Hello,
the request has public IP address in Via and Contact, matching the
source IP. If you look at the tests you do in the readme of the
nathelper module, then the request appears as not being natted, see:
-
http://kamailio.org/docs/modules/stable/modules/nathelper.html#nathelper.f.…
Test with 3 is 1 + 2 and test with 19 is 1 + 2 + 16
Cheers,
Daniel
On 14/07/15 11:56, Alberto Sagredo wrote:
I have found an issue detecting NAT of GS 1.4.23
Phones when using
Stun on this phones
Usual nat_uac_test with numbers 19 and 3 does not seems to detect is
behind nat so NATMANAGE is not called
Here its some trace. ANy clue how to handle this phones if they
activate STUN for example
U 80.26.x.x:52768 -> 192.168.0.170:8002 <http://192.168.0.170:8002>
INVITE sip:2@x.x.x.x:8002 SIP/2.0.
Via: SIP/2.0/UDP 80.26.x.x:52768;branch=z9hG4bK358742535;rport.
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1478444456.
To: <sip:2@x.x.x.x:8002>.
Call-ID: 244257786-52768-56(a)IA.CG.BIE.BCH.
CSeq: 550 INVITE.
Contact: "Anonymous" <sip:212@80.26.x.x:52768>.
X-Grandstream-PBX: true.
Max-Forwards: 70.
User-Agent: Grandstream GXP2140 1.0.4.23.
Privacy: id.
P-Preferred-Identity: <sip:212@x.x.x.x:8002>.
Supported: replaces, path, timer.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
REFER, UPDATE, MESSAGE.
Content-Type: application/sdp.
Accept: application/sdp, application/dtmf-relay.
Content-Length: 335.
.
v=0.
o=212 8000 8000 IN IP4 80.26.x.x.
s=SIP Call.
c=IN IP4 80.26.x.x.
t=0 0.
m=audio 55422 RTP/AVP 0 8 18 9 2 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:9 G722/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
# Caller NAT detection route
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19)) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(FLT_NATS);
}
#!endif
return;
}
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