I wonder under what conditions asterisk switches from simple_bridge to native_rtp and doing route optimization?
I have 2 nodes with exactly the the same asterisk configs.
Only difference is that they are connected to different SIP-trunks (2 different voip providers)
One does route optimization and the other one does’nt.
Same codec, same packetization, no features that need audio, DTMF method the same. Those are the general ones. If you enable debug in logger.conf it’ll tell you why it’s not allowing it when the decision is made.