The info on the website is incomplete.
Important to me is what do you do with your audio process?
Most conference services will "improve" the audio by having auto-gain, reducing feedback (by cutting one of the audio sources!) or some other kind of processing. For users which want to talk this can be Ok or even good, but for music classes this is terrible. What kind of audio process do you do? how much is the time delay (latency) from one end to the other?
I think this is important for music classes at distance and it should be published.