Since the transition from Vivox to WebRTC, voice quality has noticeably decreased for many users.

WebRTC itself is fully capable of excellent voice quality. I would like to see the current WebRTC/Opus audio configuration revisited.

I would like to see

* A higher default bitrate/audio quality target

* Add a user adjustable voice quality/bitrate setting (Low, Normal, High)

Second Life is a social platform; Why are we treating the one of two socializing options we have in-game as an afterthought.

I understand that increasing bitrate has bandwidth and infrastructure costs, but it is disappointing that moving to a modern WebRTC/Opus resulted in worse perceptual audio quality in exchange for relatively small bandwidth savings.

If bitrate is not the cause of the quality difference, then I would ask that the codec, bandwidth, processing, and other WebRTC audio parameters be reviewed with the goal of matching or exceeding the perceptual quality of the previous Vivox implementation.