[SR-Users] Kamailio and NAT

Alexandru Covalschi 568691 at gmail.com
Mon Dec 28 18:17:14 CET 2015


AFAIK bye is usually sent to the address stored in record_route. Try
setting changing record_route() to
record_route_preset("PUBLICIP:5060;nat=yes:)

2015-12-23 16:28 GMT+02:00 Nelson Migliaro <eng.migliaro at gmail.com>:

>
> Hello,
>
> I am running Kamailio behind NAT.
>
> Kanailio has a private IP and I am relaying NAT to internet router.
>
> I am using:
>
> - #!define WITH_NAT
> - listen=udp:PRIVATE-IP:5060 advertise PUBLIC-IP:5060
>
> - Patched RTP proxy including the advertise option
>
> And everything goes fine. I can make calls and have two way audio.
>
> The problem begins when the callee ends the call. BYE is not received in
> Kamailio (caller)
>
> I included the public IP using "add_contact_alias" because
> "set_contact_alias" was not adding the public IP. I included this in in
> NATDETECT (pre loaded router)
>
>         if(is_first_hop()) {
>             xlog("L_NOTICE","Metodo: $rm \n");
>             xlog("L_NOTICE","is first hop\n");
>             #set_contact_alias();
>                  if (!add_contact_alias("PUBLIC-IP", "$Rp", "udp")) {
>                  xlog("L_ERR", "Error in aliasing contact $ct\n");
>                     send_reply("400", "Bad request");
>                             exit;
>                     }
>         }
>
> I think the problem is related to destination that BYE is sent by the
> vendor. From what I see IP and port is taken from advertised in contact
> (PUBLIC-IP and 5060).
> The problem is that internet router changes the source port.
>
> Contact: <sip:999999999 at PRIVATE-IP:5060;alias=PUBLIC-IP~5060~1>
>
> --- Is it correcto to add_contact_alias("PUBLIC-IP", "$Rp", "udp") in
> order to received new transactions or should I follow a different
> procedure???
>
> Thank you
>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
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>
>


-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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