np160

8/10/2019 np160

1/36

np160Dennis Baron, January 15, 2008

Page 1

SIP FundimentalsIAP 2008 VoIP Series

Dennis Baron

January 15, 2008

8/10/2019 np160

2/36


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

8/10/2019 np160

3/36


Page 3

Whats 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 addressingsingle communications identity

mailto:[email protected] for email

xmpp:[email protected] for instant messaging

sip:[email protected] for voice and video

Username replaced by numbers for telephone applications

8/10/2019 np160

4/36


Page 4

Wheres SIP

Application

Transport

Network

Physical/Data Link Ethernet

IP

TCP UDP

RTSP SIP

SDP codecs

RTP DNS(SRV)

8/10/2019 np160

5/36


Page 5

SIP Components

User Agents

ClientsMake requests

ServersAccept requests

Server types

Redirect Server

Proxy Server

Registrar Server

Location Server

Gateways

8/10/2019 np160

6/36


Page 6

SIP Trapezoid

DNS

Server

Location

Server

Terminating

User Agent

Outbound

Proxy

Originating

User Agent

DNS

SIP

SIP

SIP SIP

RTP

Registrar

Inbound

Proxy

SIP

8/10/2019 np160

7/36


Page 7

SIP Triangle ?

DNS

Server

Location

Server

Terminating

User Agent

Originating

User Agent

DNS

SIP

SIP SIP

RTP

Registrar

Inbound

Proxy

SIP

8/10/2019 np160

8/36


Page 8

Terminating

User Agent

Originating

User Agent RTP

SIP SIP

B2BUA

SIP Peer to Peer !

Back-to-Back User Agent

Terminating

User Agent

Originating

User Agent

SIP

RTP

8/10/2019 np160

9/36


Page 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 users address to server

8/10/2019 np160

10/36


Page 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

8/10/2019 np160

11/36


Page 11

SIP Flows - Basic

ACK

200 - OK

INVITE: sip:18.10.0.79Calls

18.18.2.4

180 - Ringing Rings

200 - OK Answers

BYEHangs up

RTPTalking Talking

UserA

UserB

8/10/2019 np160

12/36


Page 12

SIP INVITE

INVITE joeuser.mit.edu SIP/2.0

From: "Dennis Baron";tag=1c41

To: sip:joeuser.mit.edu

Call-Id: [email protected]

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, replacesUser-Agent: Pingtel/2.1.11 (WinNT)

Date: Thu, 30 Sep 2004 00:28:42 GMT

Via: SIP/2.0/UDP 18.10.0.79

8/10/2019 np160

13/36

np160Dennis Baron, January 15, 2008Page 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)

8/10/2019 np160

14/36


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

8/10/2019 np160

15/36


CODECs

Audio G.711

8kHz sampling rate

64kbps

G.729

8kHz sampling rate

8kbps

Voice Activity Detection

Video H.264

MPEG-4

H.263

8/10/2019 np160

16/36


SIP Flows - Registration

200 - OK

REGISTER: sip:[email protected]

401 - Unauthorized

UserB MIT.EDU

Registrar

REGISTER: (add credentials)

MIT.EDU

Location

sip:[email protected]

Contact 18.10.0.79

8/10/2019 np160

17/36


SIP REGISTERREGISTER sip:mit.edu SIP/2.0

From: "Dennis Baron";tag=4561c4561To: "Dennis Baron";tag=324591026

Call-Id: 9ce902bd23b070ae0108b225b94ac7fa

Cseq: 5 REGISTER


Expires: 3600

Date: Thu, 30 Sep 2004 00:46:53 GMTAccept-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

8/10/2019 np160

18/36


SIP REGISTER401 ResponseSIP/2.0 401 Unauthorized



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

8/10/2019 np160

19/36


SIP REGISTER with CredentialsREGISTER sip:mit.edu SIP/2.0



Cseq: 6 REGISTER


Expires: 3600

Date: Thu, 30 Sep 2004 00:46:53 GMTAccept-Language: en



Content-Length: 0

Authorization: DIGEST [email protected]", REALM="mit.edu",

NONCE="f83234924b8ae841b9b0ae8a92dcf0b71096505216", URI="sip:mit.edu",RESPONSE="ae064221a50668eaad1ff2741fa8df7d", OPAQUE="reg:change4"

Via: SIP/2.0/UDP 18.10.0.79

8/10/2019 np160

20/36


SIP FlowsVia Proxy

INVITE: sip:[email protected] dbaron

@MIT.EDUINVITE:sip:[email protected]

100 - Trying

180 - Ringing

Rings180 - Ringing

200 - OK Answers

200 - OK

ACK

BYEHangs up

200 - OK

UserA

UserBMIT.EDU

Proxy

Talking TalkingRTP

8/10/2019 np160

21/36


SIP FlowsVia Gateway

INVITE: sip:[email protected] joeuser

@MIT.EDUINVITE: sip:[email protected]

100 - Trying

ACK

ACK

UserA MIT.EDU

Proxy

38400Gateway

180 - Ringing

180 - Ringing

Rings

200 - OK

200 - OK

Answers

BYEHangs up

BYE

200 - OK

200 - OK

Talking TalkingRTP

8/10/2019 np160

22/36


SIP INVITE with Record-RouteINVITE sip:[email protected] SIP/2.0

Record-Route: From: \"Dennis Baron\";tag=2c41

To: sip:[email protected]

Call-Id: [email protected]

Cseq: 1 INVITE

Contact: \"Dennis Baron\"

Content-Type: application/sdpContent-Length: 304

Accept-Language: en

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE



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

8/10/2019 np160

23/36


SIP Standards

Just a sampling of IETF standards work

IETF RFCs http://ietf.org/rfc.html

RFC3261 Core SIP specificationobsoletes RFC2543

RFC2327 SDPSession Description Protocol RFC1889 RTP - Real-time Transport Protocol

RFC2326 RTSP - Real-Time Streaming Protocol

RFC3262 SIP PRACK methodreliability for 1XX messages

RFC3263 Locating SIP serversSRV and NAPTR

RFC3264 Offer/answer model for SDP use with SIP
http://ietf.org/rfc.htmlhttp://ietf.org/rfc.html

8/10/2019 np160

24/36

np160 Dennis Baron, January 15, 2008Page 24

SIP Standards (cont.)

RFC3265 SIP event notificationSUBSCRIBE and NOTIFY

RFC3266 IPv6 support in SDP

RFC3311 SIP UPDATE methodeg. changing media

RFC3325 Asserted identity in trusted networks

RFC3361 Locating outbound SIP proxy with DHCP

RFC3428 SIP extensions for Instant Messaging

RFC3515 SIP REFER methodeg. call transfer

RFC4474 Authenticated Identity Management

SIMPLE IM/Presence - http://ietf.org/ids.by.wg/simple.html
http://ietf.org/ids.by.wg/simple.htmlhttp://ietf.org/ids.by.wg/simple.html

8/10/2019 np160

25/36


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

8/10/2019 np160

26/36


Hard phones

Soft phones

Soft and Hard SIP Clients

8/10/2019 np160

27/36


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?

8/10/2019 np160

28/36


IAP 2008 - VoIP Series

SIP Fundimentals

Dennis Baron

Tue Jan 15, 01-02:30pm, 4-149

Personal SIP Account Workshop

Dennis BaronTue Jan 22, 01-02:30pm, 4-231

Build, Test, and Deploy VoIP Applications with Asterisk

and other Open-Source Applications

Elliot EichenTue Jan 29, 01-02:30pm, 4-231

8/10/2019 np160

29/36

np160

Dennis Baron, January 15, 2008

Page 29

Questions?

8/10/2019 np160

30/36

np160


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.

8/10/2019 np160

31/36

np160


Page 31

Outline

Whats changed

What is SIP

MIT VoIP services

Questions and answers

8/10/2019 np160

32/36

np160


Page 32

Whats Changed

We used to send data over phone callsremember

modems?

A number defined who you wereand whereyou were

The Phone Company defined the servicesand we used

what they wanted to sell us

Intelligent networksdumb phones

8/10/2019 np160

33/36

np160


Page 33

Why SIP

Core protocol used for VoIP

Except Skype!

Used by

Vonage, AT&T, and other VoIP service providers

Free service providerseg. Free World Dialup

Second Life

MIT

Open peering

SIP.edu

ISN

8/10/2019 np160

34/36

np160


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

8/10/2019 np160

35/36

np160


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/

SIP Users at MIT Wiki https://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/

Not currently activemay replace with newer technology
http://mit.edu/ist/topics/telecommunications/psip/https://wikis.mit.edu/confluence/display/SIP/SIP+Users+at+MIThttps://scripts-cert.mit.edu/~sip/sip-users/https://scripts-cert.mit.edu/~sip/sip-users/https://scripts-cert.mit.edu/~sip/sip-users/https://scripts-cert.mit.edu/~sip/sip-users/https://scripts-cert.mit.edu/~sip/sip-users/https://scripts-cert.mit.edu/~sip/sip-users/https://wikis.mit.edu/confluence/display/SIP/SIP+Users+at+MIThttp://mit.edu/ist/topics/telecommunications/psip/

8/10/2019 np160

36/36

160


Page 36

Whats Changed

Plenty of bandwidthbroadband to the home

Voice (and video) are just another data stream

Everybody can be anywhereits the Internet

Get a phone number from anywhere (optional)

Anybody can provide services

If you dont like what theyre selling build your own

Anything can be an Internet phone

Your laptop, your mobile phone, your

np160

Documents