[asterisk-dev] Audio to/from Asterisk

Wed Jul 24 08:18:57 CDT 2019

On Mon, Jul 22, 2019 at 2:01 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.
>

Hmmm.   Why can't the nodejs process handling the call open a socket for
listening then provide that in the URI sent to Asterisk in the ARI call?
I.E.  tcp://node_host:12345

>
> 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 --
>>>>>
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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 --
>>>
>>> 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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