[asterisk-users] reinvite

[Jeff LaCoursiere]

On Mon, 9 Feb 2009, Jeff LaCoursiere wrote:

>
> I've never used "reinvite" in systems I have installed to date, and I have 
> finally run across a situation where it would be preferred.
>
> A remote office has a flaky Internet connection.  With G729 encoding the 
> calls to the central office over the 'net are tolerable.  One Linksys 2102 
> drives two phones at this location, and when the first one calls the second 
> one it travels to the central office and back, which is no longer tolerable.
>
> For each sip peer I have "canreinvite=yes", but I am a bit confused as to the 
> correct options on the 2102 to use this feature.  Is anyone doing this with 
> 2102s that can give me some pointers?
>

I have been playing around with this in my "lab" and cannot seem to make 
it work as expected.

I have a remote asterisk server on a public IP - 1.4.22-3 on Centos 5.

I have two Polycom IP501s on a local LAN behind a NAT gateway.

Both Polycom's register with the remote server and can call each other 
without issues.

Both SIP contexts have nat=yes, canreinvite=yes.  The caller is 223, the 
callee is 222.

eth0 is the outside (public) interface, XXX is my dynamic IP.

I trapped a conversation on the asterisk server with:

tcpdump -nli eth0 -s 0 -w /tmp/reinvite.debug host XXX and not port 22

While this was running I made a call between the two extensions for a few 
seconds then hungup.

I opened this capture in etherreal and can see the following:

223->AST  INVITE 222 at AST
AST->223  407 Proxy auth required
223->AST  ACK
223->AST  INVITE 222 at AST, with proxy-auth info
AST->223  100 Trying
AST->223  200 OK
223->AST  ACK

Then I see the RTP traffic begin back and forth.  I am confused on two 
fronts - first where is the INVITE from AST to 222?  Not sure how I missed 
capturing that side of the conversation.  And of course where is the AST 
"reinvite"?  It isn't occurring since I can clearly see the RTP traffic 
flowing via the asterisk server.

Any ideas?

Cheers,

j