[OpenSIPS-Users] fix_nated_sdp() not taking effect
Callum Guy
callum.guy at x-on.co.uk
Tue Nov 19 08:18:25 EST 2019
You might want to read up on bridge mode, it allows you to meet the finely
control which interface is presented during the SDP rewrites.
All of the information on the various use cases is available in the module
docs, I've used both successfully including some pretty complex request
routing.
The move to offer/answer with interface specifications works great, you'll
just have to fire off the offer with different params when in the failure
route so it will override your initial public/public selection from the
initial invite processing
On Tue, 19 Nov 2019, 12:27 Mark Farmer, <farmorg at gmail.com> wrote:
> Hi Răzvan
>
> My OpenSIPS/RTPProxy box has 2 interfaces, public(NAT) - for phones &
> internal - for Asterisk.
> The issue is that if a call from one registered user to another is
> rejected & goes to failure_route() then I send the call to an Asterisk box
> for voicemail which is connected via the internal interface.
>
> When the call is routed to Asterisk, I need the RTP to flow between
> RTPproxy & Asterisk on the internal interfaces so I need to have the SDP
> correct before it hits Asterisk. RTP to & from the phone needs to use the
> public interface.
>
> Initial media flow:
> phone<-->OpenSIPS/RTPproxy<-->phone
>
> Voicemail media flow:
> phone<-->OpenSIPS/RTPproxy<-->Asterisk
>
> What is the best way to achieve this?
>
> Many thanks!
> Mark.
>
>
> On Mon, 18 Nov 2019 at 12:50, Răzvan Crainea <razvan at opensips.org> wrote:
>
>> Yes, the problem is definitely the fact that you are calling
>> `rtpproxy_offer()` for the initial invite. Hence, when you run
>> `fix_nated_sdp()`, you're trying to change the same IP once again - this
>> is not possile in OpenSIPS.
>> But I wonder why you need the `fix_nated_sdp()` if you are using
>> RTPProxy. Can't you just use the `ip_address`[1] field to advertise the
>> proper IP int he c= line.
>>
>> [1]
>>
>> https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#func_rtpproxy_offer
>>
>> Best regards,
>> Răzvan
>>
>> On 11/13/19 1:51 PM, Mark Farmer wrote:
>> > Hi everyone
>> >
>> > In my failure_route I'm routing to an Asterisk box for voicemail & I
>> > need to change the SDP c/o parameters to use the correct internal IP
>> > address but using fix_nated_sdp() is not taking effect.
>> >
>> > if (t_check_status("486|408|603")) {
>> > xlog("CUSTOM_LOG: User replied $T_reply_code - Routing
>> > to Asterisk Voicemail service.");
>> > prefix("VMR_");
>> > rewritehostport("10.150.50.53:2404
>> > <http://10.150.50.53:2404>");
>> > force_send_socket(udp:10.150.50.51);
>> > fix_nated_sdp(10,"10.150.50.51");
>> >
>> > if (!t_relay()) {
>> > send_reply(500,"Internal Error");
>> > }
>> > exit;
>> > }
>> >
>> > I get the CUSTOM_LOG entry so I know that the route is executing.
>> >
>> > Maybe I'm doing something wrong with the flags, I've tried:
>> > fix_nated_sdp(2,"10.150.50.51");
>> > fix_nated_sdp(8,"10.150.50.51");
>> > fix_nated_sdp(10,"10.150.50.51");
>> >
>> > But when I examine the SDP in the resulting invite, the c/o parameters
>> > are never changed.
>> > I'm using rtpengine_offer/answer in the initial routing, could it be
>> > related to that?
>> >
>> > I'm using OpenSIPS 3.0.1
>> >
>> > Best regards
>> > Mark.
>> >
>> >
>> >
>> > _______________________________________________
>> > Users mailing list
>> > Users at lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >
>>
>> --
>> Răzvan Crainea
>> OpenSIPS Core Developer
>> http://www.opensips-solutions.com
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> --
> Mark Farmer
> farmorg at gmail.com
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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