Session Initiation Protocol (SIP)

I need more information on Session Initiation Protocol because my boss asked me to keep a presentation about VoIP technology and SIP. During googling, I have found a website that seems to be suitable to my ideas because it is full of samples and comprehensive documentations as well.
Among others, I have found a good description about SIP protocol which I wish to share:
SIP can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions. These sessions include Internet telephone calls, multimedia distribution, multimedia conferences, instant messaging, file transfer and online games.

SIP is very much like Hypertext Transfer Protocol (HTTP) as it uses similar model to the HTTP request/response transaction model. It employs similar header structure and text-based formats. Session Description Protocol (SDP) defines the SIP message bodies for phone calls.

In this way SIP transactions consist of a client request that invokes a particular method or function on the server and at least one response. SIP reuses most of the header fields, encoding rules and status codes of HTTP, and providing a readable text-based format.
SIP is primarily employed for setting up and tearing down voice or video calls. However, in the process SIP needs to be adopted with other protocols since SIP is only involved in the signaling portion of a communication session. It assists in the connection of communication channels while the factual connection is created by SDP and RTP protocols. Therefore, voice and video stream communications in SIP applications are transmitted withReal-time Transport Protocol (RTP) while parameters like port numbers, protocols, codecs for these media streams are specified and negotiated with the use of Session Description Protocol (SDP).
The identification of users in a SIP network is realized with SIP addresses. The format of these SIP addresses is the following: sip:userID@domain.com. If users want to register they need to do so with the help of a registrar server and they also need to use their SIP addresses mentioned above. After this procedure this information will be available for the location server from the registrar server, if it is requested.

SIP users are also able to move between end systems since the location server makes it possible to locate the end user by protocols. However, it can occur that users login from more than one station or it is also possible that the location server does not have the accurate information. In these cases more than one address could be returned to the user. If a SIP proxy server mediates the request, all the returned addresses will be tested until the proxy server finds the user. If it is a SIP redirect server, it transmits all the returned addresses to the caller in the contact header field of the invitation response.
If you're interested in VoIP technology then I recommend you to take a look at Ozeki’s website. You can read the mentioned article about SIP protocol further on the following webpage:
http www voip-sip-sdk com/p_230-session-initiation-protocol-voip html

Regards,
Frankenberry

it is a very good protocol in todays life for calling abroad at less rate..