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 user-
agent 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: ciscopress@cisco.com
 Sip: 4085262222@171.171.171.1
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 LOC
RED
2 - INVITE 4 - 3XX Redirect

5 - INVITE
1 - REGISTER 7 - 200 OKAY
6 - 200 OKAY
User
IPP Agent

RTP 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
 sip:jdoe@cisco.com
 E.164 addresses
 sip:14085551234@gateway.com; user=phone
 Mixed addresses
sip:14085551234@10.1.1.1; user=phone
sip:jdoe@10.1.1.1

 E.164 addresses
 tel:14085551234

Modeled after mailto URLs

 SIP Headers Explained
Received: Tells if this messages was Sent or Received
INVITE sip:23198@172.17.207.91:5060 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 jifrench@cisco.com

PS PS IPP
X1001 NetA IP NetB
SIP-GW

INVITE
INVITE
TRYING
REDIRECT

INVITE INVITE
TRYING
RINGING RINGING
RINGING
200 OK 200 OK
200 OK
Media
ACK
ACK
ACK

BYE BYE BYE

200 OK
200 OK 200 OK
 Cisco IOS Software SIP
Configuration LOC
10.1.1.1 RED
1XXX 10.1.2.1 2XXX
PBX PS PS PBX
NetA IP NetB SIP-GW
SIP-GW

dial-peer voice 1 voip dial-peer voice 1 voip

destination-pattern [1-7]... destination-pattern [1-7]…
dtmf-relay cisco-rtp dtmf-relay cisco-rtp
application session application session
session protocol sipv2 session protocol sipv2
codec g711ulaw codec g711ulaw
session target sip-server session target sip-server
! !
dial-peer voice 2 pots dial-peer voice 2 pots
application session application session
destination-pattern 1… destination-pattern 2…
port 1/1 port 1/1
! !
sip-ua sip-ua
sip-server ipv4:10.1.1.1 sip-server ipv4:10.1.2.1