Roberto just call it on NATMANAGE as shown. I route everything now thru
that route..
You have to configure rtpengine to use internal external interfaces
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;
}
2015-07-15 16:59 GMT+02:00 Roberto Fichera <ker...@tekno-soft.it>:
> 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.tr...@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
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> _______________________________________________
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> sr-users@lists.sip-router.org
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>
>
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