# WebRTC
Confidence: high
Last verified: 2026-05-22
Generation: human_only


## TL;DR

WebRTC (Web Real-Time Communication) is a W3C standard enabling peer-to-peer audio, video, and data communication between browsers without plugins. It uses DTLS-SRTP for encryption and ICE/STUN/TURN for NAT traversal.

## Core Explanation

Core APIs: `RTCPeerConnection` (media/data transport), `getUserMedia()` (camera/microphone access), `RTCDataChannel` (arbitrary binary data). Signaling (connection negotiation) is not specified — applications must implement their own signaling channel (usually via WebSocket). WebRTC uses UDP by default but can fall back to TCP.

## Further Reading

- [WebRTC 1.0 (W3C Recommendation)](https://www.w3.org/TR/webrtc/)