1. CMPE 208 FALL 2008 PROJECT
Chinmay Padhye
Amit More
Abhishek Sharma
Nihar Dandekar
SIP: SESSION INITIATION PROTOCOL

2. INTRODUCTION
Developed originally as MULTIPARTY MULTIMEDIA

SESSION CONTROL IN 1999 -- RFC 2543 (SIPv1)
Latest revision RFC 3261 thru 3265 in June 2002 (SIPv2)

A powerful alternative to H.323 protocol

Is used for:

Initiating SESSIONS of multimedia over the Internet

transport session description from caller to callees

Change of parameters in mid-session

Terminate the session


3. INTRODUCTION
LINEAGE :

OSI Model – Layer 6 (Session Layer)

TCP/IP Model – Layer 5 (Application Layer)
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Protocols supported:

RSVP

RTP

RTCP

RTSP

SAP

SDP


4. INTRODUCTION
Applications:

IP PBX

IP TELEPHONEY

INSTANT MESSEGING

INTERNET CONFERENCING

Features:

Uses the client – server model

Both the client and server can be on the same platform

Uses the concept of intelligent endpoint


5. DISTRIBUTED FUNCTIONALITY
De-centralization permits more functionality within each component.

Changes made to specific components have a minor impact on the rest

of the system. It is possible to connect one SIP phone to another with an
Ethernet cable & make calls between the sets without the aid of any
other server modules.
The other system components become useful when the network

requires more than two phones.

6. SIP - ENTITIES
SIP uses the following main Entities:
USER AGENT CLIENT

USER AGENT SERVER

PROXY SERVER

REDIRECT SERVER

REGISTRAR / LOCATION SERVER


7. ENTITIES – UAC , UAS & REGISTRAR

8. ENTITIES – PROXY & REDIRECT SERVERS

9. SIP - SYNTAX

10. SIP - METHODS
INVITE initiate call

ACK confirm final response

BYE terminate (and transfer) call

CANCEL cancel searches and “ringing”

OPTIONS features support by other side

REGISTER register with location service

INFO mid-call information (ISUP)

PRACK provisional acknowledgement

SUBSCRIBE subscribe to event

NOTIFY notify subscribers

REFER ask recipient to issue SIP request (call transfer)


11. SIP – REQUEST & RESPONSES
 In text format
Look very similar to HTTP/1.1

Requests and responses are similar except for first line

Requests and responses can contain in there message

bodies
ASCII

HTML

SESSION DESCRIPTION


12. SIP RESPONSES

13. AUTHENTICATION & ENCRYPTION
SIP supports a variety of approaches:

End to end encryption

Hop by hop encryption

End to end encryption implemented using proxy servers that

form a tunnel between peers after authentication
Responds to INVITEs with 407 Proxy-Authentication Required


14. TEST BED
User Agent Client (UAC)

- Xlite

- 3CX

- SJphone

 User Agent Server
 - 3CX
- Hamachi

 Packet analyzer
 - Wireshark

15. TEST CASES
Soft-phone registration

Simple call setup

Call accepted

Call ignored

Soft-phone unregistered

Call forwarding

To voice mail

To extension
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Call forking

2 way parallel call forking

3 way parallel call forking

Secure call connection via HAMACHI server


16. SOFT-PHONE REGISTRATION

17. SOFT-PHONE REGISTRATION

18. SIMPLE CALL SETUP

19. SIMPLE CALL SETUP – CALL ACCEPTED

20. SIMPLE CALL SETUP – CALL ACCEPTED

21. SIMPLE CALL SETUP – CALL IGNORED

22. SIMPLE CALL SETUP – CALL IRNORED

23. SIMPLE CALL SETUP – PHONE UNREGISTERED

24. SIMPLE CALL SETUP – PHONE UNREGISTERED

25. CALL FORWARDING

26. CALL FORWARDING – TO VOICEMAIL

27. CALL FORWARDING – TO VOICEMAIL

28. CALL FORWARDING – TO VOICEMAIL

29. CALL FORWARDING – TO EXTENSION

30. CALL FORWARDING – TO EXTENSION

31. 2 WAY CALL FORKING

32. 2 WAY CALL FORKING

33. 2 WAY CALL FORKING

34. 2 WAY CALL FORKING

35. 3 WAY CALL FORKING

36. 3 WAY CALL FORKING

37. 3 WAY CALL FORKING

38. SECURE CALL CONNECTION

39. SECURE CALL CONNECTION

40. SECURE CALL CONNECTION

41. CONCLUSION
SIP is:

Relatively easy to implement

Gaining vendor and carrier acceptance

Very flexible in service creation

Extensible and scalable

Appearing in products right now

SIP provides its own reliability mechanism & is therefore

independent of the packet layer and only requires an
unreliable datagram service

42. REFRENCES
[1] http://faq.programmerworld.net/voip/voip.htm

[2] http://groups.google.com/group/SJSUee284/files

[3] http://ezinearticles.com/?The-SIP-Advantage&id=270970

[4] Internet Telephony based on SIP

SMU - Dallas

April 28, May 1, 2000

Henry Sinnreich, MCI WorldCom

Alan Johnston, MCI WorldCom

[5]http://books.google.com/books?hl=en&lr=&id=VMP6gCBazzIC&oi=fnd&pg=PR17&dq=project+on+call+flow+using+SIP+protocol&ots=EtmKee0_M3&sig=bjqG

[6] Evaluating SIP Proxy Server Performance

Erich M. Nahum, John Tracey, and Charles P. Wright

IBM T.J. Watson Research Center

Hawthorne, NY, 10532

fnahum,traceyj,cpwrightg@us.ibm.com

[7] Session Initiation Protocol (SIP) and other Voice over IP (VoIP) protocols and applications Henrik Ingo1

[8] http://www.3cx.com/phone-system/

[9] http://en.wikipedia.org/wiki/Session_Initiation_Protocol

[10] http://tools.ietf.org/html/rfc3261

[11] http://www.counterpath.com/x-lite.html

[12] http://www.counterpath.com/assets/files/191/X-Lite3.0_UserGuide.pdf

[13] http://www.qgpop.net/2003fukuoka/papers/A7-3.pdf


[14] http://en.wikipedia.org/wiki/Session_Initiation_Protocol


[15] Carrier Grade VoIP - Daniel Collins – McGraw-Hill – NETWORKING eBOOK


[16]http://www.radvision.com/NR/rdonlyres/0AFA30DF-DAD6-461D-943C-ED33F3E7ABD8/0/SIPServerTechnicalOverviewWhitepaper.pdf
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
[17] http://en.wikipedia.org/wiki/Hamachi
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[18] http://www.cmpe.sjsu.edu/~fclin/
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43. QUESTIONS ?