I’m migrating from Asterisk 1.6.2 to 10.7.0. In 1.6.2, I made a small patch to allow specifying an address for RTP media. That worked. In
10.7.0, this appears to be built in with “media_address”, but it doesn’t work for me.
My Asterisk server has multiple addresses, all global address on two different /24’s with different routing policies via BGP. I’m connecting to a phone that’s over NAT. I have “nat=yes” in the “general” section of sip.conf. Everything works fine with the default.
But if I specify media_address to be the Asterisk server’s address on the other /24, I get one-way audio. I can see with “sip debug” that the proper address is being given in the SDP data. Audio from the phone is fine. Audio *to* the phone starts out with maybe 1-2 seconds of very garbled audio, then goes quiet.
Running traceroute shows that data comes from the phone *to* Asterisk on the desired /24, but goes out with a source address from the other /24 (the default address). I’m not sure if this is the problem or not, but in any event, I think the source address for RTP should be the one in
“media_address” and want it that way for my purposes anyway. Is there a way to configure this to happen? If not, where should I look to make a patch? And is this likely the reason for the one-way audio or is something else the likely cause?