Chap#11

Session Initiation Protocol (SIP)

Session Initiation Protocol 



The session initiation protocol (SIP) is an application-layer signaling-control protocol used to establish, maintain, and terminate multimedia sessions. Multimedia sessions include internet telephony, conferences, and other similar applications involving such media as audio, video, and data

Session Initiation Protocol 





You can use SIP invitations to establish sessions and carry sessions descriptions. SIP supports unicast and multicast sessions as well as point-to-point and multipoint calls. You can establish and terminate communications using the following five SIP facets: User location, user capability, user availability, call setup, and call handling

User Agents 

User agents are client end system applications that contain both a useragent client (UAC) and a user-agent (UAS), otherwise known as client and server, respectively. 



Client—Initiates SIP requests and acts as the user’s calling agent. Server –Receives requests and returns responses on behalf of the users; acts as the user-called agent.

Network Services 

Two types of sip network servers exist: proxy servers and redirect servers: 

Proxy servers 



Acts on behalf of others clients and contains both client and server functions. A proxy server interprets and can rewrite request headers before passing them on to other servers. Rewriting the headers identifies the proxy as the initiator of the request and ensures that replies follow the same path back to the proxy instead of the client.

Redirect Server 

Accepts sip requests and sends a redirect response back to the client containing the address of the next server. Redirect servers do not accept calls, nor do they process or forward sip requests.

Addressing 

SIP addresses, also called sip universal resource locators (URLs) exist in the form of users @ hosts. Similar to email addresses; a sip URL is identified by user @ host. The user portion of the address can be a user name of telephone number, and the host portion can be a domain name or network address. You can identify a user’s sip URL by his or her email address. The following examples depicts two possible sip user  

Sip: [email protected] Sip: [email protected]

Locating a Server 

A client can send a sip request either directly, to a locally configured proxy server, or to the IP address and port of the corresponding sip URL. Sending a sip request directly is relatively easy, as the end-system application knows the proxy server. Sending a sip request in the second manner is somewhat more complicates, for the following reasons

Locating a Server 







The client must determine the IP address and port number of the server for which the request is destined. If the port number is not listed in the requested sip URL, the default port is 5060. If the protocol type is not listed in the request sip URL, the client must first attempt to connect using user datagram protocol and then transmission control protocol (TCP). The client queries the domain name system (DNS) server for the host IP address. If it finds no address records, the client is unable to locate the server and cannot continue with the request.

SIP Transactions. 

After addressing is resolved, the client sends one or more sip requests and receives one or more responses from the specified server

Locating a user 

The called party might move from one to several end systems over time. He or she might move form the corporate local area network (LAN) to a home office through his or her internet service provider (ISP), or to a public internet connection while attending a conference

Session Initiation Protocol Background  







IETF RFC2543 issued in 1999 Protocol for creating, modifying, and terminating sessions (Internet multimedia conferences, Internet telephone calls, and multimedia distributions) with one or more participants Distributed call processing with intelligent endpoints Based on IETF standards (RTP, RTCP, HTTP, SDP, DNS, SAP, RTSP) Addressing by E.164, e-mail, or DNS SRV record

Cisco SIP Components 3 - INVITE

PS 2 - INVITE

4 - 3XX Redirect 5 - INVITE

7 - 200 OKAY

1 - REGISTER

6 - 200 OKAY

IPP RTP 

LOC RED

User Agent

SIP-GW

Cisco SIP Proxy Server (PS) Registration Server (REG) - Accepts registration requests from UAs Redirect Server (RED) - Maps SIP request to one or more addresses Location Server (LOC) - Provides information on a callee’s location 



User Agent (UA) Cisco SIP Gateway (SIP-GW) Cisco IP Phones (IPP) 

SIP Messages 







 

INVITE – Indicates a user or service is being invited to participate in a call session ACK – Confirms that the client has received a final response to an INVITE request BYE – Terminates a call and can be sent by either the caller or the callee CANCEL – Cancels any pending searches but does not terminate a call that currently is in progress OPTIONS – Queries the capabilities servers REGISTER – Registers the address listed in the To header field with a SIP server (not gateway)

SIP Addressing 

Fully-qualified domain names 



E.164 addresses 



sip:[email protected] sip:[email protected]; user=phone

Mixed addresses sip:[email protected]; user=phone sip:[email protected] 



E.164 addresses 

tel:14085551234

Modeled after mailto URLs

SIP Headers Explained Received: Tells if this messages was Sent or Received INVITE sip:[email protected]:5060 SIP/2.0 - Request URI line Expires: When this SIP message will expire (in seconds) Content-Type: What type of body will be attached to the SIP message Via: A list of all SIP devices that were in the signaling path; also includes Proxy “branch” parameter To: The destination of the SIP message From: The originator of the SIP message Call-ID: The unique Call Identifier CSeq: A sequence of messages in that Method Contact: Where the CallingParty can be reached for the return signaling path Content-Length: The length of the body attached to the SIP message User-Agent: The device that initiated the SIP message Accept: What application should be used to read the Sip body Record-Route: The list of SIP Proxies that must be in the return signaling path

SIP Headers Explained  

 

 

 

v=the SDP version o=the Organization of the device that originated the SDP message s=the Description of the SDP message c=the IPAddress or Hostname that the originator expects the media to arrive at t=the Time field m=the Description of the media that the originator expects to receive a=a description of the media attributes a=a description of the media attributes

SIP Signaling LOC RED

X1001

SIP-GW

NetA INVITE TRYING

PS

[email protected] PS

IP

NetB

INVITE REDIRECT

INVITE

INVITE TRYING

RINGING 200 OK ACK

RINGING

RINGING

200 OK

200 OK

Media ACK

BYE 200 OK

BYE 200 OK

ACK BYE 200 OK

IPP

Cisco IOS Software SIP Configuration 10.1.1.1

1XXX PBX SIP-GW

NetA

dial-peer voice 1 voip destination-pattern [1-7]... dtmf-relay cisco-rtp application session session protocol sipv2 codec g711ulaw session target sip-server ! dial-peer voice 2 pots application session destination-pattern 1… port 1/1 ! sip-ua sip-server ipv4:10.1.1.1

PS

LOC RED

IP

10.1.2.1 PS

2XXX NetB

PBX SIP-GW

dial-peer voice 1 voip destination-pattern [1-7]… dtmf-relay cisco-rtp application session session protocol sipv2 codec g711ulaw session target sip-server ! dial-peer voice 2 pots application session destination-pattern 2… port 1/1 ! sip-ua sip-server ipv4:10.1.2.1