<div dir="ltr"><div><div><div>Hi,<br><br></div>I'm currently using a RTMP server for audio conference.<br></div>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.<br>
<br></div><div>I'm trying to evaluate FreeSWITCH so I'll be able to support both Flash and WebRTC users in a conference.<br><br></div><div>My use case is two types of conferences:<br></div><div>- A conference with up to 5 participants all speaking.<br>
</div><div>- A conference with 20-30 participants with 2-3 speakers.<br><br><div>What part require more CPU?<br></div><div>- Encryption (DTLS-SRTP)<br></div><div>- Transcoding<br></div>- Mixing<br><br></div><div>Does FS mix a channel separately for each participant or can it reuse the same mix?<br>
</div><div>For example, if I have 1 speaker and 20 listeners, 10 with speex and 10 with opus, does FS produce 2 streams or 20?<br><br></div><div>Does FS exclude silent participants from the mix to save CPU?<br></div><div>
<br></div><div>Thanks<br></div><div><br></div><div><br><br></div></div>