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

johan johan at democon.be
Fri Sep 17 09:30:35 EST 2021

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.


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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