[OpenSIPS-Users] loose routing question

[Robert Dyck]

I did not describe the scenario accurately. I should have reviewed the issue 
before composing my response.

The problem arises when the asterisk does a re-INVITE. If the UA receiving the 
re-INVITE includes the original R-R list in the 200OK then asterisk will 
route the ACK correctly. If the UA does not include the R-R list in the 200OK 
the asterisk will create a null routeset and the ACK will be misrouted. The 
spec says that a UAS may include R-R in the in-dialog response but the UAC 
must ignore it. The same logic suggests that the lack of R-R should not be 
interpreted as a null route set.

I am sorry about the confusion. You are quite right. The UA does not send a 
R-R in a request.

On Thursday 05 March 2009, Iñaki Baz Castillo wrote:
> 2009/3/4 Robert Dyck <rob.dyck at telus.net>:
> > Twinkle as one example however does not
> > send R-R with the in-dialog request.
>
> Hi, no one SIP device sends R-R, neither in initial INVITE or
> re-INVITE. R-R (Record-Route) must be add just be proxies, and UAS
> must copy them in the 1XX/2XX responses to the *initial* INVITE.
> When creating an in-dialog request, the SIp device must insert Route
> headers, no Record-Route.
>
> > Asterisk will then create a null route set.
>
> I assume you mean that the *proxy* doesn't add R-R for in-dialog
> request (the correct behaviour) so Asterisk fails and re-set the route
> set.
>
>
> Regards.