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Introduction

Q1. What are the benefits of using Cisco Unified Border

Element ?

Q2: What type of Calls are supported in CUBE ?

Q3: How the Media is handled in CUBE ?

Q4: What Platforms are supported for CUBE?

Q5: Do we need Digital signal processors (DSPs) for CUBE?

Q6: What is the licensing requirements of the CUBE?

Q7: How to have a basic setup between CUCM and CUBE?

Q8: Do CUCM need special license for CUBE or it will be just a


SIP trunk without license?

Q9: WE have a CUBE for a SIP connection. I know there is a


license that we purchased from Cisco. Is there a command to enter on the
CUBE to view the licenses?

Q10: Whether CUBE supports Voice and Data traffic


simultaneously?

Q11: Does CUBE have to be run on a dedicated device/router?

Q12: Do we need DSP resources in CUBE if the SIP trunk from


city (external) to the SIP trunk of CUCM will be used?

Q13: Will SRST work on CUBE ?

Q14: Explain the Call flow in CUBE.

Q15: How to enable the IP-IP Calls and Protocol support in


CUBE ?

Q16: Whether CUBE supports Universal Transcoding ?

Q17: What is CAC?

Q18: Is CP-PWR-CUBE-4= compatible with the 7900 series

phones?

Q19: Im trying to find a way to route inbound SIP calls, based


on their source.

Q20: CUBE Sizing, how many simultaneous calls/ sessions are


supported on each Router model to terminate a SIP Trunk at the service
provider end?

Q21: What are the types of CUBE licenses available?

Q22: I am trying to troubleshoot the Multicast MOH on Cube,


how to do it?
Q23: Debug commands on CUBE

Verifying Fundamental Cisco Unified Border Element


Configurations

Related Information

Introduction
This document answers frequently asked questions about the Cisco Unified
Border Element (CUBE).

Q1. What are the benefits of using Cisco Unified


Border Element ?
A:
Multiple physical interconnects, intelligent OAM, Call Admission Control, Billing etc
Security Demarc: FireWall, DOS protection, VPN, etc
Signaling, Protocol & Media Interworking: H.323/SIP, transcoding, DTMF etc.
Media QoS control and monitoring
Interoperability: various network elements, CUCM, etc.
Co-existence/co-operation with TDM trunking

Q2: What type of Calls are supported in CUBE ?


A: CUBE supports,
IP-to-IP Calls
TDM Voice Calls
Gatekeeper
Routing

Q3: How the Media is handled in CUBE ?


A: Media is handled in 2 different modes

Media Flow-Through

Media Flow-Around

Q4: What Platforms are supported for CUBE?


A: Cisco CUBE is an Integrated application with Cisco IOS software.
CUBE functionality is supported in Cisco 2600XM, 2691, Cisco ISR 2800, 3800
series, Cisco VXR 7200, Cisco XR 12000, AS5400XM Universal gateways and the
Service Provider Gateways.

Q5: Do we need Digital signal processors (DSPs) for


CUBE?
A: Digital Signal Processors (DSPs) are only required for calls with dissimilar codecs

Q6: What is the licensing requirements of the CUBE?


A: If you have any existing Cisco Gateway then you can download the Cisco IOS
supported version for running CUBE on your Gateway. Also you need to buy license
for the no. of sessions in CUBE for different platform. Say for an example if you want
to have a CUBE supporting 500 sessions , you need to buy the license for this
amount.

Q7: How to have a basic setup between CUCM and


CUBE?
A: The layout will be CUCM-sip-Cube-sip-provider.

First you need to create the SIP trunk on the CUCM and point it to the cube

Bind your sip signalling to the appriprate interfaces on the CUBE

Enter the respective dial-peers on the cube for inbound and out bound calls to
and from the CUCM

Q8: Do CUCM need special license for CUBE or it will


be just a SIP trunk without license?
A: No, there is no special license for connecting CUCM to CUBE..

Q9: WE have a CUBE for a SIP connection. I know


there is a license that we purchased from Cisco. Is

there a command to enter on the CUBE to view the


licenses?
A: The CUBE licenses are not entered into the router. As of now, they are paper
licenses.

Q10: Whether CUBE supports Voice and Data traffic


simultaneously?
A: Yes. CUBE is in IOS based platform hence it can handle both Voice and Data
traffic.

Q11: Does CUBE have to be run on a dedicated


device/router?
A: Not Necessary. It depends on the voice traffic you want to route using CUBE. For
large enterprise its recommended to use standalone device for CUBE functionality. If
the traffic is not high you can run other services along with the CUBE application for
voice on the same platform simultaneously.

Q12: Do we need DSP resources in CUBE if the SIP


trunk from city (external) to the SIP trunk of CUCM will
be used?
A: DSP resources are required to simply connect to SIP Trunk. DSP resources are
required if you plan on doing transcoding between g711 to g729 codec. For
conferences if all parties are using g711 codec, then you can use the software
resources in UCM to do the conferencing. If one of the parties is using g729 codec
and if that needs to be conferenced, you will need to use voice gateways for
hardware conferencing which will require DSPs. Digital Signal Processors (DSPs)
are only required for calls with dissimilar codecs.

Q13: Will SRST work on CUBE ?


A: Yes, SRST and CUBE can be co-located on the same router.

Q14: Explain the Call flow in CUBE.


A:

1.Incoming VoIP setup message from OGW to CUBE


2.This matches an inbound VoIP dial peer 1 for characteristics such as codec, VAD,
DTMF method, protocol etc
3.CUBE then looks up called number in setup and matches outbound VoIP dial peer
2
4.Outgoing VoIP setup message from CUBE to TGW

Q15: How to enable the IP-IP Calls and Protocol


support in CUBE ?
A: Enabling the IP-to-IP Calls
CUBE#config t
CUBE(config)# voice service voip
CUBE(conf-voi-serv)#allow-connections h323 to h323
CUBE(conf-voi-serv)#allow-connections h323 to sip
CUBE(conf-voi-serv)#allow-connections sip to h323
CUBE(conf-voi-serv)#allow-connections sip to sip
It is mandatory to have Incoming and Outgoing VoIP Dial-peers with required
parameters like Protocol, Transport, Codec, CAC, QoS, etc

Q16: Whether CUBE supports Universal


Transcoding ?
A: Yes CUBE supports Universal Transcoding, any to any Voice codec. Example.
iLBC to G.711 or iLBC to G.729
List of codecs supported

g711alaw 64Kbps

g711ulaw 64Kbps

g723r53 5.3Kbps

g723r63 6.3Kbps

g723r63 6.3Kbps

g729 (all variants) 8Kbps

iLBC

Q17: What is CAC?


A: CAC is Call Admission Control. It control number of calls based on resources and
bandwidth and Proactively reserve resources for good quality video calls. Also it
ensure traffic adheres to QoS policies within each network. CUBE can provide six
different CAC mechanisms

Q18: Is CP-PWR-CUBE-4= compatible with the 7900


series phones?
A: No this power cube is for the new 8900/9900 phones, compatable with
Communications Manager. The power tip is different size, won't fit in other phones.

Q19: Im trying to find a way to route inbound SIP calls,


based on their source.
For example:
SIPCarrier1 is coming from 1.1.1.1

SIPCarrier2 is coming from 2.2.2.2


If calls originate from SIPCarrier1, I want them to be passed to my UCM Cluster.
If call originate from SIPCarrier2, I want them to be passed to my Asterisk box.
I've played around with incoming called-number, trying to make it work with IP
addresses, rather than phone numbers, but no luck.
Any ideas on how to enable source based routing?
A: Yes this can be achieved by Carrier based Routing. Refer this link for more
details.
VoIP Gateway Trunk and Carrier Based Routing Enhancements

Q20: CUBE Sizing, how many simultaneous calls/


sessions are supported on each Router model to
terminate a SIP Trunk at the service provider end?
A: Refer the given table for more details.

Q21: What are the types of CUBE licenses available?


A:Two types of Cisco Unified Border Element licenses are available: Pay-as-You-Grow, or session count, license
Platform, or flat, license
Cisco Unified Border Element licenses are available in both types on select
platforms. Cisco Gatekeeper licenses are available in the platform license type only.

Pay-as-You-Grow License
This type of license covers the right to use the feature as well as the maximum
session count allowed. An example is the FL-CUBE-25 license, which allows up to
25 sessions.
This license is designed to allow a specific number of sessions (or calls) on a
platform. Purchase only as many sessions as are required in your deployment. You
can add more licenses later as your needs expand, thus offering pay-as-you-grow
benefits. Session licenses are stackable. These licenses are available on select
platforms as given in Table 3, and are available on all software images. Examples
include the FL-CUBE-4 and FL-CUBEE-100 licenses.

Platform License
This type of license covers the right to use the feature up to the maximum session
count supported on the chosen platform. An example is the FL-INTVVSRV-2811
license, which allows the maximum number of sessions the Cisco 2811 platform
supports.
These licenses are available on select platforms. These licenses require a software
image.Examples include the FL-INTVVSRV-2801 and FL-GK-3945 licenses.

Cisco Unified Border Element Licenses


For Active/Standby redundant configurations, use the "Redundant-Platform"
licenses. For all other configurations, including single-platform Active/Active load
balancing, and Inbox redundant configurations, use the "Single-Platform" licenses.
Q: Can we Integrate CUBE with Siemens OSV. Can CUBE do it? What are the
element in specific needed for Integration and deployment ? Please share a sample
configurations?
A: We have validated the integration with Siemens HiPath 4000. More details on the
integration is:
http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns833/698642.p
df

Q22: I am trying to troubleshoot the Multicast MOH on


Cube, how to do it?
Following commands will help you to analyse and troubleshoot the MOH issue you
are facing on CUBE.
Debug commands:

debug ccm-manager music-on-old all

debug voip rtp

debug ccsip all


Show commands:

show version

show running configuration


When MMOH is being streamed, please collect the following output on CUBE:

show ccm-manager music-on-hold

show voip rtp connections

show call active voice compact


Note: Please collect the logs using buffered logging mechanism. You can configure
the following:
no logging queue-limit
logging buffered 80000000
no logging rate-limit
no logging console

Q23: Debug commands on CUBE


DEBUG COMMANDS - Make sure to clear logg before call is made & get show logg
after call is done
1. H.323 - H.323 Scenarios
debug h225 asn1
debug h225 events
debug h245 asn1
debug h245 events
debug h225 q931
debug cch323 all
debug voip ipipgw
debug voip ccapi inout
2. H.323 - SIP Scenarios
debug h225 asn1
debug h225 events
debug h245 asn1
debug h245 events
debug cch323 all
debug voip ipipgw
debug voip ccapi inout
debug ccsip all

3. SIP - SIP Scenarios


debug ccsip all
debug voip ccapi inout
4. Transcoder related scenarios
Apart from the debugs mentioned above based on the scenario
debug dspfarm all
debug sccp messages

Verifying Fundamental Cisco Unified Border Element


Configurations
To verify Cisco Unified Border Element feature configuration and operation, perform
the following steps (listed alphabetically) as appropriate.
Step 1 show call active video
Use this command to display the active video H.323 call legs.
Step 2 show call active voice
Use this command to display call information for voice calls that are in progress.
Step 3 show call active fax
Use this command to display the fax transmissions that are in progress.
Step 4 show call history video
Use this command to display the history of video H.323 call legs.
Step 5 show call history voice
Use this command to display the history of voice call legs.
Step 6 show call history fax
Use this command to display the call history table for fax transmissions that are in
progress.
Step 7 show crm
Use this command to display the carrier ID list or IP circuit utilization.
Step 8 show dial-peer voice
Use this command to display information about voice dial peers.
Step 9 show running-config
Use this command to verify which H.323-to-H.323, H.323-to-SIP, or SIP-to-SIP
connection types are supported.
Step 10 show voip rtp connections

Use this command to display active Real-Time Transport Protocol (RTP)


connections.
This FAQ Document was created from the Cisco Unified Border Element
related discussions in Cisco Support Community

Solution
Show commands to Identify the active call count on SIP:
Show

SBC03#Show call active voice compact


Number of call-legs counted during viewing: 1448
SBC03#Show voip rtp connections
Found 1459 active RTP connections
SBC03#show call active voice brief | incl call-legs
Telephony call-legs: 0
SIP call-legs: 1053
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 410
Multicast call-legs: 0
Total call-legs: 1463
Telephony call-legs: 0
SIP call-legs: 1050
H323 call-legs: 0

Call agent controlled call-legs: 0


SCCP call-legs: 410
Multicast call-legs: 0
Total call-legs: 1460
is not equal to the initial count. Some call-legs were
Number of call-legs counted during viewing: 1460
SBC03#
SBC03#
SBC03#
SBC03#sh sip-ua call su
Total SIP call legs:1066, User Agent Client:526, User Agent Server:540

SBC03#sh call active voice summary


Telephony call-legs: 0
SIP call-legs: 38
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 38
SBC03#sh sip-ua calls summary
Total SIP call legs:44, User Agent Client:19, User Agent Server:25

SBC03#sh call active voice summary


Telephony call-legs: 0
SIP call-legs: 38
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 38
Since the above example is only on SIP-SIP calls you can see from active call Summary
No. of SIP-SIP Calls = SIP call leg / 2 = 19

But there can be more on total legs as there can some calls being generated. If you have MT
take no. of SIP legs /2 for SIP-SIP calls.

- See more at: https://supportforums.cisco.com/document/123281/how-check-active-callcount-sip#sthash.QBf5sBeP.dpuf

Introduction
This document covers the Configuration procedures with deployment examples to Implement Call
Admission Control (CAC) on Cisco Unified Border Element (CUBE). Call Admission Control plays a
major role in the network to control the number of calls based on the available resources and
bandwidth.

What is CAC and Why it is required ?


CAC is nothing but Call Admission Control which
1.

Controls number of calls based on resources & bandwidth

2.

Proactively reserve resources for good quality video calls

3.

Ensure traffic adheres to QoS policies within each network

CUBE can provide six different CAC mechanisms based on,

Total calls

CPU

Memory

IP call capacity

Max-connections

Call Spike detection

Dial-peer / Interface bandwidth

RSVP

CAC mechanisms ensure good QoS for video and voice calls and help meet the SLA

CAC Implementation

1. CAC Based on Total Calls, CPU or Memory

Configura

call threshold global [total/mem/cpu] calls low xx high yy


call treatment on

Global Command

call threshold global [total-calls | cpu-5sec | cpu-avg | total-mem | low <low-threshold> high

call treatment on

call treatment cause-code ?

busy

Insert cause code indicating the GW is busy (17)

no-QoS Insert cause code indicating the GW can't provide QOS (49)
no-resource Insert cause code indicating the GW has no resource (47)

Call threshold values for total concurrent calls or CPU or memory to be


handled by CUBE can be defined

Call treatment can be turned on to handle the call once the CAC limit is
reached

2. CAC Based on IP Call Capacity

Configura

gatekeeper#
endpoint circuit-id h323id CUBE1 AA max-calls 500

CUBE#

voice service voip


allow-connections h323 to h323
h323
ip circuit max-calls 1500
ip circuit carrier-id AA reserved-calls 1000 <Note Number is twice because of 2 call legs>

IP call capacity on CUBE works in conjunction with the Cisco Gatekeeper


(GK)

Call counting mechanism does not take into account the codec type used
this is taken into account by enabling bandwidth management on the Cisco GK

This only works if at least 1 call leg on CUBE is using H.323

3. CAC Based on Max Connections per destination

Configura

CUBE#
dial-peer voice 1 voip
max-conn 2

Restricting the number of concurrent calls that can be active on a VoIP dial
peer

Max-Conn works on individual dial-peers, does not provide CAC for the entire
gateway

4. CAC based on Call Spike detection


CUBE rejects calls if call spike is detected

Configura

call spike call-number [steps number-of-steps size milliseconds]


call spike 10 steps 5 size 200

5. CAC based Dial-peer or interface bandwidth

Configura

dial-peer voice 1 voip


max-bandwidth 160

6. CAC based on RSVP

Configura

interface FastEthernet0/0
ip rsvp bandwidth 1000 1000

dial-peer voice 10 voip


destination-pattern 2...
session target ras
req-qos guaranteed-delay audio
req-qos guaranteed-delay video
acc-qos guaranteed-delay audio
acc-qos guaranteed-delay video

Synchronization of RSVP with H.323 signaling to ensure that the bandwidth


reservation is established
RSVP ensures that bandwidth is reserved before the far end phone rings

Behavior - For CAC based on Total Calls.

With the call threshold command, you can configure two thresholds, high and low, for
each
resource. Call treatment is triggered when the current value of a resource exceeds
the
configured high. The call treatment remains in effect until the current resource value
falls
below the configured low. Having high and low thresholds prevents call admission
flapping
and provides hysteresis in call admission decision making.
I have this configuration:
call treatment cause-code busy
call treatment on
call threshold global total-calls low 2 high 3

////Zero Calls
R_ISOL_HQ_7.06_36#show call threshold status
Status IF
------ --Avail N/A

Type
Value Low High Enable
---------- ---- ---- -----total-calls 0
2
3
busy&treat

R_ISOL_HQ_7.06_36#

////One Call - Ok
R_ISOL_HQ_7.06_36#show call threshold status
Status IF
------ --Avail N/A

Type
Value Low High Enable
---------- ---- ---- -----total-calls 1
2
3
busy&treat

R_ISOL_HQ_7.06_36#

////Two Calls - Ok
R_ISOL_HQ_7.06_36#show call threshold status

Status IF
------ --Avail N/A

Type
Value Low High Enable
---------- ---- ---- -----total-calls 2
2
3
busy&treat

R_ISOL_HQ_7.06_36#

////Three Calls - Ok
R_ISOL_HQ_7.06_36#show call threshold status
Status IF
------ --Avail N/A

Type
Value Low High Enable
---------- ---- ---- -----total-calls 3
2
3
busy&treat

R_ISOL_HQ_7.06_36#

////Four Calls - We have Fast Busy on the IP Phone.


Under Status under the command "show call threshold status" we have
NonAv
R_ISOL_HQ_7.06_36#show call threshold status
Status IF
------ --NonAv N/A

Type
Value Low High Enable
---------- ---- ---- -----total-calls 3
2
3
busy&treat

R_ISOL_HQ_7.06_36#
In the console router we have
Feb 24 22:11:17.512: %CALLTREAT-3-HIGH_TOTAL_CALLS: High call volume.
Processing for callID(53874) is rejected.
On SIP Level, the CUBE send, back to the CUCM.
Feb 24 22:11:17.516: //53874/B01796000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 486 Busy here
Via: SIP/2.0/TCP 177.56.25.10:5060;branch=z9hG4bK16d798bab89
From: <sip:67952@177.56.25.10>;tag=11644~afccc525-5442-4756-a44e24c835482eb3-33492681
To: <sip:0082213224@177.56.25.4>;tag=8B736658-12FD
Date: Tue, 24 Feb 2015 22:11:17 GMT

Call-ID: b0179600-4ec1f9c5-b1-a1938b1@177.56.25.10
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M7
Reason: Q.850;cause=17
Content-Length: 0

////Drop one call of three, the status still in NonAv, because the value is not under the
Low Threshold. The new calls will be droped.
R_ISOL_HQ_7.06_36#show call threshold status
Status IF
------ --NonAv N/A

Type
Value Low High Enable
---------- ---- ---- -----total-calls 2
2
3
busy&treat

R_ISOL_HQ_7.06_36#
////Drop one call of two, the status now is Avail, because the value is under the Low
Threshold, and we can reach the High Threshold again.
R_ISOL_HQ_7.06_36#show call threshold status
Status IF
------ --Avail N/A

Type
-----total-calls

Value Low High Enable


----- ---- ---- -----1
2
3
busy&treat

R_ISOL_HQ_7.06_36#

***** If we change the cause code "R_ISOL_HQ_7.06_36(config)#call treatment


cause-code no-QoS " we have the next Cause Code Error on SIP, send to the
CUCM, on the fourth call.
Feb 24 22:23:14.918: //53881/5B751A800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 580 Precondition Failed
Via: SIP/2.0/TCP 177.56.25.10:5060;branch=z9hG4bK1812367926d
From: <sip:67952@177.56.25.10>;tag=11661~afccc525-5442-4756-a44e24c835482eb3-33492692
To: <sip:0082213224@177.56.25.4>;tag=8B7E58B8-13A9
Date: Tue, 24 Feb 2015 22:23:14 GMT
Call-ID: 5b751a80-4ec1fc92-b5-a1938b1@177.56.25.10
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M7

Reason: Q.850;cause=49
Content-Length: 0
***** If we change the cause code "R_ISOL_HQ_7.06_36(config)#call treatment
cause-code no-resource " we have the next Cause Code Error on SIP, send to the
CUCM, on the fourth call.
Feb 24 22:28:39.335: //53890/1C9394800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/TCP 177.56.25.10:5060;branch=z9hG4bK197162fe198
From: <sip:67952@177.56.25.10>;tag=11683~afccc525-5442-4756-a44e24c835482eb3-33492707
To: <sip:0082213224@177.56.25.4>;tag=8B834BFC-1061
Date: Tue, 24 Feb 2015 22:28:39 GMT
Call-ID: 1c939480-4ec1fdd6-bb-a1938b1@177.56.25.10
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M7
Reason: Q.850;cause=47
Content-Length: 0

- See more at: https://supportforums.cisco.com/document/71326/call-admissioncontrol-cac-implementation-cube#sthash.jKBtrcwr.dpuf