Session Initiation Protocol - SIP

Switching Core Network Signalling Rel.4 M14/U4

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Document change history

Date Version Name Change comment

March, 03.08 July, 08.08

1 2

Pubate Satienpoch Guido Schneiders

New creation for M14 Adaptation

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Introduction : SIP (Session Initiation Protocol)

Developed by the Internet Engineering Task Force (IETF)

An application level protocol to assist in providing advanced telephony

services across the Internet like establishing, modifying and terminating sessions with one or more participants

New RFC 3261 (Jun, 02) replaced RFC 2543 (Mar,99)

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SIP in Mobile Network

3GPP has chosen SIP as the main signaling protocol for IMS.

Limitation : no method of directly interworking with PSTN no

corresponding message and functionality

Alternative method to SIP with ISUP encapsulation SIP-T (SIP for Telephony developed by IETF) SIP-I (SIP with ISUP encapsulation developed by ITU-T)

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SIP in MSS
SIP-T (IETF) and SIP-I (ITU) can
be used between MSS instead of BICC

SIP is used toward CPS UDP/TCP/SCTP can be used as

transport for SIP/SIP-T/SIP-I

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SIP interfaces

MGCF MSS
BICC / SIP Nc SIP Mg

IMS

MSS

H248

H248

Mc

GERAN

MGW
Nb

IP

MGW

UTRAN

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Mc

Important SIP specifications

IETF - RFC 3261 Session Initiation Protocol

IETF RFC 2976

IETF RFC 3262

SIP Enhancement: INFO method

SIP Enhancement: PRACK method

IETF RFC 3311

ITU-T Q.1912.5

SIP Enhancement: UPDATE method

Interworking between SIP and BICC/ISUP (SIP-I) SIP for IMS

3GPP 24.228/229

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SIP Main functions

Invite users to sessions
Find users current location to deliver invitation Transport session description transparently

Modify session characteristics Terminate sessions Sessions include, but are not limited to, IP telephony calls and multi-media
conferences

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SIP operation
SIP is based on an HTTP-like
request/response transaction model.

Each transaction consists of

a request that invokes a particular
method, or function, on the server zero or informational responses final response

Transaction is identified by a Cseq

(Command Sequence) number

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SIP Message
Both types of messages consist of a start-line, one or more header fields, an
empty line indicating the end of the header fields, and an optional message-body.

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SIP Request
Request Line structure

Each request invokes a method on a server

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Elements of a SIP Request Message

Method <sp> Request-URI <sp> SIP-version

Request Line

Message Header

INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds Max-Forwards: 70 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710@pc33.atlanta.com CSeq: 314159 INVITE Contact: <sip:alice@pc33.atlanta.com> Content-Type: application/sdp Content-Length: 142

<CRLF>

(142 Byte Message Body, not shown)

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Supported SIP Methods

INVITE ACK basic SIP BYE CANCEL Session initiation. Acknowledging final responses to INVITE. Requesting the release of the call (used in speech phase). Transaction cancelling request.

OPTIONS
UPDATE enhancements INFO

Asking for capabilities (for example, allowed methods).

Bearer-related information update/session refreshment. Sending information that does not affect call state (the main usage is tunneling ISUP messages that do not have a corresponding SIP message; it is also used to transfer out-of-band DTMF between MSC Servers). For reliable provisional response.

PRACK
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SIP Response
Status Line structure

A 3-digit integer Response-Code indicates the outcome of an attempt to

understand and satisfy a request

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Elements of a SIP Response Message

SIP-Version <sp> Status-Code <sp> Reason-Phrase

Status Line

Message Header

SIP/2.0 200 OK Via: SIP/2.0/UDP here.com:5060 From: BigGuy <sip:UserA@here.com>;tag=6744509153 To: LittleGuy <sip:UserB@there.com>;tag=8321234356 Call-ID: 12345601@here.com CSeq: 1 INVITE Contact: <sip:UserB@110.111.112.113> Content-Type: application/sdp Content-Length: 147

<CRLF> (147 Byte Message Body, not shown)

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SIP Response categories

1xx Informational
request received, continuing to process the request (=provisional responses)

2xx Success
the action was successfully received, understood and accepted

3xx Redirection
further action needs to be taken in order to complete the request

4xx Client Error

the request contains bad syntax or cannot be fulfilled at this server

5xx Server Error

the server failed to fulfil an apparently valid request

6xx Global Failure

the request cannot be fulfilled at any server

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SIP Response codes, part 1

Supported SIP responses 1xx Provisional responses 100 Trying 180 Ringing 181 Call Is Being Forwarded 182 Queued 183 Session Progress 3xx Redirection 300 Multiple Choices 301 Moved Permanently 302 Moved Temporarily 305 Use Proxy 380 Alternative Service

2xx Successful outcome 200 OK

The list contains all possible response codes. The ones that can not be sent by the MSS are marked in italics.

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SIP Response codes, part 2

4xx Client Error 400 Bad Request 401 Unauthorized 403 Forbidden 404 Not Found 405 Method Not Allowed 406 Not Acceptable 407 Proxy Authentication Required 408 Request Timeout 410 Gone 411 Length Required 413 Request Entity Too Large 414 Request-URI Too Large 415 Unsupported Media Type 420 Bad Extension 421 Extension Required 422 Session Interval Too Small 480 Temporarily Not Available
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481 Call Leg/Transaction Does Not Exist 482 Loop Detected 483 Too Many Hops 484 Address Incomplete 485 Ambiguous 486 Busy Here 487 Request Cancelled 488 Not Acceptable Here 490 Request Updated 491 Request Pending

The list contains all possible response codes. The ones that can not be sent by the MSS are marked in italics.

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SIP transactions
Client MSS A Server MSS B

transaction

INVITE 100(Trying) 180(Ringing) 200(OK)

requests

transaction

ACK

provisional responses (=1xx)

transaction

BYE 200(OK)

final responses (1xx)

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SIP Addresses

SIP Addresses SIP-URI

sip:user:password@host:port;uri-parameters <sip:john@server.com>
SIPS-URI

telephone-URI

tel: phone-number <tel:+392695550009>

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Tasks of a SIP message

A B B D

INVITE sip:bill@work.com SIP/2.0

To: Bill<sip:bill@work.com> From: John<sip:john@home.com> Call-ID: 267343@192.168.1.212 Via: SIP/2.0/UDP 192.168.1.212 Contact: john@192.168.1.212 CSeq: 1 INVITE Content-Type: application/SDP Content-Length: 124

A Defines purpose of message B Declares who the message relates to C Enables delivery of nonSIP media D Ensures request can be related to other messages in session E Records path

E
B D C C

v=0 o=192.168.1.212 IN IP4 192.168.1.212 s= Call from John c=IN IP4 192.168.1.212 m=audio 5004 RTP/AVP 8 3 0

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SDP Parameters
SDP-parameters: v: protocol version o: owner/creator and session identifier s: session name c: connection information t: time the session is active m: media name and transport address a: zero or more media attribute lines b: bandwidth information

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Codecs represented in SIP

Codec EFR FR G.711 A-law G.711 u-law G.723.1 G.723.1 Annex A SDP representation in SIP messages m=audio 1234 RTP/AVP 103 a=rtpmap:103 GSM-EFR/8000 m=audio 1234 RTP/AVP 3 a=rtpmap:3 GSM/8000 m=audio 1234 RTP/AVP 8 a=rtpmap:8 PCMA/8000 m=audio 1234 RTP/AVP 0 a=rtpmap:0 PCMU/8000 m=audio 1234 RTP/AVP 4 a=rtpmap:4 G723/8000 m=audio 1234 RTP/AVP 4 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=yes m=audio 1234 RTP/AVP 18 a=rtpmap:18 G729A/8000 m=audio 1234 RTP/AVP 18 a=rtpmap:18 G729A/8000 a=fmtp:18 annexb=yes m=audio 1234 RTP/AVP 97 a=rtpmap:97 iLBC/8000 m=audio 1234 RTP/AVP 100 a=rtpmap:100 CLEARMODE/8000

G.729a G.729a Annex B

iLBC Clearmode

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SDP Parameters in a SIP Request (example)

Content-Type:application/SDP;charset=ISO-10646 v=0 [1] o=- 0 0 IN IP4 10.8.110.141 [2] s=MSS call [3] c=IN IP4 10.10.1.85 [4] t=0 0 [4] m=audio 4568 RTP/AVP 103 [6] a=rtpmap:103 GSM-EFR/8000 [7] --4915214841081945351868528377048 Content-Type:application/ISUP;version=SIP;base=itu-t92+ [8] [9] --4915214841081945351868528377048-[10]

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MSS-MSS call with ISUP encapsulation

MSS-A
INVITE (IAM, SDP) 100 Trying 183 Session Progress (SDP) PRACK 200 OK 183 Session Progress (ACM) PRACK 200 OK 180 Ringing (CPG) PRACK 200 OK encapsulated ISUP messages

MSS-A

200 OK (ANM)
ACK

BYE (REL) 200 OK (RLC)

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ISUP versions for encapsulation

base string used itu-t18 itu-t92+ ansi88 ansi00 etsi121 etsi356 gr317 ttc87 ttc93+ Protocol ITU-T Q.761-4 (1988) ITU-T Q.761-4 (1992) ANSI T1.113-1988 ANSI T1.113-2000 ETS 300 121 ETS 300 356 BELLCORE GR-317 JT-Q761-4(1987-1992) JT-Q761-4(1993-)
e.g. ZJNM:FQDN=NAME.MSS.COM"::ISUPVR="itu-t92+";
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