We are trying to get live audio stream from extensions. Our initial solution was to follow this tutorial to setup sip devices or webphones, however, we were informed that there's limit on number of webphones and hence this is not scalable. Now we are following this tutorial to setup SIP over TCP and RTP (digital lines) but the problem is we are working with python and there's no viable python library available which works with RTP. Is it possible to setup these digital lines over SIP and WebRTC?