About

WebRTC is easy to demo and hard to run. Two laptops on the same Wi-Fi connect in twenty lines of JavaScript. Ship the same code to a few thousand people on hotel Wi-Fi, corporate VPNs and flaky LTE, and you start meeting the parts nobody put in the tutorial: symmetric NATs, TURN bills, one-way audio that only happens on Safari, video that drops to 200 kbps and stays there.

This site is a collection of practical tips, tricks, and hard-earned tweaks to make real-world WebRTC stacks run faster, fail less, and recover seamlessly. Each article focuses on actionable solutions to real-world edge cases encountered when running media in production.

How the articles are written

Articles come from hands-on debugging and production experience. Where relevant, we reference the underlying RFCs and highlight browser quirks or libwebrtc nuances, because the difference between what's specified and how browsers actually behave in the wild is where most bugs hide.

Who it's for

People building calling products, SFUs, or anything that has to keep a media stream alive across a network they don't control. Some familiarity with sockets and HTTP is assumed. Nothing about RTP is.

Corrections

WebRTC.wiki is written by Paul Grace. If you find an error, email me at errors@webrtc.wiki.