route[NATMANAGE] {
#!ifdef WITH_NAT
# if (is_request()) {
# if(has_totag()) {
# if(check_route_param("nat=yes")) {
# setbflag(FLB_NATB);
# }
# }
# }
# if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))){
# return;
# }
# route(RTPPROXY);
route(RTPPROXY);
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
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>:
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.
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_______________________________________________ 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
_______________________________________________
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