Session Initiation Protocol

Session Initiation Protocol
Matt Bynum, CCIE (Voice) #21753

SIP is a protocol for establishing sessions in an IP network.

Agenda
Protocol History
SIP 101
Cisco and SIP
(Ssshhh!) Other vendors and SIP
Future of SIP

Protocol History
To know where you’re going, you have to know where you’ve been.
- who knows? Not Google.

Setting the Stage
The Internet Engineering Task Force first met in 1986.
“The mission of the IETF is to make the Internet work better by producing high quality, relevant technical documents that influence the way people design, use, and manage the Internet. “
- http://www.ietf.org/about/mission.html
DNS dhcpIPv4 IPv6 TCP UDP RTP SMTP TELNET IGMP ICMPFTP ECHO ARP POP3 OSPF SNMP RIP
http://tools.ietf.org/html/rfc5000

IETF Meetings
The First IETF Audiocast occurred in 1992. Since then, IETF sessions were conducted on the Mbone.
Create 1
Descr.: DNS Discussion San Fran
Orig.: John Doe j.doe@com.com
Info: http://www.com.com
Start: 04.04.2001 / 09.30
End: 04.20.2001 / 16:30
Media: Audio GSM 224.1.6.7/49000
Media: Video H.263 224.1.6.8/49100
Disseminate 2
SAP/NNTP/HTTP
Invite
SMTP/SIP
Join 3
PC/Telephone
Media 4
PC/Telephone

Simple Conference Invitation Protocol
by Henning Schulzrinne
CALL
CHANGE
CLOSE
TCP/SCIP
1xx
2xx
3xx
4xx
5xx
Session Invitation Protocol
by Mark Handley and Eve Schooler
SUCCESSUNSUCCESSFUL
BUSY
DECLINE
UNKNOWN
FAILED
FORBIDDEN
RINGING
RINGING
TRYING
REDIRECT
ALTERNATIVE
UDP/SDP
NEGOTIATE

Simple Conference Invitation Protocol
SCIP/1.0 302 Callee has moved temporarily
Location: jones@salt.lab3.company.com
Location: jones@pepper.lab3.company.com
CALL hgs@lupus.fokus.gmd.de 1.0
User-Agent: coco/1.3
From: Christian Zahl <cz@cs.tu-berlin.de>
To: Henning Schulzrinne <schulzrinne@fokus.gmd.de>
Call-Id: 9510021900.AA07734@lion.cs.tu-berlin.de
Referer: ceres.fokus.gmd.de
Expires: Mon, 02 Oct 1995 18:44:11 GMT
Required: fc99cb08 audio/pcmu; port=3456; transport=RTP;
rate=16000; channels=1; pt=97; net=224.2.0.1; ttl=128,
audio/gsm; port=3456; transport=RTP; rate=8000; channels=1,
audio/lpc; port=3456; transport=RTP; rate=8000; channels=1
SIP/1.0 REQ
PA=128.16.65.19 16
AU=none
ID=128.16.65.19/32492374
FR=M.Handley@cs.ucl.ac.uk
TO=J.Crowcroft@cs.ucl.ac.uk
v=0
o=van 2353644765 2353687637 IN IP4 128.3.4.5
s=Mbone Audio
i=Discussion of Mbone Engineering Issues
e=van@ee.lbl.gov (Van Jacobsen
c=IN IP4 224.2.0.1/127
t=0 0
m=audio 3456 RTP PCMU
Session Invitation Protocol

Papa SIP
“Personal Mobility for Multimedia Services in the Internet”
by Henning Schulzrinne*, March 1996
http://www.cs.columbia.edu/~hgs/papers/Schu9603_Personal.pdf
http://www.cs.columbia.edu/~hgs/
* Developed RTP

The Internet Architect
SIP (RFC 2543, RFC 3261); SDP (RFC 2327; SAP, RFC 2974); Protocol Independent Multicast-Sparse Mode (PIM-SM, RFC 2362), TCP-Friendly Rate Control (TFRC, RFC 3448), Multicast-Scope Zone Announcement Protocol (MZAP, RFC 2776), Multicast Address Allocation (RFC 2908, RFC 2909), TCP Congestion Window Validation ( RFC 2861), Reliable Multicast ( RFC 3451, RFC 3452, RFC 3453, RFC 3048), Datagram Congestion Control Protocol ( RFC 4340, RFC 4336).
Mark Handley
Founder of XORP (www.xorp.org)
http://www.cs.ucl.ac.uk/staff/M.Handley/

SIP Drafts
http://www.cs.columbia.edu/sip/history.html
Dec. 2, 1996 draft-ietf-mmusic-sip-01
March 27, 1997 draft-ietf-mmusic-sip-02
July 31, 1997 draft-ietf-mmusic-sip-03
November 11, 1997 draft-ietf-mmusic-sip-04
May 14, 1998 draft-ietf-mmusic-sip-05
June 17, 1998 draft-ietf-mmusic-sip-06
July 16, 1998 draft-ietf-mmusic-sip-07
August 7, 1998 draft-ietf-mmusic-sip-08
September 18, 1998 draft-ietf-mmusic-sip-09
September 28, 1998 Last call
November 12, 1998 draft-ietf-mmusic-sip-10
December 15, 1998 draft-ietf-mmusic-sip-11
January 15, 1999 draft-ietf-mmusic-sip-12
February 2, 1999 Approved
March 17, 1999 RFC 2543

SIP Today
The Hitchhiker’s Guide to SIP
RFC 3261 (SIP: Session Initiation Protocol)
RFC 3263 (Session Initiation Protocol (SIP): Locating SIP Servers)
RFC 3264 (An Offer/Answer Model with Session Description Protocol (SDP))
RFC 3265 (Session Initiation Protocol (SIP)-Specific Event Notification)
RFC 3325 (Private Extensions to SIP for Asserted Identity within Trusted Networks)
RFC 3327 (SIP Extension Header Field for Registering Non-Adjacent Contacts)
RFC 3581 (An Extension to SIP for Symmetric Response Routing)
RFC 3840 (Indicating User Agent Capabilities in SIP)
RFC 4320 (Actions Addressing Issues Identified with the Non-INVITE Transaction in SIP)
RFC 4474 (Enhancements for Authenticated Identity Management in SIP)
GRUU (Obtaining and Using Globally Routable User Agent Identifiers (GRUU) in SIP)
OUTBOUND (Managing Client Initiated Connections through SIP)
RFC 4566 (Session Description Protocol)SDP-CAP (SDP Capability Negotiation)
ICE (Interactive Connectivity Establishment)
RFC 3605 (Real Time Control Protocol (RTCP) Attribute in the Session Description Protocol)
RFC 4916 (Connected Identity in the Session Initiation Protocol (SIP))
RFC 3311 (The SIP UPDATE Method)
SIPS-URI (The Use of the SIPS URI Scheme in the Session Initiation Protocol (SIP))
RFC 3665 (Session Initiation Protocol (SIP) Basic Call Flow Examples)
Don’t Panic!

SIP 101
Any sufficiently advanced technology is indistinguishable from magic.
- Arthur C. Clarke

User Agents
Client Server
Proxy
Registrar
Redirect

SIP Methods

SIP Responses

Basic Call Flow
User Agent
Proxy Server
User Agent
INVITE
INVITE
100 Trying
180 Ringing
180 Ringing
200 OK
200 OK
ACK
ACK
Media Session
BYE
BYE
200 OK
200 OK

Example SIP Request
INVITE sip:matt@ncug.org SIP/2.0
Via: SIP/2.0/UDP 216.81.194.139:5060;branch=j3mF42aV349
From: TN Lottery<sip:youwon@tnlottery.com>;tag=27fn23ask
To: Matt <sip:matt@ncug.org>
Call-ID: 393j23m9df3adv3211
Max-Forwards: 70
Cseq: 1 INVITE
Contact: sip:youwon@216.81.194.139
Content-Type: application/sdp
Contact-Length: 126
v=0
o=youwon 2890844526 2890844526 IN IP4 youwon.tnlottery.com
s=SIP Call
c=IN IP4 216.81.194.139
t=0 0
m=audio 32894 RTP/AVP 0 101
a=rtpmap: 0 PCMU/8000
a=rtpmap: 101 iLBC/8000

Example SIP Response
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=j3mF42aV349
From: Matt <sip:matt@ncug.org>;tag=32fd45d36-d4ad
To: TN Lottery<sip:youwon@tnlottery.com> ;tag=27fn23ask
Call-ID: 393j23m9df3adv3211
Max-Forwards: 70
Cseq: 1 INVITE
Contact: <sip:matt@10.0.0.1:5060>
Content-Type: application/sdp
Contact-Length: 126
v=0
o=matt 7844 125 IN IP4 10.0.0.1
s=SIP Call
c=IN IP4 10.0.0.1
t=0 0
m=audio 43589 RTP/AVP 0
a=sendrecv
a=rtpmap: 0 PCMU/8000

Uniform Resource Identifier
sip:user@domain.com

Cisco and SIP
"Cisco's multi-protocol packet voice strategy includes support for SIP, and we believe the promise of SIP has become a reality.”
- Lou Santora, former VP of Cisco’s voice technology group in 2002

Cisco Fellow

Active in IETF
Co-author of the Session Initiation Protocol (SIP), RFC 3261,
SIMPLE - SIP for presence and IM.
STUN (Simple Traversal of UDP through NAT)
TURN (Traversal Using Relay NAT)
XCAP (XML Configuration Access Protocol)
Author of 30 patents and publications, 45 Internet RFCs, and numerous Internet Drafts in the area of multimedia communications over packet networks

Jonathan Rosenberg
http://www.jdrosen.net/

SIP Enabled Cisco Products

Cisco Unified SIP Proxy
NM for the 3800 series ISR
NME-CUSP-522-K9
2 GB of RAM
160 GB hard disk
Gigabit Ethernet to the router backplane
Supported on 12.4.22T
Simplifies management of large SIP networks
CUSP uses a counted license (10, 30, and 100 calls per second)

Cisco UC Manager
Functions as a B2BUA
owns each leg of call as a separate dialog
more stateful than proxy servers
inter-work SIP with other protocols
B2BUA for all types of SIP calls (trunk and line)
Cisco’s implementation is 100% standards compatible SIP

But…
There are “extensions” to SIP implemented in CUCM for SCCP feature parity.
Leads to two modes of SIP support for phones.
Advanced
Basic

Third-Party SIP Phone Categories

Cisco Unified Border Element
Feature in IOS, since 12.3.11T (version 1.0)
was IPIPGW
up to version 1.3 as of 12.4.22YB
Allows for demarcation point in SP scenarios
Provides H.323<->SIP interoperability
Two licensing models, CUBE session licenses, or flat INTVVSRV license
At a minimum, requires the IP Voice feature set

Same ol’ Dial-peers
SIP-VG(config)# voice service voip
SIP-VG(config-voi-serv)# allow-connections sip to sip
SIP-VG(config-voi-serv)# allow-connections sip to h323
SIP-VG(config)# dial-peer voice 2111 voip
SIP-VG(config-dial-peer)# session target ipv4:10.0.0.1
SIP-VG(config-dial-peer)# session protocol sipv2
SIP-VG(config-dial-peer)# session transport tcp
SIP-VG(config-dial-peer)# destination-pattern 615[2-9]……
SIP-VG(config-dial-peer)# dtmf-relay sip-notify rtp-nte

Troubleshooting CUBE
SIP-VG# debug ccsip ?
all
calls
error
events
info
media
messages
preauth
states
transport
SIP-VG# debug voip dial-peer
SIP-VG# show sip-ua service

Cisco Unified Presence
Presence server provides SIP SUBSCRIBE/NOTIFY functionality to the Cisco Unified Personal Communicator
Integrates with CUCM via SIP Trunk
SUBSCRIBE ext 1111
NOTIFY ext 1111

Other Vendors and SIP
“Competition is not only the basis of protection to the consumer, but is the incentive to progress”
- Herbert Hoover

SIP, it’s everywhere

Future of SIP
As far as I'm concerned, progress peaked with frozen pizza.
- John McClain, Die Hard 2

What’s next?
P2P SIP (DNS SRV, end-point resolution)
Universal Personal Telecommunications
Presence as the dial-tone of the 21st century
ENUM (E.164 to SIP URI discovery)
Extensions Galore

Links
http://www.cs.columbia.edu/sip/talks.html
http://en.wikipedia.org/wiki/Universal_Personal_Telecommunications
http://www.voip-info.org/wiki/view/SIP
http://www.scribd.com/doc/6293213/SIP-Presentation
http://www.ietf.org/rfc/rfc3261.txt
http://www.sipworkbench.com/
http://engineering.columbia.edu/videos/schulzrinne/index.html
http://www.cs.columbia.edu/sip/history.html

The End