14 Jul
2015
14 Jul
'15
12:02 p.m.
On 14 Jul 2015, at 11:56, Alberto Sagredo
<alberto.sagredo(a)avanzada7.com> wrote:
I have found an issue detecting NAT of GS 1.4.23 Phones when using Stun on this phones
If they activate STUN they signal with a public IP. If they signal with a public IP,
they tell us they
can handle NAT by themself and require no help with NAT traversal from Kamailio.
Turn off STUN to get help from Kamailio.
/O
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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