Thanks Vasily i have changed a little today using a RTPPROXY route.

Thats what i have right now

But its not working as expected

What i try is to detect if i have SAVP from endpoint and translate to RTP to ASterisk an later RTP from ASterisk translate to SRTP using rtpengine

I had extrange behaviour using rtpproxy that send SRTP to Asterisk and i have SRTP calls, i though rtpproxy 2.0 could not manage SRTP calls. but it pass it to Asterisk

Using RTPengine i have tested with rtpproxy_manage as you see and also with rtpengine.

If i load both start_recording() feature is lost.

On rtpengine (behind NAT) im using it as:

INTERFACES="192.168.0.178 internal/192.168.0.178 external/192.168.0.179!EXTERN_IP



On NATMANAGE route i call directly

route(RTPPROXY);


Hope this helps


-----


route[RTPPROXY] {

        if (is_method("INVITE")){

if(ds_is_from_list(1)){

                if (is_ip_rfc1918("$si")) {

                                xlog("L_INFO", "LLamada desde los Asterisk_$si -> RTPPROXY\n");

                        if (sdp_get_line_startswith("$avp(mline)", "m="))

                        {

                                #!ifdef WITH_RTPENGINE

                                if ($avp(mline) =~ "SAVP")

                                {

                                xlog("L_INFO", "Tenemos SRTP ");

                                xlog("L_INFO", "Llamada entre Extensiones -> RTPENGINE INTERNAL");

                                rtpengine_manage("direction=internal replace-origin replace-session-connection ICE=remove");

                                return;

                                }

                                #!endif


                                if ($avp(mline) =~ "AVP")

                                {

                                xlog("L_INFO", "Tenemos RTP ");

                                xlog("L_INFO", "Llamada entre Extensiones -> RTPROXY ");


                                #!ifdef WITH_RTPPROXY

                                 set_rtp_proxy_set("1");

                                rtpproxy_manage("fwei");

                                start_recording();

                                #!endif


                                #!ifdef WITH_RTPENGINE

                                set_rtp_proxy_set("2");

                                rtpproxy_manage("ie");

                                #!endif

                                }

                        }

                        }

               }else if(!ds_is_from_list()){


                        if (sdp_get_line_startswith("$avp(mline)", "m="))

                        {

                                 #!ifdef WITH_RTPENGINE

                                 if ($avp(mline) =~ "SAVP")

                                {

                                xlog("L_INFO", "Tenemos SRTP ");

                                xlog("L_INFO", "Llamada entre Extensiones -> RTPENGINE EXTERNAL ");

                                rtpengine_manage("direction=external replace-origin replace-session-connection ICE=remove");

                                return;

                                }


                                #!endif

                                if ($avp(mline) =~ "AVP")

                                {

                                xlog("L_INFO", "Tenemos RTP ");

                                xlog("L_INFO", "Llamada entre Extensiones -> RTPROXY ");


                                #!ifdef WITH_RTPPROXY

                                set_rtp_proxy_set("1");

                                rtpproxy_manage("fwie");

                                start_recording();

                                #!endif


                                #!ifdef WITH_RTPENGINE

                                set_rtp_proxy_set("2");

                                rtpproxy_manage("ei");

                                #!endif


                                }

                        }



                }

      }


}



2015-07-14 14:24 GMT+02:00 Vasiliy Ganchev <vasiliy.ganchev@wildix.com>:
Alberto Sagredo-2 wrote
> ...
> I have been able to make SRTP To RTP to Asterisk
>
> But im not able to call between SRTP extensions, i understand also SRTP to
> RTP would work as im doing with Asterisk (Only the speak SRTP as rtpengine
> trasncode)
>
>
> If you need any more info let me know.
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list

> sr-users@.sip-router

> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Hi!
If you make SRTP to RTP to Asterisk, you possibly will need vice versa
conversion (when request coming from Asterisk to client with SRTP).

Can you describe the logic of test case: (UA-A (SRTP) --> Kamailio (make
SRTP->RTP) .... etc.

Because your explanation is difficult to understand.

Cheers!



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