Session Initiation Protocol

1
Session Initiation Protocol

• Team Members
• Manjiri Ayyar
• Pallavi Murudkar
• Sriusha Kottalanka
• Vamsi Ambati
• Girish Satya
• LeeAnn Tam

2
Agenda

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

3
Introduction

• Session Initiation Protocol (SIP)
• application layer signaling protocol
• used to create, manage and terminate sessions in
  an IP based network.
• RFC 3261

4
Circuit switched Network

• Circuit is fully established between the two
  devices before data is sent.
• Less efficient since much of the bandwidth is
  wasted.

5
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.

6
SIP applications

• VoIP
• Video Conferencing
• Instant Messaging

7
Multimedia session in a packet switched network

• A 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.

8
SIP functionality

• SIP 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

9
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)

10
SIP Network
11
Wheres SIP

• Application
• Transport
• Network
• Physical/Data Link

12
SIP Protocol Layers
Transaction User
Transaction
Transport
Syntax and Encoding
13
SIP - Messages

• start-line
• message-header
• CRLF
• message-body
• start-line Request-Line / Status-Line

14

• 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 users address to server

15
SIP - Responses

• 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

16
SIP Call Scenarios

• Session Registration Establishment , Termination
• RFC 3665

17
User A
Registrar Server
Location Server
Register sipbob_at_lab.acme.com
bob_at_lab.acme.com Contact 10.18.2.4
200 - OK
Registration binds a particular device Contact
URI with a SIP user Address of Record.
18
SIP call with Proxy Server
Host2.com proxy
Host1.com proxy
Alice
Bob
Invite F1
Invite F2
Invite F4
100 Trying F3
100 Trying F5
180 Ringing F6
180 Trying F7
180 Trying F8
200 OK F9
200 OK F10
200 OK F11
ACK F12
Media Session
Bye F13
200 OK F14
19
SIP INVITE

• INVITE sipbob_at_biloxi.com SIP/2.0
• Via SIP/2.0/UDP pc33.atlanta.combranchz9hG4bK77
  6asdhds
• Max-Forwards 70
• To Bob ltsipbob_at_biloxi.comgt
• From Alice ltsipalice_at_atlanta.comgttag1928301774
  
• Call-ID a84b4c76e66710_at_pc33.atlanta.com
• CSeq 314159 INVITE
• Contact sipalice_at_pc33.atlanta.com
• Content-Type application/sdp
• Content-Length 142

20
SIP - Response

• SIP/2.0 200 OK
• Via SIP/2.0/UDP server10.biloxi.com
  branchz9hG4bKnashds8received192.0.2.3
• Via SIP/2.0/UDP bigbox3.site3.atlanta.com
  branchz9hG4bK77ef4c2312983.1received192.0.2.2
  
• Via SIP/2.0/UDP pc33.atlanta.com
  branchz9hG4bK776asdhds received192.0.2.1
• To Bob ltsipbob_at_biloxi.comgttaga6c85cf
• From Alice ltsipalice_at_atlanta.comgttag1928301774
  
• Call-ID a84b4c76e66710_at_pc33.atlanta.com
• CSeq 314159 INVITE
• Contact ltsipbob_at_192.0.2.4gt
• Content-Type application/sdp
• Content-Length 131

21
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

22
SIP Concernscontd

• Quality of Service
• Latency, network delays (upper bound is 150ms)
• Jitter ( refers to non-uniform delays )
• Packet Loss
• Power Failure and Backup Systems
• Interoperability

23
Demo

• User Agents used Yahoo Messenger
• Call Scenarios Covered
• Register
• Call Establishment
• Call Termination