|
Mobile Computing & Wireless Hot Topics
SIP
- Session Initiation Protocol (in IP Context)
The Session Initiation Protocol (SIP) is a
signaling protocol used for establishing sessions in an IP network. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session. The ability to establish these sessions means that a host of innovative services become possible, such as voice-enriched e-commerce, web page click-to-dial, Instant Messaging with buddy lists, and IP Centrex services. Over the last couple of years, the Voice over IP community has adopted SIP as its protocol of choice for
signaling and the industry has focused a great deal of attention on this emerging standard. Currently, SIP is a draft from the Internet Engineering Task Force (IETF), the body responsible for administering and developing the mechanisms that comprise the Internet. SIP is still evolving and being extended as technology matures and SIP products are
socialized in the marketplace.
The IETF's philosophy is one of simplicity: specify only what you need to specify. SIP is very much of this mould; having been developed purely as a mechanism to establish sessions, it does not know about the details of a session, it just initiates, terminates and modifies sessions. This simplicity means that SIP scales, it is extensible, and it sits comfortably in different architectures and deployment scenarios. SIP is a request-response protocol that closely resembles two other Internet protocols, HTTP and SMTP (the protocols that power the world wide web and email); consequently, SIP sits comfortably alongside Internet applications. Using SIP, telephony becomes another web application and integrates easily into other Internet services. SIP is a simple toolkit that service providers can use to build converged voice and multimedia services. 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), to authenticate users (RADIUS, DIAMETER), to provide directories (LDAP), to be able to guarantee voice quality (RSVP, YESSIR) and to inter-work with today's telephone network. Here we will only cover SIP.
Original source: http://www.sipcenter.com
SIP-Related White Papers : Go to
SIP Trade Forum and SIPCenter.Com websites for the following papers:
SIP Server and Related Product
Vendors - Go to SIPCenter.Com
vendor pages.
More Resources for SIP-related
Information
|
