[asterisk-dev] Question about interface to sound processing library

[Dennis Guse]

Hi Rolf,

there is a nice option available in Asterisk.
Asterisk supports JACK [1], which is basically a virtual local audio
routing system that allows to send audio data from one application on the
same PC running as the same user. Using a audio processing like Puredata
[2] realtime audio processing / manipulation is quite easy.
Basically Puredata gives you a graphical programming interface for realtime
processing where you can create a small UI to apply or not-apply some
degradations.

At the moment I have configuration that can to bandpass filtering,
white+pinknoise and simulates packet-loss by adding 20ms empty frames (so
no PLC). In addition delay is quite easy to add here.

With the upcoming release of Asterisk 13, the JACK-interface is extended to
support more than 8Khz.

Best regards,

Dennis Guse

PS: To listen in such a call ChanSpy is quite useful.

[1] http://jackaudio.org/
[2] http://puredata.info/



Kind regards

Dennis Guse

Quality and Usability Lab
Telekom Innovation Laboratories
TU Berlin
Ernst-Reuter-Platz 7
D-10587 Berlin, Germany
Tel: +49 30 8353 58874
Fax: +49 30 8353 58409
E-mail: dennis.guse at telekom.de
Web: www.qu.tlabs.tu-berlin.de


On Fri, Jul 18, 2014 at 11:44 AM, Rolf-Werner Eilert <
eilert-sprachen at t-online.de> wrote:

> Hi folks,
>
> I hope I'm right here in this list. Tried to ask this as a general
> question in forum General first, but there was only a vague answer. Here I
> expect to find the guys who make the core functions of Asterisk, so I ask
> my question again.
>
> To describe the reason for my question: We are running a school for
> foreign languages training students for office communication. This includes
> telephoning in foreign languages. Up to now, we provide a simple one-box
> unit with a wireless phone for the person that leaves the classroom and a
> loudspeaker for the class to listen.
>
> There was the idea of building a telephone system that allows to simulate
> telephone calls to far destinations and to cellphones offering kind of
> simulating distortions typical to such calls (delays, cracks, echos, or
> scenarios like "cellphone at a busy street café" :D ).
>
> With Asterisk, my phantasy goes to offer the trainer an easy interface
> (which I could program myself) to choose line quality, scenarios etc. and
> to have a telephone in every room to call from and to be called. One might
> even simulate international numbers...
>
> My idea is to use some interface to the voice processing modules of
> Asterisk to be able to let them remodulate the sound stream (e. g.
> decreasing the sampling rate for 200 ms or so, then going back to normal to
> simulate typical GSM distortions like organ-like noises) or to mix in
> pre-recorded noises like from a street etc.
>
> Has anyone here ever seen something like this with Asterisk, or are there
> any plugins/modules you would consider worth taking a look at? I am new to
> Asterisk, so I don't know what to look for.
>
> Let me add this: There is a Linux server running 24/7 and a terminal for
> the teachers in every classroom.
>
> Thanks for reading up to here - and thanks a lot for all your ideas!
>
> Rolf
>
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