WebRTC (Web Real-Time Communication)

WebRTC (Web Real-Time Communication) is a collection of browser APIs and network protocols that lets web pages and apps exchange audio, video and arbitrary data directly between endpoints, with encryption mandatory and no plugin required.

Why it matters

WebRTC powers browser-based video calls, screen sharing and some file transfer or game networking. Media is sent over UDP-based RTP (secured as SRTP), and a data channel can carry app messages. Since most devices sit behind NAT, WebRTC uses ICE to find a working path, using STUN servers to discover public addresses and TURN servers to relay traffic when a direct connection is impossible.

Example

const pc = new RTCPeerConnection({
  iceServers: [{ urls: "stun:stun.example.org:3478" }]
});
const channel = pc.createDataChannel("chat");

The app must still exchange an offer, an answer and ICE candidates through its own signaling channel, such as a WebSocket. WebRTC does not define that part.

Common mistakes

Skipping a TURN server. Some users, especially on corporate or mobile networks, cannot connect peer to peer without a relay.
Assuming WebRTC has built-in signaling.
Forgetting that peer-to-peer connections can reveal a user's IP address to the other side, which matters for privacy.

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