WebRTC & WebSockets Cheat Sheet

This content was generated with the assistance of AI and should be verified against the official documentation before being relied on in production.

See the section bibliography for the reference material consulted while preparing these pages.

A single-page, colour-coded summary of the WebRTC and WebSockets essentials — a transport chooser, the WebSocket handshake, frame header and close codes, the browser API, reconnection and heartbeats, one strip per server stack, the WebRTC offer and answer sequence, perfect negotiation, media constraints, data-channel options, simulcast, getStats, mesh versus SFU versus MCU, TURN credentials, media servers, XMPP essentials, chat, videoconference and game checklists, production settings, and a strip of what changed since older tutorials — handy for quick memorization or as a printable desk reference through the downloadable PDF. It reflects the baseline of the section as checked on 2026-10-24: RFC 6455 with RFC 8441 and RFC 9220, the W3C WebRTC 1.0 Recommendation with JSEP in RFC 9429, mediasoup 3.31.0, LiveKit 1.13.10, Janus 1.4.3, socket.io 4.8.4, python-socketio 5.17.0, uWebSockets.js v20.71.0, Channels 4.3.2, ASP.NET Core 10.0.13, Spring Framework 7.0.10, Atmosphere 4.0.76, Swoole 6.2.3, Laravel Reverb 1.12.0, ejabberd 26.09 and Prosody 13.0.7. The section home page is WebRTC & WebSockets.

For the full explanation, every argument, and additional examples, see the dedicated pages:

Reference — Comparison & Decision Guide.