Hi,
On Tue, 7 Mar 2006, Mike Taht wrote:
In particular I wanted to make it easier to do a call in radio station
(see rivendell) and integrating the voice to mp3 function and music to
mp3 function struck me as asterisk with jack as a natural bridge...
If ladspa plugins could be run through asterisk or a jack compliant
sip phone you could give your outgoing voice calls a little bass boost
for that "voice of god" effect...
I'm currently working with the sofia-sip project and we have a couple of
test clients that use gstreamer for media and rtp. Gstreamer's JACK
support is unfortunately not quite up to date, but at least it has LADSPA
support, and an architecture that makes it easy to add new processing
elements to the voice send/receive chains.
- http://sofia-sip.sourceforge.net/
- http://sofia-sip.sourceforge.net/cgi-bin/twiki-bin/view/SofiaSIP/SofiaApplications
.. see the Gaim SIP plugin, Farsight/Telepathy and Tapioca apps
These are so far just clients of course, but you can use them to connect
to asterisk or to conference bridges (or use the libraries to build
your own and/or enhance asterisk).
Btw, in true ecasound-fashion ;), I'm working on a state-of-art,
console-mode SIP VoIP/IM client that uses gstreamer for RTP and media (and
hopefully JACK support at some point):
- http://sofia-sip.sourceforge.net/cgi-bin/twiki-bin/view/SofiaSIP/SofSipCli
stuff like that. It doesn't make sense to me that these two worlds -
telephony and professional audio - should be seperated.
True, lot of the audio routing and processing functionality can be used in
telephony apps, while transport of media over wide area networks is of
interest to the audio crowd as well.
--
http://www.eca.cx
Audio software for Linux!
