📡 What is WebRTC?
WebRTC (Web Real-Time Communication) is a free, open-source technology that allows real-time audio, video, and data communication between browsers and mobile apps — without plugins or external software.
✅ Think Zoom, Google Meet, or Discord voice/video — built directly into the browser.
🔧 What Can You Build with WebRTC?
🔴 Video/audio chat apps (e.g., Google Meet)
🟢 P2P file sharing
🔵 Live streaming
🟡 IoT and remote camera control
🧑🤝🧑 Online collaborative tools (whiteboards, games, etc.)
🛠️ How WebRTC Works (Simplified)
WebRTC uses peer-to-peer (P2P) communication:
📞 Signaling – Set up the connection (via WebSocket, Firebase, or server)
🧭 Session Description – Exchange media capabilities (SDP)
🧲 ICE Candidates – Find the best network path (NAT traversal)
🔗 Peer Connection – Direct media/data stream between peers
🌐 Key WebRTC Components
| Component | Purpose |
|---|---|
getUserMedia() | Access camera/microphone |
RTCPeerConnection | Handle video/audio/data connection between peers |
RTCDataChannel | Send arbitrary data (e.g., chat, files) |
MediaStream | Represents media captured or received |
ICE (STUN/TURN) | Finds the best path for P2P |
🧪 Basic Example – One-Way Video Stream
navigator.mediaDevices.getUserMedia({ video: true, audio: true })
.then(stream => {
document.querySelector('video').srcObject = stream;
})
.catch(error => console.error('Error accessing media:', error));
📦 WebRTC Architecture Overview
User A ↔ (Signaling Server) ↔ User B
↕ ↕
Exchange SDP & ICE ↕
↔ Direct P2P Media & Data Connection ↔
Signaling: You handle it (via WebSocket, Socket.IO, Firebase, etc.)
Media: WebRTC handles once connection is established
🧩 Libraries That Make WebRTC Easier
| Library | Use |
|---|---|
| 📦 Simple-Peer | Wrapper around WebRTC |
| 🔌 PeerJS | Abstraction for peer connections |
| 🧠 Mediasoup | Advanced SFU for multi-user video |
| 🧪 [Janus / Jitsi / Kurento] | Full WebRTC servers for production use |
⚠️ Challenges with WebRTC
| Challenge | Solution |
|---|---|
| 🔒 NAT/Firewall traversal | Use STUN/TURN servers |
| 📶 Bad network conditions | Add retry/ICE restarts |
| 🔄 Multi-peer scaling | Use an SFU/MCU (e.g., Jitsi, Janus) |
| 🔐 Security | Encrypted by default (DTLS, SRTP) |
| ⚙️ Signaling | Must build it yourself |
🛡️ Security in WebRTC
All media and data is encrypted by default
WebRTC enforces HTTPS origins and secure ICE
Supports identity verification with certificates and fingerprints
📚 Learn WebRTC
✅ Summary
| Feature | WebRTC |
|---|---|
| Audio/Video Chat | ✅ |
| P2P File Transfer | ✅ |
| Built-in to Browsers | ✅ |
| Requires Server for Signaling | ✅ |
| Real-time Data Transfer | ✅ |

































































































