---
title: "GitHub Copilot Real-Time WebRTC"
description: "Configures GitHub Copilot for real-time communication with WebRTC, Socket.io, and peer-to-peer networking."
category: "configuration"
tags: ["github-copilot", "real-time", "webrtc", "socketio", "p2p", "communication"]
tech_stack: ["webrtc", "socketio", "nodejs", "stun-turn", "media-capture", "p2p"]
---

This setup creates an engaging space for real-time communication using WebRTC and Socket.io. Let’s walk through the steps.

### Configuration Overview
With this configuration, you can enjoy real-time peer-to-peer communication, capture media, and host multi-user video conferences. WebRTC and Socket.io handle the signaling part for you.

### Prerequisites
Before diving in, make sure you have the following:
- **Node.js** (version 14 or higher)
- **npm** (Node package manager)
- **GitHub Copilot** (enabled in your IDE)
- **STUN/TURN server** (to help with NAT traversal)
- A basic understanding of JavaScript and WebRTC

### Installation & Setup
Let’s get started with the installation.

1. **Create a new project directory**:
   ```bash
   mkdir webrtc-communication
   cd webrtc-communication
   ```

2. **Initialize a new Node.js project**:
   ```bash
   npm init -y
   ```

3. **Install required packages**:
   ```bash
   npm install express socket.io webrtc-adapter
   ```

4. **Set up the server**:
   Create a file called `server.js` and add the following code:
   ```javascript
   const express = require('express');
   const http = require('http');
   const socketIo = require('socket.io');

   const app = express();
   const server = http.createServer(app);
   const io = socketIo(server);

   app.use(express.static('public'));

   io.on('connection', (socket) => {
       console.log('New user connected');

       socket.on('offer', (offer) => {
           socket.broadcast.emit('offer', offer);
       });

       socket.on('answer', (answer) => {
           socket.broadcast.emit('answer', answer);
       });

       socket.on('candidate', (candidate) => {
           socket.broadcast.emit('candidate', candidate);
       });

       socket.on('disconnect', () => {
           console.log('User disconnected');
       });
   });

   const PORT = process.env.PORT || 3000;
   server.listen(PORT, () => {
       console.log(`Server is running on port ${PORT}`);
   });
   ```

5. **Create the client-side files**:
   - First, create a directory called `public` and add an `index.html` file:
   ```html
   <!DOCTYPE html>
   <html lang="en">
   <head>
       <meta charset="UTF-8">
       <meta name="viewport" content="width=device-width, initial-scale=1.0">
       <title>WebRTC Communication</title>
       <script src="/socket.io/socket.io.js"></script>
       <script src="script.js" defer></script>
   </head>
   <body>
       <h1>WebRTC Communication</h1>
       <video id="localVideo" autoplay muted></video>
       <video id="remoteVideo" autoplay></video>
   </body>
   </html>
   ```

   - Next, add a `script.js` file inside the `public` directory:
   ```javascript
   const socket = io();

   let localStream;
   let peerConnection;
   const configuration = {
       iceServers: [
           { urls: 'stun:stun.l.google.com:19302' },
           { urls: 'turn:your.turn.server:3478', username: 'user', credential: 'pass' }
       ]
   };

   async function start() {
       localStream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true });
       document.getElementById('localVideo').srcObject = localStream;
       socket.emit('join');
   }

   socket.on('offer', async (offer) => {
       peerConnection = new RTCPeerConnection(configuration);
       peerConnection.setRemoteDescription(new RTCSessionDescription(offer));
       localStream.getTracks().forEach(track => peerConnection.addTrack(track, localStream));
       const answer = await peerConnection.createAnswer();
       await peerConnection.setLocalDescription(answer);
       socket.emit('answer', answer);
   });

   socket.on('answer', (answer) => {
       peerConnection.setRemoteDescription(new RTCSessionDescription(answer));
   });

   socket.on('candidate', (candidate) => {
       peerConnection.addIceCandidate(new RTCIceCandidate(candidate));
   });

   start();
   ```

### File Structure
Here’s how your project should look:
```
webrtc-communication/
├── public/
│   ├── index.html
│   └── script.js
└── server.js
```

### Key Configuration Files
- **`server.js`**: This is the main server file that manages WebSocket connections and signaling.
- **`index.html`**: This file contains the basic HTML structure for your client interface.
- **`script.js`**: This JavaScript file handles the WebRTC connections on the client side.

### Advanced Options
If you want to enhance your setup, consider these options:
- **STUN/TURN Server Configuration**: Choose a reliable STUN/TURN server for improved connectivity. Services like Twilio or Google can help.
- **Media Constraints**: Adjust the media capture settings in `getUserMedia()` to specify resolutions or audio settings.

### Troubleshooting
If you encounter issues, here are some common problems and solutions:
- **Issue**: Unable to connect peers.
  - **Solution**: Make sure your STUN/TURN server is reachable and properly configured.
- **Issue**: No audio/video stream.
  - **Solution**: Verify that your browser has permissions to access the camera and microphone.

### Best Practices
Keep these tips in mind:
- Use HTTPS for secure connections, especially in production.
- Regularly update your dependencies to ensure you have the latest security patches.
- Add error handling for WebRTC events to handle connection issues smoothly.

### Performance Optimizations
To improve performance, consider these strategies:
- Reduce bandwidth usage by adjusting video quality based on the network conditions.
- Utilize data channels for sending non-media data effectively.
- Monitor and log WebRTC connection statistics to fine-tune performance.