[Users] forcing rtpproxy on a call

[Script Head]

I created a setup on my LAN with that works great and forwards BYE all the
way to the soft phone. I verified via tcpdump that RTP is flowing via the
proxy. When I move this setup to the public Internet, it works as well,
except the BYE isn't getting forwarded. The config file is attached.



On 3/10/06, Vitaly Nikolaev <vnikolaev at intermedia.net> wrote:
>
>  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 J on_reply route is also MUST
> be there.
>
>
>
>
>  ------------------------------
>
> *From:* 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] *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) -> 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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