resides on a different IP subnet.
Dial Peer 0 If no inbound dial-peer matches are made using configured dial-peer rules, the
voice gateway will use this built-in catch-all rule.
dial plan A telephone number methodology that uses dial peers to interpret dial strings
and determine how calls are directed through IP and/or PSTN networks.
bgloss.indd 5 8/19/11 12:40:29 PM
6 Glossary
Dialed Number Identification Service (DNIS) The destination telephone number a caller
dials and wishes to reach.
Differentiated Services (DS) byte A field within the IP header that is used to mark
packets with a DSCP value.
Differentiated Services Code Point (DSCP) A ToS/DS field within the Layer 3 packet
header that marks traffic as being one of 64 (0 to 63) classes for QoS prioritization
purposes.
DiffServ A QoS model that classifies different IP traffic flows and marks them for use on
other QoS-aware devices along the traffic flow path. Classified traffic can then be given
different priorities although it is not considered to be guaranteed.
digit stripping A digit-manipulation technique that removes digits that are explicitly
defined in dial-peer rules.
Digital Signal Processor (DSP) A hardware chip installed in a voice gateway that serves
to assist in voice connectivity, conferencing, and transcoding functionality in a voice
network.
digital telephone An edge device that converts the analog signal into a digital format.
This is done to overcome distance and scalability issues inherent with analog phones.
digital trunk A PSTN connection capable of transporting multiple digitized voice streams
across a single cable.
direct inward dial (DID) A PSTN option where the carrier strips off and sends only a
portion of the dialed digits to the customer.
directory alias A command used when identifying files to be serviced by the TFTP server
when the files are organized in a directory structure. The alias helps the phones locate the
directory where the files can be stored on the flash.
discrete signal The resulting data after digital sampling has been performed on an
analog signal.
drop precedence A portion of the DSCP AF marking classification system where traffic
can be marked with a drop precedence of low, medium, or high probability.
DSP farm A voice gateway configured to use DSP resources to offload transcoding,
conferencing, and MTP from a CUCM.
DSP profile A grouping of DSP resources to serve a specific DSP-farm offloading service.
DTMF relay A method of transporting DTMF tones to better ensure that they are
accurately reconstructed at the opposite end of a VoIP network.
dual line The combining of two separate phone lines in one telephone button. This lets
users of the phone place calls on hold or receive a second call when one line is in use.
bgloss.indd 6 8/19/11 12:40:29 PM
Glossary 7
dual-tone multi-frequency (DTMF) A method of inputting telephone digits using buttons
that sends two distinct tones to the phone switch to indicate a specific dialed digit.
dynamic gatekeeper discovery The method in which an H.323 device sends gatekeeper
request (GRQ) RAS messages in a multicast to discover its local gatekeeper.
Dynamic Host Control Protocol (DHCP) A service that dynamically assigns IP addresses
and other network information to endpoint devices such as PCs and IP phones.
E
E&M port An analog port commonly used to connect two PBX systems together.
E&M signaling An analog signaling protocol used to communicate between PBX systems
or PSTN network switches.
E.164 See International Numbering Plan.
E1 A digital trunk that carries 32 TDM channels. Two channels are used for framing and
signaling so the E1 can carry up to 30 simultaneous calls. E1s are used almost everywhere
outside of North America and Japan.
echo The reflection of sound that arrives to the listener a period of time after the original
sound is heard.
encoding The process of taking the quantized samples and translating them into binary.
ephone button separator A CUCM Express command character that is used to set
different ring, call waiting, overlay, and monitor options on an ephone.
ephone extension states The six different operational states that an IP phone can be in.
ephone A CUCM Express configuration statement that represents physical phones on the
CUCM Express system running SCCP. It includes a number used to identify a particular
device within the IOS.
ephone-dn A CUCM Express configuration statement that represents the telephone
extension configured on each phone that is running SCCP.
error correction mode (ECM) A fax feature that can be enabled to better ensure the
proper receipt of all fax-transmission packets.
expansion An ephone button separator option used to expand line coverage for an
overlay button.
expedited forwarding (EF) PHB A PHB classification system used for IP data flows that
require low-latency packet loss and jitter.
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8 Glossary
explicit congestion notification (ECN) The two rightmost bits of the DS field that can be
marked by Layer 3 devices to indicate link congestion.
Extended Superframe (ESF) The newer digital framing method, which bundles 24 TDM
channel cycles together in a single frame. It also performs cyclic redundancy checks (CRC)
for better reliability compared to SF.
F
fair queuing A queuing algorithm that schedules packets for transport across the same
interface. It is used in conjunction with priority marking of packets so that lower-class
packets are not choked off completely.
fax pass-through A fax transmission method that transports fax messages the same way
that voice calls are transmitted. The only difference is that when fax pass-through is
enabled, it ensures that fax transmissions are encoded using either G.711 mu-law or G.711
a-law to provide a high-quality digital representation of the original analog source.
fax relay A fax transmission method where analog fax transmissions are terminated at
the voice gateway, which then demodulates, packetizes, and transmits the packets to the
remote voice gateway. This process is accomplished using either the Cisco fax relay or T.38
fax relay method.
fax transmission rate A static transmit rate (measured in bps) at which the fax is
transmitted over an IP network.
fidelity The accuracy of a copied signal (such as voice) compared to the original.
firmware load file A file on the CUCM Express system used to tell the registering Cisco
phones which firmware they are to download.
First-In, First-Out (FIFO) A queuing mechanism that does not place any emphasis on
packet priorities. Instead, the first packet to be placed in the queue is the first one to
come out.
fixed delay The amount of time it takes in an ideal situation where the only slowdown is
in how fast it takes electrical and optical signals to transport IP packets.
Foreign Exchange Office (FXO) An analog port commonly used to connect a voice
gateway to the PSTN.
Foreign Exchange Station (FXS) An analog port commonly used to connect analog end
devices such as a telephone or fax machine.
FRF.12 An LFI mechanism that can be configured on Frame Relay circuits.
bgloss.indd 8 8/19/11 12:40:30 PM
Glossary 9
G
gatekeeper A device whose primary function is to maintain a database mapping
telephone extensions to IP addresses. It is primarily found in environments running the
H.323 protocol.
gateway signaling protocols Protocols used to communicate signaling between voice
gateways or between a voice gateway and a call agent.
glare An occurrence when loop-start signaling is used whereby a user picks up a phone
and unexpectedly finds they are already connected to a call that came inbound.
ground-start signaling A supervisory signaling type that uses grounding wires for the
signaling of the line to be seized.
group 3 A fax standard transmission method that supports speeds up to 14.4 Kbps.
H
H.323 A suite of protocols for the signaling of voice, video, and data using a peer-to-peer
architecture.
H.323 Early Media An H.323 feature used in concert with H.323 fast start to provide
early communication channels for media such as broadcast announcements and music on
hold (MOH).
H.323 fast start An H.323 call-initiation process that sets up an H.245 channel during
the call setup stage and does not wait for the call proceeding, alerting, and connect stages
to complete.
H.323 gatekeeper An H.323 component that breaks up an H.323 network into multiple
zones. It can also be configured for other services, including RSVP-based CAC.
H.323 proxy server A server that works as a head end for call setup and teardown of one
or more H.323 endpoints.
H.323 slow start An H.323 call-initiation process that sets up an H.245 channel after the
call setup, call proceeding, alerting, and connect stages have completed.
high-complexity codec A codec that requires a large amount of DSP processing power
typically because of higher compression rates while maintaining call clarity.
house wiring The customers internal telephone wiring that it is responsible for
maintaining.
huntstop A command that tells CUCM Express to look for the next preferred ephone-DN
if the most preferred phone is busy.
hybrid system The combination of PBX and key-system functionalities.
bgloss.indd 9 8/19/11 12:40:30 PM
10 Glossary
I
impedance The ratio between voltage and electrical current.
inbound dial peer A dial peer that matches number strings coming into the voice gateway.
informational signaling Feedback generated from the telephone switch to the user in the
form of tones or voice messages to inform the phone user what state a call is in.
inject-tone A voice port-testing method used to send a tone across the port at a specific
frequency that is used to determine proper settings as optimal impedance settings.
inline power (ILP) A Cisco proprietary PoE option integrated into switches.
Integrated Services Digital Network (ISDN) A standard suite of protocols that operates
on Layers 13 of the OSI model. It utilizes PSTN circuits running CCS for the transport of
voice, data, and video.
Intelligent Power Management (IPM) A Cisco method using CDP to negotiate power
allocation of 802.3af PoE devices.
intercluster trunk A VoIP call type where call setup signaling is transferred between the
CUCMs at each site in order to establish the call.
interexchange network The level within the PSTN hierarchy where national long-distance
charges are incurred.
interface binding A configuration method used to associate a virtual interface with
multiple physical interfaces. This is commonly implemented on voice networks to eliminate
a single physical point of failure.
international network The level within the PSTN hierarchy where international long-
distance charges are incurred.
International Numbering Plan A telephone numbering plan used by all countries around
the world. The plan breaks numbers into three categories: country code, national
destination code, and subscriber code.
Internet Telephony Service Provider (ITSP) A public telephone service provider that uses
the IP network to connect customers to the PSTN. It offers telephony services similar to the
PSTN; the primary difference is that connection between the private organization and the
service provider uses VoIP as opposed to legacy analog or digital circuits.
interoffice trunk Backhaul connections that interconnect central offices. Calls made
between interconnected COs are considered to be local.
IntServ A QoS model that guarantees the quality of service for specific traffic types. It
can provide a guarantee by reserving a specific amount of bandwidth for a flow from
end to end.
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Glossary 11
IOS feature set A license that determines the different features that can be run on an
IOS-capable device.
IP Precedence A technique that uses the three leftmost bits of the ToS byte to mark
packets with a value of 07 for QoS classification purposes.
ISDN Basic Rate Interface (BRI) Digital circuit that has three channels of 64 Kbps each.
The two channels used for transport are called B channels, and the one channel that
out-of-band signaling uses is called the D channel.
J
jitter The variation in the time between the receipt of each voice packet. For voice, it is
recommended that jitter be reduced to 30 ms or less, on average.
K
key system A telephone system used in small businesses for the sharing of external PSTN
lines on multiple phones.
L
lightweight registration A feature in H.323v2 that uses modified RRQ messages that are
smaller and consume less bandwidth when notifying the gatekeeper that the end device is
still alive.
line seizure A telephone line state when a phone transitions from an on-hook to an
off-hook state.
link efficiency A QoS method used to make the transport of data flows more efficient,
including techniques such as compression and LFI techniques.
link fragmentation interleaving (LFI) A link-efficiency compression technique that takes
large data frames and fragments them into smaller, more manageable sizes. It then
interleaves these smaller fragmented frames with other small frames such as voice.
listener echo The reflection of sound that is an echo of an echo; the listening party hears
the talker two times during different time intervals.
local call A VoIP call type where source and destination phones are connected to the same
call-processing agent or voice gateway.
local loop The physical wiring between a customers private phone equipment and the
PSTN central office (CO).
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12 Glossary
local zones H.323 zones that are managed by the local gatekeeper.
loop-start signaling A supervisory signaling type that uses a two-wire method for line
seizure.
Low Latency Queuing (LLQ) A queuing mechanism that can be configured to offer
priority queuing (PQ) for traffic such as voice and CBWFQ queuing for other types
of traffic.
low-complexity codec A codec that requires a low amount of DSP processing power.
M
Mean Opinion Score (MOS) A subjective ITU-T method of ranking call clarity for
various audio codecs. Each codec is judged for various quality aspects and is given an
averaged score between 1 and 5, where 5 is considered excellent quality.
media flow-around Media streams flow freely between the two networks and find their
own path to the destination instead of being forced through a CUBE.
media flow-through Voice/video streams come into and are proxied by the CUBE.
Media Gateway Control Protocol (MGCP) A newer voice gateway protocol that uses a
client-server architecture. It is very easy to set up but limited in its features.
Media Termination Point (MTP) A method used to set up logical terminations to offload
voice duties such as call hold, transfer, park, conference calling, and DTMF generation.
medium-complexity codec A codec that requires a moderate amount of DSP processing
power.
MGCP fallback A MGCP failover feature that lets gateways fall back to the H.323
protocol when communications are lost, which renders MGCP useless.
MGCP residential gateway A type of MGCP gateway where the protocol is responsible
for providing signaling between the IP network and analog voice ports including FXS,
FXO, and E&M.
MGCP trunking gateway A type of MGCP gateway where the protocol is responsible for
providing signaling between the IP network and PSTN trunked ports such as ISDN BRI
and PRI circuits.
Modular QoS CLI (MQC) A Cisco QoS configuration method that uses a modular three-step
hierarchical approach to configuring classes and policies and applying the policies to
interfaces.
monitor line An ephone button separator option used to monitor the status (on- or off-
hook) of a single ephone-DN.
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Glossary 13
Multilink PPP (MLP) A Layer 2 transport mechanism defined in RFC 1990 that
encapsulates Layer 3 traffic over point-to-point links including ISDN.
multiplexing Combining multiple analog or digital signals over a shared physical
medium.
multipoint control units (MCU) Devices used to control and facilitate H.323 multimedia
content such as audio and video for a point-to-multipoint communication.
N
named signaling events (NSE) A message used to communicate resources such as codec
choice when transporting fax transmissions. In this book, NSE messages can be either
Cisco proprietary or ITU-T standard messages.
narrowband Describes an audio sample taken using a smaller frequency range than
wideband but that collects the vast majority of audio. Narrowband commonly collects
signals between 300 and 3400 Hz.
Network Time Protocol (NTP) A service that synchronizes the internal clocks on
networked equipment.
non-standard facilities (NSF) Proprietary (non-T.30) capability codes that are exchanged
between two fax machines to determine possible fax transmission methods.
North American Numbering Plan (NANP) The numbering plan used in the United States,
Canada, and parts of the Caribbean. The plan uses a fixed format of 10 digits divided into
three categories: area code, central office code, and subscriber code.
number expansion A digit-manipulation technique that matches one string of digits and
then substitutes different digits before forwarding them to the next destination.
Nyquist sampling theorem A mathematical equation used to find the optimal method
for sampling the human voice for transport on a telephone network.
O
octo-line The combining of eight separate phone lines in one telephone button. This lets
users of the phone place calls on hold or receive a second call when the first line is in use.
off-hook When the analog circuit between the ring and tip is connected and the ring
powers the tip.
off-net A VoIP call type where source and destination phones are on different networks
where the PSTN must be utilized to complete the call.
bgloss.indd 13 8/19/11 12:40:31 PM
14 Glossary
one-stage dialing When a voice network is configured so that a call is not terminated
until the endpoint phone is reached.
on-hook When the analog circuit between the ring and tip is severed and the battery
(ring) cannot power the tip lead.
on-net A VoIP call type where source and destination phones are on the same network
but traverse more than one voice gateway.
outbound dial-peer rule A dial peer that matches number strings before exiting the voice
gateway.
out-of-band signaling The process of using a separate voice channel that is reserved for
signaling. That way, signaling is kept completely separate from any voice or data
transmissions.
overlay An ephone button separator option used to associate multiple ephone-DNs with a
single-line button.
P
packet loss An occurrence when network hardware queues fill up and packets are
dropped. For voice, it is recommended that packet loss not exceed 1 percent.
packet loss concealment Software used to intelligently guess what the payload should be
for lost packets. The software then generates a substitute packet to fill in for the one that
was lost on the network.
peak information rate (PIR) A byte setting that sets an absolute maximum rate above and
beyond the CIR.
peer-to-peer architecture A model in which both voice peers have intelligence to route
calls from one point to another.
Perceptual Evaluation of Speech Quality (PESQ) An objective ITU-T method that
produces a highly reproducible score of voice codec quality. It is similar to PSQM but takes
network issues such as latency, jitter, and packet loss into the scoring equation. Scores are
graded using a scoring method similar to MOS (1 to 5) so the two scores can be easily
compared.
Perceptual Objective Listening Quality Analysis (POLQA) An objective ITU-T method
that is used to score next-generation codecs in terms of voice quality. The standard will
eventually be the replacement to PESQ because of its ability to offer more advanced
benchmarking for sideband codecs as well as more advanced wireless networks.
Perceptual Speech Quality Measure (PSQM) An objective ITU-T method that produces
a highly reproducible score of voice-codec quality. It has since been replaced with the more
accurate PESQ scoring system.
bgloss.indd 14 8/19/11 12:40:32 PM
Glossary 15
per-hop behavior (PHB) A term used to describe DSCP subsets created by the IETF that
define a structured methodology for marking packets with DSCP.
phantom power Using the same wiring to power devices as Ethernet uses to transmit and
receive data.
phone branch exchange (PBX) A telephone switch that lets a business run an internal
and private voice network.
policy maps The second tier of a class-based QoS policy that associates traffic class types
with one or more QoS operations.
POTS dial peer Voice gateway configuration command that provides routing information
for connecting to traditional telephony devices such as analog phones, fax machines, and
any off-network calls that are routed out to the PSTN using either analog or digital
interfaces connected to the voice gateway.
power brick Standard 110v AC unit that plugs directly into a single phone endpoint.
Power over Ethernet (PoE) A method of providing end devices with power using the
same Ethernet cable used for the transport of data.
powered patch panel A device that sits in between an IP phone and a non-PoE-capable
switch that provides power to multiple phone endpoints.
predictor compression A link-efficiency compression technique that attempts to predict
the next sequence of characters in a data stream by using an index in the compression
dictionary.
prefix adding A digit-manipulation technique that adds digits to the beginning of a
number string before it is forwarded out of the voice gateway.
Primary Rate Interface (PRI) Digital circuit that uses either 23 or 32 channel T1/E1
ports. Out-of-band signaling is used.
priority queuing (PQ) A strict queuing mechanism that is used to give priority explicitly
to certain traffic types.
Private Line Automatic Ringdown (PLAR) An autodialing mechanism that is used to
associate a port with a single destination.
private numbering plan Numbering plan for the configuration of private telephone
networks within an organization.
propagation delay The amount of time it takes a packet to travel from source to
destination on a network. For voice, it is recommended that delay not exceed 150 ms.
protocol internetworking The ability of a CUBE to terminate and reinitiate IP voice
sessions between devices that run H.323, SIP, or H.323-to-SIP.
bgloss.indd 15 8/19/11 12:40:32 PM
16 Glossary
proxy A device that acts as an intermediary for connected clients.
Public Safety Answering Point (PSAP) The name for the CO that is the first hop
connecting a private voice network CAMA trunk to E911 services.
public switched telephone network (PSTN) A network that interconnects private home
and business phones. Customers pay a service fee to use the PSTN.
pulse dialing A method of inputting telephone digits using a rotary disk with a
mechanical motion to perform on- and off-hook transitions to specify a digit.
pulse-amplitude modulation (PAM) The process of sampling, quantizing, and encoding
an analog voice signal.
pulse-code modulation (PCM) The process of translating sampled analog signals into a
numbering system. This is also referred to as quantization.
Q
Q signaling (QSIG) A signaling protocol that uses Q.931 as its underlying signaling
protocol but modifies the signals so proprietary ISDN signaling protocols can be used by
nonproprietary equipment on the other end of the connection.
Q.931 ITU-T standard sub-signaling protocol that is responsible for the setup and
teardown of B channel connections whether they are voice or data connections.
Quality of Service (QoS) A set of traffic-control mechanisms used to give time-sensitive
traffic priority on the network to limit delay, jitter, and packet loss.
quantization The process of translating sampled analog signals into a numbering system.
This is also referred to as pulse-code modulation.
R
Random Early Detection (RED) A congestion-avoidance technique that randomly
drops packets when congestion is detected. It is set into motion when queues begin to
fill up and packets need to be discarded on bottleneck interfaces. RED cannot
differentiate between traffic types, so any packets could potentially be dropped.
Real-time Transport Control Protocol (RTCP) An out-of-band supporting protocol for
RTP. Its primary purpose is to track statistics for QoS adjustments.
Real-time Transport Protocol (RTP) An IETF RFC 1889 and 3050 protocol designed to
transport real-time IP payloads.
Registration Admission and Status (RAS) An H.225 message protocol that is used to
communicate the registration process between H.323 gatekeepers and H.323 endpoints and
voice gateways.
bgloss.indd 16 8/19/11 12:40:33 PM
Glossary 17
remote zones H.323 zones that are not configured locally and are handled by an external
gatekeeper.
reset A full reboot of a Cisco IP phone.
Resource Reservation Protocol (RSVP) A Transport layer protocol that is designed to
reserve bandwidth resources dynamically across an IP network.
restart A partial reset of a Cisco IP phone.
ringing time The amount of time that an echo canceller waits to listen for echo on the
receiving (Rx) line of the tail circuit.
robbed-bit signaling (RBS) The technique of taking bits from SF framing channels 6 and
12 and ESF framing channels 6, 12, 18, and 24 for sending signaling data from one end of
the digital circuit to the other. This stealing of bits is done to maximize the number of calls
a CAS T1 can handle.
S
Secure RTP (sRTP) Protocol that provides authentication, data encryption, and relay
protection for RTP packets.
seizure The process of taking a telephone connection off-hook, which reserves the line for
a telephone call.
service policy The third tier of a class-based QoS policy that is used to apply policy maps
to router interfaces including subinterfaces and virtual circuits.
Session Description Protocol (SDP) An RFC 2327 protocol that uses standard ASCII
codes for describing and negotiating multimedia sessions.
Session Initiation Protocol (SIP) A peer-to-peer transport protocol that uses a
distributed call-processing architecture. The protocol messages are sent in ASCII format,
and addressing looks similar to an email address.
shared line The term used to describe an ephone-DN that is applied to two or more IP
phones.
SIP delayed offer A method of exchanging SDP messages using SIP where the target
device sends the initial request in a SIP OK message.
SIP early offer A method of exchanging SDP messages using SIP where the initiating
device sends the initial request in a SIP INVITE message.
SIP proxy server Device that takes the responsibility of forwarding INVITE messages for
the UACs.
SIP register server Device that maintains a database mapping phone numbers to IP
addresses on a SIP network.
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18 Glossary
SIP secure (SIPS) A configuration feature to secure SIP communication.
SIP source-bind A SIP feature used to statically assign an IP address to a specific voice
gateway interface to be used for the signaling and/or media source IP address.
SIP user agent (UA) SIP endpoint device that can be considered either a UAC or a UAS device.
SIP user agent client (UAC) Device that sends INVITE messages to a remote peer to
establish a SIP connection.
SIP user agent server (UAS) Device that responds to UAC INVITE messages.
site code dialing A dial plan method that uses a digit or multiple digits to specify a
specific location on a voice network. Site codes are useful in situations where you have
overlapping telephone extensions.
Skinny Client Control Protocol (SCCP) Ciscos proprietary voice signaling protocol. It is
primarily used as an endpoint-to-call-agent protocol for signaling but can be used on voice
gateways for signaling. It uses a client-server architecture with centralized call control.
source IP address The IP address that defines the location of the CUCM Express
call-processing unit.
stacker compression A link-efficiency compression technique that uses a special encoded
dictionary, which both routers possess. The router replaces streaming data with much
smaller codes found in the dictionary.
Super Frame (SF) The former digital framing method, which bundles 12 TDM channel
cycles together in a single frame.
Super Group 3 (SG3) A fax standard transmission method that supports speeds up to
33.6 Kbps.
Supervisory Signaling Signaling that detects changes in the status of the telephone
physical loop or trunk and is then used to set up and tear down calls. Loop-start and
ground-start analog signaling fall within this signaling category.
survivable remote site telephony (SRST) A voice backup method that allows the voice
gateway to temporarily act as the call-processing agent in the event that a WAN connection
is lost and there are phones that cannot communicate with the CUCM.
T
T.30 An ITU-T standard for the transmission of fax messages over POTS lines.
T.37 store-and-forward fax A fax transmission method that uses SMTP email messages
as transport for the fax transmission. The transmission is obtained and converted into a
bgloss.indd 18 8/19/11 12:40:33 PM
Glossary 19
TIFF file by the T.37-capable device. It is then attached to an email and sent to one or more
recipients.
T.38 fax relay An ITU-T standard fax relay method for transmitting fax messages over IP
networks.
T1 A digital trunk that carries 24 TDM channels. Depending on the signaling type used, a
T1 can carry either 23 or 24 voice calls simultaneously. T1s are used primarily in North
America.
tail end hop off (TEHO) The voice design of configuring your IP network to transport
calls as far as possible on the IP WAN before letting them hop off onto the PSTN. This is an
extension of toll bypass that can work in geographically dispersed IP networks.
talker echo The reflection of sound that arrives back at the originating talker where they
hear themselves repeated.
TCP synchronization A phenomenon that occurs when interface queues fill up. All TCP
flows passing through the congested interface will back off and begin sending packets more
slowly. They will eventually speed back up and cause the same congestion, causing a seesaw
effect in traffic flow.
technology prefix A special E.164 prefix number that can be dialed by endpoints to take
advantage of special H.323 features.
telephony service event (TSE) Special messages that can provide a way to communicate
telephony events between MGCP gateways.
tie trunk (or tie line) A dedicated voice circuit that directly connects two PBX switches.
time-division multiplexing (TDM) A strict time-based method for sharing a single cable
to transport multiple voice signals.
token A metaphor used to describe a routers permission for a source device to send a
specified amount of bits out to the network.
token bucket A metaphor used to describe interface queues.
toll bypass The voice design of configuring your IP network to utilize WAN connections
for voice transport and therefore avoiding long-distance charges.
tone suppression A configuration method used to block fax tone transmissions so they
can be transported at a lower rate.
Tool Command Language (TCL) A scripting language that can be used within the IOS to
script events for processes such as the T.37 store-and-forward fax method.
traffic classification The process of identifying traffic based on different characteristics
in order to group same-traffic types together for QoS.
bgloss.indd 19 8/19/11 12:40:34 PM
20 Glossary
traffic marking The process of flagging critical packets so that the rest of the network can
properly identify them and give them priority over all other traffic.
traffic policing A QoS technique that sets a strict maximum transmission rate to a certain
group of traffic. If traffic flows go above the configured rate, the traffic is dropped or
retagged.
traffic queuing The process of ordering certain types of traffic for transport over LAN/
WAN interfaces.
traffic shaping A QoS technique that sets a maximum transmission rate to a certain
group of traffic. If traffic flows go above the configured rate, the traffic is put into queues
when available. That means the data will still be sent, but it will be delayed.
transcoding The process of translating data between two different codecs.
translation rule A digit-manipulation technique that nests multiple rules in a translation
rule set, which then can be called within dial peers or POTS ports to match and manipulate
number strings.
translation rule regular expressions Defined characters used to provide an easy and
structured method to match number strings used for matching and manipulating number
strings within translation rules.
Transmission Control Protocol (TCP) An IP suite protocol often used in data applications
that benefit from features such as reconstructing unordered packets at the destination, and
the retransmission of missing packets.
trust boundary A term used to describe the point within a network topology where you
start trusting QoS markings contained within a packet or frame.
two-stage dialing When a voice network is configured so that a caller dials digits, which
are accepted by a voice gateway, and the call terminates at a second hop along the
connection where a second dial tone is given. The caller must then enter a second series of
digits to complete the intended call.
Type of Service (ToS) byte A field within the IP header that is used to mark packets with
an IP Precedence value.
U
Unity Express A hardware device that provides voicemail services. It integrates directly
into CUCM Express on an open network module.
User Datagram Protocol (UDP) An IP suite protocol often used by voice because it
provides no error-correction mechanisms, which real-time traffic cannot utilize.
bgloss.indd 20 8/19/11 12:40:34 PM
Glossary 21
V
VLAN trunk A link between two Layer 2 switches that can transport traffic from multiple
VLANs. It keeps the traffic between the VLANs separate by tagging each frame.
Voice Activity Detection (VAD) Software used to detect silence on a phone call and
prevent the sending of silent packets across the network to conserve bandwidth.
voice clipping A side effect of VAD in which the first few milliseconds of a users voice
are not transmitted to the remote party.
voice gateway A router that connects IP and PSTN voice networks. The gateway is
responsible for translation, transcoding, and compression, when needed.
voice register dn A CUCM Express configuration statement that represents the
telephone extension configured on each phone that is running SIP.
voice register pool A CUCM Express configuration statement that represents physical
phones on the CUCM Express system running SIP. It includes a number used to identify a
particular device within the IOS.
voice VLAN A dedicated VLAN specifically used for voice communications on an IP
network.
VoIP dial peer Voice gateway configuration command that provides routing information
for devices connecting to each other through an IP network.
W
watch phone An ephone button separator option used to monitor the status (on- or off-
hook) of all ephone-DNs assigned to a phone.
weighted fair queuing (WFQ) A queuing mechanism that uses byte sizes to fairly
distribute traffic out of queues. This lets smaller packets such as voice have preferential
treatment over larger data packets.
Weighted Random Early Detection (WRED) A congestion-avoidance technique that uses
RED but adds an extra layer of intelligence to better determine which packets should have a
higher probability of being dropped based on QoS markings. It is a Cisco proprietary
advancement of RED to make the dropping of packets less random by selecting packets that
are marked lower than others.
wideband Describes an audio sample taken using a large frequency range that captures
more of the audio signal than narrowband methods. Wideband commonly collects signals
between 50 and 7000 Hz.
bgloss.indd 21 8/19/11 12:40:34 PM
22 Glossary
wildcard One or more characters used as a placeholder to describe a range of telephone
digits. Wildcards are used in configurations (such as dial peers) to limit the number of rules
that need to be created.
wink A term used to describe the on-off-on hook transition when using E&M wink-start
supervisory signaling.
WRED See Weighted Random Early Detection (WRED).
Z
zone In H.323, a gatekeeper is used to break up a large network into logical units known
as zones for better management and policy enforcement.
zone prefix Configured on H.323 gateways, it is used to identify a zone by an E.164
number.
bgloss.indd 22 8/19/11 12:40:35 PM
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bmedinst.indd 560 9/21/11 2:59:32 PM

CVOICE 8.0: Implementing Cisco Unified Communications
Voice over IP and QoS v8.0 Study Guide
642-437-AF-CVOICE Objectives
OBJECTIVE CHAPTER
Describe a Dial Plan
Describe a numbering plan. 1, 4
Describe digit manipulation. 4
Describe path selection. 4, 9
Describe calling privileges. 4, 9
Describe the Basic Operation and Components Involved in a VoIP Call
Describe VoIP call ows. 3, 9
Describe RTP, RTCP, cRTP, and sRTP. 3
Describe H.323. 3, 7
Describe MGCP. 3, 7
Describe Skinny Call Control Protocol. 3
Describe SIP. 3, 7
Identify the appropriate gateway signaling protocol for a given scenario. 3, 7
Choose the appropriate codec for a given scenario. 5
Describe and Congure VLANs. 5
Implement Cisco Unied Communications Manager Express to Support Endpoints Using CLI
Describe the appropriate software components needed to support endpoints. 8
Congure DHCP, NTP, and TFTP. 8
Describe the differences between the different types of ephones and ephone-dns. 8
Congure Cisco Unied Communications Manager Express endpoints. 8
Describe the Components of a Gateway
Describe the function of gateways. 1, 4, 5, 7
Describe DSP functionality. 5
Describe the different types of voice ports and their usage. 2, 5, 6
Describe dial peers and the gateway call routing process. 4, 9
Describe codecs and codec complexity. 5
Perf.indd 1 8/29/11 2:33:16 PM
OBJECTIVE CHAPTER
Implement a Gateway
Congure analog voice ports. 2, 6
Congure digital voice ports. 2, 6
Congure dial peers. 4, 6
Congure digit manipulation. 4, 7, 9
Congure calling privileges. 7
Verify a dial-plan implementation. 4, 6, 7
Implement fax support on a gateway. 9
Implement Cisco Unied Border Element
Describe the Cisco Unied Border Element features and functionality. 10
Congure Cisco Unied Border Element to provide address hiding. 10
Congure Cisco Unied Border Element to provide protocol and media internetworking. 10
Congure Cisco Unied Border Element to provide call admission control. 10
Verify Cisco Unied Border Element conguration and operation. 10
Describe the Need to Implement QoS for Voice and Video
Describe the causes of voice and video quality issues. 5, 11
Describe how to resolve voice and video quality issues. 11
Describe QoS requirements for voice and video trafc. 5, 11
Describe and Congure the DiffServ QoS Model
Describe the DiffServ QoS model. 11
Describe marking based on CoS, DSCP, and IP Precedence. 11
Congure Layer 2 to Layer 3 QoS mapping. 12
Describe trust boundaries. 11
Congure trust boundary on Cisco switches. 12
Describe the operations of the QoS classications and marking mechanisms. 11, 12
Describe Low Latency Queuing. 11, 12
Describe the operations of the QoS WAN link efciency mechanisms. 11, 12
Enable QoS mechanisms on switches using AutoQoS. 12
Congure Low Latency Queuing. 12
Perf.indd 2 8/29/11 2:33:17 PM

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indd ii 9/21/11 11:51:43 AM

642-437 CVOICE v8.0

642-447 CIPT1 v8.0

642-457 CIPT2 v8.0

642-427 TVOICE v8.0

642-467 CAPPS v8.0

642-437 CVOICE v8.0

A thorough knowledge of analog, digital, and IP voice technologies including but

A solid understanding of QoS fundamentals and how to implement them on Cisco

Chapter 1, An Introduction to Traditional Telephony and Cisco Unified

Chapter 5, VoIP Design Options, exposes readers to VoIP network design

Chapter 7, Configuring Voice Gateway Signaling Protocols, is an in-depth look at

Chapter 10, Configuring and Managing CUBE and H.323 Gateways, is an

Chapter 11, Introduction to Quality of Service, fully introduces the concepts

Chapter 12, Configuring Quality of Service, continues our coverage of QoS,

Multiple-choice multiple-answerCisco will always tell you to choose two or three,

Cisco voice router gateway/gatekeeper/CUCM Express, switch or CUBE simulations

A monospaced font indicates the contents of configuration files, messages displayed at

Italicized monospaced text indicates a variableinformation that differs from one

Describe the function of gateways.

Describe a numbering plan.

An Introduction to Traditional Telephony and Communications

An Introduction to Traditional Telephony and Communications

An Introduction to Traditional Telephony and Communications

An Introduction to Traditional Telephony and Communications

An Introduction to Traditional Telephony and Communications

An Introduction to Traditional Telephony and Communications

An Introduction to Traditional Telephony and Communications

An Introduction to Traditional Telephony and Communications

An Introduction to Traditional Telephony and Communications

An Introduction to Traditional Telephony and Communications

An Introduction to Traditional Telephony and Communications

An Introduction to Traditional Telephony and Communications

An Introduction to Traditional Telephony and Communications

An Introduction to Traditional Telephony and Communications

An Introduction to Traditional Telephony and Communications

Describe the different types of voice ports and their usage.

Configure analog voice ports.

Configure digital voice ports.

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Understanding Analog and Digital Voice

Describe VoIP call flows.

Describe RTP, RTCP, cRTP, and sRTP.