[Freeswitch-users] Web sockets and gRPC modules?

[Anthony Minessale]

You would want to probably do it real-time by using a media bug similar to
how mod_flite and mod_pocketsphinx are designed.



On Mon, Aug 15, 2016 at 11:37 AM, Michael Jerris <mike at jerris.com> wrote:

> this would have to be implemented as a module, similar to mod_unimrcp or
> mod_pocketsphix, that implements these protocols
>
>
> On Aug 15, 2016, at 12:14 PM, Leonardo Lima Ribeiro <llribeiro90 at gmail.com>
> wrote:
>
> Michael, thank you for the response!
>
> Actually I was thinking about copy the function “recordFile” but instead
> of save a file in the system etc, just the streams to my STT service…
>
> Example of usage in LUA:
> text = session:asrApi(“IBM_WATSON”, max_len_secs, silent_threshold,
> silence_secs, keywords)
>
> Which “IBM_WATSON” points me to a XML conf file (eg.
> conf/asrapi/IBM_WATSON.xml) which have all the settings needed to stream to
> that API like auth token, endpoints urls, supported protocols (initially
> web sockets and gRPC) etc.
> max_len_secs, silent_threshold, silence_secs - exactly like the current
> recordFile function params.
> keywords - optional field, some of these apis can use a list of words to
> improve the accuracy and speed of recognition...
>
> For now, if my approach is correct, I found the following functions, to
> have an idea on how to start:
> switch_ivr_play_say.c - switch_ivr_record_file
> switch_cpp.cpp - CoreSession::recordFile
>
> So yeah, I've setup already a development environment for me, following
> the instructions on (Creating New Modules - Confluence page).
> To be honest it’s my first time developing on C, so if you have good
> sources to improve my learning curve on “how to create a module in
> FreeSWITCH” would be awesome!
>
> Can you give me a feedback on my approach and also the
> directions/instructions in what modules/functions should I use to do that?
>
> Thank you,
> Leo
>
> On Aug 15, 2016, at 11:33 AM, freeswitch-users-request@
> lists.freeswitch.org wrote:
>
> we would welcome for review someone submitting an module to support
> these.. it should not require a new module format, we already have those
> for speech recognition.. it would require a module that implemented the
> interfaces.
>
>
> On Aug 14, 2016, at 5:47 AM, Leonardo Lima Ribeiro <llribeiro90 at gmail.com>
> wrote:
>
> Hello guys!
>
> Is there any modules that we can use to make ASR using IBM Bluemix Watson
> Speech To Text (which accepts web sockets for streaming recognition) or
> Google Speech to Text API (which uses gRPC for streaming recognition)?
>
> If there are none, do you think it’s worth it to build them (like in a new
> module format)? I know the current standard way is to use unimrcp, but
> unfortunately I could not get good results for my business cases using
> Sphinx, and the good services appears to be pricey… Said that, I was
> thinking about build modules that allows us to easily integrate to Google
> STT and IBM Watson STT Apis, which are the most popular and accessible
> services...
>
> Thank you,
>
>
>
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-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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