AeThex-Connect/PHASE4-QUICK-START.md

Phase 4: Voice & Video Calls - Quick Reference

What Was Implemented

✅ Database Schema

• Extended calls table with WebRTC fields (type, sfu_room_id, recording_url, quality_stats)
• Extended call_participants table with media state and connection quality
• New turn_credentials table with auto-cleanup function

✅ Backend Services

• callService.js - Complete call lifecycle management (390+ lines)
  • initiateCall, answerCall, endCall
  • TURN credential generation with HMAC-SHA1
  • SFU room management (Mediasoup placeholder)
  • Media state updates
  • Call statistics and quality monitoring

✅ API Routes - 7 RESTful endpoints

• POST /api/calls/initiate - Start a call
• POST /api/calls/:id/answer - Answer call
• POST /api/calls/:id/reject - Reject call
• POST /api/calls/:id/end - End call
• PATCH /api/calls/:id/media - Update media state
• GET /api/calls/turn-credentials - Get TURN credentials
• GET /api/calls/:id - Get call details

✅ WebSocket Signaling

• call:offer, call:answer, call:ice-candidate
• call:incoming, call:ended
• call:participant-joined, call:participant-left
• call:media-state-changed

✅ Frontend WebRTC Manager (utils/webrtc.js)

• Peer connection management
• Local/remote stream handling
• Audio/video controls (toggle, mute)
• Screen sharing
• ICE candidate exchange
• Connection quality monitoring
• ~550 lines of WebRTC logic

✅ Call React Component (components/Call/)

• Full-featured call UI with controls
• Local and remote video display
• Call status indicators
• Media controls (mute, video, screen share)
• Connection quality indicator
• Responsive design with CSS

✅ Documentation

• PHASE4-CALLS.md - Complete technical documentation
• API endpoint specifications
• WebSocket event documentation
• TURN server setup guide (Coturn)
• Testing checklist
• Troubleshooting guide

Files Created

Backend

• src/backend/database/migrations/004_voice_video_calls.sql
• supabase/migrations/20260110140000_voice_video_calls.sql
• src/backend/services/callService.js
• src/backend/routes/callRoutes.js

Backend Updated

• src/backend/services/socketService.js - Added call signaling handlers
• src/backend/server.js - Registered call routes

Frontend

• src/frontend/utils/webrtc.js
• src/frontend/components/Call/index.jsx
• src/frontend/components/Call/Call.css

Documentation

• PHASE4-CALLS.md
• .env.example - Updated with TURN config

Quick Start

1. Run Database Migration

Apply the Supabase migration:

# Using Supabase CLI
supabase db push

# Or apply SQL manually in Supabase Dashboard
# File: supabase/migrations/20260110140000_voice_video_calls.sql

2. Configure Environment

Add to .env:

# TURN Server Configuration
TURN_SERVER_HOST=turn.example.com
TURN_SERVER_PORT=3478
TURN_SECRET=your-turn-secret-key
TURN_TTL=86400

For local development, you can skip TURN and use public STUN servers (already configured in webrtc.js).

3. Use Call Component

import Call from './components/Call';

function Chat() {
  const [inCall, setInCall] = useState(false);
  
  return (
    <>
      <button onClick={() => setInCall(true)}>
        Start Call
      </button>
      
      {inCall && (
        <Call
          socket={socket}
          conversationId={conversationId}
          participants={participants}
          onCallEnd={() => setInCall(false)}
        />
      )}
    </>
  );
}

Testing Without TURN Server

For development and testing behind the same network:

1. Use Public STUN servers (already configured)

   • stun:stun.l.google.com:19302
   • stun:stun1.l.google.com:19302

2. Test locally - Both users on same network work fine with STUN only

3. Behind NAT/Firewall - You'll need a TURN server for production

Architecture

User A (Browser)                    User B (Browser)
     |                                   |
     |--- HTTP: Initiate Call --------->|
     |<-- Socket: call:incoming --------|
     |                                   |
     |--- Socket: call:offer ---------->|
     |<-- Socket: call:answer -----------|
     |                                   |
     |<-- Socket: call:ice-candidate -->|
     |--- Socket: call:ice-candidate -->|
     |                                   |
     |<======= WebRTC P2P Media =======>|
     |        (Audio/Video Stream)       |

Media Controls API

// Toggle audio
webrtcManager.toggleAudio(false); // mute
webrtcManager.toggleAudio(true);  // unmute

// Toggle video
webrtcManager.toggleVideo(false); // camera off
webrtcManager.toggleVideo(true);  // camera on

// Screen sharing
await webrtcManager.startScreenShare();
webrtcManager.stopScreenShare();

// Get connection stats
const stats = await webrtcManager.getConnectionStats(userId);
console.log('RTT:', stats.connection.roundTripTime);
console.log('Bitrate:', stats.connection.availableOutgoingBitrate);

Supported Call Types

• Audio Call: Voice only, ~32-64 kbps per user
• Video Call: Audio + video, ~500kbps-2Mbps per user
• Screen Share: Replace camera with screen capture

Browser Support

• ✅ Chrome/Edge (Chromium) - Best support
• ✅ Firefox - Full support
• ✅ Safari - iOS 14.3+ required for WebRTC
• ❌ IE11 - Not supported (WebRTC required)

Next Steps

1. Apply Supabase migration for database schema
2. Test 1-on-1 calls with audio and video
3. Configure TURN server for production (optional for local dev)
4. Implement Mediasoup SFU for efficient group calls (future)
5. Add call history UI to display past calls

Known Limitations

• Group calls use mesh topology (all participants connect to each other)
  • Works well for 2-5 participants
  • Bandwidth intensive for 6+ participants
  • Mediasoup SFU implementation planned for better group call performance
• Call recording infrastructure in place but not implemented
• No call transfer or hold features yet

Production Checklist

• Set up TURN server (Coturn)
• Configure firewall rules for TURN
• Set TURN_SECRET environment variable
• Test calls across different networks
• Monitor bandwidth usage
• Set up call quality alerts
• Implement call analytics dashboard
• Add error tracking (Sentry, etc.)

Troubleshooting

No audio/video: Check browser permissions for camera/microphone

Connection fails:

• Verify both users are online
• Check Socket.io connection
• Test with public STUN servers first

Poor quality:

• Monitor connection quality indicator
• Check network bandwidth
• Reduce video resolution
• Switch to audio-only

Echo/Feedback:

• Use headphones
• Ensure echo cancellation is enabled
• Check for multiple audio sources

Support Resources

• Full documentation: PHASE4-CALLS.md
• WebRTC docs: https://webrtc.org
• Coturn setup: https://github.com/coturn/coturn
• Mediasoup: https://mediasoup.org

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Phase 4 Complete! Ready for testing and integration.