WebRTC

Browser and mobile softphone via SIP over WebSocket

PBX supports WebRTC calling, allowing users to make and receive calls directly from a web browser or mobile app without installing any SIP software.

Architecture

Browser (JsSIP/SIP.js)
  ↕ WSS (SIP over WebSocket)
Drachtio SIP Server
  ↕ RTP via rtpengine
PSTN / Other Extensions

WebRTC calls use:

SIP over WebSocket (RFC 7118) for signaling via wss://sip.pbx.dev:8443
DTLS-SRTP for encrypted media
ICE/STUN/TURN for NAT traversal
OPUS codec for high-quality audio, transcoded to G.711 for PSTN

ICE Servers

GET/v1/pbx/ice-servers

Get STUN/TURN server configuration

Requires:pbx:webrtc:read

curl -X GET "https://api.do.dev/v1/pbx/ice-servers" \
  -H "Authorization: Bearer do_live_your_key_here"

{
  "iceServers": [
    { "urls": "stun:turn.pbx.dev:3478" },
    {
      "urls": ["turn:turn.pbx.dev:3478", "turns:turn.pbx.dev:443"],
      "username": "1711460000:ext_101",
      "credential": "generated_credential_here"
    }
  ],
  "expiresAt": "2026-03-26T15:00:00Z"
}

TURN credentials are time-limited and auto-rotated. Request fresh credentials before each call.

Client Setup (JsSIP)

import JsSIP from "jssip";

const socket = new JsSIP.WebSocketInterface("wss://sip.pbx.dev:8443");
const ua = new JsSIP.UA({
  sockets: [socket],
  uri: "sip:101_a1b2c3@pbx.dev",
  password: "sip_password_here",
  display_name: "Front Desk",
});

ua.on("registered", () => console.log("Registered"));
ua.on("newRTCSession", (data) => {
  const session = data.session;
  // Handle incoming/outgoing call
});

ua.start();

Making a Call

const session = ua.call("sip:102@pbx.dev", {
  mediaConstraints: { audio: true, video: false },
  pcConfig: {
    iceServers: iceServers, // from /ice-servers endpoint
  },
});

Mobile App

The PBX mobile app (React Native) uses the same WebRTC infrastructure. It registers via WebSocket, receives push notifications for incoming calls, and renders a native calling UI.