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SIP stands for Session Initiation Protocol.
It is an internet protocol which has been developed and designed within
the Internet Engineering Task Force (IETF).
The protocol is used for establishing, and controlling sessions with
one or more devices (for example PCs, Telephones or PDA). Examples of a
session can include Voice Over IP (VoIP) calls, Instant Messaging,
distributed computer games, etc.
SIP should not be confused with H.323. Though similar in concept, and
both have been used in initiating and managing communications, the H.323
protocol was actually developed as part of the International
Telecommunication Union (ITU) and targeted at videoconferencing over
ISDN lines. Because of this limitation in scale, the H.323 protocol is
not used widely for Voice Over IP services.
SIP is not the only protocol that the
communicating devices will require in order to successfully route a VoIP
call. The purpose of SIP is just to
initiate and control the communication, the communication itself occurs
via other protocols. Two of the most common protocols that are used for VoIP
along with SIP are
Real Time Protocol (RTP) and Session
Description Protocol (SDP)
RTP: In association with several codecs that can convert a
user’s voice into computer data, the RTP
protocol is used to carry the real-time multimedia data.
SDP: This is the protocol by which the capabilities and
available codecs upon the participant’s devices are exchanged.
A SIP session is initiated by a client (sometimes called a User Agent
Client – or UAC), and is routed by entities in the network to the
receiving client. The client can take on the form of a piece of software
on a Personal Computer, or hardware such as an IP phone, or even a
traditional phone with an Analog Telephone Adapter (ATA).
SIP has three major goals that are
essential in ensuring that communications can occur in a consistent and
simple manner. These are as follows:
Name resolution and routing
SIP clients are identified by a unique
address, commonly known as a Uniform Resource Identifier (URI). This URI
can take many forms, but in practice it is typically similar to a phone
number or an email address. Name resolution takes sessions targeted for
a URI, and maps that to a specific UAC, and routes the request via
network servers that understand how to route SIP.
Capability negotiation
UAC’s will have different capabilities,
such as the ability to have high or low quality audio, or the ability to
guarantee a certain quality of service. As part of the session
initiation, these capabilities can be negotiated between the UACs, such
that a consistent set of capabilities are agreed upon.
Participant management
Within a session there are typically two
participants (or two independent UACs). However there are cases whereby
additional participants may be required to be part of the session. An
example of this would be establishing a conference session and adding
additional users. SIP provides mechanisms to control who can include
additional participants, and how they can be added or removed.
SIP makes VoIP possible, not only technically but also making it usable and
cost effective. The use of SIP allows your voice and data to travel on
the same internet connection, and therefore not require a
dedicated phone line. This is the main reason why VoIP can save you $500
a year on your home bill.
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