SIP stands for Session Initiation Protocol.
SIP is a widely used signaling protocol in the VoIP industry, heavily used in telecommunications for establishing, modifying, and terminating voice and video calls. It is defined in IETF RFC 3261.
SIP and WebRTC
WebRTC does not use SIP natively – it leaves signaling to the application developer. However, SIP plays an important role in the WebRTC ecosystem:
SIP over WebSocket: To bridge WebRTC with SIP infrastructure, SIP over WebSocket (RFC 7118) was developed. This allows a web browser to run a SIP user agent, connecting directly to SIP infrastructure while using WebRTC for media. Libraries like JsSIP and SIP.js implement this in JavaScript.
WebRTC-SIP gateways: SBCs and gateways translate between WebRTC signaling and SIP, enabling WebRTC endpoints to communicate with SIP phones, PSTN networks, and enterprise PBX systems.
SIP vs WebRTC signaling
While SIP is feature-rich, many WebRTC applications choose simpler signaling approaches (plain WebSocket with JSON messages) for pure WebRTC-to-WebRTC communication, only using SIP when interoperability with legacy telephony is needed.
