
What is SIP?
Introduction
SIP (Session Initiation Protocol) is a protocol developed to assist in providing advanced
telephony services across the Internet. Internet telephony is evolving from its use as a
"cheap" (but low quality) way to make international phone calls to a serious business
telephony capability. SIP is one of a group of protocols required to ensure that this
evolution can occur.
SIP is part of the IETF standards process and is modeled upon other Internet protocols such
as SMTP (Simple Mail Transfer Protocol) and HTTP (Hypertext Transfer Protocol.) It is used
to establish, change and tear down (end) calls between one or more users in an IP-based
network. In order to provide telephony services there is a need for a number of different
standards and protocols to come together - specifically to ensure transport (RTP), signaling
inter-working with today’s telephony network, to be able to guarantee voice quality
(RSVP, YESSIR), to be able to provide directories (LDAP), to authenticate users (RADIUS,
DIAMETER), and to scale to meet the anticipated growth curves. This introduction covers
only SIP, but at the end of the paper there is a brief overview on associated standards.
SIP is described as a control protocol for creating, modifying and terminating sessions with
one or more participants. These sessions include Internet multimedia conferences, Internet
(or any IP Network) telephone calls and multimedia distribution. 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 allow participants to agree on a set of compatible
media types. It also supports user mobility by proxying and redirecting requests to the
user's current location. 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 they are located. Carrying out any mapping of descriptive information to
location information. Ensuring that details of the nature of the call (Session) are
supported.
Feature Negotiation - This allows the group involved in a call (this may be a multi-party
call) to agree on the features supported – recognizing that not all the parties can support
the same level of features. For example video may or may not be supported; as any form
of MIME type is supported by SIP, there is plenty of scope for negotiation.
Call Participant Management - During a call a participant can bring other users onto the
call or cancel connections to other users. In addition, users could be transferred or placed
on hold.
Call feature 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 in the
course of the call, the users may need to enable a video function. A third party joining a
call may require different features to be enabled in order to participate in the call
Protocol Components
There are two components within SIP. The SIP User Agent and the SIP Network Server. The
User Agent is effectively the end system component for the call and the SIP Server is the
network device that handles the signaling 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.
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