I've just discovered SRT -> Streaming Server -> WebRTC on mass for ~500ms end to end latency. (Even to iphone/ipad without an app needed for playback).

There is a hiccup with that, seems WebRTC uses Opus audio only, with H.264 or VP9 video.

Wondering if Opus could be supported as an audio codec under the SRT settings in Vmix? Would make life easier not having to transcode the audio on a streaming server, reducing latency/problems in that area.

Of course I would love it if WebRTC supported AAC, but don't think that's likely. If there is already a way, please let me know, I could have overlooked something.