[SR-Users] RTPproxy in bridge mode question

[Uriel Rozenbaum]

Daniel,

Maybe my question is silly, but in this case the contact should remain
intact? (I mean in bridge mode).
I understand the destination UA should read the Record-route Headers and
ignore the contents of the Contact Header, but I think this is not what´s
happening.

I'm not using force_socket because the gateway already knows how to route
the calls and I'm detecting the outgoing interface before calling
force_rtp_proxy with flags.

Should I replace the contact using REGEX?

Thanks,
Uriel

On Fri, Apr 30, 2010 at 3:03 PM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>  Hello,
>
> it might not be the solution, because they should route based on
> Record-Route headers, not on Contact header. Anyhow changing the Contact
> will break the routing, so you will need to store somehow the original
> contact.
>
> You can do manual detection in case you do bridging, by checking the
> receiving interface, $Ri is the local IP where the request was received,
> therfore you will be sending on the other interface. Are you doing force
> send socket to select outgoing interface? If yes, then is where you know the
> local ip for sending.
>
> Cheers,
> Daniel
>
>
>
> On 4/30/10 6:32 PM, Uriel Rozenbaum wrote:
>
> Guys,
>
> I'm successfully using a Kamailio + RTPproxy setup in bridge mode with most
> of my Gateways. My setup includes two different interfaces one with a public
> IP and teh other with the private IP.
>
> Now I'm facing some slight issue. Some providers won't accept my calls (or
> calls will have some strange behavior) if the Contact header has an IP out
> of immediate range.
>
> I tried to use fix_nated_contact() function but as per my topology, this
> function will not change the contact header because the IP is already the
> one on the interface.
>
> Example:
> U 192.168.200.X:5060 -> 192.168.200.Y:5060
> INVITE sip:111160911097 at 192.168.200.Y SIP/2.0.
> Via: SIP/2.0/UDP 192.168.200.X:5060;branch=z9hG4bK096baacc;rport.
> From: "Uriel Rozenbaum" <sip:60911100 at 192.168.200.X><sip:60911100 at 192.168.200.X>
> ;tag=as32794d5e.
> To: <sip:111160911097 at 192.168.200.Y> <sip:111160911097 at 192.168.200.Y>.
> Contact: <sip:60911100@*192.168.200.X*>.
>
> U 200.A.A.A:5060 -> 200.B.B.B:5060
> INVITE sip:898960911097 at 200.B.B.B SIP/2.0.
> Record-Route: <sip:200.A.A.A;r2=on;lr=on;ftag=as32794d5e>.
> Record-Route: <sip:192.168.200.Y;r2=on;lr=on;ftag=as32794d5e>.
> Via: SIP/2.0/UDP 200.A.A.A;branch=z9hG4bK5222.14fbf4f7.0.
> Via: SIP/2.0/UDP
> 192.168.200.X:5060;received=192.168.200.X;branch=z9hG4bK096baacc;rport=5060.
> From: "Uriel Rozenbaum" <sip:60911100 at 192.168.200.X><sip:60911100 at 192.168.200.X>
> ;tag=as32794d5e.
> To: <sip:111160911097 at 192.168.200.Y> <sip:111160911097 at 192.168.200.Y>.
> Contact: <sip:60911100@*192.168.200.X*>.
>
> Is there any way to let know Kamailio the outgoing IP I'll be using and fix
> the contact accordingly?
> I can trigger this change after I know the destination IP.
>
> Thanks!
> Uriel
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierla
> * http://www.asipto.com/
> * http://twitter.com/miconda
> * http://www.linkedin.com/in/danielconstantinmierla
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20100430/ff45fc9f/attachment.htm>