[asterisk-users] Native bridging to SIP endpoints on the same NAT'd network

Hi, I have the following setup: Asterisk - Nat - Internet - Nat - 2 x SIP endpoints With directmedia=no I can make a call between the two SIP endpoints; the RTP stream being passed through the Asterisk box. Obviously, this is sub-optimal. I attempted to enable bridging of the call between the

[asterisk-users] RTP stream when * and Xlite are both behind Nat devices.

Hi, I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone behind a different NAT network. Asterisk - Nat - Internet - Nat - Softphone. I can register my softphone to the asterisk box ok via SIP but the RTP stream from the asterisk box is addressed to the private