[asterisk-dev] Audio to/from Asterisk

George Joseph gjoseph at digium.com
Wed Jul 24 09:03:11 CDT 2019


On Wed, Jul 24, 2019 at 7:11 AM Dan Cropp <dan at 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-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> 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> 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> 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> 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> 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> 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
>
> ulexus at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20190724/879d1aad/attachment-0001.html>


More information about the asterisk-dev mailing list