[asterisk-dev] Audio to/from Asterisk

Matt Fredrickson creslin at digium.com
Thu May 16 15:36:12 CDT 2019


On Wed, May 15, 2019 at 7:13 PM Sylvain Boily <sylvain at wazo.io> wrote:
>
> Hello,
>
> On 2019-01-25 4:14 p.m., Sylvain Boily wrote:
> > Hello,
> >
> > On 2018-10-16 3:18 p.m., Sylvain Boily wrote:
> >> Hello,
> >>
> >> On 2018-10-15 3:29 p.m., Matt Fredrickson wrote:
> >>> On Tue, Oct 9, 2018 at 12:09 PM Seán C. McCord <ulexus at gmail.com>
> >>> wrote:
> >>>> Because several people raised the issue at DevCon, I figured it may
> >>>> be worth mentioning this: app_audiosocket.  I haven't submitted it
> >>>> mainly due to the thought that no one else would fine it
> >>>> interesting.  There exist other, similar ways to get audio out:
> >>>> app_jack, app_unimrcp, etc.  I built this because of some special
> >>>> needs, and it is very convenient due to its extremely light weight.
> >>>>
> >>>> Regardless, should anyone be interested, here it is:
> >>>>
> >>>> https://github.com/CyCoreSystems/audiosocket
> >>>>
> >>>> The idea is to create a TCP socket to somewhere, pass some
> >>>> extremely simple metadata (a UUID), and broker audio between the
> >>>> channel and the socket.  It is as simple as possible.
> >>> For those who aren't aware, getting this pushed out to the -dev list
> >>> was an AstriDevCon 2018 takeaway action item with regards to interop
> >>> with web-based speech recognition APIs.  I'd love to see more
> >>> discussion and work on this topic, as I think that there stands much
> >>> to be improved in Asterisk to better interoperate with some of the
> >>> major speech recognition vendors.
> >>>
> >>
> >> Will be nice to have this on ARI, like GET
> >> /channels/channelId/stream. We can help to develop this feature!
> >
> > We did a 3 days Wazo hackathon this week and we developed a module to
> > get audio from a channel_id to a websocket in asterisk.
> >
> > Our project has been to get a realtime voice communication, send it to
> > an STT and with the result to prioritize a call in a mini emergency
> > call center before someone get the call. The source code of this
> > project is on my github. [1]
> >
> > It works well but it's a proof of concept (no test). I will be nice to
> > have input, tests and other suggestions to put it on Asterisk in the
> > future. Actually, the concept is you open a websocket with a
> > subprotocol channel-stream and a Channel-ID http header with the
> > channel_id of the channel you want to have the stream. The module use
> > audiohook in Asterisk, transcode and send it in PCM 16k to the websocket.
> >
> I talked with Matthew at the Kamailio World 2019 about this module and
> he said to me George will start to work on this feature. Do you have
> feedback or comment about this module?

Hey Sylvain,

Sorry, sounds like we had a bit of a misunderstanding.  George did
some research on prospective architectures around this functionality,
but we have not committed to or engaged upon any work on it at this
time.  I did mention that George might have some new perspective on
any submitted implementation or continued discussion due to some of
the information he gained during his research though :-)

Best wishes, and sorry about any confusion in our conversation.

-- 
Matthew Fredrickson
Digium - A Sangoma Company | Asterisk Project Lead
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA



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