- Add backend endpoint to fetch ICE server credentials from Cloudflare
- Implement dynamic ICE server configuration in frontend
- Add fallback to public STUN servers when Cloudflare unavailable
- Create comprehensive test suite for WebRTC API endpoint
- Update environment configuration with Cloudflare TURN credentials
Backend changes:
- New route: GET /api/webrtc/ice-servers (authenticated)
- Fetches temporary credentials from Cloudflare API with 24h TTL
- Returns formatted ICE servers for RTCPeerConnection
- Graceful fallback to Google STUN servers on errors
Frontend changes:
- Remove hardcoded ICE servers from useWebRTC hook
- Fetch ICE servers dynamically from backend on mount
- Store servers in ref for peer connection initialization
- Add webrtcAPI service for backend communication
Tests:
- 9 comprehensive tests covering all scenarios
- 100% coverage for webrtc.js route
- Tests authentication, success, and all fallback scenarios
- Add CONNECTION_STATE (disconnected, connecting, connected, failed)
- Add SUGGESTION_TYPE (toBeRecorded, toRecord)
- Update useWebRTC.js to use CONNECTION_STATE
- Update MatchChatPage.jsx to use CONNECTION_STATE
- Update RecordingTab.jsx to use SUGGESTION_TYPE
- Add openrelay.metered.ca TURN servers for testing
- Support connections through symmetric NAT (mobile networks)
- Add TCP transport fallback for strict firewalls
- Enables P2P file transfer across different networks
Changed WebRTC config from localhost-only to STUN servers:
- Removed rtcConfigLocalhost (no longer needed)
- Using rtcConfig with Google STUN servers
- Enables NAT traversal for users on different networks
- Removed unnecessary iceCandidatePoolSize config
- Link sharing remains as fallback for blocked users
Implemented complete WebRTC peer-to-peer file transfer system for match chat:
**Core WebRTC Implementation:**
- Created useWebRTC hook with RTCPeerConnection and RTCDataChannel
- P2P file transfer with 16KB chunking for large files (tested up to 700MB)
- Real-time progress monitoring for sender and receiver
- Automatic file download on receiver side
- End-to-end encryption via DTLS (native WebRTC)
- ICE candidate exchange via Socket.IO signaling
- Support for host candidates (localhost testing)
**WebRTC Detection & User Experience:**
- Automatic WebRTC capability detection on page load
- Detects if ICE candidates can be generated (fails in Opera, privacy-focused browsers, VPNs)
- User-friendly warning component with fix suggestions
- Graceful degradation: disables WebRTC button when blocked
- Suggests alternative methods (video links via Google Drive/Dropbox)
**Socket.IO Improvements:**
- Fixed multiple socket instance creation issue
- Implemented socket instance reuse pattern
- Disabled React.StrictMode to prevent reconnection loops in development
**Technical Details:**
- RTCPeerConnection with configurable STUN servers (currently using localhost config)
- RTCDataChannel with ordered delivery
- Comprehensive logging for debugging (ICE gathering, connection states, signaling)
- Match room-based signaling relay via Socket.IO
- Authorization checks for all WebRTC signaling events
**Files Changed:**
- frontend/src/hooks/useWebRTC.js - Complete WebRTC implementation
- frontend/src/utils/webrtcDetection.js - WebRTC capability detection
- frontend/src/components/WebRTCWarning.jsx - User warning component
- frontend/src/pages/MatchChatPage.jsx - WebRTC integration
- frontend/src/services/socket.js - Socket instance reuse
- frontend/src/main.jsx - Disabled StrictMode for Socket.IO stability
**Testing:**
- ✅ Verified working in Chrome (ICE candidates generated)
- ✅ Tested with 700MB file transfer
- ✅ Detection working in Opera (shows warning when WebRTC blocked)
- ✅ P2P connection establishment and DataChannel opening
- ✅ File chunking and progress monitoring
**TODO:**
- Add STUN server configuration for production (NAT traversal)
- Consider server-based upload fallback for blocked users
Add complete WebRTC peer-to-peer file transfer functionality:
Backend changes:
- Add WebRTC signaling events to Socket.IO (offer, answer, ICE candidates)
- Implement authorization checks for match participants
- Add signaling relay between matched users
Frontend changes:
- Create useWebRTC hook for RTCPeerConnection management
- Implement RTCDataChannel with 16KB chunking for large files
- Add real-time progress monitoring for sender and receiver
- Implement automatic file download on receiver side
- Add connection state tracking and error handling
- Integrate WebRTC with MatchChatPage (replace mockup)
Configuration:
- Add Vite allowed hosts configuration via VITE_ALLOWED_HOSTS env var
- Support comma-separated host list or 'all' for development
- Add .env.example with configuration examples
- Update docker-compose.yml with default allowed hosts
Documentation:
- Add comprehensive WebRTC testing guide with troubleshooting
- Add quick test checklist for manual testing
- Document WebRTC flow, requirements, and success criteria
Features:
- End-to-end encrypted P2P transfer (DTLS)
- 16KB chunk size optimized for DataChannel
- Buffer management to prevent overflow
- Automatic connection establishment with 30s timeout
- Support for files of any size
- Real-time progress tracking
- Clean connection lifecycle management
