Individual skill
webrtc-pro
Build real-time audio/video with WebRTC: signaling, peer connections, media devices, and TURN infrastructure. Use for video calls and peer-to-peer media.
skills/webrtc-pro/SKILL.md
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# WebRTC Pro
A practical guide to WebRTC: browser-to-browser real-time audio, video, and data — signaling, peer connections, NAT traversal (STUN/TURN), media handling, and the infrastructure decisions that determine whether calls actually connect.
## Overview
WebRTC enables **peer-to-peer** real-time communication in the browser: audio/video calls, screen sharing, and data channels — with encryption (DTLS-SRTP) built in. The catch: browsers can't find each other alone. You need **signaling** (your server exchanging session descriptions/ICE candidates) and **NAT traversal** (STUN to discover addresses, TURN to relay when direct fails). Media flows peer-to-peer; signaling and TURN are your infrastructure.
## When to use
- 1:1 or group video/voice calls.
- Screen sharing.
- Peer-to-peer data transfer (files, game state) via data channels.
- Live broadcasting with sub-second latency (WHIP/WHEP or custom).