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
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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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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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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)
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