WebRTC Video Call, Chat, and File Sharing Application

I'll create a complete WebRTC application with video calling, chat, and file sharing capabilities that works across devices.

How the WebRTC Application Works

This application provides a complete peer-to-peer communication platform with video calling, chat, and file sharing capabilities using WebRTC technology. Here's how it works:

1. Connection Establishment:

   • Users can create or join a room using a unique room ID
   • The application uses WebRTC's RTCPeerConnection to establish direct connections between peers
   • STUN/TURN servers are configured to handle NAT traversal, ensuring connections work across different networks

2. Video Calling:

   • Accesses user's camera and microphone using getUserMedia()
   • Streams video and audio between peers using WebRTC media tracks
   • Provides controls to toggle audio/video and end calls

3. Chat Messaging:

   • Uses WebRTC data channels for real-time text messaging
   • Messages are displayed with timestamps and sender information
   • The interface automatically scrolls to show new messages

4. File Sharing:

   • Allows sending files of any type between peers
   • Files are chunked and transferred via data channels
   • Shows progress indicators during transfer
   • Provides download/open options for completed transfers

5. Error Handling:

   • Implements ICE connection state monitoring
   • Automatically attempts to restart ICE if connections fail
   • Provides user-friendly error messages
   • Handles permission issues for camera/microphone access

The application is designed to work across different devices (PC to PC, PC to phone) and includes responsive UI elements that adapt to different screen sizes.

Technical Implementation Details

• Signaling: In a production environment, you would need a signaling server to exchange connection information between peers. This example simulates signaling for demonstration purposes.
• NAT Traversal: The application uses multiple STUN servers and a TURN server to handle complex network scenarios where direct connections might be blocked.
• Data Channels: WebRTC data channels are used for both chat messages and file transfers, providing a secure and efficient way to exchange data.
• Media Handling: The application properly manages media streams, including stopping tracks when connections end to free up resources.
• Responsive Design: The UI is fully responsive and works well on both desktop and mobile devices.

Running the WebRTC Application Locally

Yes, this application will run on your local machine and can connect with another PC or phone, but there are some important considerations to make it work properly:

How to Run It Locally

1. Set up a Next.js project:

   npx create-next-app@latest my-webrtc-app cd my-webrtc-app

2. Install the required dependencies:

   npm install lucide-react
3. Copy the code from the code blocks above into your project

4. Start the development server:

   npm run dev
5. Access the application at http://localhost:3000

Connecting Between Devices

For the application to work between different devices (PC to PC or PC to phone), you need to consider:

1. Local Network Connections

If both devices are on the same local network (same WiFi):

• Access the app using your computer's local IP address (e.g., http://192.168.1.100:3000)
• You can find your IP by running ipconfig (Windows) or ifconfig (Mac/Linux)
• Both devices must be on the same network

2. Internet Connections (Important)

For connections across the internet (different networks), you need:

1. A real signaling server: The current code simulates signaling for demonstration. For real-world use, you need to implement an actual signaling server (using WebSockets or a service like Firebase).
2. HTTPS: WebRTC requires secure contexts (HTTPS) in most browsers when not on localhost.
3. NAT traversal: The code includes STUN/TURN server configurations, but you might need to use your own TURN server for reliable connections across restrictive networks.

Making It Work Across Devices

To make this application work reliably across different devices:

1. Replace the simulated signaling:

   // Replace this in video-call.tsx: // simulateSignaling(peerConnection, roomId) // With a real signaling implementation using WebSockets or a service like Firebase
2. Deploy the application to a hosting service that provides HTTPS (like Vercel, Netlify, or GitHub Pages)
3. Use a free TURN server service or set up your own if connections fail between networks with strict firewalls

Quick Test on Local Network

For a quick test between devices on the same network:

1. Run the app on your computer
2. Find your local IP address
3. On another device, open a browser and navigate to http://[your-local-ip]:3000
4. Create a room on one device and join with the same room ID on the other

The current implementation will work for basic testing in this scenario, but for reliable connections across different networks, you'll need to implement a proper signaling server.