[Users] forcing rtpproxy on a call

Vitaly Nikolaev vnikolaev at intermedia.net
Fri Mar 10 16:23:01 CET 2006


Add

 

if (!method=="REGISTER") record_route();    

 

Somewhere in ur route (before relay but after loose_route)

 

do u have loose_route anyway ?

 

________________________________

From: users-bounces at openser.org [mailto:users-bounces at openser.org] On
Behalf Of Script Head
Sent: Thursday, March 09, 2006 6:44 PM
To: users at openser.org
Subject: Re: [Users] forcing rtpproxy on a call

 

Now that my rtpproxy actually passes traffic I stumbled on another
problem. When the called party hangs up the call (asterisk command
Hangup()) the soft phone remains connected. Yet, when I click the Hangup
button on the softphone, SER receives BYE messages. 

On 3/9/06, Vitaly Nikolaev <vnikolaev at intermedia.net> wrote:

Looks like forward includes relay in it. And by putting force_rtpproxy
AFTER forward you does not give it a chance :-) on_reply route is also
MUST be there.

 

 

________________________________

From: users-bounces at openser.org [mailto: users-bounces at openser.org
<mailto:users-bounces at openser.org> ] On Behalf Of Script Head
Sent: Thursday, March 09, 2006 12:59 PM
To: users at openser.org
Subject: Re: [Users] forcing rtpproxy on a call

 

Thank you guys, it's working now.

Apparently, rewritehostport("<ip>:<port>") works great with rptproxy
while forward does exactly that, forwards the call to the destination
bypassing the force_rtp_proxy request. This should be documented
somewhere. 

ScriptHead

On 3/9/06, Vitaly Nikolaev <vnikolaev at intermedia.net> wrote:

1.	I never used forward, see my example, I do not know if it
actually relay call or not
2.	if you do not have NAT between client and server you do not need
force_rport, and try to avoid any nat_uac_test, etc if you are actually
working on private ips without nat
3.	you MUST enable proxy also for reply

 

route[x] {

 

.....

 

force_rtp_proxy();

t_on_reply("1");

rewritehostport("x.x.x.x:5060");

if (!t_relay()) {

                sl_reply_error();

};

}

 

 

onreply_route[1] {

        if (!(status=~"183" || status=~"200"))

                break;

        force_rtp_proxy("");

}

 

________________________________

From: users-bounces at openser.org [mailto: users-bounces at openser.org
<mailto:users-bounces at openser.org> ] On Behalf Of Script Head
Sent: Wednesday, March 08, 2006 6:29 PM
To: users at openser.org
Subject: [Users] forcing rtpproxy on a call

 

Hello everyone,

I am trying to debug why my rtpproxy isn't working. I have the following
setup, on my LAN. 

softphone (192.168.1.100) -> openser/rtpproxy ( 192.168.1.10
<http://192.168.1.10> ) -> asterisk (192.168.1.12)



The rtpproxy is running and I see commands flying thru it.

the following route works

        if(method=="INVITE") { 
                if(uri=~"^sip:[0-9]{6}1[0-9]*{10}@") {

                        forward(192.168.1.12,5060);


                };
         }

when I replace it with this route 

        if(method=="INVITE") {
                if(uri=~"^sip:[0-9]{6}1[0-9]*{10}@") {

                        forward(192.168.1.12,5060);


                }; 
                force_rport();
                force_rtp_proxy();
        }

I get dead air while asterisk logs show that my test message is playing.
How should I proceed to debug this?

ScriptHead

 

 

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