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

[Mark Allen]

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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