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An Introduction to SIP
Moshe Sambol
Services Research Lab
November 18, 1998
November 18, 1998 GTE Laboratories SRL Presentation 2
The Session Initiation Protocol
Communications protocol under development by the IETF’s MMUSIC working group. Last call for comments is over, SIP is about to become a standard.
Text-based IP application-layer protocol (derived from HTTP) for Internet multi-party multi-media conferencing establishment and management.
November 18, 1998 GTE Laboratories SRL Presentation 3
SIP Basics
Separation of call data and connection control as opposed to PSTN: DTMF / pulses and on-
hook/off-hook signals carried in band with call.
Carries service and participant information provides natural support for advanced
telephony services.
November 18, 1998 GTE Laboratories SRL Presentation 4
SIP Basics continuedSimpler language than H.323 supports
flexibility, expandability, greater expression
Media and/or participants can be added to or removed from existing conferences
Supports personal mobility, degrees of security...
November 18, 1998 GTE Laboratories SRL Presentation 5
The SIP Call Model Session: A set of multimedia senders and receivers and
the data which flows between them. Conference: A multimedia session identified by a
session description. Composed of one or more calls.Call: All participants in a conference invited by a single
source.Call Leg: All communication between one caller and
callee within a call.Transaction: All messages within a call leg from a first
request until a final* response.
November 18, 1998 GTE Laboratories SRL Presentation 6
Actors in a SIP ConferenceUser Agent
Client application which initiates and accepts invitations; interacts with media applications.
SIP Proxy Server Intermediary which relays requests after possible lookup
and/or translation.
SIP Redirect Server Maps a request to a set of alternate addresses and returns
those to the client without initiating requests of its own.
November 18, 1998 GTE Laboratories SRL Presentation 7
SIP Actors, continued
Location Service Not defined by SIP but used by its servers to
locate other SIP servers.
No Gatekeeper Gatekeeper traditionally provides admission
control, address translation. These functions provided by other SIP actors.
November 18, 1998 GTE Laboratories SRL Presentation 8
SIP Operation in Proxy Mode
All illustrations courtesy of H.Schulzrinne
November 18, 1998 GTE Laboratories SRL Presentation 9
SIP Operation in Redirect Mode
November 18, 1998 GTE Laboratories SRL Presentation 10
SIP Requests
REGISTER Notify server of user location and availability
OPTIONS Determine user capabilities
INVITE CANCEL Call setup Transaction termination
BYE Call termination / transfer
November 18, 1998 GTE Laboratories SRL Presentation 11
SIP Responses
SIP Provides a rich set of response codes which make it well suited to support advanced telephony services:
1xx: Provisional responses (ringing, queued, etc.) 2xx: Final success responses 3xx: Redirect responses (moved perm/temp, mult. choices, etc.) 4xx: Client-Error (formatting, authorization, timeout, loop, etc.) 5xx: Server-Error 6xx: Global-Failure (busy, decline, does not exist, etc.)
November 18, 1998 GTE Laboratories SRL Presentation 12
An Example SIP MessageINVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell <sip:[email protected]>
To: T. Watson <sip:[email protected]>
Call-ID: [email protected]
Cseq: 1 INVITE
Subject: Mr. Watson, come here.
Content-Type: application/sdp
Content-Length: …
v=0
o=bell 53655765 2353687637 IN IP4 128.3.4.5
m=audio 3456 RTP/AVP 0 3 4 5
November 18, 1998 GTE Laboratories SRL Presentation 13
… Sample ResponseSIP/2.0 180 Ringing
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell <sip:[email protected]>
To: T. Watson <sip:[email protected]>
Call-ID: [email protected]
Cseq: 1 INVITE
Content-Length: 0
-----------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell <sip:[email protected]>
To: T. Watson <sip:[email protected]>
Call-ID: [email protected]
Cseq: 1 INVITE
Contact: sip:[email protected]
Content-Type: …
m=audio 5004 RTP/AVP 0 3 …
November 18, 1998 GTE Laboratories SRL Presentation 14
Sample Services Multi-party call transfers generic addressing directed pick-up multiple pick-up hold music control third party call control camp-on click to dial back availability notification
SIP Supports Multicast UDP + TCP Privacy (authentication, encryption, path hiding)
November 18, 1998 GTE Laboratories SRL Presentation 15
Transfer Example: Telemarketing
November 18, 1998 GTE Laboratories SRL Presentation 16
State of SIP
SIP implementations are being worked on at: ObjectSoftware Nokia Research Lucent Mediatrix ISI HP Labs GMD Fokus Ericsson Dynamicsoft others?
My own graduate work was on...
November 18, 1998 GTE Laboratories SRL Presentation 17
Open IssuesBillingMappings to telephony APIsInterfacing to external media applicationsLocation services
Topics for Future PresentationsSIP vs. H.323 SDP
Security Richer examples!
November 18, 1998 GTE Laboratories SRL Presentation 18
Additional Info At:
Professor Henning Schulzrinne: http://www.cs.columbia.edu/~hgs/sip http://www.cs.columbia.edu/~hgs/internet/sdp.html
Working group mailing list: [email protected]
Struggling ex-graduate student: [email protected]