WebRTCReal-timeReact Native

Master React Native WebRTC: Build Real-Time Video & Audio Apps Like a Pro 2025

By Viewlytics Team16 min readMay 30, 2025

Ready to build the next Zoom, WhatsApp, or Discord? React Native WebRTC is your gateway to creating powerful real-time video calling, audio streaming, and peer-to-peer communication apps. This comprehensive guide covers everything from basic setup to advanced optimization techniques for production-ready WebRTC applications.

1. Understanding React Native WebRTC

React Native WebRTC brings the power of real-time communication to mobile apps, enabling peer-to-peer video calls, audio streaming, screen sharing, and data channels. Built on top of Google's WebRTC library, it provides native performance with cross-platform compatibility.

What Makes WebRTC Special?

Peer-to-Peer Communication: Direct connection between devices without intermediary servers
Low Latency: Real-time audio/video with minimal delay
NAT Traversal: Works behind firewalls and routers using STUN/TURN servers
Adaptive Quality: Automatic adjustment based on network conditions
Multiple Media Types: Audio, video, screen capture, and arbitrary data

Current WebRTC Revision & Platform Support

React Native WebRTC currently uses WebRTC M124 revision with support for:

Android: armeabi-v7a, arm64-v8a, x86, x86_64
iOS: arm64, x86_64
tvOS: arm64
Unified Plan: Modern SDP format (Plan B deprecated)
Simulcast: Multiple video quality streams

Use Cases for React Native WebRTC

Video calling applications (Zoom, FaceTime alternatives)
Voice chat and audio streaming platforms
Live streaming and broadcasting apps
Screen sharing and remote desktop solutions
Gaming with real-time voice communication
Telehealth and remote consultation platforms
Educational apps with live interaction

2. Installation and Platform Setup

Basic Installation

Install the React Native WebRTC package:

# Using yarn
yarn add react-native-webrtc

# Using npm
npm install react-native-webrtc

# Using pnpm
pnpm install react-native-webrtc

iOS Configuration

Add camera and microphone permissions to ios/YourApp/Info.plist:

<key>NSCameraUsageDescription</key>
<string>This app needs access to camera for video calls</string>
<key>NSMicrophoneUsageDescription</key>
<string>This app needs access to microphone for voice calls</string>

Run pod install:

cd ios && pod install

Android Configuration

Add permissions to android/app/src/main/AndroidManifest.xml:

<uses-permission android:name="android.permission.CAMERA" />
<uses-permission android:name="android.permission.RECORD_AUDIO" />
<uses-permission android:name="android.permission.INTERNET" />
<uses-permission android:name="android.permission.ACCESS_NETWORK_STATE" />
<uses-permission android:name="android.permission.CHANGE_NETWORK_STATE" />
<uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />

Add ProGuard rules if using code obfuscation in android/app/proguard-rules.pro:

-keep class org.webrtc.** { *; }
-dontwarn org.webrtc.**
-keep class com.oney.WebRTCModule.** { *; }

3. Core WebRTC Concepts and Implementation

Basic WebRTC Flow

Understanding the WebRTC connection process:

Get User Media: Access camera and microphone
Create Peer Connection: Initialize WebRTC peer connection
Create Offer/Answer: Exchange session descriptions
Exchange ICE Candidates: Share network connection info
Establish Connection: Start real-time communication

Getting User Media

Request access to camera and microphone:

import { mediaDevices } from 'react-native-webrtc';

const getUserMedia = async () => {
  try {
    const stream = await mediaDevices.getUserMedia({
      audio: true,
      video: {
        width: { min: 640, ideal: 1280, max: 1920 },
        height: { min: 360, ideal: 720, max: 1080 },
        frameRate: { min: 15, ideal: 30, max: 60 },
        facingMode: 'user', // 'user' for front camera, 'environment' for back
      },
    });
    
    console.log('Local stream obtained:', stream);
    return stream;
  } catch (error) {
    console.error('Error accessing media devices:', error);
    throw error;
  }
};

Creating Peer Connection

Initialize WebRTC peer connection with STUN/TURN servers:

import { RTCPeerConnection } from 'react-native-webrtc';

const createPeerConnection = () => {
  const configuration = {
    iceServers: [
      { urls: 'stun:stun.l.google.com:19302' },
      { urls: 'stun:stun1.l.google.com:19302' },
      {
        urls: 'turn:your-turn-server.com:3478',
        username: 'your-username',
        credential: 'your-password',
      },
    ],
    iceCandidatePoolSize: 10,
  };

  const peerConnection = new RTCPeerConnection(configuration);

  // Handle ICE candidates
  peerConnection.addEventListener('icecandidate', (event) => {
    if (event.candidate) {
      // Send candidate to remote peer via signaling server
      sendIceCandidate(event.candidate);
    }
  });

  // Handle remote stream
  peerConnection.addEventListener('addstream', (event) => {
    console.log('Remote stream received:', event.stream);
    setRemoteStream(event.stream);
  });

  return peerConnection;
};

Signaling Server Implementation

Simple WebSocket-based signaling server for exchanging offers, answers, and ICE candidates:

class SignalingService {
  constructor(url) {
    this.ws = new WebSocket(url);
    this.listeners = {};
    
    this.ws.onmessage = (event) => {
      const data = JSON.parse(event.data);
      this.emit(data.type, data);
    };
  }

  on(event, callback) {
    if (!this.listeners[event]) {
      this.listeners[event] = [];
    }
    this.listeners[event].push(callback);
  }

  emit(event, data) {
    if (this.listeners[event]) {
      this.listeners[event].forEach(callback => callback(data));
    }
  }

  send(type, data) {
    this.ws.send(JSON.stringify({ type, ...data }));
  }

  sendOffer(offer, roomId) {
    this.send('offer', { offer, roomId });
  }

  sendAnswer(answer, roomId) {
    this.send('answer', { answer, roomId });
  }

  sendIceCandidate(candidate, roomId) {
    this.send('ice-candidate', { candidate, roomId });
  }
}

4. Building a Complete Video Calling Component

Video Call Hook

Custom hook to manage WebRTC video call state:

import { useState, useEffect, useRef } from 'react';
import { RTCPeerConnection, RTCView, mediaDevices } from 'react-native-webrtc';

export const useVideoCall = (roomId, signalingService) => {
  const [localStream, setLocalStream] = useState(null);
  const [remoteStream, setRemoteStream] = useState(null);
  const [isConnected, setIsConnected] = useState(false);
  const [isMuted, setIsMuted] = useState(false);
  const [isVideoEnabled, setIsVideoEnabled] = useState(true);
  
  const peerConnection = useRef(null);

  useEffect(() => {
    initializeCall();
    setupSignalingListeners();
    
    return () => {
      cleanup();
    };
  }, [roomId]);

  const initializeCall = async () => {
    try {
      // Get user media
      const stream = await getUserMedia();
      setLocalStream(stream);

      // Create peer connection
      peerConnection.current = createPeerConnection();
      
      // Add local stream to peer connection
      stream.getTracks().forEach(track => {
        peerConnection.current.addTrack(track, stream);
      });

    } catch (error) {
      console.error('Failed to initialize call:', error);
    }
  };

  const setupSignalingListeners = () => {
    signalingService.on('offer', handleOffer);
    signalingService.on('answer', handleAnswer);
    signalingService.on('ice-candidate', handleIceCandidate);
  };

  const createOffer = async () => {
    try {
      const offer = await peerConnection.current.createOffer();
      await peerConnection.current.setLocalDescription(offer);
      signalingService.sendOffer(offer, roomId);
    } catch (error) {
      console.error('Error creating offer:', error);
    }
  };

  const handleOffer = async ({ offer }) => {
    try {
      await peerConnection.current.setRemoteDescription(offer);
      const answer = await peerConnection.current.createAnswer();
      await peerConnection.current.setLocalDescription(answer);
      signalingService.sendAnswer(answer, roomId);
    } catch (error) {
      console.error('Error handling offer:', error);
    }
  };

  const handleAnswer = async ({ answer }) => {
    try {
      await peerConnection.current.setRemoteDescription(answer);
      setIsConnected(true);
    } catch (error) {
      console.error('Error handling answer:', error);
    }
  };

  const handleIceCandidate = async ({ candidate }) => {
    try {
      await peerConnection.current.addIceCandidate(candidate);
    } catch (error) {
      console.error('Error adding ICE candidate:', error);
    }
  };

  const toggleMute = () => {
    if (localStream) {
      localStream.getAudioTracks().forEach(track => {
        track.enabled = isMuted;
      });
      setIsMuted(!isMuted);
    }
  };

  const toggleVideo = () => {
    if (localStream) {
      localStream.getVideoTracks().forEach(track => {
        track.enabled = !isVideoEnabled;
      });
      setIsVideoEnabled(!isVideoEnabled);
    }
  };

  const endCall = () => {
    cleanup();
    // Navigate away or reset state
  };

  const cleanup = () => {
    if (localStream) {
      localStream.getTracks().forEach(track => track.stop());
    }
    if (peerConnection.current) {
      peerConnection.current.close();
    }
    setLocalStream(null);
    setRemoteStream(null);
    setIsConnected(false);
  };

  return {
    localStream,
    remoteStream,
    isConnected,
    isMuted,
    isVideoEnabled,
    createOffer,
    toggleMute,
    toggleVideo,
    endCall,
  };
};

Video Call UI Component

Complete video calling interface:

import React from 'react';
import { View, TouchableOpacity, Text, StyleSheet } from 'react-native';
import { RTCView } from 'react-native-webrtc';
import Icon from 'react-native-vector-icons/MaterialIcons';

const VideoCallScreen = ({ route }) => {
  const { roomId } = route.params;
  const signalingService = new SignalingService('ws://your-server.com');
  
  const {
    localStream,
    remoteStream,
    isConnected,
    isMuted,
    isVideoEnabled,
    createOffer,
    toggleMute,
    toggleVideo,
    endCall,
  } = useVideoCall(roomId, signalingService);

  return (
    <View style={styles.container}>
      {/* Remote Video */}
      {remoteStream ? (
        <RTCView
          style={styles.remoteVideo}
          streamURL={remoteStream.toURL()}
          objectFit="cover"
        />
      ) : (
        <View style={styles.waitingContainer}>
          <Text style={styles.waitingText}>
            {isConnected ? 'Connected' : 'Waiting for remote user...'}
          </Text>
        </View>
      )}

      {/* Local Video */}
      {localStream && (
        <RTCView
          style={styles.localVideo}
          streamURL={localStream.toURL()}
          objectFit="cover"
          mirror={true}
        />
      )}

      {/* Controls */}
      <View style={styles.controls}>
        <TouchableOpacity
          style={[styles.controlButton, isMuted && styles.mutedButton]}
          onPress={toggleMute}
        >
          <Icon 
            name={isMuted ? 'mic-off' : 'mic'} 
            size={24} 
            color="white" 
          />
        </TouchableOpacity>

        <TouchableOpacity
          style={[styles.controlButton, !isVideoEnabled && styles.disabledButton]}
          onPress={toggleVideo}
        >
          <Icon 
            name={isVideoEnabled ? 'videocam' : 'videocam-off'} 
            size={24} 
            color="white" 
          />
        </TouchableOpacity>

        <TouchableOpacity
          style={[styles.controlButton, styles.endCallButton]}
          onPress={endCall}
        >
          <Icon name="call-end" size={24} color="white" />
        </TouchableOpacity>

        {!remoteStream && (
          <TouchableOpacity
            style={[styles.controlButton, styles.callButton]}
            onPress={createOffer}
          >
            <Icon name="call" size={24} color="white" />
          </TouchableOpacity>
        )}
      </View>
    </View>
  );
};

const styles = StyleSheet.create({
  container: {
    flex: 1,
    backgroundColor: '#000',
  },
  remoteVideo: {
    flex: 1,
    backgroundColor: '#000',
  },
  localVideo: {
    position: 'absolute',
    top: 50,
    right: 20,
    width: 120,
    height: 160,
    borderRadius: 8,
    borderWidth: 2,
    borderColor: '#fff',
  },
  waitingContainer: {
    flex: 1,
    justifyContent: 'center',
    alignItems: 'center',
  },
  waitingText: {
    color: '#fff',
    fontSize: 18,
  },
  controls: {
    position: 'absolute',
    bottom: 50,
    left: 0,
    right: 0,
    flexDirection: 'row',
    justifyContent: 'center',
    alignItems: 'center',
  },
  controlButton: {
    width: 60,
    height: 60,
    borderRadius: 30,
    backgroundColor: 'rgba(255, 255, 255, 0.3)',
    justifyContent: 'center',
    alignItems: 'center',
    marginHorizontal: 10,
  },
  mutedButton: {
    backgroundColor: '#ff4444',
  },
  disabledButton: {
    backgroundColor: '#666',
  },
  endCallButton: {
    backgroundColor: '#ff4444',
  },
  callButton: {
    backgroundColor: '#44ff44',
  },
});

5. Advanced Features and Optimization

Screen Sharing Implementation

Add screen sharing capability to your app:

import { mediaDevices } from 'react-native-webrtc';

const useScreenShare = (peerConnection) => {
  const [isScreenSharing, setIsScreenSharing] = useState(false);
  const [screenStream, setScreenStream] = useState(null);

  const startScreenShare = async () => {
    try {
      const stream = await mediaDevices.getDisplayMedia({
        video: true,
        audio: true, // Include system audio if supported
      });

      // Replace video track in peer connection
      const videoTrack = stream.getVideoTracks()[0];
      const sender = peerConnection.current
        .getSenders()
        .find(s => s.track && s.track.kind === 'video');
      
      if (sender) {
        await sender.replaceTrack(videoTrack);
      }

      setScreenStream(stream);
      setIsScreenSharing(true);

      // Handle screen share ending
      videoTrack.addEventListener('ended', stopScreenShare);
    } catch (error) {
      console.error('Error starting screen share:', error);
    }
  };

  const stopScreenShare = async () => {
    if (screenStream) {
      screenStream.getTracks().forEach(track => track.stop());
      
      // Get camera stream back
      const cameraStream = await mediaDevices.getUserMedia({
        video: true,
        audio: false,
      });
      
      // Replace with camera track
      const videoTrack = cameraStream.getVideoTracks()[0];
      const sender = peerConnection.current
        .getSenders()
        .find(s => s.track && s.track.kind === 'video');
      
      if (sender) {
        await sender.replaceTrack(videoTrack);
      }

      setScreenStream(null);
      setIsScreenSharing(false);
    }
  };

  return {
    isScreenSharing,
    startScreenShare,
    stopScreenShare,
  };
};

Data Channels for Real-time Messaging

Implement peer-to-peer text messaging:

const useDataChannel = (peerConnection) => {
  const [dataChannel, setDataChannel] = useState(null);
  const [messages, setMessages] = useState([]);
  const [isChannelOpen, setIsChannelOpen] = useState(false);

  useEffect(() => {
    if (peerConnection.current) {
      setupDataChannel();
    }
  }, [peerConnection.current]);

  const setupDataChannel = () => {
    // Create data channel
    const channel = peerConnection.current.createDataChannel('messages', {
      ordered: true,
    });

    channel.addEventListener('open', () => {
      console.log('Data channel opened');
      setIsChannelOpen(true);
    });

    channel.addEventListener('close', () => {
      console.log('Data channel closed');
      setIsChannelOpen(false);
    });

    channel.addEventListener('message', (event) => {
      const message = JSON.parse(event.data);
      setMessages(prev => [...prev, { ...message, isOwn: false }]);
    });

    setDataChannel(channel);

    // Handle incoming data channels
    peerConnection.current.addEventListener('datachannel', (event) => {
      const incomingChannel = event.channel;
      
      incomingChannel.addEventListener('message', (event) => {
        const message = JSON.parse(event.data);
        setMessages(prev => [...prev, { ...message, isOwn: false }]);
      });
    });
  };

  const sendMessage = (text) => {
    if (dataChannel && isChannelOpen) {
      const message = {
        id: Date.now(),
        text,
        timestamp: new Date().toISOString(),
        sender: 'local',
      };

      dataChannel.send(JSON.stringify(message));
      setMessages(prev => [...prev, { ...message, isOwn: true }]);
    }
  };

  return {
    messages,
    sendMessage,
    isChannelOpen,
  };
};

Adaptive Video Quality

Implement adaptive bitrate based on network conditions:

const useAdaptiveQuality = (peerConnection) => {
  const [networkQuality, setNetworkQuality] = useState('good');
  const [currentBitrate, setCurrentBitrate] = useState(1000000); // 1 Mbps

  useEffect(() => {
    const interval = setInterval(checkNetworkQuality, 5000);
    return () => clearInterval(interval);
  }, []);

  const checkNetworkQuality = async () => {
    if (!peerConnection.current) return;

    try {
      const stats = await peerConnection.current.getStats();
      let bytesReceived = 0;
      let packetsLost = 0;
      let jitter = 0;

      stats.forEach(report => {
        if (report.type === 'inbound-rtp' && report.mediaType === 'video') {
          bytesReceived = report.bytesReceived || 0;
          packetsLost = report.packetsLost || 0;
          jitter = report.jitter || 0;
        }
      });

      // Calculate network quality
      const quality = calculateQuality(packetsLost, jitter);
      setNetworkQuality(quality);

      // Adjust bitrate based on quality
      adjustBitrate(quality);
    } catch (error) {
      console.error('Error checking network quality:', error);
    }
  };

  const calculateQuality = (packetsLost, jitter) => {
    if (packetsLost > 5 || jitter > 50) return 'poor';
    if (packetsLost > 2 || jitter > 30) return 'fair';
    return 'good';
  };

  const adjustBitrate = async (quality) => {
    const bitrateMap = {
      poor: 300000,   // 300 Kbps
      fair: 600000,   // 600 Kbps
      good: 1000000,  // 1 Mbps
    };

    const newBitrate = bitrateMap[quality];
    if (newBitrate !== currentBitrate) {
      await setBitrate(newBitrate);
      setCurrentBitrate(newBitrate);
    }
  };

  const setBitrate = async (bitrate) => {
    const sender = peerConnection.current
      .getSenders()
      .find(s => s.track && s.track.kind === 'video');

    if (sender) {
      const params = sender.getParameters();
      params.encodings[0].maxBitrate = bitrate;
      await sender.setParameters(params);
    }
  };

  return {
    networkQuality,
    currentBitrate,
  };
};

6. Performance Optimization and Best Practices

Memory Management

Properly manage WebRTC resources to prevent memory leaks:

const useWebRTCCleanup = () => {
  const streamsRef = useRef([]);
  const peerConnectionsRef = useRef([]);

  const addStream = (stream) => {
    streamsRef.current.push(stream);
  };

  const addPeerConnection = (pc) => {
    peerConnectionsRef.current.push(pc);
  };

  const cleanup = useCallback(() => {
    // Stop all media tracks
    streamsRef.current.forEach(stream => {
      stream.getTracks().forEach(track => {
        track.stop();
      });
    });

    // Close all peer connections
    peerConnectionsRef.current.forEach(pc => {
      pc.close();
    });

    // Clear refs
    streamsRef.current = [];
    peerConnectionsRef.current = [];
  }, []);

  // Cleanup on unmount
  useEffect(() => {
    return cleanup;
  }, [cleanup]);

  return {
    addStream,
    addPeerConnection,
    cleanup,
  };
};

Error Handling and Recovery

Implement robust error handling:

const useWebRTCErrorHandling = () => {
  const [errors, setErrors] = useState([]);

  const handleWebRTCError = (error, context) => {
    const errorInfo = {
      id: Date.now(),
      message: error.message,
      context,
      timestamp: new Date().toISOString(),
    };

    setErrors(prev => [...prev.slice(-9), errorInfo]); // Keep last 10 errors

    // Log to analytics service
    console.error(`WebRTC Error in ${context}:`, error);

    // Handle specific error types
    switch (error.name) {
      case 'NotAllowedError':
        // User denied permissions
        return 'Please grant camera and microphone permissions';
      
      case 'NotFoundError':
        // No media devices found
        return 'No camera or microphone found';
      
      case 'OverconstrainedError':
        // Constraints cannot be satisfied
        return 'Camera settings not supported';
      
      case 'NotReadableError':
        // Hardware error
        return 'Camera/microphone hardware error';
      
      default:
        return 'An error occurred with the video call';
    }
  };

  const retryConnection = async (peerConnection, createOfferFn) => {
    try {
      // Close existing connection
      peerConnection.close();
      
      // Create new connection
      const newPeerConnection = createPeerConnection();
      
      // Retry offer creation
      await createOfferFn(newPeerConnection);
      
      return newPeerConnection;
    } catch (error) {
      handleWebRTCError(error, 'retry-connection');
      throw error;
    }
  };

  return {
    errors,
    handleWebRTCError,
    retryConnection,
  };
};

Network Connectivity Handling

Handle network changes gracefully:

import NetInfo from '@react-native-async-storage/async-storage';

const useNetworkHandling = (peerConnection, reconnectFn) => {
  const [isOnline, setIsOnline] = useState(true);
  const [connectionType, setConnectionType] = useState('unknown');

  useEffect(() => {
    const unsubscribe = NetInfo.addEventListener(state => {
      setIsOnline(state.isConnected);
      setConnectionType(state.type);

      if (state.isConnected && !isOnline) {
        // Network restored, attempt reconnection
        handleNetworkRestore();
      } else if (!state.isConnected) {
        // Network lost
        handleNetworkLoss();
      }
    });

    return unsubscribe;
  }, [isOnline]);

  const handleNetworkLoss = () => {
    console.log('Network connection lost');
    // Optionally pause video or show reconnecting UI
  };

  const handleNetworkRestore = async () => {
    console.log('Network connection restored');
    
    try {
      // Check if peer connection is still valid
      if (peerConnection.current.connectionState === 'disconnected') {
        await reconnectFn();
      }
    } catch (error) {
      console.error('Failed to reconnect:', error);
    }
  };

  return {
    isOnline,
    connectionType,
  };
};

7. Testing and Debugging WebRTC Applications

WebRTC Statistics Monitoring

Monitor call quality and performance:

const useWebRTCStats = (peerConnection) => {
  const [stats, setStats] = useState({});

  useEffect(() => {
    if (!peerConnection.current) return;

    const interval = setInterval(async () => {
      try {
        const reports = await peerConnection.current.getStats();
        const parsedStats = parseStats(reports);
        setStats(parsedStats);
      } catch (error) {
        console.error('Error getting stats:', error);
      }
    }, 1000);

    return () => clearInterval(interval);
  }, [peerConnection.current]);

  const parseStats = (reports) => {
    const stats = {
      video: { bitrate: 0, packetsLost: 0, jitter: 0 },
      audio: { bitrate: 0, packetsLost: 0, jitter: 0 },
      connection: { state: 'unknown', rtt: 0 },
    };

    reports.forEach(report => {
      switch (report.type) {
        case 'inbound-rtp':
          if (report.mediaType === 'video') {
            stats.video.bitrate = report.bytesReceived * 8 / 1000; // kbps
            stats.video.packetsLost = report.packetsLost || 0;
            stats.video.jitter = report.jitter || 0;
          } else if (report.mediaType === 'audio') {
            stats.audio.bitrate = report.bytesReceived * 8 / 1000;
            stats.audio.packetsLost = report.packetsLost || 0;
            stats.audio.jitter = report.jitter || 0;
          }
          break;
        
        case 'candidate-pair':
          if (report.state === 'succeeded') {
            stats.connection.rtt = report.currentRoundTripTime * 1000; // ms
          }
          break;
      }
    });

    stats.connection.state = peerConnection.current.connectionState;
    return stats;
  };

  return stats;
};

// Stats display component
const StatsDisplay = ({ stats }) => (
  <View style={styles.statsContainer}>
    <Text>Connection: {stats.connection?.state}</Text>
    <Text>RTT: {stats.connection?.rtt?.toFixed(0)}ms</Text>
    <Text>Video: {stats.video?.bitrate?.toFixed(0)} kbps</Text>
    <Text>Packets Lost: {stats.video?.packetsLost}</Text>
  </View>
);

Common Issues and Solutions

Troubleshooting guide for common WebRTC problems:

Connection Failures: Check STUN/TURN server configuration and firewall settings
Audio Echo: Implement echo cancellation and proper audio constraints
Poor Video Quality: Adjust bitrate and resolution based on network conditions
One-way Audio/Video: Verify media stream tracks are properly added to peer connection
Connection Drops: Implement ICE restart and connection recovery mechanisms

8. Production Deployment Considerations

TURN Server Setup

For production apps, you'll need reliable TURN servers for NAT traversal:

# Using Coturn (open source TURN server)
# Install on Ubuntu/Debian
sudo apt-get update
sudo apt-get install coturn

# Configure in /etc/turnserver.conf
listening-port=3478
tls-listening-port=5349
realm=your-realm.com
server-name=your-turn-server.com
external-ip=YOUR_PUBLIC_IP

# Create TURN credentials
turnadmin -a -u username -r your-realm.com -p password

Signaling Server Architecture

Production-ready signaling server with Socket.IO and Redis:

// Node.js signaling server
const express = require('express');
const http = require('http');
const socketIo = require('socket.io');
const redis = require('redis');

const app = express();
const server = http.createServer(app);
const io = socketIo(server, {
  cors: { origin: "*" }
});

const redisClient = redis.createClient();

io.on('connection', (socket) => {
  console.log('User connected:', socket.id);

  socket.on('join-room', async (roomId) => {
    socket.join(roomId);
    
    // Store user in room
    await redisClient.sadd(`room:${roomId}`, socket.id);
    
    // Notify others in room
    socket.to(roomId).emit('user-joined', socket.id);
  });

  socket.on('offer', (data) => {
    socket.to(data.roomId).emit('offer', {
      offer: data.offer,
      senderId: socket.id
    });
  });

  socket.on('answer', (data) => {
    socket.to(data.roomId).emit('answer', {
      answer: data.answer,
      senderId: socket.id
    });
  });

  socket.on('ice-candidate', (data) => {
    socket.to(data.roomId).emit('ice-candidate', {
      candidate: data.candidate,
      senderId: socket.id
    });
  });

  socket.on('disconnect', async () => {
    // Clean up user from all rooms
    const rooms = await redisClient.keys('room:*');
    rooms.forEach(async (room) => {
      await redisClient.srem(room, socket.id);
    });
  });
});

server.listen(3000, () => {
  console.log('Signaling server running on port 3000');
});

Security Considerations

TURN Authentication: Use time-limited credentials and rotate them regularly
Signaling Security: Implement authentication and authorization for room access
Media Encryption: WebRTC provides built-in encryption (DTLS/SRTP)
Rate Limiting: Prevent abuse of signaling endpoints
Network Security: Use HTTPS/WSS for signaling communication

Conclusion

React Native WebRTC opens up endless possibilities for real-time communication in mobile apps. From simple video calls to complex multi-party conferences with screen sharing and data channels, the technology provides the foundation for building modern communication experiences.

Key takeaways for successful WebRTC implementation:

Start with basic peer-to-peer connections before adding advanced features
Implement robust error handling and network recovery mechanisms
Monitor call quality and adapt to network conditions
Use proper resource cleanup to prevent memory leaks
Plan for production infrastructure including TURN servers and signaling architecture
Test extensively on different devices and network conditions

The WebRTC landscape continues to evolve with new features like simulcast, advanced codecs, and improved mobile support. Stay updated with the latest developments to provide the best possible real-time communication experience in your React Native applications.

Whether you're building the next video conferencing platform, adding voice chat to your gaming app, or creating innovative real-time collaboration tools, React Native WebRTC provides the performance and flexibility needed for production-ready applications.

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