[Freeswitch-users] FreeSwitch and WebRTC

[Adam Ben-Ayoun]

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
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