@aethex.os/webrtc

WebRTC manager for voice/video calls, screen sharing, and voice activity detection. Uses socket.io for signaling — drop it in next to your existing socket server.

Install

npm install @aethex.os/webrtc

Usage

Start a call

import { WebRTCManager } from '@aethex.os/webrtc';

const rtc = new WebRTCManager('https://your-signaling-server.com');

// Called when remote stream arrives
rtc.onRemoteTrack = (peerId, stream) => {
  videoElement.srcObject = stream;
};

// Start voice call
await rtc.initiateCall('peer_456', { audio: true, video: false });

Answer a call

Incoming calls are handled automatically via socket events. Attach handlers before connecting:

rtc.onRemoteTrack = (peerId, stream) => {
  remoteAudio.srcObject = stream;
};

rtc.onPeerDisconnected = (peerId) => {
  console.log(`${peerId} left`);
};

Controls

rtc.toggleAudio(false);   // mute
rtc.toggleVideo(false);   // camera off

await rtc.startScreenShare('peer_456');
await rtc.stopScreenShare('peer_456');

rtc.endCall('peer_456');

Voice Activity Detection

await rtc.initializeLocalStream({ audio: true, video: false });

setInterval(() => {
  if (rtc.isSpeaking()) {
    showSpeakingIndicator();
  }
}, 100);

Custom ICE servers

const rtc = new WebRTCManager('https://signaling.example.com', {
  iceServers: [
    { urls: 'stun:stun.example.com' },
    { urls: 'turn:turn.example.com', username: 'user', credential: 'pass' },
  ],
});

Cleanup

rtc.destroy(); // closes all connections, stops media, disconnects socket

Signaling protocol

Your socket server needs to handle these events:

| Event | Direction | Payload | |---|---|---| | call:offer | client → server → peer | { to, offer } | | call:answer | client → server → peer | { to, answer } | | call:ice-candidate | client → server → peer | { to, candidate } | | call:end | client → server → peer | { to } |

Part of the @aethex.os ecosystem

See @aethex.os/core for the full package list.