Real-time voice along with video communication can be attained with the internet standard SIP (Session Initiation Protocol). That was initiated by the IETF (Internet Engineering Task Force) and you can find it in published form from RFC 3261. As for live communications, internet protocol “SIP” is used for establishing voice and video calls and within an IP network, one or more than one participant can whether create, alter, or end sessions with the help of this signaling protocol. It is very important in explanation of the functionality of the VoIP technology as one of the Voice over IP protocols. But at this point of learning about SIP, first you have to understand the term “session” within a communicating network. Actually, it is a clear-cut two-way phone call procedure. But in case of multi-media conference session, it can be consisted on loads of participating persons.

Session initiation protocol working group is at the present in the authority to make improvements in it and to maintain the standard of this text based SIP such as upholding its function of starting interactive communication session for the users. Moreover, this group is doing their best for maintaining the basic architecture of SIP model by using on hand internet protocols so to retain the architecture and the simplicity of model etc.

Peep to peer protocol SIP is required just a simple but scalable (network ability to deal with growing amount of exertion in a proper and capable manner) central network along with intellect distributed to the edge of network, fixed in the endpoints (terminating hardware of software devices). Signaling protocol “Session Initiation Protocol” (SIP) is designed especially for a series of services such as: internet conferencing, instant messaging, IP telephony, presence, voice contact, video communication, data alliance, live gaming, distribution of application and much more tasks are possible with this protocol assistance. “SIP” acts in the same way for the real-time unicast or multicast communication as HTTP protocol takes steps for the web.

In addition to this, SIP forking is referred to the course of dividing a particular SIP call into several SIP endpoints. With this powerful SIP feature, a sole call can ring on a lot of endpoints simultaneously e.g. with it, you can at the same time ring your deskphone as well as Android phone sip. Most of all, in both devices cases no rules for forwarding are required in order to make them ring. Best example of SIP forking can be as: an office device with this protocol can let the secretary to reply all calls to the phone extension of boss whenever he/she is out of office. Such SIP telephone system is featured with secure, considerable cost-saving and improved user’s mobility and efficiency functions facilities.

In short, SIP was originally developed within the IETF (Internet Engineering Task Force) MMUSIC (multiparty multimedia session control) group. But a number of regularities, associations and other groups are considerably using SIP such as: IETF PINT working group, IMTC, ETSI Tiphon, PacketCable DCS etc. The Application layer SIP protocol is designed to be self regulating of the transport layer. It can be run over the TCP, UDP, or SCTP and it is incorporated several features of the HTTP and the SMTP.