Taken from www.sipcenter.com:
The SessionInitiationProtocol (SIP) is a signalling 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 signalling 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 socialised 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.
-- TomO'Hern
Other resorces for SIP
Interesting artical comparing H.323 and SIP. --TomO'Hern
http://www.ietf.org/html.charters/sip-charter.html - IETF SIP Working Group
http://www.cs.columbia.edu/sip/
http://www.cs.columbia.edu/sip/implementations.html - SIP implementaions
http://www.linphone.org/?lang=us&rubrique=1 - Linux Webphone that uses SIP
http://www.sipforum.org - Slash site that is trying to promote awareness and provide inforamtion about SIP
http://iptel.org - dedicated to the promotion of InterNet Telephony.
http://www.hotsip.com/sip/tutorial/cartoon/index.html - A non-technical cartoon on how SIP works
http://www.vovida.org/ - a communications community site dedicated to providing a forum for open source software used in datacom and telecom environments.
http://www.cs.columbia.edu/~xiaotaow/sipc/ - SIP user agent (not free) for Windows 95/98/NT/2000/XP, Linux and Solaris. Does voice and video.
Apps:
http://www.cs.columbia.edu/IRT/cinema/ - CINEMA is a set of SIP-based Internet multimedia servers for creating an enterprise Internet telephony and multimedia system
http://sourceforge.net/projects/siphon/ - Siphon will ultimately be a Software Voice-Over-IP phone using the SIP protocol.
http://www.linphone.org/ - LinPhone is a SIP compatible web-phone, running under Linux.
http://www.cs.columbia.edu/sip/indigo.html - Fully Java based, supports audio, video and chat, available as both API and Application, advanced security scheme, web/e-mail integration. Indigo Software is developing a Presence and Instant Messaging system based entirely on SIP.
Phone:
http://www.sipphone.com - Same guy from MP3.com and Lindows fame is now marketing 2 SIP phones for $129
