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Session Initiation Protocol - SIPSwitching Core Network Signalling Rel.4

M14/U4

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Nokia Siemens Networks Academy

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Intellectual Property Rights

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Siemens Networks has the sole right to copy, distribute, amend, modify, develop, license,sublicense, sell, transfer and assign the Nokia Siemens Networks training material. Individualscan use the Nokia Siemens Networks training material for their own personal self-developmentonly, those same individuals cannot subsequently pass on that same Intellectual Property toothers without the prior written agreement of Nokia Siemens Networks. The Nokia SiemensNetworks training material cannot be used outside of an agreed Nokia Siemens Networkstraining session for development of groups without the prior written agreement of NokiaSiemens Networks.

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Document change history

Date Version Name Change comment

March, 03.08 1 Pubate Satienpoch New creation for M14

July, 08.08 2 Guido Schneiders 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 telephonyservices across the Internet like establishing, modifying and terminatingsessions 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 nocorresponding 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 ofBICC

SIP is used toward CPS

UDP/TCP/SCTP can be used astransport for SIP/SIP-T/SIP-I

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SIP interfaces

MSS

MGW

MGCF

MGWIP

GERAN

UTRAN

BICC / SIP

H24

8

H24

8

MSS

IMSSIPNc Mg

Mc Mc

Nb

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Important SIP specifications

IETF - RFC 3261 Session Initiation Protocol

IETF RFC 2976 SIP Enhancement: INFO method

IETF RFC 3262 SIP Enhancement: PRACK method

IETF RFC 3311 SIP Enhancement: UPDATE method

ITU-T Q.1912.5 Interworking between SIP and BICC/ISUP

(SIP-I)

3GPP 24.228/229 SIP for IMS

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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-mediaconferences

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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 particularmethod, 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 Request-URI SIP-version

INVITE sip:bob@biloxi.com SIP/2.0

Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhdsMax-Forwards: 70

To: Bob

From: Alice ;tag=1928301774

CSeq: 314159 INVITE

Contact:

Content-Type: application/sdpContent-Length: 142

(142 Byte Message Body, not shown)

Message Header

Request Line

Call-ID: a84b4c76e66710@pc33.atlanta.com

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Supported SIP Methods

INVITE Session initiation.

ACK Acknowledging final responses to INVITE.

BYE Requesting the release of the call (used in speech phase).

CANCEL Transaction cancelling request.

OPTIONS Asking for capabilities (for example, allowed methods).

UPDATE Bearer-related information update/session refreshment.

INFO Sending information that does not affect call state

(the main usage is tunneling ISUP messages that donot have a corresponding SIP message; it is also

used to transfer out-of-band DTMF between MSC Servers).

PRACK For reliable provisional response.

basicSIP

enhance

ments

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SIP Response

Status Line structure

A 3-digit integer Response-Code indicates the outcome of an attempt tounderstand and satisfy a request

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Elements of a SIP Response Message

SIP-Version Status-Code Reason-Phrase

SIP/2.0 200 OKVia: SIP/2.0/UDP here.com:5060

From: BigGuy ;tag=6744509153

To: LittleGuy ;tag=8321234356Call-ID: 12345601@here.com

CSeq: 1 INVITEContact:

Content-Type: application/sdp

Content-Length: 147

Message Header

Status Line

(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 Successthe action was successfully received, understood and accepted

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

4xx Client Errorthe request contains bad syntax or cannot be fulfilled at this server

5xx Server Errorthe server failed to fulfil an apparently valid request

6xx Global Failurethe 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

2xx Successful outcome

200 OK

3xx Redirection

300 Multiple Choices

301 Moved Permanently

302 Moved Temporarily

305 Use Proxy

380 Alternative Service

The list contains all possible response codes.

The ones that can notbe sent by the MSS are

marked in italics.

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SIP Response codes, part 2

4xx Client Error

400 Bad Request401 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

481 Call Leg/Transaction Does Not Exist482 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 notbe sent by the MSS are

marked in italics.

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SIP transactions

Client Server

200(OK)

BYE

ACK

180(Ringing)

100(Trying)

INVITE

200(OK)transaction

transaction

transaction

requests

provisional

responses

(=1xx)

finalresponses

(1xx)

MSS A MSS B

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SIP Addresses

SIP-URI

SIPS-URI

sip:user:password@host:port;uri-parameters

telephone-URI

tel: phone-number

SIP Addresses

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Tasks of a SIP message

Defines purpose of

message

Declares who the message

relates to

Enables delivery of non-

SIP media

Ensures request can be

related to other messages

in session

Records path

AA

B

B

CC

C

C

D

D

D

E

B

B

E

INVITE sip:bill@work.com SIP/2.0

To: Bill

From: John

Call-ID: 267343@192.168.1.212

Via: SIP/2.0/UDP 192.168.1.212

CSeq: 1 INVITE

Content-Type: application/SDP

Content-Length: 124

v=0o=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

Contact: john@192.168.1.212

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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 activem: media name and transport address

a: zero or more media attribute lines

b: bandwidth information

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Codecs represented in SIP

Codec SDP representation in SIP messages

EFR m=audio 1234 RTP/AVP 103

a=rtpmap:103 GSM-EFR/8000

FR m=audio 1234 RTP/AVP 3

a=rtpmap:3 GSM/8000

G.711 A-law m=audio 1234 RTP/AVP 8

a=rtpmap:8 PCMA/8000

G.711 u-law m=audio 1234 RTP/AVP 0

a=rtpmap:0 PCMU/8000

G.723.1 m=audio 1234 RTP/AVP 4

a=rtpmap:4 G723/8000

G.723.1 Annex A m=audio 1234 RTP/AVP 4

a=rtpmap:4 G723/8000

a=fmtp:4 annexa=yes

G.729a m=audio 1234 RTP/AVP 18

a=rtpmap:18 G729A/8000

G.729a Annex B m=audio 1234 RTP/AVP 18

a=rtpmap:18 G729A/8000

a=fmtp:18 annexb=yes

iLBC m=audio 1234 RTP/AVP 97

a=rtpmap:97 iLBC/8000

Clearmode m=audio 1234 RTP/AVP 100

a=rtpmap:100 CLEARMODE/8000

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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 MSS-A

INVITE (IAM, SDP)

PRACK

200 OK

183 Session Progress (ACM)

PRACK

180 Ringing (CPG)

PRACK

200 OK

200 OK (ANM)

200 OK

100 Trying

183 Session Progress (SDP)

ACK

BYE (REL)

200 OK (RLC)

encapsulated

ISUP messages

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ISUP versions for encapsulation

base string used Protocol

itu-t18 ITU-T Q.761-4 (1988)

itu-t92+ ITU-T Q.761-4 (1992)

ansi88 ANSI T1.113-1988

ansi00 ANSI T1.113-2000

etsi121 ETS 300 121

etsi356 ETS 300 356

gr317 BELLCORE GR-317

ttc87 JT-Q761-4(1987-1992)

ttc93+ JT-Q761-4(1993-)

e.g. ZJNM:FQDN=NAME.MSS.COM"::ISUPVR="itu-t92+";