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What is Session Initiation Protocol (SIP)

Encyclopedia of Networked and Virtual Organizations
Provides SIPS communications between communication servers
Published in Chapter:
A Managerial Perspective of Mobile VoIP
Mahesh S. Raisinghani (TWU School of Management, USA)
Copyright: © 2008 |Pages: 8
DOI: 10.4018/978-1-59904-885-7.ch115
Abstract
Voice over IP (VoIP) is the convergence of Internet technology and mobile telephones that offers low-cost voice communication services. It is still in its infancy but has a potential to be adopted by masses in the future, as the technology matures. VoIP is a technology that is used to make telephone calls using a broadband internet connection instead of a regular phone line. There are some services that allow the users to make calls to other people with the same service; other services allow the user to make calls to any number (FCC Web site). VoIP traffic can be deployed on any IP network. It also includes the ones not having connection with the rest of the Internet such as local area network (LAN) (Wikepedia.org).
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More Results
Performance Analysis of Multimedia Traffic
a standard from the Internet Engineering Task Force (IETF) for initiating, modifying and terminating interactive user sessions that involve multimedia communications, such as audio, video, instant messaging, online games, simulation environments and virtual reality. SIP is a peer-to-peer protocol. Thus, it requires only a very simple (and highly scalable) core network with intelligence distributed to the network edge, embedded in endpoints or terminals.
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The Playout Control Management: An Issue for the IP Telephony Service Providers
The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. It can be used to create two-party, multiparty, or multicast sessions that include Internet telephone calls, multimedia distribution, and multimedia conferences. (cit. RFC 3261). SIP is designed to be independent of the underlying transport layer; it can run on TCP, UDP, or SCTP. It was originally designed by Henning Schulzrinne (Columbia University) and Mark Handley (UCL) starting in 1996. The latest version of the specification is RFC 3261 from the IETF SIP Working Group. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IMS architecture (from wikipedia.org).
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Open APIs and Protocols for Services and Applications in Telecoms
An application-layer control (signalling) protocol for creating, modifying, and terminating sessions with one or more participants.
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Minimizing Risk of Disputes Among Telecommunication Carriers With Blockchain Technologies
A signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video and messaging applications.
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