SIP over WebSocket
SIP over WebSocket is how browser-based softphones and WebRTC-SIP gateways work. If you're building a web-based communications app that bridges to PSTN or SIP trunks, this is the transport spec.
What It Defines
Defines the WebSocket transport binding for SIP. Enables browser-based SIP clients (WebRTC phone apps) to connect to SIP servers without a native SIP stack. Specifies the SIP/WS and SIP/WSS URI schemes, framing within WebSocket messages, and Via header handling for SIP/WS.
Canonical (Normative)
Convenient (Practical)
The canonical publication point for finalized RFCs. If a protocol is standardized as an RFC, the RFC Editor text is the normative final reference. Published by the IETF, IRTF, IAB, and independent stream.
Related Specs
SIP is the PSTN of the internet. Every VoIP platform (Twilio, Vonage, AWS Chime, Asterisk, FreeSWITCH) speaks SIP. Building any communications product requires understanding INVITE flows, registration, and codec negotiation via SDP.
SDP is the handshake language between any two endpoints setting up a call. WebRTC, SIP, and RTSP all use it. You can't debug codec mismatches, ICE failures, or video call setup without reading SDP.
Real-time features — chat, live dashboards, collaborative editing, notifications, trading UIs — run on WebSockets. Every serious app eventually needs them.