Making media flow

Now that the SIP messages are flowing properly from FreeSWITCH to the phone, what about the media? A phone call or a videocall is not very eventful if you can't even hear each other, right? We had many problems where the calls would set up properly until the point where NAT would strike the RTP packets that provide the actual media of the call, rendering the call with one-way-audio or even no-way-audio in some cases. In light of this injustice, we created a separate feature that is always enabled and only needs to be manually disabled in a very few set of cases inspired by the fourth pitfall. This feature is called RTP auto-adjust. The reason we need it is because when the phone tries to call us from behind NAT, it will naively advertise its unreachable LAN private address to FreeSWITCH as to where to send the media streams (eg the address written in the...