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