Hi Fellow VoIP experts,
We are working on a leading web-conferencing open source platform for online learning.
This platform uses Freeswitch with Opus codec through WebRTC; so directly through the browser.
We made some tests and the audio quality is a lot better on Jitsi, we want to achieve the same quality through our platform.
If you are a certified VoIP engineer or have a lot of experience configuring Freeswitch for web usage with opus codec through WebRTC we would be glad if you could help us.
So your mission is to help us understand Freeswitch mod_conf opus codec and achieve a better audio quality result.