Voice over IP (VoIP)

Overview

Why VoIP?
Voice Codec
Signaling & Control Protocols
Quality of Service
Design a VoIP network

Why VOIP ?
Cost reduction
Toll by-pass

WAN Cost Reduction

Operational Improvement
Common network infrastructure
Simplification of Routing Administration

Business Tool Integration

Voice mail, email and fax mail integration
Web + Call
Mobility using IP

Yesterdays Network

Converged Network

Components of VOIP
Coding & Decoding of Analog Voice
Analog-to-Digital and Digital-to-Analog conversions
Compression
Signaling
Call setup & tear down
Resource & coding negotiation
Transport of Bearer Traffic
Voice packet transmission
Routing
Support of quality of service
Numbering
Phone number, IP address

What Protocols are Required ?

Signaling Protocol: To establish presence, locate user, set up, modify

and tear down sessions.

Media Transport Protocols: To transmit packetized audio/video signal.

Supporting Protocols: Gateway Location, QoS, AAA,

Address Translation, etc.

VOIP Protocols
H.323:
ITU-T standard, latest version v4
Peer-to-peer protocol that supports terminals communicating
over packet based networks

SIP:
IETF standard, RFC 3261
Peer-to-peer protocol for initiation, modification termination of
communication sessions between users

MGCP:
ITU-T and IETF collaboration, RFC 3435
Master/slave protocol for media gateway controller to control media gateway.

VOIP Protocol Stacks

VOIP Using H.323

ITU-T standard, latest version v4
Peer-to-peer protocol that supports terminals communicating over packet based
networks
H.323 includes
Call signaling H.225
Media control H.245
Audio coding G.711, G.722, G.723, G.728, G.729
Video coding H.261, H263
Data sharing T.120
Media transport RTP, RTCP

Powerful for video-conferencing

Widely deployed

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H.323 Components

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Components Defined in H.323

H.323 defines four major components for a packet network based multimedia communication
H.323 Terminal: Client end points that provides real-time communications with other H.323
entities. Functions performed include (1) signaling and control, (2) real-time communications,
and (3) codec.
Gateway: Provides the connection path between the packet switched network and the
switched circuit network.
Gatekeeper: Performs address translation, admission control, bandwidth control, zone
management, call control signaling, call authorization, bandwidth management, call
management.
Multipoint control unit (MCU): Supports conferencing between three or more endpoints.
MCU consists of multipoint controller (MC) and multipoint processor (MP).
The collection of all terminals, gateways, and MCUs managed by a single gatekeeper is
called a zone.

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SIP Registration

SIP Registration
Establishes presence of user with an
address (eg.dave@abc.com)
Binds this address to users current
location (199.147.77.23)

Register sip:abc.com SIP/2.0

From sip:dave@abc.com
Contact <sip:199.147.67.23>
Expires 3600 1

dave@ 199.147.67.23

Location sever

SIP register
SIP / 2.0 200 OK

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SIP Operation with Proxy Server

Location Server

OK 200
VIA: dave@abc.com
FROM: sip:dave@abc.com
TO: sip:john@xyz.com
Call-id: 1234@abc.com

Where is john?

INVITE sip:john@xyz.com
VIA: dave@abc.com
FROM: sip:dave@abc.com
TO: sip:john@xyz.com
Call-ID: 1234@abc.com

john@195.127.75.123

3
4

INVITE sip:john@195.127.75.123
VIA: proxy@internet.com
VIA: dave@abc.com
FROM: sip:dave@abc.com
TO: sip:john@xyz.com
Call-id: 1234@abc.com

5
Proxy Server

ACK sip:john@xyz.com
VIA: dave@abc.com
FROM: sip:dave@abc.com
TO: sip:john@xyz.com
Call-id: 1234@abc.com

Media Streams
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OK 200
VIA: proxy@internet.com
VIA: dave@abc.com
FROM: sip:dave@abc.com
TO: sip:john@xyz.com
Call-id: 1234@abc.com

SIP operation with Redirect Server

Location Server

302 moved temporarily

Contact john@def.com
ACK john@def.com

Where is
john

john@def.
com

INVITE sip:john@xyz.com
VIA dave @abc.com
1 From sip:dave@abc.com
To sip:john@xyz.com
Call ID 1234@abc.com

Redirect Server
4
INVITE sip:john@def.com
VIA dave@abc.com
To sip:john@def.com
Call ID 1234@abc.com

5
6

OK 200

8
ACK john@def.com
Media Stream
9

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Session Initiation Protocol (SIP)

Peer-to-peer protocol for initiation, modification termination of communication
sessions between users
Two components: user agents and network servers
User agents: client end-system applications
User-agent client (UAC): originate calls
User-agent server (UAS): listen for incoming calls

Network servers:
Proxy server: relay calls, act as both client and server
Redirect server: redirect calls to other servers
Registrar: accept user registration

Simple text based messages

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SIP Messages
SIP is a text-based protocol with message syntax and header fields identical to HTTP
Message header includes:
General header
Entity header

Two kinds of messages:

Requests initiated by client
Responses returned by server

Methods defined in RFC 2543:

INVITE: to invite the server to participate in a session
ACK: to accept the INVITE to participate
OPTIONS: to inquire capability
BYE: to terminate a session
CANCEL: to cancel any in-progress request
REGISTER: a client to register location information with a server

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SIP Messages

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