The core of WebRTC implementing P2P video calls is to understand the collaboration methods of various components, rather than building wheels from scratch; 2. Establish direct connections through signaling exchange of SDP and ICE candidate paths to reduce latency and server costs; 3. Use STUN to penetrate NAT, and deploy TURN when necessary to ensure connection reliability; 4. The code needs to correctly handle media stream addition, offer/answer exchange and ICE candidate transmission; 5. Pay attention to common pitfalls such as HTTPS restrictions, mobile compatibility and no built-in fallback mechanism - mastering these can build efficient real-time communication applications.
Building a peer-to-peer video chat app with WebRTC isn't about reinventing the wheel—it's about understanding how the pieces fit together. If you're diving into real-time communication, WebRTC is your go-to tech. It's built into modern browsers and handles the heavy lifting of audio/video streaming between users without needing a central server for the media itself. Here's how it actually works in practice:
1. The Core Idea: Peer-to-Peer, Not Server-to-Peer
Unlike traditional video apps that route media through a server (like Zoom or old-school RTMP), WebRTC establishes a direct connection between two browsers (peers). That means lower latency and less server cost—once the connection is set up, your server isn't carrying the video stream at all.
But how do two browsers find each other in the first place? That's where signaling comes in.
2. Signaling: The “Handshake” You Have to Build Yourself
WebRTC doesn't handle signaling—it's up to you. Signaling is how peers exchange:
- Session descriptions (SDP): info about media capabilities (eg, codecs, resolution)
- ICE candidates: potential network paths (IPs, ports) for the connection
You can use WebSockets, Socket.IO, or even HTTP long-polling for this. Example flow:
- User A clicks "call," creates an offer via
RTCPeerConnection.createOffer()
- Sends the offer via your signaling channel to User B
- User B sets the remote description, creates an answer, sends it back
- Both peers exchange ICE candidates as they're discovered (
onicecandidate
)
This part is often confusing because WebRTC doesn't dictate how signaling works—you choose the transport. But it's essential for discovery and negotiation.
3. ICE, STUN, TURN: Getting Through Firewalls and NATs
Most users are behind routers or firewalls. That's where ICE (Interactive Connectivity Establishment) comes in—it finds the best path between peers using:
- STUN servers : Help peers discover their public IP (eg, Google's
stun.l.google.com:19302
) - TURN servers : If direct connection fails (eg, symmetric NAT), TURN acts as a relay (media goes through it—slower but reliable)
You don't need to run your own STUN/TURN at first—public STUNs are fine for testing. But for production, especially with enterprise users, a TURN server is a must-have.
4. Code Snippets That Actually Work
Here's the bare minimum to get video flowing:
const pc = new RTCPeerConnection({ iceServers: [{ urls: 'stun:stun.l.google.com:19302' }] }); // Add local stream (from getUserMedia) navigator.mediaDevices.getUserMedia({ video: true, audio: true }) .then(stream => { stream.getTracks().forEach(track => pc.addTrack(track, stream)); }); // Send offer or answer via your signaling channel pc.createOffer().then(offer => { pc.setLocalDescription(offer); signalingChannel.send(offer); // eg, socket.emit('offer', offer) });
On the other side, handle the incoming offer, set it as remote description, create an answer, and set that locally too. Then ICE candidates flow automatically.
5. Common Gotchas
- HTTPS required : Browsers block
getUserMedia
on HTTP (except localhost) - ICE failures : If peers can't connect directly, you'll need TURN
- No built-in fallback : If WebRTC fails, you're on your own—no automatic fallback to server-based streaming
- Mobile quirks : iOS Safari is picky about constraints and permissions
Bottom line: WebRTC gives you the power of real-time P2P video, but you have to gleat the pieces together—signaling, ICE, and media handling. Once it clicks, it's surprisingly elegant.
The above is the detailed content of WebRTC Explained: Building a Peer-to-Peer Video Chat App. For more information, please follow other related articles on the PHP Chinese website!

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