[Asterisk-Dev] Jackd and Asterisk

[Steven]

On Wed, 2005-08-31 at 14:17 +0300, Tzafrir Cohen wrote:
> On Tue, Aug 30, 2005 at 11:17:44PM -0500, Steven wrote:
> > On Tue, 2005-08-30 at 20:52 -0700, Mike Taht wrote:
> > > I am curious if anyone has tried to create a jackd (jack audio
> > > connection kit) <-> asterisk plugin? It looks like a straightforward
> > > way (with admittedly a lot of up and downsampling) to interface odd
> > > things into asterisk's sound processing loop  - graphically watch
> > > sound quality on a conference for example - mix in a little extra bass
> > > response on an outgoing call (adding a sense of authoritay to the 
> > 
> > Jackd doesn't force up or downsampling onto anything itself. The trouble
> > you would run into is that to run any of the graphical portions would
> > require X. X isn't conducive to a well functioning asterisk machine. 
> 
> Does jack require X? Even so, an X app and the X server need not reside
> on the same computer. And it is the X server that competes with
> Asterisk, not the X app.

No, jackd doesn't require X, just graphical apps. Jackd doesn't even
need graphical apps to setup and configure. There are nice curses or cli
apps to handle what ever is needed in jackd. 

While the Xserver is usually a bad thing for asterisk as most people
want the GUI to be as responsive as possible, the xclients aren't all
that light if they are doing much in the way of screen drawing. Consider
the amount of compute load and network traffic you would generate if the
audio where passed through a scope or other visualization tool. 

My comments there where not to discourage the use of jackd, but rather
certain functions in it.

> Anyway, jack is not the only sound server. gstreamer provides similar
> functionality. esound probably perfoms not as well but may have better
> networking support in it?

Jack is more than a sound server. Jack is essentially an audio plumbing
tool. The idea being that not all sound being routed around the service
actually needs to find itself outputting to a sound card. Many outputs
could be routed to other applications and stop there. This is why I
think this is a good idea for asterisk. You could register up multiple
"ports" with jack and inside of asterisk, you could link those to the
functions necessary. Such as, ports for MOH coming in, ports for
intercom going out, maybe the call centers would like a monitoring out
port that they could connect to.

gstreamer doesn't provide anything like this, and esound is tied to a
sound card. with a jack.udp transport, I don't think network support is
a problem.

-- 
Steven Critchfield
critch at basesys.com
KI4KTY