[asterisk-dev] Audio to/from Asterisk

Mon Jul 22 03:01:11 CDT 2019

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
>>>>>
>>>> --
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>>>
>>>
>>>
>>> --
>>> *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 --
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>>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
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