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?
Thank you
Sylvain
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