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

[johan]

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)? 
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20210917/478e8279/attachment.html>