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Session Initiation Protocol (SIP)

Over the last few years, the VoIP community has adopted SIP as its signaling protocol of choice. SIP is an RFC standard (RFC 3621) from the Internet Engineering Task Force (IETF), the body responsible for administering and developing the mechanisms that comprise the Internet. The approach adopted by IETF is one of simplicity – specify only that which needs to be specified and SIP is a good example of this philosophy. It is purely a mechanism which initiates, modifies and terminates sessions, which means that it scales well. SIP is similar to HTTP and SMTP (the protocols that power the world wide web and e mail), so it works well alongside Internet applications. By employing SIP, voice becomes just another web application.

SIP is not meant to be the solution to all problems concerning voice over IP – this is a virtue, given the difficulties encountered by its all encompassing predecessor, H.323. SIP is designed to be flexible, which makes it ideal for the rapidly developing technology environment of the Internet.

SIP is designed to be a modular component of larger IP telephony solutions, interacting with other protocols which control features such as resource reservation and quality of service, but it goes beyond this by co-existing with legacy protocols, such as H.323 and MGCP.
SIP has also been adopted as the protocol of choice for video conferencing, messaging and collaboration systems. This is allowing an ever increasing array of equipment from diverse manufacturers to interoperate with eachother.

SIPcom’s offers the widest range of protocol support of any provider in the industry today. This includes not only, as the name suggests, SIP, but also the widely used MGCP and most importantly, the leading proprietary VOIP protocol, Cisco’s own SCCP (Skinny). This allows SIPcom to support the widest range handsets and features possible.