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

[Kai Vehmanen]

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!