Hello Klaus,
I have changed the rtpproxy_manage() in the route(NATMANAGE) to no avail. The
c= in the Invite that arrives to Asterisk still points to the public IP of the
Phone1.
Code is as follows:
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
if (src_ip == ip.pbx.priva.te) {
rtpproxy_manage();
} else {
rtpproxy_manage("","ip.kamailio.priva.te");
}
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
The next test has been to comment out the rtpproxy_manage at NATMANAGE function
and to put it both at route[RELAY] and onreply(route) following your post in
this list from January 2013:
http://lists.sip-router.org/pipermail/sr-users/2013-January/076254.html.
Now the media flows from Phone1 to Kamailio, from Kamailio to Asterisk and
back, but it gets stuck at Kamailio. I cannot see it flow towards the public IP
of the Phone2.
The force_send_socket you used could be of any use here?
Thank you
> ----- Original Message -----
> From: Klaus Darilion
> Sent: 01/21/14 07:25 AM
> To: Kamailio (SER) - Users Mailing List
> Subject: Re: [SR-Users] Kamailio behind NAT
>
> Yes, when calling rtpproxy (whatever function you use,
> manage/offer/answer/force), every function accepts as second parameter
> the IP address which should be written into the SDP c=... line.
>
> So, detect the direction (eg based on source-ip) and put the respective
> internal or external IP address into the SDP.
>
> Of course the NAT must be a static 1:1 NAT without any PAT.
>
> Btw: you have the same problem with some SIP headers, like Record-Route
> or Via. Kamailio has to insert the respective public/private IP,
> depending on where to send the message. If Kamailio uses only a single
> socket, then you have to do this manually. The automatic approach is to
> configure 2 listen=... sockets. One for the internal side, and one for
> the external side. On the "external" socket you add the "advertise"
> option - then Kamailio will automatically put the correct IP addresses
> in RR and Via, eg:
>
> # public IP address of Kamailio/rtpproxy: 3.3.3.3
> # internal socket
> listen=udp:10.10.0.2
> # external socket
> listen=udp:10.10.0.3 advertise 3.3.3.3:5060
> mhomed=0
>
> You could use separate subnets for the 2 sockets, or you could also use
> only a single IP address with 2 different ports for the 2 sockets.
>
>
> regards
> Klaus
>
>
> On 21.01.2014 14:50, John Smith wrote:
> > The intended setup is as you described in option a.
> >
> > And the behaviour of rewriting SDP using private/public ports is what I
> > understood the rtpproxy_manage was tasked to do.
> >
> > As it is not working the way I expected, what is the best way to proceed?
> > To check if the IP is from the outside and then rewrite via rtpproxy_offer
> > in the NATMANAGE block?
> >
> > Thank you
> >
> >> ----- Original Message -----
> >> From: Klaus Darilion
> >> Sent: 01/21/14 05:25 AM
> >> To: Kamailio (SER) - Users Mailing List
> >> Subject: Re: [SR-Users] Kamailio behind NAT
> >>
> >> On 21.01.2014 13:24, John Smith wrote:
> >>> I might be making wrong assumptions regarding this traffic flow. Is that
> >>> correct?
> >>
> >> That depends on your policy. It is up to you to define how RTP should be
> >> routed. There are basically 2 choices:
> >>
> >> a) RTP from clients is handled by rtpproxy:
> >>
> >> phone1 <-nat-> rtpproxy <--> Asterisk <--> rtpproxy <-nat-> phone2
> >>
> >> In this case, only the private IP of Kamailio and rtpproxy (can be the
> >> same IP address) must be mapped to a public IP address.
> >>
> >>
> >> b) RTP directly to Asterisk:
> >>
> >> phone1 <-nat-> Asterisk <-nat-> phone2
> >>
> >> In this case, the private IPs of Kamailio and Asterisk must be mapped to
> >> a public IP address.
> >>
> >>
> >> When using version a) you have to make sure to set the proper IP address
> >> in the SDP. For example, SDPs in messages from Kamailio to the phone
> >> must contains the PUBLIC IP of rtpproxy in the c=... line. SDPs in
> >> messages from Kamailio to Asterisk must contain the PRIVATE IP of
> >> rtpproxy in the c=... line.
> >>
> >> regards
> >> Klaus
> >>
> >> _______________________________________________
> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> >> sr-users@lists.sip-router.org
> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >
> >
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> > sr-users@lists.sip-router.org
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >
>
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> sr-users@lists.sip-router.org
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