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<font face="monospace">Hi Mark,<br>
<br>
But using the media_exchange you can "fork" (as a new SIP call)
only one of the RTP streams - of course, the TTS engine should be
able to accept pure SIP calls.<br>
<br>
Regards,<br>
</font>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="https://www.opensips-solutions.com">https://www.opensips-solutions.com</a>
OpenSIPS eBootcamp 2021
<a class="moz-txt-link-freetext" href="https://opensips.org/training/OpenSIPS_eBootcamp_2021/">https://opensips.org/training/OpenSIPS_eBootcamp_2021/</a></pre>
<div class="moz-cite-prefix">On 9/17/21 3:50 PM, Mark Allen wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CADaqb1sp+oOCFQc0kQSa6KTNZ-PKXvObmd_8pir7ij5u3s8hUw@mail.gmail.com">
<meta http-equiv="content-type" content="text/html; charset=UTF-8">
<div dir="ltr">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?...
<div><br>
</div>
<blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">"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." - <a
href="https://www.ibm.com/docs/en/voice-gateway?topic=gateway-about-voice"
moz-do-not-send="true">https://www.ibm.com/docs/en/voice-gateway?topic=gateway-about-voice</a><br>
</blockquote>
<div> </div>
</div>
<br>
<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">On Fri, 17 Sept 2021 at 10:33,
johan <<a href="mailto:johan@democon.be"
moz-do-not-send="true">johan@democon.be</a>> wrote:<br>
</div>
<blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div>
<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>On 17/09/2021 11:04, Mark Allen wrote:<br>
</div>
<blockquote type="cite">
<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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