Session Initiation Protocol Team Members: Manjiri Ayyar Pallavi Murudkar Sriusha Kottalanka Vamsi...

Session Initiation Protocol

Team Members:

Manjiri AyyarPallavi Murudkar

Sriusha KottalankaVamsi AmbatiGirish SatyaLeeAnn Tam

Agenda Introduction to SIP Overview of functionality SIP components SIP protocol layers SIP call flows SIP concerns Demo Conclusion

Introduction

Session Initiation Protocol (SIP) application layer signaling protocol used to create, manage and terminate

sessions in an IP based network. RFC : 3261

Circuit switched Network

• Circuit is fully established between the two devices before data is sent.

• Less efficient since much of the bandwidth is wasted.

Packet switched network

• No fixed path is established between devices• Data broken into packets.• Packets may take multiple paths to reach the destination device.• More efficient.

SIP applications

• VoIP

• Video Conferencing

• Instant Messaging

Multimedia session in a packet switched networkA typical real-time multimedia session requires

Session management : Users may move from terminal to terminal with different capabilities. To set up communication session between two or more users, a signaling protocol is needed.

Media transport : RTP is used for transmitting real-time data like audio and video.

End-to-End delivery : Underlying IP layer which connects the whole world.

SIP functionalitySIP is limited to only the setup, modification and termination of sessions.

Establishment of user location Feature negotiation

Call management

Changing features while a session is in progress

All of the other key functions are done with other protocols

SIP components

The key components in a SIP network are

SIP Clients : SIP Phones (User-Agents) SIP servers SIP PSTN gateways Application servers (such as media

servers)

SIP Network

Application

Transport

Network

Physical/Data Link

Ethernet

IP

TCP UDP

RTSP SIP

SDP codecs

RTP DNS(SRV)

Transport

Transaction

Transaction User

Syntax and Encoding

start-line*message-headerCRLF[ message-body ]start-line = Request-Line / Status-Line

INVITE  Requests a session ACK   Final response to the INVITE OPTIONS  Ask for server capabilities CANCEL  Cancels a pending request BYE   Terminates a session REGISTER  Sends user’s address to server

1XX  Provisional  100 Trying

2XX  Successful  200 OK 

3XX  Redirection  302 Moved Temporarily 

4XX  Client Error  404 Not Found

5XX  Server Error  504 Server Time-out

6XX  Global Failure  603 Decline

Session Registration Establishment , TerminationRFC 3665

User A Registrar Server Location Server

Register sip:bob@lab.acme.com

bob@lab.acme.comContact 10.18.2.4

200 - OK Registration binds a particular device Contact URI with a SIP user Address of Record.

Alice Host1.com proxy

Host2.com proxy Bob

Invite F1Invite F1

Invite F2Invite F2Invite F4Invite F4

100 Trying F5100 Trying F5100 Trying F3100 Trying F3

180 Ringing F6180 Ringing F6

180 Trying F8180 Trying F8

180 Trying F7180 Trying F7

ACK F12ACK F12

200 OK F9200 OK F9

200 OK F11200 OK F11200 OK F10200 OK F10

Media Session

Bye F13Bye F13

200 OK F14200 OK F14

INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP

pc33.atlanta.com;branch=z9hG4bK776asdhds Max-Forwards: 70 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710@pc33.atlanta.com CSeq: 314159 INVITE Contact: sip:alice@pc33.atlanta.comContent-Type: application/sdp Content-Length: 142

SIP/2.0 200 OK Via: SIP/2.0/UDP

server10.biloxi.com ;branch=z9hG4bKnashds8;received=192.0.2.3 Via: SIP/2.0/UDP

bigbox3.site3.atlanta.com ;branch=z9hG4bK77ef4c2312983.1;received=192.0.2.2

Via: SIP/2.0/UDP pc33.atlanta.com ;branch=z9hG4bK776asdhds ;received=192.0.2.1

To: Bob <sip:bob@biloxi.com>;tag=a6c85cf From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710@pc33.atlanta.com CSeq: 314159 INVITE Contact: <sip:bob@192.0.2.4> Content-Type: application/sdp Content-Length: 131

SIP Concerns Security

◦ Authentication of signaling data using HTTP digest

authentication

◦ TLS usage (over TCP)

◦ Usage of IPSec (SIP VPN Scenario)

◦ Use SecureRTP for Media

◦ Use S/MIME to enable mechanisms like public key

distribution, authentication, integrity and

confidentiality of SIP signaling data

SIP Concerns…contd Quality of Service

◦ Latency, network delays (upper bound is 150ms)

◦ Jitter ( refers to non-uniform delays )

◦ Packet Loss

◦ Power Failure and Backup Systems

◦ Interoperability

Demo

User Agents used : Yahoo Messenger

Call Scenarios Covered:

◦Register

◦Call Establishment

◦Call Termination