Real-time / Web

SIP over WebSocket (WSS)

SIP over Secure WebSocket (WSS) is a transport binding (RFC 7118) that lets browsers and JavaScript softphones speak SIP signaling natively. Combined with WebRTC media, it is how zero-install browser phones work.

Why WebSocket?

SIP traditionally runs on UDP/TCP/TLS. Browsers cannot open raw UDP or TCP sockets, so SIP-over-WebSocket was standardized to fit the browser's networking model. WSS = WebSocket Secure (over TLS).

The wire format is plain SIP messages framed by WebSocket's text-frame boundary — the same INVITE / 200 OK / BYE you would send over UDP, just wrapped in a WebSocket frame.

Connection example

// Using sip.js
const ua = new SIP.UserAgent({
  uri: SIP.UserAgent.makeURI("sip:user@didhub.io"),
  transportOptions: {
    server: "wss://wss.didhub.io",
  },
  authorizationUsername: "trunk_xxxxx",
  authorizationPassword: "<PASSWORD>",
});
await ua.start();

Common gotchas

Mixed-content: a page served over HTTPS cannot connect to ws:// — only wss://.
Origin allow-list: the WSS server should accept the browser's Origin header. DIDHub WSS accepts any origin for trunked customers.
Keepalives: NATs and proxies kill idle WebSockets. Set CRLF keepalives every 30s.