[asterisk-dev] Audio to/from Asterisk

[Seán C. McCord]

Just as a mention, though it uses my AudioSocket rather than what George is
talking about, I do have a complete example of bidirectional communication
with Google TTS and speech-to-text.

Https://GitHub.com/CyCoreSystems/asterisk-k8d-demo


On Thu, Aug 1, 2019, 14:44 Mohit Dhiman <mohitdhiman736 at gmail.com> wrote:

> This is getting really interesting now because just a few days ago I
> started exploring about how can I get a continuous media stream (without
> blocking the channel in the Dialplan) out of Asterisk and feed it to the
> Google ASR engine.
> I hope this feature comes out soon as it will be a massive help towards my
> project.
>
> Thanks
> Mohit Dhiman
>
>
> On Thu, 1 Aug 2019 at 23:48, George Joseph <gjoseph at digium.com> wrote:
>
>>
>>
>> On Thu, Aug 1, 2019 at 12:10 PM George Joseph <gjoseph at digium.com> wrote:
>>
>>>
>>>
>>> On Thu, Aug 1, 2019 at 9:56 AM marek <cervajs64 at gmail.com> wrote:
>>>
>>>> is there someone who can write/share small HOWTO test it with
>>>> https://cloud.google.com/speech-to-text/  ?
>>>>
>>> You won't be able to use the new capability directly with Google or any
>>> other public speech to text service provider as they all have different
>>> access mechanisms and protocol constraints.  We also wouldn't know what to
>>> do with the returned transcription.  Instead, you'd write an ARI
>>> application using the technology of your own choosing to act as the proxy
>>> between Asterisk and your chosen service provider.  Most of the service
>>> providers have api toolkits to help with that.  What you then do with the
>>> returned transcription is up to you.
>>>
>>
>> Actually, I just talked to the boss (Matt Fredrickson) and we agreed that
>> we could provide a bare-bones example ARI app to talk to Google.
>>
>>
>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>> Dne 01/08/2019 v 16:54 George Joseph napsal(a):
>>>>
>>>>
>>>>
>>>> On Thu, Aug 1, 2019 at 7:36 AM Joshua C. Colp <jcolp at digium.com> wrote:
>>>>
>>>>> On Thu, Aug 1, 2019, at 10:28 AM, George Joseph wrote:
>>>>> > So here's where we're at with adding this capability...
>>>>> >
>>>>> > Initial release:
>>>>> >  * Two new ARI endpoints, one on channel and one on bridge:
>>>>> >    * /channels/<channel_id>/externalMedia
>>>>> >    * /bridges/<bridge_id>/externalMedia
>>>>>
>>>>> What do these return? How do you stop external media at a future time?
>>>>>
>>>>
>>>> They'd return an ExternalMedia object which would contain an ID along
>>>> with other pertinent data that can be gleaned from the underlying
>>>> provider.  For chan_rtp, it could be the local IP address and local port.
>>>> To stop the streaming, you'd make a DELETE  request on the ExternalMedia
>>>> resource.
>>>>
>>>> This is similar to how we do Playback and Record today.
>>>>
>>>>
>>>>
>>>>
>>>>>
>>>>> --
>>>>> Joshua C. Colp
>>>>> Digium - A Sangoma Company | Senior Software Developer
>>>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>>>>> Check us out at: www.digium.com & www.asterisk.org
>>>>>
>>>>> --
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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
>>>>
>>>>
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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
>>>
>>>
>>
>> --
>> *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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>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
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>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>
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