Hello friends
I want to pass real-time rtp stream to another server to convert audio to text
We have this service (voice to text conversion module) but i dont know is there any way in asterisk to pass rtp to that service during the call.
It would be so appritiated if you could help me.
You will need to use ARI to control your application. With that you can then use an external media channel to send the RTP stream to your audio server:
https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
There’s an example on how to do this using Javascript:
You can use several different languages with ARI, not just Javascript.
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