SIP means Session Initiation Protocol developed and designed by IETF (Internet Engineering Task Force) to establish and control sessions with one or more devices like telephones, computers, PDA's etc. for example VoIP (Voice over IP) call and today's Instant Messaging are all forms of SIP based communication. Online gaming also falls under same category where gamers can connect to a gaming server using any SIP enabled device over internet to play and enjoy.

SIP came after H323 protocol which was specifically designed for VoIP but later SIP became device independent that people around world adopted it as there primary communications protocol. Infact H323 was developed by the ITU (International Telecommunication Union) as a standard for precisely Videoconferencing device running over ISDN (Integrated Services Digital Network) lines which was a limitation in itself and no every company was using H323 to fulfill their VoIP requirements.

VoIP call are all routed over SIP which is all a device needs in order to communicate with another SIP enabled device no matter if they match in brand, or structure. As long as they can be linked over SIP, they can use there device embedded features to sort out a way of VoIP communication as this is what SIP does, it initiates the session and then controls it. Although, the actual communication is performed by other protocols which communicates over SIP, like Audio, Video, Remote desktop based on RTP (Real Time Protocol) and SDP (Session Description Protocol) etc. Make sure you have broadband internet connection in place.

RTP: is the only protocol which is associated with several other codec's which can convert the Audio codec into computer formatted data which is then used as a interpreter between two user sessions based on SIP. RTP can also carry real-time multimedia data which can be either streamed or direct play.

SDP: it defines the exchange of data between two SIP enabled devices by picking up the right codec.

A user agent or a client initiates the SIP session with another SIP enabled user agent over a router network where both could acknowledge there session ID's and based on the application that will be used during there session, RTP prepares itself to open a specific codec which then goes to SDP. SDP contacts the destination SDP and request for a specific codec exchange which interns contact destination RTP. That's how a session is build upon successful data link between two SIP enabled devices irrespective of there make and model.

SIP also makes sure the following steps are working along the path of communication

Routing and Name resolution: SIP enabled clients are identified by SIP URI (Uniform Resource Locators) which are nothing but unique identity of each device for example a phone number or a MSN sign-in name like user@msn.com or user1@hotmail.com etc. In order to get the exact identify of this SIP URI, DNS name resolution is performed which get the exact IP address of the URI and then find the shortest path to reach the destination IP address and build a route which remains persistent until the conversation is active.

Capability negotiation: with its sensing feature, SIP can recognize the connection strength and based on that it can boost up the connection. Say in case of audio call, depending upon the bandwidth the call may get high or low quality. This capability has to be negotiated before the call actually builds up over SIP.

Participant management: SIP session contains two parties or UAC (User Agent Client) and any additional participant can be added to an existing SIP session just like a videoconferencing and conference VoIP Call.

Therefore, save up your home telephone bill and use SIP based VoIP for calling the world.