Audio/Video Calls

WebRTC-based audio and video calls with end-to-end encryption via Insertable Streams.

Initialize Call Module

The call module requires user identity keys. Initialize after authentication:

import { loadIdentity, deriveIdentity } from '@daomessage_sdk/sdk';

const stored = await loadIdentity();
const identity = deriveIdentity(stored.mnemonic);

client.initCalls({
  signingPrivKey: identity.signingKey.privateKey,
  signingPubKey: identity.signingKey.publicKey,
  myAliasId: stored.aliasId,
  alwaysRelay: false,  // true = force TURN relay (paid feature)
});

Make a Call

// Audio call
await client.calls.call('u87654321', { audio: true, video: false });

// Video call
await client.calls.call('u87654321', { audio: true, video: true });

Receive a Call

client.calls.onIncomingCall = (fromAlias) => {
  // Show incoming call UI
  showIncomingCallDialog(fromAlias);
};

// User accepts
await client.calls.answer();

// User rejects
client.calls.reject();

Handle State Changes

client.calls.onStateChange = (state) => {
  // state: 'idle' | 'calling' | 'ringing' | 'connecting' | 'connected' | 'hangup' | 'rejected' | 'ended'
  updateCallUI(state);
};

Access Media Streams

// Remote video/audio
client.calls.onRemoteStream = (stream) => {
  videoElement.srcObject = stream;
};

// Local preview
client.calls.onLocalStream = (stream) => {
  localVideoElement.srcObject = stream;
};

// Or get streams directly
const local = client.calls.getLocalStream();
const remote = client.calls.getRemoteStream();

Hang Up

client.calls.hangup();

Error Handling

client.calls.onError = (err) => {
  console.error('Call error:', err.message);
};

Call Flow

Caller                    Relay Server                   Callee
  │                           │                            │
  │── call_offer (SDP+sig) ──▶│── call_offer ──────────────▶│
  │                           │                            │
  │◀── call_answer (SDP) ─────│◀── call_answer (SDP+sig) ──│
  │                           │                            │
  │◀──────── ICE candidates ──│◀── ICE candidates ─────────│
  │── ICE candidates ────────▶│── ICE candidates ──────────▶│
  │                           │                            │
  │◀═══════ WebRTC P2P (or TURN relay) ═══════════════════▶│

E2EE for Calls

The SDK uses Insertable Streams (WebRTC Encoded Transform) to encrypt audio/video frames:

import { setupE2EETransform } from '@daomessage_sdk/sdk';

// Applied automatically by CallModule — no manual setup needed
// Each RTP frame is encrypted with AES-256-GCM before leaving the device

Important Notes

• Calls use the relay server's TURN credentials for NAT traversal
• ICE configuration is fetched from GET /api/v1/calls/ice-config
• Call signaling goes through WebSocket, not a separate channel
• The alwaysRelay option forces all traffic through TURN (prevents IP leak)