[linux-audio-user] Re: GNU Audio Community Conference Room

[Esben Stien]

Dan Mills <dmills at spamblock.demon.co.uk> writes:

> A sip <-> jack "hybrid" would be way cool, but while that covers the
> audio side of the problem, it leaves the call setup and control
> side.

The way this now with freeswitch is that you send messages to the
server. No, not using OSC, though I have mentioned it to them;). They
are using a simple text protocol over Jabber. With this you can
transfer your call to a conference room, then call other participants
and raise and lower their volumes, f.ex, just by sending these
messages. (bind them to a midi controller)

> Perhaps a daemon that could connect multiple sip "lines" to jackd
> and provided a couple of fifos to communicate line status and to
> handle dialing?

This is not really necessary as you really only need one bus to send
and one to receive, using ardour;) covering the mic and different
feeds with a midi controller (like feed0, feed1) if you do all
call management on the server, by sending these messages.

-- 
Esben Stien is b0ef at e     s      a             
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