WebRTC Connection Library

This library is developed to establish peer-to-peer (1-1) connections between browsers using WebRTC technology, share video/audio streams, and enable messaging through data channels.

Features

• 🎥 Camera sharing
• 🖥️ Screen sharing
• 📨 Messaging via data channel
• 🔄 RTC connection management
• ⚙️ Codec preference settings
• 🚀 Bandwidth optimization

Installation

// Include the library in your project
import WebRTC from './WebRTC.js';

Usage

Creating a Connection

// Create a WebRTC instance
const webrtc = new WebRTC({
    configuration: {
        iceServers: [{
            urls: ['turn:sunucu_adresi:port'],
            username: 'kullanıcı_adı',
            credential: 'şifre'
        }]
    }
});

Create First Connection

await webRTC.createConnection();
const offer = await webrtc.createOffer(); // Send the offer to the other peer (HTTP, WebSocket, etc.)

Join Connection an Offer Data

await webrtc.createConnection();
const answerData = await webrtc.createAnswer(JSON.parse(offer)) // Offer Data

Complete Connection

// First Client
const answer = await webrtc.createAnswer(JSON.parse(answerData)); // Answer Data

Screen Sharing

// Start screen sharing
await webRTC.startScreenShare();

// Stop screen sharing
webRTC.stopScreenShare();

Camera Sharing

// Start camera sharing
await webRTC.startCamera();

// Stop camera sharing
webRTC.stopCamera();

Messaging

// Send a message
webRTC.sendMessage({
  cmd: 'custom-command',
  data: 'Hello world'
});

// When data message received
webrtc.onDataMessage = (message) => {
    console.log('Data Message: ', message);
};

Events

// When audio stream starts
webrtc.onTrackAudio = (event, streamType) => {
    console.log('Audio Track: ', event)
    document.getElementById('audioElement').srcObject = event.streams[0];
    document.getElementById('audioElement').play().catch(e => console.error('Audio play error:', e));
};

// When video stream starts
webrtc.onTrackVideo = async (event, streamType) => {
    if (streamType === 'screen') {
        document.getElementById('screenVideoElement').srcObject = event.streams[0];
        document.getElementById('screenVideoElement').play().catch(e => console.error('Video play error:', e));
    } else {
        document.getElementById('videoElement').srcObject = event.streams[0];
        document.getElementById('videoElement').play().catch(e => console.error('Video play error:', e));
    }
};

// When video stream stops
webrtc.onStopVideo = (streamType, streamId) => {
    console.log('onStopVideo: ', streamType, streamId);

    if (streamType === 'screen') {
        document.getElementById('screenVideoElement').srcObject = null;
    } else {
        document.getElementById('videoElement').srcObject = null;
    }
};

// When connection is established
webrtc.onConnect = () => console.log('onConnect');

// When connection is closed
webrtc.onDisconnect = () => console.log('onDisconnect');

Browser Support

This library works in versions of modern browsers that support the WebRTC API:

• Chrome
• Firefox
• Edge
• Safari (iOS 11+)

Screen Sharing Support

To check for screen sharing support:

if (WebRTC.isScreenShareSupport()) {
  // Screen sharing is supported
}

Code Optimization

The library provides a setting that prefers the VP8 video codec and operates at a default bit rate of 5 Mbps. These parameters can be modified if needed.

Notes

• WebRTC works over HTTPS or on localhost.
• STUN/TURN servers may be required for firewall and NAT traversal.
• Additional permissions may be required in some browsers.