[OpenSIPS-Users] Forcing reinvite for t38 on different pstn gw

[Alex Balashov]

Uwe Kastens wrote:

> Is it possible to handle reinvites in that way, that I can send them to
> a special pstn gw? This looks a little tricky, since I need to drop the
> 1st invite.

No, that would not be in the slightest bit compatible with SIP protocol 
mechanics as described per the RFC.  The initial INVITE establishes the 
dialog, and without that initial request there cannot be sequential 
in-dialog requests - and therefore, no re-INVITEs.

In-dialog requests must be routed to the dialog peer that was 
established by the initial INVITE;  you can't route them somewhere else 
instead once the dialog has been established.

The following scenario plays out - and even then, only if Record-Route 
is turned on and sequential requests flow through the proxy:


    A                       Proxy                           B

    ---- INVITE 123 at Proxy ---->
    <------- 100 Trying -------
                              -------- INVITE 123 at B -------->
                              <-------- 100 Trying ----------
                              <---------- 180 Ringing -------
    <------ 180 Ringing -------
                              <---------- 200 OK w/Contact URI
    <------ 200 OK + contact --
    -------> ACK contact at B --->
                              ---------- ACK contact at B ----->
    ----- INVITE contact at B --->
                              ------- INVITE contact at B ----->


The only way you can pull this off is to decide in advance whether the 
call needs to go through a special PSTN GW when routing the initial INVITE.


-- 
Alex Balashov
Evariste Systems
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