Session Initiation Protocol (SIP)
#1

Session Initiation Protocol
Session Initiation Protocol (SIP) is a protocol developed by IETF MMUSIC Working Group and proposed standard for initiating, modifying, and terminating an interactive user session that involves multimedia elements such as video, voice, instant messaging, online games, and virtual reality.
SIP clients traditionally use TCP and UDP port 5060 to connect to SIP servers and other SIP endpoints. SIP is primarily used in setting up and tearing down voice or video calls. However, it can be used in any application where session initiation is a requirement. These include, Event Subscription and Notification, Terminal mobility and so on. There are a large number of SIP-related RFCs that define behavior for such applications. All voice/video communications are done over RTP.
A motivating goal for SIP was to provide a signaling and call setup protocol for IP-based communications that can support a superset of the call processing functions and features present in the public switched telephone network (PSTN).
SIP enabled telephony networks can also implement many of the more advanced call processing features present in Signalling System 7 (SS7), though the two protocols themselves are very different. SS7 is a highly centralized protocol, characterized by highly complex central network architecture and dumb endpoints (traditional telephone handsets). SIP is a peer-to-peer protocol.

SIP network elements

Hardware endpoints, devices with the look, feel, and shape of a traditional telephone, but that use SIP and RTP for communication, are commercially available from several vendors. Some of these can use Electronic Numbering (ENUM) or DUNDi to translate existing phone numbers to SIP addresses using DNS, so calls to other SIP users can bypass the telephone network, even though your service provider might normally act as a gateway to the PSTN network for traditional phone numbers (and charge you for it).

SIP makes use of elements called proxy servers to help route requests to the user s current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users.
SIP also provides a registration function that allows users to upload their current locations for use by proxy servers.
Since registrations play an important role in SIP, a User Agent Server that handles a REGISTER is given the special name registrar.
It is an important concept that the distinction between types of SIP servers is logical, not physical.
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#2
i want seminar report for sip
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#3
its latest seminar topic in m knowledge no ready made seminar report available

please make one

i can give some good tutor see
http://cse.iitd.ernet~pkalra/siv864/SIP-overview.pdf
http://old.iptelsip/siptutorial.pdf
http://radvisionNR/rdonlyres/51855E82-BD7C-4D9D-AA8A-E822E3F4A81F/0/RADVISIONSIPProtocolOverview.pdf
http://sctpsipsctp.pdf
http://en.wikipediawiki/Session_Initiation_Protocol


SIP ( Session Initiation Protocol ), the protocol can provide converged and unified communication services, such as voice and video conversations. Like HTTP, SIP is a text-based protocol, which makes it easy to write applications that incorporate the technology,


i hope these all help you,
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#4

if you wanna read more about please read following pages
http://studentbank.in/report-implementat...7#pid47637
http://studentbank.in/report-session-ini...otocol-sip
http://studentbank.in/report-session-ini...-sip--2633
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