Building a Modern Video Call Application: React, Go, WebRTC, and Redis
Published September 17, 2025
Video calling has become an essential part of modern communication. In this comprehensive guide, we'll build a full-featured video call application similar to Google Meet, using modern web technologies. Our application will feature real-time peer-to-peer communication, signaling server coordination, and a polished user interface.
Technology Stack
- Frontend: React 18 with TypeScript, Tailwind CSS
- Backend: Go with Gorilla WebSocket and Mux router
- Database: Redis for real-time data and pub/sub messaging
- WebRTC: For peer-to-peer audio/video communication
- Containerization: Docker and Docker Compose
Architecture Overview
Our video call application follows a signaling server architecture:
- Frontend (React): Handles user interface and WebRTC peer connections
- Signaling Server (Go): Manages room creation, user coordination, and message relay
- Redis: Provides real-time messaging and session management
- WebRTC: Establishes direct peer-to-peer connections for media streaming
┌─────────────┐ WebSocket ┌─────────────────┐ Redis ┌─────────┐
│ React │◄────────────────►│ Go Server │◄────────────►│ Redis │
│ Frontend │ │ (Signaling) │ │ Pub/Sub │
└─────────────┘ └─────────────────┘ └─────────┘
▲
│
│ WebRTC P2P
▼
┌─────────────┐
│ React │
│ Frontend │
│ (Peer) │
└─────────────┘Step 1: Setting Up the Development Environment
Prerequisites
First, ensure you have the following installed:
- Node.js (18+ recommended)
- Go (1.21+)
- Docker and Docker Compose
Project Structure
Create the following directory structure:
video-call-app/ ├── frontend/ # React TypeScript application ├── backend/ # Go signaling server ├── docker-compose.yml # Container orchestration └── README.md
Step 2: Backend Implementation (Go Signaling Server)
Setting Up the Go Module
Create the backend directory and initialize a Go module:
mkdir backend && cd backend go mod init video-call-backend
Main Server Setup
The main server file sets up our HTTP server with WebSocket and REST endpoints. See the complete main.go implementation
WebSocket Handlers
The heart of our signaling server handles WebSocket connections and message routing. View the WebSocket handler code
Redis Integration
Add Redis support for scalability and persistence. See Redis integration code
Step 3: Frontend Implementation (React)
Setting Up React Application
Create a new React TypeScript application:
npx create-react-app frontend --template typescript cd frontend npm install @radix-ui/react-avatar @radix-ui/react-slot class-variance-authority clsx lucide-react tailwind-merge tailwindcss-animate socket.io-client
WebRTC Service
This service handles all WebRTC logic including peer connections, signaling, and media streams. View the complete WebRTC service
Main App Component
The main App component manages routing between the home page and video call room. See App component code
Home Page Component
The home page provides room creation and joining functionality. View HomePage component
Step 4: Docker Configuration
Docker Compose Setup
Set up the complete application stack with Docker Compose. View docker-compose.yml
Backend Dockerfile
Multi-stage Docker build for the Go backend. See Dockerfile
Step 5: Running the Application
Development Setup
- Start Redis:
docker-compose up redis -d - Start Backend: Navigate to backend directory and run
go run . - Start Frontend: Navigate to frontend directory and run
npm start
Production Deployment
For production deployment:
- Full Stack with Docker:
docker-compose up --build - Environment Variables:
- Set appropriate CORS origins
- Configure STUN/TURN servers for NAT traversal
- Use HTTPS certificates for WebRTC
Key Features Implemented
1. Real-time Signaling
- WebSocket-based communication between peers
- Message routing for offers, answers, and ICE candidates
- Room management and user coordination
2. WebRTC Peer-to-Peer Communication
- Direct audio/video streaming between clients
- ICE candidate exchange for NAT traversal
- Connection state monitoring
3. Modern UI/UX
- Google Meet-inspired interface
- Responsive design with Tailwind CSS
- Real-time participant management
4. Scalable Backend
- Go-based signaling server
- Redis integration for session management
- RESTful API for room operations
Security Considerations
1. WebRTC Security
- Peer-to-peer connections encrypt media streams automatically
- DTLS encryption for data channels
- SRTP for media encryption
2. Authentication
- Implement proper user authentication
- Room access controls and permissions
- Rate limiting for API endpoints
3. Production Requirements
- HTTPS required for getUserMedia API
- STUN/TURN servers for production NAT traversal
- Secure WebSocket connections (WSS)
Troubleshooting Common Issues
1. getUserMedia Errors
Ensure HTTPS or localhost for development:
// Ensure HTTPS or localhost for development
if (location.protocol !== 'https:' && location.hostname !== 'localhost') {
console.error('WebRTC requires HTTPS in production');
}2. Connection Issues
- Check firewall settings for WebRTC ports
- Verify STUN/TURN server accessibility
- Monitor browser console for WebRTC errors
3. Performance Optimization
- Implement adaptive bitrate for different network conditions
- Add connection quality monitoring
- Optimize video resolution based on available bandwidth
Conclusion
This comprehensive guide demonstrates how to build a modern video calling application using WebRTC, React, and Go. The architecture provides:
- Scalability: Redis-backed signaling server
- Real-time Communication: WebRTC peer-to-peer connections
- Modern UI: React with TypeScript and Tailwind CSS
- Production-Ready: Docker containerization and proper error handling
The application can be extended with additional features like screen sharing, chat messaging, recording capabilities, and advanced room management. The modular architecture makes it easy to add new functionality while maintaining code quality and performance.
For production deployment, consider implementing user authentication, room permissions, connection quality monitoring, and proper STUN/TURN server configuration for reliable connectivity across different network environments.