Post on 27-May-2015
SIP: SESSION INITIATION PROTOCOL
CMPE 208 FALL 2008 PROJECT
Chinmay PadhyeAmit MoreAbhishek SharmaNihar Dandekar
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
INTRODUCTION
LINEAGE : OSI Model – Layer 6 (Session Layer) TCP/IP Model – Layer 5 (Application Layer)
Protocols supported: RSVP RTP RTCP RTSP SAP SDP
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
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.
SIP - ENTITIES
SIP uses the following main Entities: USER AGENT CLIENT USER AGENT SERVER PROXY SERVER REDIRECT SERVER REGISTRAR / LOCATION SERVER
ENTITIES – UAC , UAS & REGISTRAR
ENTITIES – PROXY & REDIRECT SERVERS
SIP - SYNTAX
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)
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
SIP RESPONSES
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
TEST BED
User Agent Client (UAC) - Xlite - 3CX - SJphone
User Agent Server - 3CX - Hamachi
Packet analyzer - Wireshark
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
SOFT-PHONE REGISTRATION
SOFT-PHONE REGISTRATION
SIMPLE CALL SETUP
SIMPLE CALL SETUP – CALL ACCEPTED
SIMPLE CALL SETUP – CALL ACCEPTED
SIMPLE CALL SETUP – CALL IGNORED
SIMPLE CALL SETUP – CALL IRNORED
SIMPLE CALL SETUP – PHONE UNREGISTERED
SIMPLE CALL SETUP – PHONE UNREGISTERED
CALL FORWARDING
CALL FORWARDING – TO VOICEMAIL
CALL FORWARDING – TO VOICEMAIL
CALL FORWARDING – TO VOICEMAIL
CALL FORWARDING – TO EXTENSION
CALL FORWARDING – TO EXTENSION
2 WAY CALL FORKING
2 WAY CALL FORKING
2 WAY CALL FORKING
2 WAY CALL FORKING
3 WAY CALL FORKING
3 WAY CALL FORKING
3 WAY CALL FORKING
SECURE CALL CONNECTION
SECURE CALL CONNECTION
SECURE CALL CONNECTION
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
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/
QUESTIONS ?