I haven’t yet understood all the concepts in audio streaming world, but my current understanding is that I need a plain RTP server.
Something like this:
The telephone PBX is Asterisk and it has a capability to stream the phone call over RTP. The most straight forward way to connect my service to the PBX would be to use SIP protocol and act like a endpoint device (VoIP telephone). But this comes with the disadvantage that my service is a SIP client and I need to connect from the service to the PBX, not the other way around. So I cannot really create a server application to which multiple Asterisks can connect.