feat: implement WebRTC P2P file transfer with detection and fallback

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
This commit is contained in:
Radosław Gierwiało
2025-11-15 16:12:02 +01:00
parent 664a2865b9
commit d23a12e5e3
6 changed files with 374 additions and 41 deletions

View File

@@ -0,0 +1,99 @@
import { AlertTriangle, X } from 'lucide-react';
import { useState } from 'react';
/**
* WebRTC Warning Component
*
* Shows a dismissible warning when WebRTC is not available or blocked
*/
const WebRTCWarning = ({ detection, onDismiss }) => {
const [isDismissed, setIsDismissed] = useState(false);
if (isDismissed) {
return null;
}
// Don't show if WebRTC is working fine
if (detection.supported && detection.hasIceCandidates) {
return null;
}
const handleDismiss = () => {
setIsDismissed(true);
if (onDismiss) {
onDismiss();
}
};
const getMessage = () => {
if (!detection.supported) {
return {
title: 'WebRTC Not Supported',
description: 'Your browser does not support WebRTC. P2P video transfer is disabled.',
suggestions: [
'Update your browser to the latest version',
'Try using Chrome, Firefox, or Edge',
],
};
}
if (!detection.hasIceCandidates) {
return {
title: 'WebRTC Blocked',
description: 'WebRTC is blocked by browser settings or extensions. P2P video transfer is disabled.',
suggestions: [
'Check browser privacy settings (e.g., Opera: Settings → Privacy → WebRTC)',
'Disable VPN extensions that block WebRTC',
'Try using Chrome or Firefox',
'Use incognito/private mode without extensions',
],
};
}
return {
title: 'WebRTC Error',
description: detection.error || 'Unknown WebRTC error',
suggestions: ['Try refreshing the page', 'Try a different browser'],
};
};
const { title, description, suggestions } = getMessage();
return (
<div className="bg-yellow-50 border border-yellow-200 rounded-lg p-4 mb-4">
<div className="flex items-start justify-between">
<div className="flex items-start space-x-3 flex-1">
<AlertTriangle className="w-5 h-5 text-yellow-600 flex-shrink-0 mt-0.5" />
<div className="flex-1">
<h3 className="text-sm font-semibold text-yellow-800 mb-1">
{title}
</h3>
<p className="text-sm text-yellow-700 mb-2">
{description}
</p>
<div className="text-sm text-yellow-700">
<p className="font-medium mb-1">How to fix:</p>
<ul className="list-disc list-inside space-y-1 ml-2">
{suggestions.map((suggestion, index) => (
<li key={index}>{suggestion}</li>
))}
</ul>
</div>
<p className="text-xs text-yellow-600 mt-2 italic">
You can still send video links via Google Drive, Dropbox, etc. using the "Link" button.
</p>
</div>
</div>
<button
onClick={handleDismiss}
className="text-yellow-400 hover:text-yellow-600 transition-colors flex-shrink-0 ml-2"
aria-label="Dismiss warning"
>
<X className="w-5 h-5" />
</button>
</div>
</div>
);
};
export default WebRTCWarning;

View File

@@ -8,6 +8,13 @@ const rtcConfig = {
{ urls: 'stun:stun1.l.google.com:19302' },
{ urls: 'stun:stun2.l.google.com:19302' },
],
iceTransportPolicy: 'all', // Use all candidates (host, srflx, relay)
iceCandidatePoolSize: 10, // Pre-gather candidates
};
// Alternative config for localhost testing (no STUN)
const rtcConfigLocalhost = {
iceServers: [], // No STUN - use only host candidates for localhost
};
// File chunk size (16KB recommended for WebRTC DataChannel)
@@ -29,6 +36,14 @@ export const useWebRTC = (matchId, userId) => {
const peerConnectionRef = useRef(null);
const dataChannelRef = useRef(null);
const socketRef = useRef(null);
const matchIdRef = useRef(matchId);
const userIdRef = useRef(userId);
// Update refs when props change
useEffect(() => {
matchIdRef.current = matchId;
userIdRef.current = userId;
}, [matchId, userId]);
// File transfer state
const fileTransferRef = useRef({
@@ -53,18 +68,37 @@ export const useWebRTC = (matchId, userId) => {
return peerConnectionRef.current;
}
const pc = new RTCPeerConnection(rtcConfig);
// Use localhost config for testing (no STUN servers)
const pc = new RTCPeerConnection(rtcConfigLocalhost);
peerConnectionRef.current = pc;
console.log('🔧 Using rtcConfigLocalhost (no STUN servers)');
// ICE candidate handler
pc.onicecandidate = (event) => {
if (event.candidate && socketRef.current) {
if (event.candidate) {
console.log('🧊 ICE candidate generated:', event.candidate.type);
if (socketRef.current) {
socketRef.current.emit('webrtc_ice_candidate', {
matchId,
matchId: matchIdRef.current,
candidate: event.candidate,
});
console.log('📤 Sent ICE candidate');
} else {
console.error('❌ Socket not available to send ICE candidate');
}
} else {
console.log('🧊 ICE gathering complete (candidate is null)');
}
};
// ICE gathering state handler
pc.onicegatheringstatechange = () => {
console.log('🧊 ICE gathering state:', pc.iceGatheringState);
};
// Signaling state handler
pc.onsignalingstatechange = () => {
console.log('📡 Signaling state:', pc.signalingState);
};
// Connection state change handler
@@ -84,8 +118,9 @@ export const useWebRTC = (matchId, userId) => {
};
console.log('✅ RTCPeerConnection initialized');
console.log('🔍 Initial states - Connection:', pc.connectionState, 'ICE:', pc.iceConnectionState, 'Signaling:', pc.signalingState);
return pc;
}, [matchId]);
}, []);
/**
* Create data channel for file transfer
@@ -200,9 +235,11 @@ export const useWebRTC = (matchId, userId) => {
const offer = await pc.createOffer();
await pc.setLocalDescription(offer);
console.log('✅ Local description set (offer). ICE gathering should start now...');
console.log('🔍 SDP has candidates:', pc.localDescription.sdp.includes('candidate:'));
socketRef.current.emit('webrtc_offer', {
matchId,
matchId: matchIdRef.current,
offer: pc.localDescription,
});
@@ -211,7 +248,7 @@ export const useWebRTC = (matchId, userId) => {
console.error('Failed to create offer:', error);
setConnectionState('failed');
}
}, [matchId, initializePeerConnection, createDataChannel]);
}, [initializePeerConnection, createDataChannel]);
/**
* Handle incoming offer and create answer
@@ -230,12 +267,15 @@ export const useWebRTC = (matchId, userId) => {
};
await pc.setRemoteDescription(new RTCSessionDescription(offer));
console.log('✅ Remote description set (offer)');
const answer = await pc.createAnswer();
await pc.setLocalDescription(answer);
console.log('✅ Local description set (answer). ICE gathering should start now...');
console.log('🔍 SDP has candidates:', pc.localDescription.sdp.includes('candidate:'));
socketRef.current.emit('webrtc_answer', {
matchId,
matchId: matchIdRef.current,
answer: pc.localDescription,
});
@@ -244,7 +284,7 @@ export const useWebRTC = (matchId, userId) => {
console.error('Failed to handle offer:', error);
setConnectionState('failed');
}
}, [matchId, initializePeerConnection, setupDataChannelHandlers]);
}, [initializePeerConnection, setupDataChannelHandlers]);
/**
* Handle incoming answer
@@ -258,7 +298,7 @@ export const useWebRTC = (matchId, userId) => {
}
await pc.setRemoteDescription(new RTCSessionDescription(answer));
console.log('✅ Remote description set');
console.log('✅ Remote description set (answer). ICE should connect now...');
} catch (error) {
console.error('Failed to handle answer:', error);
setConnectionState('failed');
@@ -385,36 +425,67 @@ export const useWebRTC = (matchId, userId) => {
socketRef.current = socket;
// Listen for WebRTC signaling events
socket.on('webrtc_offer', ({ from, offer }) => {
console.log('📥 Received WebRTC offer from:', from);
if (from !== userId) {
// Create stable handlers using current values from refs
const onOffer = ({ from, offer }) => {
console.log('📥 Received WebRTC offer from:', from, 'My userId:', userIdRef.current);
if (from !== userIdRef.current) {
handleOffer(offer);
} else {
console.log('⏭️ Ignoring offer from self');
}
});
};
socket.on('webrtc_answer', ({ from, answer }) => {
console.log('📥 Received WebRTC answer from:', from);
if (from !== userId) {
const onAnswer = ({ from, answer }) => {
console.log('📥 Received WebRTC answer from:', from, 'My userId:', userIdRef.current);
if (from !== userIdRef.current) {
handleAnswer(answer);
} else {
console.log('⏭️ Ignoring answer from self');
}
});
};
socket.on('webrtc_ice_candidate', ({ from, candidate }) => {
const onIceCandidate = ({ from, candidate }) => {
console.log('📥 Received ICE candidate from:', from);
if (from !== userId) {
if (from !== userIdRef.current) {
handleIceCandidate(candidate);
}
});
};
// Cleanup on unmount
// Attach listeners
const attachListeners = () => {
socket.off('webrtc_offer', onOffer);
socket.off('webrtc_answer', onAnswer);
socket.off('webrtc_ice_candidate', onIceCandidate);
socket.on('webrtc_offer', onOffer);
socket.on('webrtc_answer', onAnswer);
socket.on('webrtc_ice_candidate', onIceCandidate);
console.log('✅ WebRTC Socket.IO listeners attached (socketId:', socket.id, ')');
};
// Attach initially
attachListeners();
// Reattach on reconnect
const onReconnect = () => {
console.log('🔄 Socket reconnected - reattaching WebRTC listeners');
attachListeners();
};
socket.on('connect', onReconnect);
// Cleanup on unmount only
return () => {
socket.off('webrtc_offer');
socket.off('webrtc_answer');
socket.off('webrtc_ice_candidate');
console.log('🧹 Removing WebRTC Socket.IO listeners');
socket.off('connect', onReconnect);
socket.off('webrtc_offer', onOffer);
socket.off('webrtc_answer', onAnswer);
socket.off('webrtc_ice_candidate', onIceCandidate);
cleanupConnection();
};
}, [userId, handleOffer, handleAnswer, handleIceCandidate, cleanupConnection]);
// eslint-disable-next-line react-hooks/exhaustive-deps
}, [userId]); // Only re-run if userId changes
return {
connectionState,

View File

@@ -1,10 +1,9 @@
import { StrictMode } from 'react'
import { createRoot } from 'react-dom/client'
import './index.css'
import App from './App.jsx'
// StrictMode disabled for development - causes Socket.IO reconnection issues
// TODO: Re-enable in production or fix Socket.IO to handle double mounting
createRoot(document.getElementById('root')).render(
<StrictMode>
<App />
</StrictMode>,
)

View File

@@ -6,6 +6,8 @@ import { matchesAPI } from '../services/api';
import { Send, Video, Upload, X, Check, Link as LinkIcon, Loader2 } from 'lucide-react';
import { connectSocket, getSocket } from '../services/socket';
import { useWebRTC } from '../hooks/useWebRTC';
import { detectWebRTCSupport } from '../utils/webrtcDetection';
import WebRTCWarning from '../components/WebRTCWarning';
const MatchChatPage = () => {
const { slug } = useParams();
@@ -19,6 +21,7 @@ const MatchChatPage = () => {
const [showLinkInput, setShowLinkInput] = useState(false);
const [videoLink, setVideoLink] = useState('');
const [isConnected, setIsConnected] = useState(false);
const [webrtcDetection, setWebrtcDetection] = useState(null);
const messagesEndRef = useRef(null);
const fileInputRef = useRef(null);
@@ -33,6 +36,21 @@ const MatchChatPage = () => {
cleanupConnection,
} = useWebRTC(match?.id, user?.id);
// Detect WebRTC support on mount
useEffect(() => {
const runDetection = async () => {
const result = await detectWebRTCSupport();
setWebrtcDetection(result);
if (result.hasIceCandidates) {
console.log('✅ WebRTC detection: Working correctly');
} else {
console.warn('⚠️ WebRTC detection:', result.error);
}
};
runDetection();
}, []);
// Fetch match data
useEffect(() => {
const loadMatch = async () => {
@@ -88,13 +106,15 @@ const MatchChatPage = () => {
return;
}
// Socket event listeners
socket.on('connect', () => {
// Helper to join match room
const joinMatchRoom = () => {
setIsConnected(true);
// Join match room using numeric match ID for socket
socket.emit('join_match_room', { matchId: match.id });
console.log(`Joined match room ${match.id}`);
});
};
// Socket event listeners
socket.on('connect', joinMatchRoom);
socket.on('disconnect', () => {
setIsConnected(false);
@@ -105,9 +125,14 @@ const MatchChatPage = () => {
setMessages((prev) => [...prev, message]);
});
// Join immediately if already connected
if (socket.connected) {
joinMatchRoom();
}
// Cleanup
return () => {
socket.off('connect');
socket.off('connect', joinMatchRoom);
socket.off('disconnect');
socket.off('match_message');
};
@@ -315,6 +340,13 @@ const MatchChatPage = () => {
</div>
<div className="flex flex-col h-[calc(100vh-320px)]">
{/* WebRTC Warning */}
{webrtcDetection && (
<div className="p-4 pb-0">
<WebRTCWarning detection={webrtcDetection} />
</div>
)}
{/* Messages */}
<div className="flex-1 overflow-y-auto p-4 space-y-4">
{messages.length === 0 && (
@@ -472,7 +504,12 @@ const MatchChatPage = () => {
/>
<button
onClick={() => fileInputRef.current?.click()}
disabled={isTransferring || selectedFile}
disabled={isTransferring || selectedFile || (webrtcDetection && !webrtcDetection.hasIceCandidates)}
title={
webrtcDetection && !webrtcDetection.hasIceCandidates
? 'WebRTC is blocked - see warning above'
: 'Send video via P2P WebRTC'
}
className="flex-1 px-4 py-2 bg-primary-600 text-white rounded-md hover:bg-primary-700 transition-colors disabled:opacity-50 disabled:cursor-not-allowed flex items-center justify-center space-x-2"
>
<Video className="w-4 h-4" />

View File

@@ -12,10 +12,14 @@ export function connectSocket() {
return null;
}
if (socket && socket.connected) {
// Return existing socket instance (even if not connected yet)
// This prevents creating multiple socket instances during React StrictMode
if (socket) {
console.log('♻️ Reusing existing socket instance');
return socket;
}
console.log('🔌 Creating new socket instance');
socket = io(SOCKET_URL, {
auth: {
token,

View File

@@ -0,0 +1,123 @@
/**
* WebRTC Detection Utility
*
* Detects if WebRTC ICE candidate generation is working.
* This can fail due to:
* - Browser privacy settings (Opera, Brave)
* - VPN extensions blocking WebRTC
* - Corporate firewalls
* - Browser not supporting WebRTC
*/
/**
* Test if WebRTC can generate ICE candidates
* @returns {Promise<{supported: boolean, hasIceCandidates: boolean, error: string|null}>}
*/
export async function detectWebRTCSupport() {
const result = {
supported: false,
hasIceCandidates: false,
error: null,
};
// Check if RTCPeerConnection is available
if (!window.RTCPeerConnection) {
result.error = 'RTCPeerConnection not available in this browser';
return result;
}
result.supported = true;
// Test ICE candidate generation
try {
const pc = new RTCPeerConnection({
iceServers: [], // No STUN servers for quick test
});
let candidateReceived = false;
// Wait for ICE candidates
const candidatePromise = new Promise((resolve) => {
pc.onicecandidate = (event) => {
if (event.candidate) {
candidateReceived = true;
resolve(true);
} else if (event.candidate === null) {
// Gathering complete
resolve(candidateReceived);
}
};
// Timeout after 3 seconds
setTimeout(() => {
resolve(candidateReceived);
}, 3000);
});
// Create a data channel and offer to trigger ICE gathering
pc.createDataChannel('test');
const offer = await pc.createOffer();
await pc.setLocalDescription(offer);
// Wait for candidates
result.hasIceCandidates = await candidatePromise;
// Cleanup
pc.close();
if (!result.hasIceCandidates) {
result.error = 'ICE candidates not generated - WebRTC may be blocked by browser settings or extensions';
}
return result;
} catch (error) {
result.error = `WebRTC test failed: ${error.message}`;
return result;
}
}
/**
* Get user-friendly error message based on detection result
* @param {{supported: boolean, hasIceCandidates: boolean, error: string|null}} detection
* @returns {string}
*/
export function getWebRTCErrorMessage(detection) {
if (!detection.supported) {
return 'Your browser does not support WebRTC. Please use a modern browser like Chrome, Firefox, or Edge.';
}
if (!detection.hasIceCandidates) {
return 'WebRTC is blocked by your browser settings or extensions. P2P file transfer is disabled. Please check your privacy settings or try a different browser.';
}
if (detection.error) {
return `WebRTC error: ${detection.error}`;
}
return 'WebRTC is working correctly.';
}
/**
* Get suggestions to fix WebRTC issues
* @param {{supported: boolean, hasIceCandidates: boolean, error: string|null}} detection
* @returns {string[]}
*/
export function getWebRTCFixSuggestions(detection) {
if (!detection.supported) {
return [
'Update your browser to the latest version',
'Try using Chrome, Firefox, or Edge',
];
}
if (!detection.hasIceCandidates) {
return [
'Check browser privacy settings (e.g., Opera: Settings → Privacy → WebRTC)',
'Disable VPN extensions that block WebRTC',
'Try using Chrome or Firefox with default settings',
'Use incognito/private mode to test without extensions',
];
}
return [];
}