[OpenSIPS-Users] losing subsequent requests - loose routing question?

[Iñaki Baz Castillo]

2009/3/4 Brett Nemeroff <brett at nemeroff.com>:
> Ok, I'm really confused..
>
> My carrier, who is running some variant of OpenSER says that I have to
> change the value of my "Contact:" header to indicate the IP of my own
> server. Otherwise subsequent requests within the dialog (ie: BYE) will
> not go via me.

This is the typicall reply coming from the ignorance. Don't trust him at all.


> Well I thought this was the purpose of record-route? no? I am record
> routing the INVITE that establishes the dialog. Isn't that good
> enough?

SURE


> I was under the impression that messing with the contact header will
> break all sorts of dialog matching. Any ideas out there?

A proxy SHOULD NOT change the Contact. Well, this is false since in
case the request comes from NAT (with no STUN and so) the Contact will
be a private address, so it won't be reachable for sending in-dialog
requests. Then, the proxy replaces the private Contact address with
the real reecived public address, but this is not the proxy address of
course.

Your carrier has no idea of how SIP works.


-- 
Iñaki Baz Castillo
<ibc at aliax.net>