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Session Initiation Protocol - In depth analysis

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  • 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)  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  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   [17] http://en.wikipedia.org/wiki/Hamachi   [18] http://www.cmpe.sjsu.edu/~fclin/ 
  • 43. QUESTIONS ?