[OpenSIPS-Users] OpenSIPS and Speech-to-Text

[Mark Allen]

Thanks for that Johan - I hadn't thought about that aspect. All theoretic
at the moment, but IBM Voice Gateway, at least, does claim to be able to
handle it using SIPREC - so maybe they are confident about their ability to
differentiate between caller and callee in a single stream?...

"The voice gateway provides the ability to transcribe caller and callee
> (e.g. contact-center agent) audio from an active phone call in real time
> using the SIPREC protocol." -
> https://www.ibm.com/docs/en/voice-gateway?topic=gateway-about-voice
>


On Fri, 17 Sept 2021 at 10:33, johan <johan at democon.be> wrote:

> The issue with siprec (based on rtpproxy) is that you have only 1 stream
> containing the voice from caller to callee and callee to caller. So that
> will give a hard time on the ASR :-).  I do know that rtpengine has
> something similar to siprec but I don't know the details.
>
>
> Bottom line, in my opinion, you need to have 2 separate streams before you
> can start STT.
>
>
> wkr,
>
>
> On 17/09/2021 11:04, Mark Allen wrote:
>
> I'm just starting to look at Speech-to-Text (STT) processing for calls -
> initially recordings but moving on to real-time. I would see this working
> along the lines of either:
>
> - a call is recorded, and when the call ends an event is triggered to
> initiate transcription of the recording
> - a call starts, the RTP is forked to the STT engine which sends real-time
> transcription
>
> I can see that with OpenSIPS, the SIPREC and Media Exchange modules allow
> for forking of the RTP, providing a means of sending the data for
> processing, but is anybody actually doing this? If so, what has been
> your experience? Is there a toolset that works well with this (e.g. IBM
> Voice Gateway, Google, Amazon etc)?
>
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