Hi Mark,
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.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS eBootcamp 2021
https://opensips.org/training/OpenSIPS_eBootcamp_2021/
On 9/17/21 3:50 PM, Mark Allen wrote:
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
<https://www.ibm.com/docs/en/voice-gateway?topic=gateway-about-voice>
On Fri, 17 Sept 2021 at 10:33, johan <jo...@democon.be
<mailto:jo...@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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