What is WebRTC?
WebRTC (Web Real-Time Communication) is an open-source project that provides web browsers and mobile applications with real-time communication capabilities via APIs. It enables audio, video, and data to be transmitted peer-to-peer between browsers without requiring plugins, downloads, or intermediate servers for media relay (in ideal conditions).
Originally developed by Google and standardized by W3C and IETF, WebRTC has revolutionized web-based communication. It powers applications like Google Meet, Facebook Messenger video calls, Discord voice chat, and thousands of other real-time communication applications. WebRTC handles the complex networking, codec negotiation, and media processing automatically.
WebRTC Performance Calculator
Memory: 150 MB
Packet Loss: 0.5%
Jitter: 5ms
WebRTC Architecture
MediaStream API
Captures audio/video from user devices
⢠getDisplayMedia()
⢠Audio/Video tracks
⢠Device enumeration
⢠Constraints handling
RTCPeerConnection
Core API for peer-to-peer connections
⢠ICE negotiation
⢠Media transport
⢠DTLS encryption
⢠Statistics/monitoring
RTCDataChannel
Peer-to-peer data transfer
⢠Low latency messaging
⢠Reliable/unreliable modes
⢠Binary/text data
⢠Gaming and file transfer
Connection Flow
1. Signaling Server Setup
ā
2. Media Capture (getUserMedia)
ā
3. Create PeerConnection
ā
4. Add Local Media Streams
ā
5. Create Offer/Answer (SDP)
ā
6. Exchange SDP via Signaling
ā
7. ICE Candidate Exchange
ā
8. DTLS/SRTP Handshake
ā
9. Media/Data Flow Established
NAT Traversal & ICE
The NAT Problem
Most devices are behind NAT/firewalls, making direct peer connections impossible.
Solution: ICE framework with STUN/TURN servers
STUN Servers
Session Traversal Utilities for NAT
⢠Determine NAT type
⢠Enable hole punching
⢠Lightweight and fast
⢠Works for ~80% of cases
TURN Servers
Traversal Using Relays around NAT
⢠Fallback for symmetric NAT
⢠Higher bandwidth cost
⢠99%+ success rate
⢠Increased latency
ICE Candidate Types
⢠Direct LAN connections
⢠Lowest latency
⢠NAT hole punching
⢠Most common type
⢠Guaranteed connectivity
⢠Highest latency/cost
Signaling & Session Negotiation
Offer/Answer Model
Peers negotiate capabilities through SDP (Session Description Protocol) exchange.
// Caller creates offer
const offer = await peerConnection.createOffer();
await peerConnection.setLocalDescription(offer);
// Send offer via signaling server
signalingSocket.send(JSON.stringify({
type: 'offer',
sdp: offer
}));
// Callee receives offer, creates answer
await peerConnection.setRemoteDescription(offer);
const answer = await peerConnection.createAnswer();
await peerConnection.setLocalDescription(answer);
// Send answer back
signalingSocket.send(JSON.stringify({
type: 'answer',
sdp: answer
}));
SDP Example
Session Description Protocol contains media capabilities and network information.
o=- 123456789 2 IN IP4 192.168.1.100
s=-
t=0 0
a=group:BUNDLE 0 1
m=audio 9 UDP/TLS/RTP/SAVPF 111
a=rtpmap:111 opus/48000/2
a=sendrecv
m=video 9 UDP/TLS/RTP/SAVPF 96
a=rtpmap:96 VP8/90000
a=sendrecv
a=candidate:1 1 UDP 2130706431 192.168.1.100 54400
WebRTC Implementation Example
Basic Video Call Setup
class VideoCall {
constructor(signalingSocket) {
this.socket = signalingSocket;
this.peerConnection = new RTCPeerConnection({
iceServers: [
{ urls: 'stun:stun.l.google.com:19302' },
{
urls: 'turn:turn.example.com:3478',
username: 'user',
credential: 'pass'
}
]
});
this.setupEventHandlers();
}
async startCall() {
// Get user media
const stream = await navigator.mediaDevices.getUserMedia({
video: { width: 1280, height: 720 },
audio: true
});
// Add tracks to peer connection
stream.getTracks().forEach(track => {
this.peerConnection.addTrack(track, stream);
});
// Create and send offer
const offer = await this.peerConnection.createOffer();
await this.peerConnection.setLocalDescription(offer);
this.socket.send(JSON.stringify({ type: 'offer', offer }));
}
setupEventHandlers() {
// Handle ICE candidates
this.peerConnection.onicecandidate = (event) => {
if (event.candidate) {
this.socket.send(JSON.stringify({
type: 'ice-candidate',
candidate: event.candidate
}));
}
};
// Handle remote stream
this.peerConnection.ontrack = (event) => {
const remoteVideo = document.getElementById('remoteVideo');
remoteVideo.srcObject = event.streams[0];
};
// Handle connection state
this.peerConnection.onconnectionstatechange = () => {
console.log('Connection state:', this.peerConnection.connectionState);
};
}
}
Data Channel Example
// Create data channel
const dataChannel = peerConnection.createDataChannel('messages', {
ordered: true,
maxRetransmits: 3
});
dataChannel.onopen = () => {
console.log('Data channel opened');
dataChannel.send('Hello from peer!');
};
dataChannel.onmessage = (event) => {
console.log('Received:', event.data);
};
// Handle incoming data channel
peerConnection.ondatachannel = (event) => {
const channel = event.channel;
channel.onmessage = (event) => {
console.log('Data received:', event.data);
};
};
Scaling WebRTC Applications
Mesh Topology
Every peer connects to every other peer
⢠Low latency
⢠High bandwidth usage (N²)
⢠Limited to ~4-6 participants
⢠CPU scales poorly
SFU (Selective Forwarding Unit)
Central server forwards streams between peers
⢠Bandwidth efficient
⢠Low server CPU usage
⢠Simulcast support
⢠Quality adaptation
MCU (Multipoint Control Unit)
Server mixes all streams into composite output
⢠Lowest bandwidth
⢠High server CPU usage
⢠Centralized processing
⢠Layout control
Hybrid Approaches
Combine multiple topologies based on needs
⢠P2P + SFU fallback
⢠Regional distribution
⢠Edge computing
⢠CDN integration
WebRTC in Production
Google Meet
Large-scale video conferencing with advanced features.
- ⢠SFU architecture with simulcast
- ⢠AI-powered noise cancellation
- ⢠Adaptive bitrate streaming
- ⢠100+ participants support
Discord Voice
Low-latency voice chat for gaming communities.
- ⢠Opus codec optimization
- ⢠Push-to-talk and voice activity
- ⢠Global edge server network
- ⢠Mobile battery optimization
Telemedicine
HIPAA-compliant video consultations.
- ⢠End-to-end encryption
- ⢠Screen sharing for records
- ⢠Recording capabilities
- ⢠Bandwidth adaptation
Online Education
Interactive classrooms and virtual labs.
- ⢠Whiteboard collaboration
- ⢠Screen sharing and recording
- ⢠Breakout room support
- ⢠Mobile-friendly interfaces
WebRTC Best Practices
ā Do
- ⢠Implement proper error handling and fallbacks
- ⢠Use both STUN and TURN servers
- ⢠Implement bandwidth adaptation
- ⢠Add connection quality monitoring
- ⢠Handle mobile/battery considerations
- ⢠Use simulcast for scalability
- ⢠Implement proper security measures
ā Don't
- ⢠Rely only on P2P for large groups
- ⢠Ignore network conditions
- ⢠Skip TURN server provisioning
- ⢠Forget about mobile constraints
- ⢠Neglect codec negotiation
- ⢠Hardcode media constraints
- ⢠Skip connection state monitoring