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Another sneaky WebRTC optimisation only known by Google Meet (RemoteEstimate RTCP packets)

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Google Meet has consistently delivered superior quality among WebRTC applications (at least for web applications). This is especially true compared with typical open-source solutions, but it stands even for most commercial solutions. The reason is that Google has the team that understands the media stack they have built very well and can make it behave in ways that solve their problems in the browser with knobs hidden in WebRTC that only they are aware of. Some of us still remember how simulcast support was added to WebRTC approximately 12 years ago with SDP munging and without any documentation or note behind the x-google-conference flag, or how “Audio Network Adaptation” was added with a secret string encoding an undocumented protobuf schema to tune settings to improve audio quality. Today I was trying to debug why a WebRTC application had a less stable bandwidth estimation than Google Meet. The scenario was quite simple: in a perfect network with plenty of bandwidth, add 50ms of ex...