How to get audio stream properties (format and sample rate) of my VoIP provider?

Question

I need to integrate Speech To Text (STT) service with IP telephony. STT service support just 5 particular input formats of audio (MuLaw/ALaw/Linear16/RawOpus/MP3) and 10 sample rates (8/11/12/16/22/24/32/44/48/96k).

How to check which format and sample rate does my VoIP provider use?

Can I do that with linphone(linphone-cli) or pyVOIP or SIP.js or pjsip?

I will be grateful for code or commands examples

Answer 1

Your provider doesn't use anything by itself. The question is what the devices you will call will support.