On Wed, May 15, 2019 at 7:13 PM Sylvain Boily <sylv...@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 <ule...@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 -- _____________________________________________________________________ -- 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