I was only able to use a crude method by having another peer stream back a virtual camera of the desktop via OBS. Only I was using the ASIO API. My client read a ping of approx 15ms while the other was around 30ms. Playing back a 60fps recording of this video feedback loop, I counted 24 frames of latency compared to around 13 frames of audio latency. Any resources for determining a better methodology?
There seems to be a lot of knowledge around reducing latency overhead for audio streaming for remote synchronous rehearsals, but I didn’t find much luck for the video portion. Are the Virtual Studios using WebRTC for video streaming?
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Welcome @happyman2570 !! Hopefully, @miked will have some answers for you.
Hi @happyman2570, yes we are using WebRTC for video. It will certainly have more latency than audio (via desktop app or bridge device), and we currently do not have any way to optimize it.
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