How do you configure sip.conf and extensions.conf for you to receive voice and video. using asterisk 13
This is when you use nat=, normally force_rport,comedia.
You may need to make adjustments on the router and the phones as well.
force_rport causes Asterisk to respond to the port number from which it actually received the signalling, rather than the one from which the request claims to have come. comedia causes it to wait for incoming media before it sends any media, and to use the address and port in the actual media for the reverse path.