[Asterisk-Users] A solution for SIP and NAT

[John Todd]

You may be correct about the Via: header, but you're incorrect in the 
concept as to how it relates to Asterisk, notably in your reversal of 
what side of the transaction is putting data in the Via: header to 
make SIP work correctly.

This is cluttering up the list.  Talk to me off line if you want a 
better understanding of how NAT and SIP work with Cisco devices.

Again, for those of you who might be trying to figure out what the 
result of this conversation is:  SIP clients behind NAT works fine in 
both directions (incoming and outgoing calls), Asterisk makes it 
work, it's not using STUN.  Cisco devices work especially well.

JT


At 8:58 PM -0400 7/1/03, Michael Kane wrote:
>
>Your correct, Cisco devices stuff the WAN address in the Via: header which
>in turn allows the proxy to correctly register the UA for an incoming call
>attempt to that UA.  If Mark is mentioning STUN as I said before, the only
>devices I'm aware of are the SNOM 100 and Grandstream 101.  These devices
>rely on an external mechanism to properly construct the Via: header
>otherwise the proxy has the incorrect return IP address of the UA.
>
[snip]