[*] SIP and nat

[John Lange]

My impression is that nat= does all sorts of weirdness.

Among other things I believe it causes Asterisk:

- to ignore what IP address the device claims to be in the SIP headers.
- to do asymmetric RTP. (Asterisk will not send RTP until it gets at
least one RTP packet from the endpoint first)

And other stuff.

John

On Sat, 2006-12-09 at 18:00 -0600, Bill Reid wrote:
> Sean Walberg wrote:
> > If a peer is marked as "nat=yes", does this mean that Asterisk will use 
> > the externip value in the SDP packets when setting up the media path 
> > between the Asterisk server and the peer?  - Yes, testing shows this.
> > 
> I suspect you are right. I just reconfigured a phone last night and it was not
> working. It was on the same network as the server but the conf file file was
> marked nat=yes.  I could not register but could receive calls but no audio. I
> did not bother tracing packages but I think what you are saying would explain
> what I was seeing.
> 
> SIP and NAT is certainly not handled well by Asterisk. Asterisk should be able
> to NAT by looking at the data stream. NAT works correctly if youi have guessed
> correctly. I see that Ollie is writing a SIP3 which wil hopefully address this
> and other issues.
> 
> -- Bill
> 
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