[OpenSIPS-Users] Force rtp proxy

[michel freiha]

Dear Bogdan,

Do you mean doing something like that?

if(!cr_route("default", "0", "$rU", "$rU", "call_id")){
       sl_send_reply("403", "Not allowed");
     } else {
         # In cas of failure, re-route the request
          t_on_failure("1");

force_rtp_proxy();
          t_relay();
route(2) ;
}


Please let me know how can I force it on route2 for the 200OK reply

Thanks Bogdan


On Fri, Feb 27, 2009 at 11:54 AM, Bogdan-Andrei Iancu <
bogdan at voice-system.ro> wrote:

> Hi Michel,
>
> You have to call twice force_rtp_proxy() in order to complete the RTP
> session (and have RTP flowing). First for the INVITE (as you already do) and
> second for the 200 OK reply.
>
> So install a onreply_route and if the reply is 200 OK, call again
> force_rtp_proxy.
>
> Regards,
> Bogdan
>
> michel freiha wrote:
>
>> Dear All,
>>
>> I need to make all my rtp traffic through OpenSips to pass through rtp
>> proxy...I have the following route:
>>
>>
>> if(!cr_route("default", "0", "$rU", "$rU", "call_id")){
>>       sl_send_reply("403", "Not allowed");
>>     } else {
>>         # In cas of failure, re-route the request
>>          t_on_failure("1");
>> force_rtp_proxy();
>>          t_relay();
>>   }
>> The call is working fine but with no audio...How can i fix this issue in
>> order to have 2 way audio through rtpproxy?
>>
>> Regards
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.opensips.org/pipermail/users/attachments/20090227/e21906d1/attachment.htm