SIP + dst nat == fix up headers?

[jflemingeds]

Does anyone know if the server side of a SIP session should have headers rewritten? I have a src and dst nat ( i know, asking for trouble ). I can see the client headers are being rewritten correctly, but the server side isn't. For example the contact: field in the message header has the server's real address and not the nat IP of the the server. To be clear the IP (layer 3) of the server was correctly natted.

If i can't get this to work right i think i'm going to have to do a double vpn. Haven't done that before but its all setup in a lab replication so i have a play ground setup.


assuming i'm goin double vpn this is what i'm thinking.

phone fw1 < -VPN1 -> fw2 < - VPN2 - > sip server

where vpn1 is really across the internal network and VPN2 is across the inet.

I'll take my answer off air, thanks!

[jflemingeds]

So in my case the destination is a proxy as all the phones call the voip server and thus a dest nat. Once support cleared that up definition they pointed me back to the nat restrictions SK and it says its not supported (could be bidirectional or proxy nat).