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SIP Protocol
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Chapter 4
SIP Protocol
INTRODUCTION
The Session Initiation Protocol (SIP) is a signalling protocol used for
establishing sessions in an IP network. A session could be a simple two-way
telephone call or it could be a collaborative multi-media conference session.
Since SIP is part of a IETF (
Internet Engineering Task Force
) specification, it
extends the open-standard spirit of the Internet to messaging, enabling remote
computers, phones, televisions and software to communicate.
Members in a session can communicate via multicast or via a mesh of unicast
relations, or via a combination of these. SIP supports session descriptions that
enable participants to agree on a set of specific media types. SIP is not tied to
any particular conference control protocol. In essence, SIP has to provide or
enable the following functions:
Name translation and user location:
ensuring that the call reaches the
called party wherever it is located. Carrying out any mapping of descriptive
information to location information. Ensuring that details of the nature of
the call are fully supported.
Feature negotiation
: this allows the group involved in a call to agree on
the features supported, recognizing that not all the parties can support the
same level of features.
Call participant management
: during a call, a participant can bring other
users onto the call or cancel connections to other users. Besides, users could
be transferred or placed on hold.
Call features changes
: a user should be able to change the call
characteristics during the course of the call. For example, a call may have
been set up as 'voice-only', but the users may need to enable a video
function.
SIP fulfils these functions and re-uses other web elements to make it flexible
and scalable.
Rather than defining a new type of addressing system, SIP addresses users by
an email-like address. Each user is identified through a hierarchical URL that is
built around elements such as a user's phone number or host name (for
example,
). This means that it is just as easy to
redirect someone to another phone as it is to redirect someone to a webpage.
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In this sense, SIP borrows from the email model, using the Domain Name
System, to deliver requests to the server that can appropriately cope with
them. This also simplifies the integration of voice and email.
SIP is also transport layer indipendent. Therefore, the underlying transport
could be IP over ATM. SIP uses the User Datagram Protocol (UDP) as well as
the Transmission Control Protocol (TCP), flexibly connecting users independent
of the underlying infrastructure.
SIP provides the necessary protocol mechanisms so that end systems and
proxy servers can provide services:
User location
User capabilities
User avaliability
Call setup
Call handling
Call forwarding
Callee and calling number delivery
Personal mobility, i.e. the ability to reach a called party under a
single, location-independent address regardless the change of
terminal by the user
Terminal type negotiation and selection
Terminal capability negotiation
Caller and callee authentication
Blind and supervised call transfer
Invitation to multicast conferences.
Protocol Components
There are four basic components within SIP:
1)
SIP user agent
2)
SIP network server
3)
SIP Registration service
4)
SIP Event and Presence server
The user agent is the end system component for the call and the SIP server is
the network device that handles the signalling associated with multiple calls.
The user agent itself has a
client element
, the User Agent Client (UAC) and a
server element
, the User Agent Server (UAS). The client element initiates the
calls and the server element answers the calls. This allows peer-to-peer calls
to be made using a client-server protocol.
The main function of the SIP servers is to provide name resolution and user
location, since the caller is unlikely to know the IP address or host name of the
called party, and to pass on messages to other servers using next hop routing
protocols.
SIP servers can operate in three different modes:
Stateful proxy mode
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Stateless proxy mode
Re-direct server
The difference between these modes is that a
server in a stateful mode
remembers the incoming requests it receives, along with the responses it sends
back and the outgoing requests it sends on.
A
server in a stateless mode
forgets all information once it has sent a request.
These stateless servers are likely to be the backbone of the SIP infrastructure
while stateful-mode servers are likely to be the local devices close to the user
agents, controlling domains of users.
A
re-direct server
receives the request, but instead of passing it onto the next
server, it sends a response to the caller party, indicating the address for the
called user. This method enables the provisioning of the address for the caller,
in order make it possible to contact the called party directly at the next server.
The
SIP Registration Service
provides a means for a particular device to
register to use a SIP address. As seen at the beginning of this chapter, SIP
addresses use ‘URLs’ based on the same addressing scheme used in the web
and similar in form to an email address. The SIP address provides a single
address of record for the user that delivers a one number service for all
communications applications. Users can dynamically register the devices
through which they may be contacted for all types of applications. As a result,
people will no longer have to hand out multiple contact addresses as the system
will automatically handle the distribution of all types of calls appropriately
through the proxy and redirect servers.
SIP Event and Presence Servers
allow the effective sharing of information about
and between users and/or applications.
SIP Signalling
Here is reproduced a simple procedure of a call set-up:
a)
The caller (User Agent Caller) sends a request with the SIP URL
of the called party.
b)
If the client knows the location of the other party, is sends the
request directly to its IP address. If the location is unknown, the
client can send it to a SIP network server, which has been
locally configured.
c) The server will try to resolve the called user's location and
sends the request to them. There are many ways it can do this:
searching the DNS or accessing databases. Alternatively, the
server may be a redirect server that may return the called user
location to the calling client for it to try directly. During the
course of locating a user, one SIP network server can proxy or
redirect the call to additional servers until it arrives at one that
definitely knows the IP address where the called user can be
found.
d) Once found, the request is sent to the user. In the simplest
case, the user's telephony client receives the request, that is,
the user's phone rings. If the user takes the call, the client
responds to the invitation with the designated capabilities of
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the client software and a connection is established. If the user
declines the call, the session can be redirected to a voice mail
server or to another user.
SIP has two additional significant features.
1)
A stateful SIP server's ability to split an incoming call so that several
extensions can be rung at once. The first extension to answer takes the
call.
2)
SIP's unique ability to return different media types within a single
session.
SIP Methods
The commands that SIP uses are called methods. SIP defines the following
methods:
Protocol Header Structure
The protocol is composed of:
a)
A start line
b)
A message header
c)
An empty line
d)
An optional message body.
R
EQUEST
M
ESSAGES
The format of the Request Packet header is shown in the following figure:
SIP Method
Description
INVITE
Invites a user to a call
ACK
Used to facilitate reliable message exchange for
INVITEs
BYE
Terminates a connection between users or declines a
call
CANCEL
Terminates a request, or search, for a user
OPTIONS
Solicits information about a server’s capabilities
REGISTER
Registers a user’s current location
INFO
Used for mid-session signalling
M
ETHOD
R
EQUEST
URI
SIP V
ERSION

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