Hi,

I'm currently using a RTMP server for audio conference.
There are no performance or latency issues because there is no encryption, transcoding or mixing involved. A 1GB VPS server can handle more than 100 users without a problem.

I'm trying to evaluate FreeSWITCH so I'll be able to support both Flash and WebRTC users in a conference.

My use case is two types of conferences:
- A conference with up to 5 participants all speaking.
- A conference with 20-30 participants with 2-3 speakers.

What part require more CPU?
- Encryption (DTLS-SRTP)
- Transcoding
- Mixing

Does FS mix a channel separately for each participant or can it reuse the same mix?
For example, if I have 1 speaker and 20 listeners, 10 with speex and 10 with opus, does FS produce 2 streams or 20?

Does FS exclude silent participants from the mix to save CPU?

Thanks