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Hey, are you using Opus for audio? I remember the whole reason Opus was so cool, is that it was the first codec that combined low latency with high quality. Most voice chat software uses Opus nowadays so I'm surprised to hear existing solutions are so bad.

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Yes Opus - the other benefit is that it can handle a degree of missing packets which is very handy.

And that feature adds ~20 msec more latency

You must have set it up wrong. It doesn't add any latency, but you do need a higher bitrate to get similar quality.

I googled "opus packet loss" and found https://www.asterisk.org/asterisk-opus-packet-loss-fec/, which says "an Opus encoder can embed redundant data about the preceding packet in-band in the current packet". That method would cause 1 packet of delay. Are you using something different, or avoiding the delay through another method like tiny packets?

Here's probably the answer: https://opus-codec.org/docs/opus_api-1.2/group__opus__decode...

"Lost packets can be replaced with loss concealment by calling the decoder with a null pointer and zero length for the missing packet."

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