Hi!

I noticed a minor problem when I set up the STUN config of my phone.

When the server is set, I am unable to reach the endpoint from kamailio (Phone1 -> Kamailio -> Asterisk -> same Kamilio -> no outgoing SIP flow to Phone2).

If I do not set it, it seems to fallback to it's RFC1918 address which is then detected by nat_uac_test() and I can reach the phone.

During my tests with Asterisk directly, this did not make any difference. Seems like Asterisk checks more than FLAG 19/8 in nat_uac_test().

If I am right, I strictly used the SIP source port for answers instead of any self-detection using STUN or similar.

My kamailio node is connected to the internet directly and 95 percent of (planned) phones are behind random NATs.

In example kamailio.cfg there are these settings:

# Caller NAT detection
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
if(is_first_hop()) {
set_contact_alias();
}
}
setflag(FLT_NATS);
}
#!endif
return;
}

AND (modified for direct RTP flow):

# RTPProxy control and signaling updates for NAT traversal
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;
if(nat_uac_test("8")) {
fix_nated_sdp("15");
if (is_request()) {
if (!has_totag()) {
if(t_is_branch_route()) {
add_rr_param(";nat=yes");
}
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
if(is_first_hop())
set_contact_alias();
}
}
#!endif
return;
}

I checked the docs but I am not sure which FLAGs are safe to use (or break things).

Any thoughts or hints are welcome.

Thank you very much!

Kind regards

Kevin