On 07/15/2015 08:44 AM, Alberto Sagredo wrote:
Hi Alberto,
can you also share part of the relevant place where you are calling that route?
Cheers, Roberto Fichera.
Hi Daniel
Kamailio is for hard people and fun :)
Thanks Visily i finnaly got it working with your tip. You were right about internal external options instead direction=...
Here its some code to someone could need it
route[RTPPROXY] {
if (is_method("INVITE")){
if(ds_is_from_list(1)){
if (is_ip_rfc1918("$si")) { if (sdp_get_line_startswith("$avp(mline)", "m=")) { #!ifdef WITH_RTPENGINE if ($avp(mline) =~ "SAVP") { xlog("L_INFO", "We got SRTP "); rtpengine_manage("trust-address internal external replace-origin
replace-session-connection ICE=remove ");
return; } #!endif if ($avp(mline) =~ "AVP") { xlog("L_INFO", "We got RTP "); #!ifdef WITH_RTPPROXY set_rtp_proxy_set("1"); rtpproxy_manage("fwei"); start_recording(); #!endif #!ifdef WITH_RTPENGINE rtpengine_manage("trust-address internal external replace-origin
replace-session-connection ICE=remove ");
#!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", "We got SRTP "); rtpengine_manage("external internal replace-origin replace-session-connection
ICE=remove RTP AVP");
return; } #!endif if ($avp(mline) =~ "AVP") { xlog("L_INFO", "We got RTP "); #!ifdef WITH_RTPPROXY set_rtp_proxy_set("1"); rtpproxy_manage("fwie"); start_recording(); #!endif #!ifdef WITH_RTPENGINE rtpengine_manage("external internal replace-origin replace-session-connection
ICE=remove RTP AVP");
#!endif } } } }
}
2015-07-14 18:46 GMT+02:00 Daniel Tryba <d.tryba@pocos.nl mailto:d.tryba@pocos.nl>:
On Tuesday 14 July 2015 18:19:02 Alberto Sagredo wrote: > In my tests rtpproxy recording waste less resources than asterisk > > That was one of the reasons How much time have you spend so far on a problem that asterisk can handle out of the box? ;) I'd love to do this with kamailio/rtpengine (I don't record), but sofar the blunt quickfix is to use asterisk. I needed a transcoder anyway and handling RTP/SRTP conversions when either endpoint needs it is simple. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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