WebRTC Real-Time Chat & Video Calling Platform

A full-featured real-time chat application with WebRTC-based peer-to-peer video calling.

Features

Real-Time Chat

• WebSocket-based instant messaging
• Multiple chat rooms
• User presence and online status
• Typing indicators
• Message history (last 100 messages per room)
• System notifications for join/leave events

Video Calling

• WebRTC peer-to-peer video calls
• One-on-one video calls between users
• Audio/video controls (mute, camera toggle)
• Screen sharing capability
• Call signaling through WebSocket
• STUN server support for NAT traversal

User Interface

• Modern, responsive design
• Real-time user list
• Incoming call notifications
• In-call controls
• Beautiful gradient animations

Quick Start with Docker

# Build and run
docker-compose up --build

# Or build manually
docker build -t webrtc-chat .
docker run -p 8000:8000 webrtc-chat

Usage

1. Open http://localhost:8000 in your browser
2. Enter your username and room name
3. Join the room to start chatting
4. Click "Call" next to any user to start a video call
5. Use in-call controls to mute, toggle video, or share screen

API Endpoints

• GET / - Web interface
• WS /ws - WebSocket connection for real-time communication
• GET /api/rooms - List active rooms
• GET /api/stats - Platform statistics

WebSocket Message Types

Client -> Server

• join - Join a room
• chat - Send chat message
• typing - Send typing indicator
• start_call - Initiate video call
• answer_call - Answer incoming call
• ice_candidate - WebRTC ICE candidate
• end_call - End call
• decline_call - Decline incoming call

Server -> Client

• history - Message history
• users_list - Online users
• user_joined - User joined notification
• user_left - User left notification
• chat - Chat message
• typing - Typing indicator
• incoming_call - Incoming video call
• call_answered - Call was answered
• ice_candidate - WebRTC ICE candidate
• call_ended - Call ended
• call_declined - Call was declined

Technical Details

• Backend: FastAPI with WebSocket support
• Frontend: Vanilla JavaScript with WebRTC API
• Signaling: WebSocket for call coordination
• Media: WebRTC for peer-to-peer audio/video
• STUN Servers: Google's public STUN servers
• Container: Docker with Python 3.12

Browser Requirements

• Modern browser with WebRTC support (Chrome, Firefox, Safari, Edge)
• Camera and microphone permissions
• HTTPS or localhost required for media devices

Future Enhancements

• TURN server for better NAT traversal
• Group video calls
• File sharing
• Message reactions and emojis
• User authentication
• Persistent message storage (database)
• Message search
• Video recording
• Chat moderation tools
• Mobile apps (React Native)