WebRTC

English | 简体中文 | 📖 Online Docs

A production-ready WebRTC learning platform — from basic peer-to-peer calls to advanced multi-party Mesh architecture.

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Table of Contents

• Features
• Quick Start
• Why This Project?
• Architecture
• Documentation
• Configuration
• Deployment
• Contributing
• License

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Features

Feature	Description
🌐 WebSocket Signaling	Gorilla WebSocket with heartbeat, room management, and message relay
👥 Multi-party Mesh	Up to 50 participants per room with automatic peer management
💬 DataChannel Chat	Peer-to-peer text messaging without server relay
🎥 Media Controls	Mute/unmute, camera toggle, screen sharing
📹 Local Recording	Browser-side MediaRecorder with WebM export
🔒 Security	Origin validation, identity binding, rate limiting
🐳 Docker Ready	Multi-stage builds with HTTPS/TURN support

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Quick Start

Prerequisites

• Go 1.22+
• Modern browser (Chrome 90+, Firefox 88+, Safari 14+)
• Docker (optional)

Run Locally

git clone https://github.com/LessUp/webrtc.git
cd webrtc
go mod tidy
go run ./cmd/server

Open http://localhost:8080 and start calling!

Run with Docker

docker build -t webrtc .
docker run --rm -p 8080:8080 webrtc

Run with Docker Compose (Production)

export DOMAIN=your-domain.com
docker compose up -d

Visit https://your-domain.com with automatic HTTPS.

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Why This Project?

• 🎓 Learning-Oriented — Progressive complexity from 1-on-1 to multi-party
• 🔐 Security-First — Origin validation, identity binding, connection limits
• 📦 Zero-Dependency Frontend — Pure vanilla JavaScript, no frameworks
• 🚀 Docker-Ready — Multi-stage builds for minimal image size
• 📝 Well-Documented — Bilingual docs (EN/ZH), architecture diagrams, troubleshooting guides

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Architecture

Module Structure

webrtc/
├── cmd/server/              # HTTP + WebSocket entry point
├── internal/signal/         # Signaling logic
│   ├── hub.go               # Room management, message relay
│   ├── hub_test.go          # Unit tests
│   └── message.go           # Message type definitions
├── web/                     # Frontend (vanilla JS)
│   ├── index.html           # UI
│   ├── app.js               # Main entry
│   ├── app.*.js             # Modular components
│   └── styles.css           # Responsive styles
├── docs/                    # Documentation (EN/ZH)
├── changelog/               # Version history
└── .github/workflows/       # CI/CD

System Architecture

┌──────────────────────────────────────────────────────┐
│  Browser A                                           │
│  ┌──────────┐    ┌──────────┐    ┌────────────────┐ │
│  │  HTML UI  │──→│  app.js  │──→│  getUserMedia   │ │
│  └──────────┘    └────┬─────┘    └──────┬─────────┘ │
└───────────────────────┼─────────────────┼───────────┘
                        │ WebSocket       │ WebRTC P2P
                 ┌──────▼──────┐          │
                 │  Go Server   │          │
                 │ ┌──────────┐│          │
                 │ │Signal Hub││          │
                 │ └──────────┘│          │
                 └──────┬──────┘          │
                        │ WebSocket       │
┌───────────────────────┼─────────────────┼───────────┐
│  Browser B            │                 │           │
│  ┌──────────┐    ┌────▼─────┐    ┌──────▼─────────┐│
│  │  HTML UI  │──→│  app.js  │──→│  getUserMedia   ││
│  └──────────┘    └──────────┘    └────────────────┘│
└─────────────────────────────────────────────────────┘

Flow	Path	Description
Signaling	Browser ↔ WebSocket /ws ↔ Hub	Offer/Answer/ICE relay
Media	Browser ↔ WebRTC P2P ↔ Browser	Audio/video streams
DataChannel	Browser ↔ WebRTC P2P ↔ Browser	Text chat

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Documentation

Complete documentation available in English and 简体中文:

Document	Description
📘 Guide	Architecture, implementation details
🚀 Deployment	Docker, HTTPS, TURN setup
📡 Signaling	WebSocket protocol specification
🔧 API Reference	Configuration, environment variables
🔍 Troubleshooting	Common issues and solutions

📖 Online Docs: https://lessup.github.io/webrtc/

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Configuration

Environment Variables

Variable	Default	Description
ADDR	:8080	HTTP listen address
WS_ALLOWED_ORIGINS	*	Comma-separated origins; * for all
RTC_CONFIG_JSON	Public STUN	JSON ICE/TURN config passed to browser

Custom ICE/TURN Configuration

export RTC_CONFIG_JSON='{
  "iceServers": [
    { "urls": ["stun:stun.l.google.com:19302"] },
    { "urls": ["turn:turn.example.com:3478"], "username": "user", "credential": "pass" }
  ]
}'

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Deployment

Production Checklist

• Set WS_ALLOWED_ORIGINS to your domain
• Configure TURN server for NAT traversal
• Enable HTTPS (Caddy handles this automatically)
• Set up monitoring and logging

Docker Compose (Recommended)

# docker-compose.yml
services:
  webrtc:
    build: .
    environment:
      - WS_ALLOWED_ORIGINS=yourdomain.com
      - RTC_CONFIG_JSON={"iceServers":[{"urls":"turn:yourdomain.com:3478"...}]}
  
  caddy:
    image: caddy:2-alpine
    ports:
      - "80:80"
      - "443:443"
  
  coturn:
    image: coturn/coturn:latest
    network_mode: host

See Deployment Guide for detailed instructions.

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Contributing

We welcome contributions! Please see:

• Contributing Guidelines — Setup and workflow
• Roadmap — Future plans
• Changelog — Version history

Development Setup

# Install dependencies
go mod tidy

# Run tests
go test -race ./...

# Run linter
golangci-lint run

# Start with hot reload
air

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Tech Stack

Category	Technology
Backend	Go 1.22+, net/http, Gorilla WebSocket
Frontend	HTML5 + Vanilla JavaScript + CSS3
Media	WebRTC APIs (getUserMedia, RTCPeerConnection, DataChannel)
Container	Docker (multi-stage)
CI/CD	GitHub Actions

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Security

• Origin whitelist validation
• Server-verified client identities
• Connection limits (50 clients/room, 1000 rooms)
• Input validation and sanitization
• See Security Policy

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License

MIT License © LessUp

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Made with ❤️ by LessUp