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: [email protected] Web: www.qu.tlabs.tu-berlin.de On Fri, Jul 18, 2014 at 11:44 AM, Rolf-Werner Eilert < [email protected]> 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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
