There’s a lot, and specifically a lot of machine learning talk and features in the 1.5 release of Opus - the free and open audio codec.
Audible and continuous (albeit jittery) talk on 90% packet loss is crazy.
Section WebRTC Integration → Samples has an example where you can test out the 90 % packet loss audio.
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I wonder if/when this will be available in voice chat clients like TeamSpeak.
