one way audio problem

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one way audio problem

Nandini
i configured and installed opensips successfully,i have registerd two clients:-

6005 :- 192.168.2.48
6006:- 192.168.2.50

i can hear only one way audio.iam using wifi standalone router to communicate with clients.i have searched in the blogs to slove the problem.
by seeing the blogs i came to know that rtptproxy would slove problem.

i have integrated opensips with rtpproxy module successfully,
but still the problem is not sloved.

iam attaching the sip trace file and opensips.cfg file.

please help me from this issue.

Thank you
Nandini

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Re: one way audio problem

aamir chougule
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 <[hidden email]>
Sender: [hidden email]
Date: Tue, 7 May 2013 13:49:04
To: <[hidden email]>
Reply-To: OpenSIPS users mailling list <[hidden email]>
Subject: [OpenSIPS-Users] one way audio problem

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




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Re: 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 <[hidden email]>
To: OpenSIPS users mailling list <[hidden email]>
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 <[hidden email]>
Sender: [hidden email]
Date: Tue, 7 May 2013 13:49:04
To: <[hidden email]>
Reply-To: OpenSIPS users mailling list <[hidden email]>
Subject: [OpenSIPS-Users] one way audio problem

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: one way audio problem

Nandini
Thanku very much for your prompt response,

iam new to opensips, please tell me where to add these lines in opensips.cfg

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


The above lines i have added in opensips.cfg under Routing logic,
when i start opensips server,iam getting errors,

please help me.

Nandini


On Tue, May 7, 2013 at 2:30 PM, aamir chougule <[hidden email]> wrote:
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 <[hidden email]>

To: OpenSIPS users mailling list <[hidden email]>
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 <[hidden email]>
Sender: [hidden email]
Date: Tue, 7 May 2013 13:49:04
To: <[hidden email]>
Reply-To: OpenSIPS users mailling list <[hidden email]>
Subject: [OpenSIPS-Users] one way audio problem

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: one way audio problem

aamir chougule
It should not be under main route block, just place it outside main route block.

For e.g.

route {

...

}

Place it here...

Or else paste your config file here.

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

From: sermj 2012 <[hidden email]>
Date: Tue, 7 May 2013 18:01:45 +0530
To: aamir chougule<[hidden email]>; OpenSIPS users mailling list<[hidden email]>
Subject: Re: [OpenSIPS-Users] one way audio problem

Thanku very much for your prompt response,

iam new to opensips, please tell me where to add these lines in opensips.cfg

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


The above lines i have added in opensips.cfg under Routing logic,
when i start opensips server,iam getting errors,

please help me.

Nandini


On Tue, May 7, 2013 at 2:30 PM, aamir chougule <[hidden email]> wrote:
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 <[hidden email]>

To: OpenSIPS users mailling list <[hidden email]>
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 <[hidden email]>
Sender: [hidden email]
Date: Tue, 7 May 2013 13:49:04
To: <[hidden email]>
Reply-To: OpenSIPS users mailling list <[hidden email]>
Subject: [OpenSIPS-Users] one way audio problem

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users