WebRTC

Web Real-Time Communications (WebRTC) — is an open-source technology for voice, text, and video communication in real-time through browsers over an IP network.

How it Works

• WebRTC is built into all modern browsers (Chrome, Firefox, Safari, Edge) and provides JavaScript APIs for developers.
• To establish a direct peer-to-peer (P2P) connection, especially between devices behind firewalls (NAT), it uses special servers:
  • STUN: Helps a device discover its public IP address.
  • TURN: Relays traffic if a direct P2P connection fails (using the ICE protocol).

1. Client selection — Use any WebRTC-compatible client.

   Client	Platform	Features
   JsSIP	Browser	No installation required, works in browser
   Linphone	Windows, macOS, Linux, iOS, Android	Multi-platform, open source
   Jitsi Meet	All platforms	For conferences and video calls
   Siperb	iOS	Optimized for iPhone/iPad

    

2. Data is available in the KOMPaaS Account in the ‘Telephony’ section → 'SIP settings' tab:

   • Login and Password.

   • Server: webrtc.kompaas.tech

   • Port (WebRTC WSS): 35060 

3. As a result, the connection string will look like this:

   wss://webrtc.kompaas.tech:35060
   

Example with JsSIP (open-source WebRTC softphone for browser).

import JsSIP from 'jssip';

// Specify WebRTC socket
const socket = new JsSIP.WebSocketInterface('wss://webrtc.kompaas.tech:35060');

// SIP account configuration
const configuration = {
  sockets  : [ socket ],
  uri      : 'sip:36XXXXXXXXX@webrtc.kompaas.tech', // login from personal account
  password : 'your_SIP_password'
};

// Create SIP agent
const ua = new JsSIP.UA(configuration);

// Start
ua.start();

// Register events
ua.on('connected', () => console.log('Connected to KOMPaaS'));
ua.on('registered', () => console.log('SIP registration successful'));
ua.on('newRTCSession', (data) => {
  console.log('New call:', data);
});