[OpenSIPS-Users] one way audio problem

[aamir chougule]

Hi Nandini,

The parameters and modules that you need to turn ON in your opensips.cfg file:

loadmodule "nat_traversal.so"

The above line load the module and the given below paragraph will set to test the parameters.


route[nat_check] {
    if (client_nat_test("3")) {
        force_rport();
        fix_contact();
        nat_keepalive();
    }
}

Everytime you route a call first test the calls through the route(nat_check) which will fix all the NAT handling parameters.

For e.g. if you are gonna route INVITE request then you need to do it like this:

    if(is_method("INVITE")) {
        route(invite_requests);
        exit;
    }


route[invite_requests] {
    route(nat_check);

    if(!lookup("location")) {
        sl_send_reply("404", "User Not registered");
        exit;
    }
        t_on_reply("user_reply");
        t_relay();
    exit;
    }


Its just an example that how I do it and always you can explore things and read the modules provided by OpenSIPS and upgrade yourself to use this server in all possible cases.
 
Regards,

Aamir Chougule
Cell: 09167989111



________________________________
 From: Aamir <aamir_ryu at yahoo.com>
To: OpenSIPS users mailling list <users at lists.opensips.org> 
Sent: Tuesday, 7 May 2013 1:58 PM
Subject: Re: [OpenSIPS-Users] one way audio problem
 

Hi Nandini,

You actually need to turn on the nat_traversal module I guess, will pass you the parameters if I get time to do so.

--Aamir
--- Sent from My BlackBerry ---

-----Original Message-----
From: sermj 2012 <sermj2012 at gmail.com>
Sender: users-bounces at lists.opensips.org
Date: Tue, 7 May 2013 13:49:04 
To: <users at lists.opensips.org>
Reply-To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: [OpenSIPS-Users] one way audio problem

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