WebSocket or WebTransport support to stream audio to/from Asterisk (for transcription, AI, TTS, etc)

[daninmadison]

Improvement request.
Using ARI ExternalMedia and Snoop channels to stream audio between Asterisk and another IP:Port works, but it has a significant limitation for cloud and other solutions. The RTP audio is not encrypted and there are scenarios where setting up VPN between Asterisk and the cloud runs into problems.

Describe the solution you'd like
Ideally, a way to use WebSocket or possibly WebTransport over QUIC to be able to stream audio to/from Asterisk to a box in the cloud. Make sure the data is encrypted to meet security and HIPAA requirements. Not sure how feasible it would be via the dial plan, but it would be nice to be able to establish and end this via ARI. WebTransports would provide some possible future expansion.

Describe alternatives you've considered
We initially started down the ARI ExtenalMedia and Snoop channel approach. We we having the audio streamed to a box living on the same network to get around some security issues. This box would then pass the audio to/from the cloud. It works, but it's extra hops for the audio.

Now we are looking at SIP options. Instead, we would bridge the audio in asterisk with a SIP call to the cloud. Cloud and Asterisk will pass audio both directions via RTP. As you know, SIP is complex.

Additional context
Entire tech industry is rushing toward AI. The ability to stream the audio out of the Asterisk box (may be on premise or in the cloud) to platforms in the cloud requires securing the audio.

[voicecomms]

Truly this would be a good feature to have in asterisk box but in a simplified manner.