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OUTLINE
INTRODUCTION ADVANTAGES OF VoIP POPULAR VoIP PROTOCOLS H.323 SIP MGCP SUPPORTING PROTOCOLS TECHNICAL ISSUES HARWARE REQUIREMENTS SOFTWARE REQUIREMENTS PRODUCTS SERVICES FUTURE DEVELOPMENTS CONCLUSION
INTRODUCTION
VoIP - The ability to carry toll quality voice using compression techniques and packet switching over the IP packet network.
Voice
CODEC: CODEC: Digital to Analog
Voice analog
analog
Analog to Digital
Compress
Decompress
Add Header
digital
Process Header
digital
INTRODUCTION (contd)
Real time voice traffic can be carried over IP networks in three different ways
1. PC to PC
2.
PC to Phone
3.
Phone to Phone
INTRODUCTION (contd)
Protocols commonly implemented by Voice over IP 1. H.323 2. SIP (Session Initiation Protocol) 3. MGCP (Media Gateway Control Protocol) 4. RSVP (Resource Reservation Protocol)
ADVANTAGES OF VoIP
INTEGRATION OF VOICE AND DATA: Web servers capable of interacting with voice, data and images. SIMPLIFICATION: Allows more standardization and less equipment management.
NETWORK EFFICIENCY: Provides bandwidth consolidation. COST REDUCTION: Slashes high charges for long distance calls.
H.323
H.323 is a standard that specifies the components, protocols and procedures that provide multimedia communication services such as real-time audio, video, and data communications over packet networks, including Internet Protocol (IP) based networks.
COMPONENTS OF H.323
1. 2. TERMINALS : Can be either a personal computer or a stand-alone device GATEWAYS : A H.323 gateway provides connectivity between an H.323 network and a nonH.323 network. GATEKEEPERS : Provide call control services such as address translation, bandwidth management, admission control and zone management. MULTIPOINT CONTROL UNITS (MCU) : Manage conference resources, negotiate between terminals for the purpose of determining the audio or video coder/decoder to use, and may handle the media stream.
3.
4.
CONTROL SIGNALING 1. The H.245 standard provides the call control mechanism that allows H.323-compatible terminals to connect to each other. 2. The control messages that it carries relate to: Opening and closing of logical channels used to carry media streams, preference requests, flow-control messages and general commands and indications.
SIP
The Session Initiation Protocol (SIP) is an application layer signaling protocol that defines initiation, modification and termination of interactive multimedia communication sessions between users. It was developed by the IETF and is explained in RFC 2453. It was approved in early 1999.
Utility Media
CODEC DNS SDP RTP/RTCP UDP IP Physical TCP
Signaling
SIP
SIP ARCHITECTURE
SIP Servers
Registrar
Redirect
IP Network
3xx Redirection They moved, try this address REGISTER Here I am INVITE I want to talk to another UA
Proxy Server
sip:hostname@192.168.10.1 sip:14083831088@vovida.org
SIP Gateway
SIP-GW
PSTN
SIP Redirect & Location Servers (Ieee.org) 3 SIP Proxy 2 5 6 7 (sjsu.edu) 11 10 12 9 8 (play.com) 4 SIP Proxy
9. Ringing ok 10. Ringing ok 11. ACK 12. ACK
5. Bob moved. Temporarily contact bob@play.com 6. ACK 7. INVITE bob@play.com 8. INVITE bob@play.com
Signaling and Media UDP and TCP for signaling, RTP for Media Conferencing Client Relationship Security Session Description Multicasting. No restrictions on no. of users Intelligent User Agents Peer-to-peer Registration with Registrar SDP
MGCP
Media Gateway Control Protocol is a master-slave protocol that defines communication between telephony Gateways and external call control elements called Media Gateway Controllers or Call Agents. It was developed by the IETF and explained in RFC 2705. It assumes limited intelligence at endpoints and concentrates it in the core of the network. Call Agent (master) provides call signaling, control and processing intelligence to the Gateway sends and receives commands to/from Gateway Gateway (slave) provides translations between packet and circuit switched networks sends notification to the call agent about endpoint events.
MGCP ARCHITECTURE
SUPPORTING PROTOCOLS
SIP
RSVP
RTCP
RTP
SAP/SDP
H.323
UDP
TCP
TECHNICAL ISSUES
Quality of service Delay, jitter, congestion, echo, packet loss, misordered packet arrival Measure of QoS The mean opinion score is widely used Algorithms: PSQM, PAMS and PESQ Bandwidth consumption A quality call requires at least 64 kbps. It is impossible to dedicate so much for voice on data network Speech compression techniques are used. For example, silence compression which brings down the bandwidth to 5-6 kbps
HARDWARE REQUIREMENTS
Minimum Requirements PC 386 or higher Sound card Full duplex capability Network card or connection to internet or other kind of interface to allow communication between 2 PCs Companies offering hardware Quicknet, Lucent, 3COM, Cisco, Nortel, Alcatel Hardware accelerating cards Quicknet PhoneJack Quicknet LineJack VoiceTronix V4PCI VoiceTronix VPB4 VoiceTronix VPB8L
SOFTWARE REQUIREMENTS
Operating Systems Windows 95, 98, 2000, ME and XP Linux Gateway Internet Switch Board PSTNGW (Packet Switching Transfer Network Gateway) Gatekeeper
PRODUCTS
Gateways: MICOM V/IP Gateway, Nortel Networks CVX SS7 Gateway, Lucent Technologies Pathstar Access Server, Cisco Systems DE-30+ Gateway, 3Com Gateway, VocalTec Series 2000 Gateway, Nuera Solutions Access plus F200 IP Gatekeepers: Eriksson H.323 gatekeeper, VocalTec Gatekeeper, Nortel Netwroks IPConnect, Elemedia H.323 gatekeeper GK2000S
SERVICES
IP telephones: Cisco's IP phones, Selsius IP phones, Nokia Systems IPCourier PC based software phones: VocalTec IPhone, Netscapes CoolTalk, Microsoft NetMeeting, WhitePines CU-SeeME Pro
FUTURE DEVELOPMENTS
Directory services over telephones Inter office trunking over the corporate intranet Remote access to voice, data and fax services of office from home Fax over IP Conference bridging Voice/data synchronization Text to speech conversion
CONCLUSION
VoIP sends voice over data networks instead of data over voice network Internet along with TCP/IP are driving forces for VoIP technology Ideal for computer based communications Market for VoIP is established and is rapidly growing VoIP cuts communication costs and improves efficiency Needs QoS for acceptable quality
REFERENCES
www.protocols.com www.cis.ohiostate.edu/~jain/refs/ref_voip.htm www.iec.org/online/tutorials/vfoip/ www.nwfusion.com/research/voip.html SIP Understanding the Session Initiation Protocol by Alan B. Johnston
Q&A
Thank You!