Alberto Sagredo-2 wrote > 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@ > >: > >> 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! >> >> >> >> -- >> View this message in context: >> http://sip-router.1086192.n5.nabble.com/Issue-handling-SRTP-and-RTP-with-rtpproxy-and-rtpengine-tp139488p139521.html >> Sent from the Users mailing list archive at Nabble.com. >> >> _______________________________________________ >> 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 >> > > _______________________________________________ > 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 What about ICE, where it has to work? (client->Kamailio - yes, Kamailio->Asterisk - no) or somehow else. For your description, I think you need to add something like this: - Kamailio -> Asterisk rtpengine_manage("...............RTP/AVP"); ///// this will change profile to RTP/AVP - Asterisk -> Kamailio rtpengine_manage("...............RTP/SAVPF"); ///// this will make backward changes Also read thoroughly the meaning and usage of "direction" parameter, I think you have little misunderstanding of how it works (maybe I'm wrong and you use it as it has to be, but re-read it anyway) -- View this message in context: http://sip-router.1086192.n5.nabble.com/Issue-handling-SRTP-and-RTP-with-rtpproxy-and-rtpengine-tp139488p139556.html Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users