Audio pipe to external program

I am in a very similar situation regarding both

  1. my telephony expertise and
  2. that I want to use Asterisk solely for sending/receiving the caller’s audio to/from an external server as a stream (realtime audio passing i.e. no recordings) - meaning the external server could basically be replaced by a human on another phone.

From what I’ve read so far, I also get the impression that a lot of approaches out there aim to solve either speech-to-text (STT) or text-to-speech (TTS) but not both (STS). For me, it’s also unclear which interface should be used for what purpose…

Which of the following approaches is the easiest/best for Asterisk for STS?


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