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