[OpenSIPS-Users] Call forking, branches, Record-routing

[Karsten Wemheuer]

Hi,

using record_route(), lookup("location) and t_relay() seems to work,
but not in all cases.
I have the 3 use cases:
1) phones reachable in local network via UDP
2) phones reachable in local network via TLS
3) phones in public internet behind NAT via TLS

The last case works except if you end the call from the phone. The
Record-Route header contains the IP address of the LAN not the address
reachable by the phone. The BYE doesn't reach the proxy.

Is it possible to change this, maybe somewhere in the branch route?
(Sometimes a call is directed towards phones of cases (2) and (3) in
parallel).

Thanks in advance,

Karsten


Am Donnerstag, dem 31.03.2022 um 17:22 +0200 schrieb Karsten Wemheuer:
> Hi Bogdan-Andrei,
>
> many thanks for Your help.
>
> I tried with record_route. It doesn't work for me, as I set
> "advertised_address" and "advertised_port" to the natted address of
> the
> (only) interface. I wasn't able to avoid this. It seemed to be
> required
> to be able to reflect the path "phone -> proxy -> pbx".
>
> I removed the "advertised"-stuff and checked again the call with
> record_route. Now this seems to work.
>
> I think, I have to fix the other call flow to avoid the global
> setting
> of the advertised address and port.
>
> Best regards,
>
> Karsten
>
> Am Donnerstag, dem 31.03.2022 um 17:44 +0300 schrieb Bogdan-Andrei
> Iancu:
> > Hi Karsten,
> >
> > See my prev email, just to record_route() before the t_relay() for
> > the
> > initial INVITE. And the loose_route() stuff for whatever
> > sequential/in-dialog requests.
> >
> > Best regards,
> >
> > Bogdan-Andrei Iancu
> >
> > OpenSIPS Founder and Developer
> >    https://www.opensips-solutions.com
> > OpenSIPS eBootcamp 23rd May - 3rd June 2022
> >    https://opensips.org/training/OpenSIPS_eBootcamp_2022/
> >
> > On 3/31/22 2:50 PM, Karsten Wemheuer wrote:
> > > Hi*,
> > >
> > > I have a understanding problem regarding branches and call
> > > forking.
> > > A call from a PBX is to be routed to phone(s) via OpenSIPS. The
> > > phones
> > > are registered to OpenSIPs.
> > >
> > > INVITE --> lookup ----> 1. Destination
> > >                     |
> > >                     \--> 2. Destination
> > >
> > > When the call is terminated by the caller, the BYE request shall
> > > take
> > > the same path. Currently, the BYE is sent from the PBX directly
> > > to
> > > the
> > > Contact URI (which is not reachable by the PBX).
> > >
> > > Is it possible to use record_route in the branch_route so that
> > > different record route headers are used? Or is there another way?
> > >
> > > Thanks in advance,
> > >
> > > Karsten
> > >
> > >
> > > _______________________________________________
> > > Users mailing list
> > > Users at lists.opensips.org
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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