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<p>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. <br>
</p>
<p><br>
</p>
<p>Bottom line, in my opinion, you need to have 2 separate streams
before you can start STT. <br>
</p>
<p><br>
</p>
<p>wkr, <br>
</p>
<p><br>
</p>
<div class="moz-cite-prefix">On 17/09/2021 11:04, Mark Allen wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CADaqb1to7yyVB=2sudGYO9Oek3eue4jzhyHaSLB=PtYPuEtdpw@mail.gmail.com">
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<div dir="ltr">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:
<div><br>
</div>
<div>- a call is recorded, and when the call ends an event is
triggered to initiate transcription of the recording</div>
<div>- a call starts, the RTP is forked to the STT engine which
sends real-time transcription<br>
<div><br>
</div>
<div>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)? </div>
</div>
</div>
<br>
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