On Wed, Jul 24, 2019 at 7:11 AM Dan Cropp <d...@amtelco.com> wrote: > Out of curiosity, would this be an alternative to unimrcp’s asterisk > support for MRCP (TTS/ASR)? >
Well it wasn't intended to implement MRCP but yes, it's intended to provide the same very-high-level functionality controlled via ARI. > > > > > *From:* asterisk-dev <asterisk-dev-boun...@lists.digium.com> *On Behalf > Of *Luca Pradovera > *Sent:* Monday, July 22, 2019 3:12 AM > *To:* Asterisk Developers Mailing List <asterisk-dev@lists.digium.com> > *Subject:* Re: [asterisk-dev] Audio to/from Asterisk > > > > Hello, > > I remember this being talked about, and it's essentially tied to the > mechanism that would allow streaming ASR/TTS services to be used. > > +1 on this feature! > > > > On Mon, Jul 22, 2019 at 10:01 AM Dan Jenkins <d...@nimblea.pe> wrote: > > Also coming back to this with some real-life case issues I'm currently > facing and why I can't use audiosocket :( > > > > I need to be able to ask the ARI/AGI/AMI for an IP/port combo and for an > external app to then connect into asterisk rather than asterisk connecting > to a URI elsewhere. Lets say I already have a nodejs (or any other > language) process taking care of controlling that call via ARI or even AGI > (all the different ways) - I need that same process to handle the media I'm > sending and receiving to/from asterisk and so if I already have that > process up and then Asterisk calls out to a generic URI then that media > will never make it back to the original nodejs process. > > > > I think its of upmost importance that I be able to ask asterisk for a > host:port pair and then be able to connect to that port from my external > application. > > > > What do people think? I thought we'd talked about this mechanism at devcon? > > > > Dan > > > > On Sat, Jul 20, 2019 at 2:39 PM Dan Jenkins <d...@nimblea.pe> wrote: > > Just going to chime in and say I don't see a one way audio solution as > enough and I'd worry that doing that would lead to "oh but only so many > people need to chuck audio in". This has been discussed at at least 3 > devcons now. > > > > On Thu, Jul 18, 2019 at 2:09 PM Seán C. McCord <ule...@gmail.com> wrote: > > I certainly don't mind if a better-designed system comes along and > obviates my AudioSocket implementation. I built it because I needed it. > However, bidirectional audio flow is critical for me (speech synthesis, > external interfacing, real-time processed audio, processed injections, > etc). While I would actually prefer a system which was a bit beefier than > my own (simple protocol aside, there's a good deal of range between my > protocol and MRCP), my meagre C skills (and patience) don't allow me to > venture forth into those difficult waters. > > > > I do like the separate connection (unlike Wazo's) and the support of TLS > (unlike mine)... and yours is certainly (even without looking) more > performant. Mine also probably needs a multi-threaded, dedicated-receiver > approach like most of the other channels which handle socket-received > media, rather than the simple non-blocking I/O with null frame insertion. > No perfect solution yet. > > > > > > > > On Thu, Jul 18, 2019 at 8:01 AM George Joseph <gjos...@digium.com> wrote: > > Hey Guys, > > > > I was on vacation when this thread happened but I'm also working on this > now. The implementation uses the existing ARI channel and bridge recording > endpoints ands add the ability to specify a URI in the form of > (udp|tcp|tls)://hostname:port to stream the media. This makes use of the > existing chan_bridge_media driver and only requires a scheme handler > similar to Seán's res_audiosocket. This implementation is more targeted > to real-time speech recognition/transcription/captioning and is therefore > one way (outbound). A future enhancement is planned that would send each > channel in a bridge as a separate audio channel in a multi-channel > container. > > > > I'm not suggesting that this should replace Seán's audiosocket stuff but I > did want to let you know what was in the pipeline. > > > > george > > > > On Fri, Jul 5, 2019 at 7:38 AM Seán C. McCord <ule...@gmail.com> wrote: > > Solutions such as Jack are non-network oriented and severely limited in > scalability. There are, of course, many other options, but the closest are > chan_rtp and chan_nbs. RTP is a good option except for the difficulty for > non-telephony people to interact with it. NBS is deprecated, undocumented, > and unsupported by any locatable resources. > > > > While the original app interface from last year required dialplan, the > channel interface does not. It is a plain channel which can be used by ARI > directly. > > > > > > On Fri, Jul 5, 2019, 15:28 Sylvain Boily <sylv...@wazo.io> wrote: > > Hello Seán, > > On 2019-07-05 4:45 a.m., Seán C. McCord wrote: > > A brief update: > > > > I have adapted my app_audiosocket from last year to become > chan_audiosocket, a full bidirectional audio channel interface for Asterisk > to any AudioSocket service (which itself is a dead-simple implementation). > I'll be demoing it in my talk next week at CommCon, for anyone who might be > interested. I'm going to try to have it ready to push to gerrit for review > this weekend. > > > I'll be there. > > > > For now, you can see it in the 'channel' branch of > github.com/CyCoreSystems/audiosocket. > > > This is very different from what we did. You need dialplan to use it. In > our case we don't need any dialplan to use it, it's more ARI approach. > > Sylvain > > -- > _____________________________________________________________________ > -- 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 > > > > > -- > > *George Joseph* > > Digium - A Sangoma Company | Software Developer | Software Engineering > > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > > direct/fax: +1 256 428 6012 > > Check us out at: https://digium.com · https://sangoma.com > > > > > > > -- > > Seán C. McCord > > ule...@gmail.com > > CyCore Systems > > -- > _____________________________________________________________________ > -- 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 > > -- > _____________________________________________________________________ > -- 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 -- *George Joseph* Digium - A Sangoma Company | Software Developer | Software Engineering 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct/fax: +1 256 428 6012 Check us out at: https://digium.com · https://sangoma.com
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