Hi,

We are using FreeSwitch as an MCU for audio-only WebRTC (basically, users connect to conference rooms). We have full control over the client application (it's a mobile app we are developing) and we solely rely on the client to choose the codec and do all the intelligent media related stuff.

My question is, how can we minimize the audio processing/transcoding on FreeSwitch/mod_conference and mitigate the related overhead? Can we achieve that with tuning mod_conference's "rate" and "interval" parameters in that sense?

Any other recommendations regarding FreeSwitch and WebRTC?

Thanks,

Adam