OAA000007 Call control protocol SIP protocol

ISSUE 2.0

Fixed Network Curriculum Development Section

Objective

After this session, you will learn:

Functions of SIP protocol SIP commands Meanings and usage of the parameters in SIP commands

Message interaction process of SIP

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References
Product Manual Technical Manual Signaling and

Protocols Chapter 6 SIP

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About this Session

Overview Message Structure Call Flow

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Overview

SoftX3000 supports such call control protocols as ISUP,

SIP, H.323, etc. ISUP is used for interworking with the

existing PSTN while SIP and H.323 protocols are used for interworking with SIP domain, H.323 domain or other softswitch systems

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Concept of SIP

The Session Initiation Protocol (SIP) is an application-layer control protocol used to establish, modify and terminate multimedia sessions or calls. These multimedia sessions include multimedia conferences, remote education, Internet telephony and similar applications.

SIP can be used to initiate sessions as well as inviting members to sessions that have been advertised and

established by other means.

SIP transparently supports name mapping and redirection services, allowing the implementation of ISDN, IN, and subscriber mobile services.

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Support of SIP to Multimedia

SIP is a light-weight signaling protocol, which can be used in conjunction with other call setup and signaling protocols.

SIP does not offer conference control services such as

site control or voting and does not prescribe how a conference is to be managed, but SIP can be used to introduce conference control protocols. SIP does not allocate multicast addresses.

SIP can invite users to sessions with and without resource reservation. SIP does not reserve resources itself, but can convey to the invited system the information necessary to do this.
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Support of SIP to Multimedia

SIP supports five facets of establishing and terminating

multimedia communications:

User location: determination of the end system to be used

for communication.

User capabilities: determination of the media and media parameters to be used.

User availability: determination of the willingness of the called party to engage in communications.

Call setup: "ringing", establishment of call parameters at both called and calling party.

Call control : establish, modify and terminate multimedia sessions or calls.

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Terms

Location Service

to obtain information about a called subscriber's possible location(s).

ProxyProxy sever Redirect server

A redirect server is a server that accepts an SIP

request, maps the address into zero or more new addresses and returns these addresses to the client.

Registrar

A registrar is a server that accepts REGISTER requests. A registrar is typically co-located with a proxy or redirect server and MAY offer location services.
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Terms

User Agent Client

A user agent client is a client application that initiates the SIP request.

User Agent Server

A user agent server is a server application that

contacts the user when a SIP request is received

and that returns a response on behalf of the user. The response accepts, rejects or redirects the request.

User Agent

An application which contains both a user agent client and user agent server.
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Protocol Stack
H.263 etc. RTP

H.323

SIP

RTSP

RSVP

RTCP

TCP

UDP

IP

PPP Sonet

AAL3/4

AAL5 ATM Ethernet

PPP V.34
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SIP Implementation in Softx3000

SoftX3000
SIP/SIP-T

SoftX3000

IP

IP

IP

IP

IP Core
SoftPhone SoftPhone

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About this Session

Overview Message Structure Call Flow

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Classification of SIP Messages

There are two types of messages :

Request messages:
sent by the client to the server including : INVITE, ACK, BYE, CANCEL, REGISTER,OPTION, etc.

Response messages:
Used to display the responses of the server to the requests of clients including 1xx, 2xx, 3xx, 4xx, 5xx and 6xx and ACK.

Both the request and response messages contain SIP

header fields and SIP message fields.

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Request Messages
Message INVITE ACK Function To initiate a session request The response message to the INVITE message

BYE
CANCEL REGISTER OPTIONS INFO

End session command

To cancel an unfinished request Register To query the servers capabilities. Interactive processed. contents when a call is being

MESSAGE

Short message

SUBSCRIBE
NOTIFY
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To subscribe an event
To send event notice
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Response Messages
Message 1XX 2XX 3XX 4XX 5XX 6XX Function Temporary response Success Redirection Client error Server error Global failure

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SIP Request Command Format

Command name Peer-end URL
Call-id:value via: value From:value To: value Contact:value Cseq value : Content-Length: value Max-Forward: value Content-Type: value White Space SDP
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Protocol version

Message Header

Parameter line

Example of SIP Request Message

INVITE sip:0109@127.0.0.1:5060;User=phone SIP/2.0

Call-ID:01E04633512400000@127.0.0.1
Via:SIP/2.0/UDP 127.0.0.1:5061 From:<sip:010203@127.0.0.1:5061;User=phone>;tag=29005

358336B534F610A000
To:<sip:0109@127.0.0.1:5060;User=phone> Contact: sip:010203@127.0.0.1:5061 CSeq:1 INVITE

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Example of SIP Request Message (Cont.)

Max-Forwards:70

Content-Type: application/SDP
Content-Length:168 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com s=Session SDP c=IN IP4 192.0.0.1 t=0 0

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Example of SIP Request Message (Cont.)

m=audio 49172 RTP/AVP 0 8

a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=sendonly

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Response Format
SIP Protocol version Call-id: value via: value From:value To: value Contact:value Response message header

Message header

Cseq: value
parameter Content-Length: value Max-Forward: value Content-Type: value White Space SDP
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Example of SIP Response Message

SIP/2.0 200 OK Content-Type:application/SDP Via:SIP/2.0/UDP 127.0.0.1:5061 Call-ID:01EF351F8140000000000@127.0.0.1

CSeq:1 INVITE
From:<sip:010203@127.0.0.1:5061;User=phone>;tag=29005 358336B534F610A000

To:<sip:0109@127.0.0.1:5060;User=phone>;tag=5358336B5
34F2900CD1B0000 Contact:<sip:0109@127.0.0.1:55061>
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Example of SIP Response Message (Cont.)

Content-Length:156 v=0 o=HuaweiSoftX3000 127.0.0.1 1073741824 1073741824 IN IP4

s=Sip Call
c=IN IP4 110.111.112.113 t=0 0

m=audio 5060 RTP/AVP 0

a=rtpmap:0 PCMU/8000
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About this Session

Overview Message Structure Call Flow

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Registration Flow
SIP Phone Register 401 Unauthorized SoftX3000

Register 200 OK

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SIP Phone to SIP Phone

SIP PhoneA SoftX3000 SIP PhoneB

1 2

INVITE 100 Trying 3 4

INVITE
100 Trying

6 8

180 Ringing 200 OK

5 7

180 Ringing 200 OK

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SIP Phone to SIP Phone (cont.)

SIP PhoneA SoftX3000 ACK 10 Conversation 11 12 BYE ACK SIP PhoneB

200 OK 13
14 BYE 200 OK

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Call Between SIP Trunk

SoftX3000 1 2 3 4 5 6 7 INVITE 100 Trying 180 Ringing 200 OK ACK BYE 200 OK SoftX3000

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Successful call flow with SIP-T

SGA IAM SoftX3000A 1 SoftX3000B INVITE SGB

IAM
ACM

2 100 Trying 3 180 Ring ACM ANM 4 ANM 5

200 OK
ACK Conversation

REL 6

BYE

REL
RLC 7 RLC
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200 OK

Summary

SIP is used to establish, modify and terminate a multimedia conference, such as conference call over Internet. SIP can be

used to initiate sessions as well as inviting members to

sessions that have been advertised and established by other means.

SIP request messages: SIP messages sent by a client to the server on the basis of designated operation for activation, which include such messages as INVITE, PRACK, BYE, CANCEL, UPDATE, etc.

SIP Response Messages: Used to display the status of the

requests sent by clients to the server, including the 1xx, 2xx, 3xx, 4xx, 5xx and 6xx responses and ACK.

The functions of each command in SIP call process should be mastered as the key points. Level: Internal Security

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