1. np160 Dennis Baron, January 15, 2008 Page 1 SIP Fundimentals IAP 2008 VoIP Series Dennis Baron January 15, 2008

2. np160 Dennis Baron, January 15, 2008 Page 2 Outline What is SIP SIP system components SIP messages and responses SIP call flows SDP basics/CODECs IS&T Services Questions and answers

3. np160 Dennis Baron, January 15, 2008 Page 3 What’s SIP IETF Standard defined by RFC 3261 “The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol for creating, modifying and terminating sessions with one or more participants.” Can be used for voice, video, instant messaging, gaming, etc., etc., etc. Uses URIs for addressing – single communications identity –mailto:dbaron@MIT.edu for email –xmpp:dbaron@MIT.EDU for instant messaging –sip:dbaron@MIT.EDU for voice and video Username replaced by numbers for telephone applications

4. np160 Dennis Baron, January 15, 2008 Page 4 Where’s SIP Application Transport Network Physical/Data Link Ethernet IP TCPUDP RTSP SIP SDPcodecs RTPDNS (SRV)

5. np160 Dennis Baron, January 15, 2008 Page 5 SIP Components User Agents –Clients – Make requests –Servers – Accept requests Server types –Redirect Server –Proxy Server –Registrar Server –Location Server Gateways

6. np160 Dennis Baron, January 15, 2008 Page 6 SIP Trapezoid DNS Server Location Server Terminating User Agent Outbound Proxy Originating User Agent DNS SIP RTP Registrar Inbound Proxy SIP

7. np160 Dennis Baron, January 15, 2008 Page 7 SIP Triangle ? DNS Server Location Server Terminating User Agent Originating User Agent DNS SIP RTP Registrar Inbound Proxy SIP

8. np160 Dennis Baron, January 15, 2008 Page 8 Terminating User Agent Originating User Agent RTP SIP B2BUA SIP Peer to Peer ! Back-to-Back User Agent Terminating User Agent Originating User Agent SIP RTP

9. np160 Dennis Baron, January 15, 2008 Page 9 SIP Methods INVITERequests a session ACKFinal response to the INVITE OPTIONSAsk for server capabilities CANCELCancels a pending request BYETerminates a session REGISTERSends user’s address to server

10. np160 Dennis Baron, January 15, 2008 Page 10 SIP Responses 1XXProvisional100 Trying 2XXSuccessful200 OK 3XXRedirection302 Moved Temporarily 4XXClient Error404 Not Found 5XXServer Error504 Server Time-out 6XXGlobal Failure603 Decline

11. np160 Dennis Baron, January 15, 2008 Page 11 SIP Flows - Basic ACK200 - OK INVITE: sip:18.10.0.79 “Calls” 18.18.2.4 180 - RingingRings200 - OKAnswersBYEHangs up RTP Talking User A User B

12. np160 Dennis Baron, January 15, 2008 Page 12 SIP INVITE INVITE joeuser.mit.edu SIP/2.0 From: "Dennis Baron" ;tag=1c41 To: sip:joeuser.mit.edu Call-Id: call-1096504121-2@18.10.0.79 Cseq: 1 INVITE Contact: "Dennis Baron" Content-Type: application/sdp Content-Length: 304 Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE Supported: sip-cc, sip-cc-01, timer, replaces User-Agent: Pingtel/2.1.11 (WinNT) Date: Thu, 30 Sep 2004 00:28:42 GMT Via: SIP/2.0/UDP 18.10.0.79

13. np160 Dennis Baron, January 15, 2008 Page 13 Session Description Protocol IETF RFC 2327 “SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation.” SDP includes: –The type of media (video, audio, etc.) –The transport protocol (RTP/UDP/IP, H.320, etc.) –The format of the media (H.264 video, MPEG video, etc.) –Information to receive those media (addresses, ports, formats and so on)

14. np160 Dennis Baron, January 15, 2008 Page 14 SDP v=0 o=Pingtel 5 5 IN IP4 18.10.0.79 s=phone-call c=IN IP4 18.10.0.79 t=0 0 m=audio 8766 RTP/AVP 96 97 0 8 18 98 a=rtpmap:96 eg711u/8000/1 a=rtpmap:97 eg711a/8000/1 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:18 g729/8000/1 a=fmtp:18 annexb=no a=rtpmap:98 telephone-event/8000/1

15. np160 Dennis Baron, January 15, 2008 Page 15 CODECs Audio –G.711 8kHz sampling rate 64kbps –G.729 8kHz sampling rate 8kbps Voice Activity Detection Video –H.264 MPEG-4 –H.263

16. np160 Dennis Baron, January 15, 2008 Page 16 SIP Flows - Registration 200 - OKREGISTER: sip:dbaron@MIT.EDU401 - Unauthorized User B MIT.EDU Registrar REGISTER: (add credentials) MIT.EDU Location sip:dbaron@MIT.EDU Contact 18.10.0.79

17. np160 Dennis Baron, January 15, 2008 Page 17 SIP REGISTER REGISTER sip:mit.edu SIP/2.0 From: "Dennis Baron" ;tag=4561c4561 To: "Dennis Baron" ;tag=324591026 Call-Id: 9ce902bd23b070ae0108b225b94ac7fa Cseq: 5 REGISTER Contact: "Dennis Baron" Expires: 3600 Date: Thu, 30 Sep 2004 00:46:53 GMT Accept-Language: en Supported: sip-cc, sip-cc-01, timer, replaces User-Agent: Pingtel/2.1.11 (WinNT) Content-Length: 0 Via: SIP/2.0/UDP 18.10.0.79

18. np160 Dennis Baron, January 15, 2008 Page 18 SIP REGISTER – 401 Response SIP/2.0 401 Unauthorized From: "Dennis Baron" ;tag=4561c4561 To: "Dennis Baron" ;tag=324591026 Call-Id: 9ce902bd23b070ae0108b225b94ac7fa Cseq: 5 REGISTER Via: SIP/2.0/UDP 18.10.0.79 Www-Authenticate: Digest realm="mit.edu", nonce="f83234924b8ae841b9b0ae8a92dcf0b71096505216", opaque="reg:change4" Date: Thu, 30 Sep 2004 00:46:56 GMT Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, NOTIFY, SUBSCRIBE, INFO User-Agent: Pingtel/2.2.0 (Linux) Accept-Language: en Supported: sip-cc-01, timer Content-Length: 0

19. np160 Dennis Baron, January 15, 2008 Page 19 SIP REGISTER with Credentials REGISTER sip:mit.edu SIP/2.0 From: "Dennis Baron" ;tag=4561c4561 To: "Dennis Baron" ;tag=324591026 Call-Id: 9ce902bd23b070ae0108b225b94ac7fa Cseq: 6 REGISTER Contact: "Dennis Baron" Expires: 3600 Date: Thu, 30 Sep 2004 00:46:53 GMT Accept-Language: en Supported: sip-cc, sip-cc-01, timer, replaces User-Agent: Pingtel/2.1.11 (WinNT) Content-Length: 0 Authorization: DIGEST USERNAME=“dbaron@mit.edu", REALM="mit.edu", NONCE="f83234924b8ae841b9b0ae8a92dcf0b71096505216", URI="sip:mit.edu", RESPONSE="ae064221a50668eaad1ff2741fa8df7d", OPAQUE="reg:change4" Via: SIP/2.0/UDP 18.10.0.79

20. np160 Dennis Baron, January 15, 2008 Page 20 SIP Flows – Via Proxy INVITE: sip:dbaron@MIT.EDU “Calls” dbaron @MIT.EDU INVITE:sip:dbaron@18.10.0.79100 - Trying 180 - Ringing Rings180 - Ringing200 - OKAnswers 200 - OK ACK BYEHangs up200 - OK User A User B MIT.EDU Proxy Talking RTP

21. np160 Dennis Baron, January 15, 2008 Page 21 SIP Flows – Via Gateway INVITE: sip:joeuser@MIT.EDU “Calls” joeuser @MIT.EDU INVITE: sip:38400@18.162.0.25100 - TryingACK User A MIT.EDU Proxy 38400 Gateway 180 - Ringing Rings 200 - OK Answers BYEHangs up BYE 200 - OK Talking RTP

22. np160 Dennis Baron, January 15, 2008 Page 22 SIP INVITE with Record-Route INVITE sip:37669@18.162.0.25 SIP/2.0 Record-Route: From: \"Dennis Baron\" ;tag=2c41 To: sip:37669@mit.edu Call-Id: call-1096505069-3@18.10.0.79 Cseq: 1 INVITE Contact: \"Dennis Baron\" Content-Type: application/sdp Content-Length: 304 Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE Supported: sip-cc, sip-cc-01, timer, replaces User-Agent: Pingtel/2.1.11 (WinNT) Date: Thu, 30 Sep 2004 00:44:30 GMT Via: SIP/2.0/UDP 18.7.21.118:5080;branch=z9hG4bK2cf12c563cec06fd1849ff799d069cc0 Via: SIP/2.0/UDP 18.7.21.118;branch=z9hG4bKd26e44dfdc2567170d9d32a143a7f4d8 Via: SIP/2.0/UDP 18.10.0.79 Max-Forwards: 17

23. np160 Dennis Baron, January 15, 2008 Page 23 SIP Standards Just a sampling of IETF standards work… IETF RFCshttp://ietf.org/rfc.htmlhttp://ietf.org/rfc.html RFC3261Core SIP specification – obsoletes RFC2543 RFC2327SDP – Session Description Protocol RFC1889RTP - Real-time Transport Protocol RFC2326RTSP - Real-Time Streaming Protocol RFC3262SIP PRACK method – reliability for 1XX messages RFC3263Locating SIP servers – SRV and NAPTR RFC3264Offer/answer model for SDP use with SIP

24. np160 Dennis Baron, January 15, 2008 Page 24 SIP Standards (cont.) RFC3265SIP event notification – SUBSCRIBE and NOTIFY RFC3266IPv6 support in SDP RFC3311SIP UPDATE method – eg. changing media RFC3325Asserted identity in trusted networks RFC3361Locating outbound SIP proxy with DHCP RFC3428SIP extensions for Instant Messaging RFC3515SIP REFER method – eg. call transfer RFC4474Authenticated Identity Management SIMPLEIM/Presence - http://ietf.org/ids.by.wg/simple.htmlhttp://ietf.org/ids.by.wg/simple.html

25. np160 Dennis Baron, January 15, 2008 Page 25 IS&T Services MITvoip –Desktop VoIP telephones to replace traditional 5ESS telephones –New voice mail system –Web interface for user control –Transition over 2 to 2.5 years Personal SIP accounts –Bring your own devices/software –Limited support

26. np160 Dennis Baron, January 15, 2008 Page 26 “Hard phones” “Soft phones” Soft and Hard SIP Clients

27. np160 Dennis Baron, January 15, 2008 Page 27 Asterisk Open source phone system Runs on Linux, Mac OS X, OpenBSD, FreeBSD and Solaris Supports SIP (and other VoIP protocols) –Cisco SCCP, H.323, IAX Highly customizable Hardware telephone interfaces available MIT applications –Shuttletrack IVR –Media Lab Owl Project –SIPB VoIP Scripts?

28. np160 Dennis Baron, January 15, 2008 Page 28 IAP 2008 - VoIP Series SIP Fundimentals Dennis Baron Tue Jan 15, 01-02:30pm, 4-149 Personal SIP Account Workshop Dennis Baron Tue Jan 22, 01-02:30pm, 4-231 Build, Test, and Deploy VoIP Applications with Asterisk and other Open-Source Applications Elliot Eichen Tue Jan 29, 01-02:30pm, 4-231

29. np160 Dennis Baron, January 15, 2008 Page 29 Questions?

30. np160 Dennis Baron, January 15, 2008 Page 30 Abstract Until the 1990s, if you wanted to make telephone hardware do your bidding you had to do it at the level of signal processing, EE, and physical-layer analog protocols. Now MIT and the rest of the world are switching to Voice-over-IP, based on RFC-documented protocols on the familiar IETF stack, and the opportunity is opening for software hackers to work their magic on the oldest extant medium in telecommunications. A SIPB project in the scripts tradition aiming to provide infrastructure for members of the MIT community to serve up their own innovations, is still in the early stages and welcoming new participants. This cluedump will give a technical grounding in the architecture of the protocols governing voice-over-IP and in their implementation at MIT.

31. np160 Dennis Baron, January 15, 2008 Page 31 Outline What’s changed What is SIP MIT VoIP services Questions and answers

32. np160 Dennis Baron, January 15, 2008 Page 32 What’s Changed We used to send data over phone calls – remember modems? A number defined who you were – and where you were The Phone Company defined the services – and we used what they wanted to sell us Intelligent networks – dumb phones

33. np160 Dennis Baron, January 15, 2008 Page 33 Why SIP Core protocol used for VoIP –Except Skype! Used by –Vonage, AT&T, and other VoIP service providers –Free service providers – eg. Free World Dialup –Second Life –MIT Open peering –SIP.edu –ISN

34. np160 Dennis Baron, January 15, 2008 Page 34 Personal SIP Accounts in Detail Uses your MIT SIP communications identity One account per person Allows you to use your own hardware or software for placing and receiving Internet calls Assigns a traditional telephone number for receiving calls Web interface for customizing your account “Experimental” service aimed at early technology adopters Not intended as a replacement for other telephone services IS&T support limited to activating accounts and web page –No support at this time for clients

35. np160 Dennis Baron, January 15, 2008 Page 35 Personal SIP Support Model Self service account activation –https://voip.mit.edu/cgi-bin/personal/sipmgr/ IS&T Documentation –http://mit.edu/ist/topics/telecommunications/psip/http://mit.edu/ist/topics/telecommunications/psip/ SIP Users at MIT Wiki –https://wikis.mit.edu/confluence/display/SIP/SIP+Users+at+MIThttps://wikis.mit.edu/confluence/display/SIP/SIP+Users+at+MIT –Your contributions to the wiki are supported and encouraged! SIP Users Forum –https://scripts-cert.mit.edu/~sip/sip-users/https://scripts-cert.mit.edu/~sip/sip-users/ –Not currently active – may replace with newer technology

36. np160 Dennis Baron, January 15, 2008 Page 36 What’s Changed Plenty of bandwidth – broadband to the home –Voice (and video) are just another data stream Everybody can be anywhere – it’s the Internet –Get a phone number from anywhere (optional) Anybody can provide services –If you don’t like what they’re selling build your own Anything can be an Internet phone –Your laptop, your mobile phone, your …