After experimenting and reaching the point where we can send and receive samples from a native app running, it seems feasible to use WebRTC to create gateway to bridge calls. Using the native API this can be done without forking using a webRTC audio device module. We can already demo the work in progress with using an audiodevicemodule to play and record.

The second leg of the call bridge could potentialy be implemented using WebRTC, by using the API to disable RTCP MUX, DTLS and controlling ICE. another alternative using MediaStreamer2/oRTP could be used.

The signaling gateway part will be rudimentary and and example using a provided rudimentary Kamailio module is provided.