Session Initiation Protocol (SIP)

What is SIP?

The Session Initiation Protocol is a signaling protocol that enables the Voice Over Internet Protocol (VoIP) by defining the messages sent between endpoints and managing the actual elements of a call. SIP supports voice calls, video conferencing, instant messaging, and media distribution. Check this blog post for in-depth information on SIP architecture, messages, and methods!

SIP is just one method of deploying VoIP; its primary benefit is the fact that it provides a direct connection between private or local telephone systems (private branch exchanges, or PBX) and the public telephone network. This way, individuals and businesses don't need a legacy telephone line in order to connect.

Other VoIP deployment methods include the Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), and Session Description Protocol (SDP).

SIP versus VoIP

VoIP is a family of technologies that all support sending or receiving voice messages over the internet. SIP is an application protocol used to carry all forms of digital media, including voice messages—so SIP is a specific technology that supports VoIP calls.

What is an SIP trunk or trunking?

An SIP trunk is the interconnection between two domains of the Unified Communications network. By creating these interconnections, SIP trunking allows us to partition the network into public and private domains.

Public domains are generally managed by an internet telephone service provider (ITSPs), while private domains are connected to someone's personal server. ITSPs use SIP trunking to securely deliver telephone and streaming media services to users equipped with private branch exchange.