> I'm trying to set up a secure audio conferencing system. I can see two
> 1. Jitsi only, with one Jitsi user running the conference.
Yes. This is the most painless approach since it doesn't require
anything on the server (other than free transit for the media streams).
It does however, place a requirement on minimum CPU and bandwidth
availability at the conference host.
Not sure which "server" you mean, but in general, there would be no media server at all,
right? Each participant does a register with some SIP registrar like getonsip.com,
and then the SRTP media streams go direct between participants, not through
any central media server at all.
The only required server is for the SIP, not for media streams.
Though of course some participants (particularly in corporate networks) might have a far
more complicated setup within their own intranet.
> 2. Jitsi + Freeswitch, with Freeswitch running the conference and being a trusted MITM.
This is indeed a valid option but I have no personal experience with it
so I can't really comment.
> For the "Jisti only" solution I could imagine there being some problems with the
> host user running out of bandwidth, and/or having problems doing direct peer-to-peer
> connections to other users through their NAT.
Our SIP support does not currently use ICE and relies entirely on
transparent server-side relaying / latching. I wouldn't expect any NAT
traversal issues since most SIP servers do this properly.
Here I assume by "server" you mean the SIP registrars such as getonsip.com or whatever each participant
We do have ICE with XMPP so you can also use that for conferencing.
You'd have to host a Jingle Nodes relay or a TURN server but you'd still
be handling less server-side traffic than with ICE-less SIP.
But we'd also give up the security of ZRTP, right?
Most modern PCs and broadband subscriptions should be enough for hosting
Jitsi managed conf calls for < 10 users.
For a "Jitsi only" SIP conference, which codec will they negotiate, with no settings changes?
What is the (symmetric) bandwidth required for media per call with that codec?