8 Mar
2006
8 Mar
'06
12:20 p.m.
Dan Mills <dmills(a)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@e s a
http://www. s t n m
irc://irc. b - i . e/%23contact
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