How Chromium Manages WebRTC Bitrate

Have you ever wondered how the magic works that changes the resolution of video and ruins your screen sharing?

In the WebRTC industry, there are many myths about this: some say it’s all about losses, others claim that losses are not relevant, and some insist that RTT and jitter should be the top priorities.

They’re all somewhat right. Recently, I had the opportunity to rewrite a fun library that deals with exactly this issue, directly in Chromium browsers. It’s called goog-cc, and I rewrote it from C++ to AssemblyScript. But that’s not the important part.

What matters is that we can delve into the composite blocks and algorithms used by this library and how they come together to determine your outgoing bitrate and whether there’s congestion right now.

And let’s also talk about the fact that there really isn’t just one goog-cc, and that’s a bit sad.