[*] SIP and nat

[Bill Reid]

Sean Walberg wrote:

> What about a local peer (ie IP phone) for which Asterisk is setting up a 
> call? Is there a way to have Asterisk use externip in the SDP info so 
> that the RTP from the phone builds the NAT channel?  It seems that 
> despite the NAT setting on the peer, Asterisk always sends the real 
> address in SDP.  The only way around this is canreinvite=no to force 
> native bridging, or to have the FW fix up the SDP info. So the nat=xxx 
> setting is only for RTP streams terminating on the * box itself

Am I correct in understanding that this is a call between two local phones 
behind the Asterisk server? If that is the case then I suspect what you are 
saying is correct. Things seem to work the best to always have Asterisk in the 
RTP stream.

-- Bill