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

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. 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
for
for instant messaging
for voice and video
Username replaced by numbers for telephone applications

4. Where’s SIP Application Transport Network Physical/Data Link Ethernet
TCP
UDP
RTSP
SIP
SDP
codecs
RTP
DNS(SRV)

5. SIP Components User Agents Server types Gateways
Clients – Make requests
Servers – Accept requests
Server types
Redirect Server
Proxy Server
Registrar Server
Location Server
Gateways

6. SIP Trapezoid DNS Server Location Server Terminating User Agent
Outbound Proxy
Originating User Agent
DNS
SIP
RTP
Registrar
Inbound Proxy

7. SIP Triangle ? DNS Server Location Server Terminating User Agent
Originating User Agent
DNS
SIP
RTP
Registrar
Inbound Proxy

8. Back-to-Back User Agent
SIP Peer to Peer !
Terminating User Agent
Originating User Agent
SIP
RTP
Back-to-Back User Agent
Terminating User Agent
Originating User Agent
RTP
SIP
B2BUA

9. SIP Methods 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 user’s address to server

10. 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

11. SIP Flows - Basic RTP User A User B INVITE: sip:18.10.0.79
“Calls”
180 - Ringing
Rings
200 - OK
Answers
ACK
RTP
Talking
BYE
Hangs up
200 - OK

12. SIP INVITE INVITE joeuser.mit.edu SIP/2.0
From: "Dennis
To: sip:joeuser.mit.edu
Call-Id:
Cseq: 1 INVITE
Contact: "Dennis
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/ (WinNT)
Date: Thu, 30 Sep :28:42 GMT
Via: SIP/2.0/UDP

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. SDP v=0 o=Pingtel 5 5 IN IP4 18.10.0.79 s=phone-call
c=IN IP
t=0 0
m=audio 8766 RTP/AVP
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. CODECs Audio Video G.711 G.729 H.264 H.263 8kHz sampling rate 64kbps
Voice Activity Detection
Video
H.264
MPEG-4
H.263

16. SIP Flows - Registration
User B
MIT.EDU
Registrar
MIT.EDU
Location
REGISTER:
401 - Unauthorized
REGISTER: (add credentials)
Contact
200 - OK

17. SIP REGISTER REGISTER sip:mit.edu SIP/2.0
From: "Dennis
To: "Dennis
Call-Id: 9ce902bd23b070ae0108b225b94ac7fa
Cseq: 5 REGISTER
Contact: "Dennis
Expires: 3600
Date: Thu, 30 Sep :46:53 GMT
Accept-Language: en
Supported: sip-cc, sip-cc-01, timer, replaces
User-Agent: Pingtel/ (WinNT)
Content-Length: 0
Via: SIP/2.0/UDP

18. SIP REGISTER – 401 Response
SIP/ Unauthorized
From: "Dennis
To: "Dennis
Call-Id: 9ce902bd23b070ae0108b225b94ac7fa
Cseq: 5 REGISTER
Via: SIP/2.0/UDP
Www-Authenticate: Digest realm="mit.edu", nonce="f b8ae841b9b0ae8a92dcf0b ", opaque="reg:change4"
Date: Thu, 30 Sep :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. SIP REGISTER with Credentials
REGISTER sip:mit.edu SIP/2.0
From: "Dennis
To: "Dennis
Call-Id: 9ce902bd23b070ae0108b225b94ac7fa
Cseq: 6 REGISTER
Contact: "Dennis
Expires: 3600
Date: Thu, 30 Sep :46:53 GMT
Accept-Language: en
Supported: sip-cc, sip-cc-01, timer, replaces
User-Agent: Pingtel/ (WinNT)
Content-Length: 0
Authorization: DIGEST REALM="mit.edu", NONCE="f b8ae841b9b0ae8a92dcf0b ", URI="sip:mit.edu", RESPONSE="ae064221a50668eaad1ff2741fa8df7d", OPAQUE="reg:change4"
Via: SIP/2.0/UDP

20. “Calls” dbaron @MIT.EDU
SIP Flows – Via Proxy
User A
MIT.EDU
Proxy
User B
INVITE:
“Calls”
100 - Trying
180 - Ringing
Rings
200 - OK
Answers
ACK
Talking
RTP
BYE
Hangs up
200 - OK

21. “Calls” joeuser @MIT.EDU
SIP Flows – Via Gateway
User A
MIT.EDU
Proxy
Gateway
38400
INVITE:
“Calls”
INVITE:
100 - Trying
180 - Ringing
Rings
200 - OK
Answers
ACK
Talking
RTP
BYE
Hangs up
200 - OK

22. SIP INVITE with Record-Route
INVITE SIP/2.0
Record-Route: <sip: :5080;lr;a;t=2c41;s=b07e28aa8f94660e a44b9ed50>
From: \"Dennis
To:
Call-Id:
Cseq: 1 INVITE
Contact: \"Dennis
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/ (WinNT)
Date: Thu, 30 Sep :44:30 GMT
Via: SIP/2.0/UDP :5080;branch=z9hG4bK2cf12c563cec06fd1849ff799d069cc0
Via: SIP/2.0/UDP ;branch=z9hG4bKd26e44dfdc d9d32a143a7f4d8
Via: SIP/2.0/UDP
Max-Forwards: 17

23. SIP Standards Just a sampling of IETF standards work…
IETF RFCs
RFC3261 Core SIP specification – obsoletes RFC2543
RFC2327 SDP – Session Description Protocol
RFC1889 RTP - Real-time Transport Protocol
RFC2326 RTSP - Real-Time Streaming Protocol
RFC3262 SIP PRACK method – reliability for 1XX messages
RFC3263 Locating SIP servers – SRV and NAPTR
RFC3264 Offer/answer model for SDP use with SIP

24. SIP Standards (cont.)
RFC3265 SIP event notification – SUBSCRIBE and NOTIFY
RFC3266 IPv6 support in SDP
RFC3311 SIP UPDATE method – eg. changing media
RFC3325 Asserted identity in trusted networks
RFC3361 Locating outbound SIP proxy with DHCP
RFC3428 SIP extensions for Instant Messaging
RFC3515 SIP REFER method – eg. call transfer
RFC4474 Authenticated Identity Management
SIMPLE IM/Presence -

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. Soft and Hard SIP Clients
“Hard phones”
“Soft phones”

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. IAP 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. Questions?

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. Outline What’s changed What is SIP MIT VoIP services
Questions and answers

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. 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. 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. Personal SIP Support Model
Self service account activation
IS&T Documentation
SIP Users at MIT Wiki
Your contributions to the wiki are supported and encouraged!
SIP Users Forum
Not currently active – may replace with newer technology

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 …