[asterisk-dev] Audio to/from Asterisk

Wed Jul 24 07:52:20 CDT 2019

Out of curiosity, would this be an alternative to unimrcp’s asterisk support for MRCP (TTS/ASR)?

From: asterisk-dev <asterisk-dev-bounces at lists.digium.com> On Behalf Of Luca Pradovera
Sent: Monday, July 22, 2019 3:12 AM
To: Asterisk Developers Mailing List <asterisk-dev at 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 <dan at nimblea.pe<mailto:dan at 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 <dan at nimblea.pe<mailto:dan at 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 <ulexus at gmail.com<mailto:ulexus at 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 <gjoseph at digium.com<mailto:gjoseph at 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 <ulexus at gmail.com<mailto:ulexus at 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 <sylvain at wazo.io<mailto:sylvain at 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<http://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
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