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Controlling Overload in Network of SIP servers
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INTRODUCTION
The Session Initiation Protocol, or SIP, is a new IETF signaling protocol for establishing real-time calls and conferences over Internet Protocol networks. Each session may include different types of data such as audio and video although currently most of the SIP extensions address audio communication. As a traditional text-based Internet protocol, it resembles the hypertext transfer protocol (HTTP) and simple mail transfer protocol (SMTP). SIP uses Session Description Protocol (SDP) for media description.
SIP is independent of the packet layer. The protocol is an open standard and is scalable. It has been designed to be a general-purpose protocol. However, extensions to SIP are needed to make the protocol truly functional in terms of interoperability. Among SIP basic features, the protocol also enables personal mobility by providing the capability to reach a called party at a single, location-independent address.
SIP IN PROTOCOL STACK
Figure-1
COMMUNICATION SESSION
Figure-2
FACETS OF COMMUNICATION:
User location: determination of the end system to be used for communication;.
User capabilities: determination of the media and media parameters to be used.
User availability: determination of the willingness of the called party to engage in communications.
Call setup: "ringing", establishment of call parameters at both called and calling party.
Call handling: including transfer of data.
Call Termination: To terminate the session.
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