On 08/13/2013 03:32 PM, Roberto Fichera wrote:
On 08/13/2013 03:25 PM, Daniel-Constantin Mierla wrote:
Can you get a ngrep trace for a registration as well (for the phone using tcp)?
Ok! I'll use pjsua from my local machine connecting in the same way as the
TCP client was doing. The TCP client it's an iPhone using the same pjlib
library.
I can confirm that the default cfg for TCP client doesn't work for me. My cfg
file is attached.
The TCP client doesn't receive any package at INVITE. Finally in
/var/log/message I get this log below:
Aug 13 14:05:37 proxy /usr/sbin/kamailio[8401]: ERROR: <core>
[tcp_main.c:4247]: handle_tcpconn_ev(): connect
94.94.X.X:1274 failed
Contact::
<sip:528@94.94.X.X:1274;transport=TCP;ob>;q=;expires=38;flags=0x0;cflags=0x0;state=0;socket=<tcp:178.79.X.X:5060>;methods=0x1FDF;user_agent=<PJSUA
v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17>;reg-id=0
[root@proxy ~]# kamctl ul show 512
Contact::
<sip:512@94.94.X.X:5060>;q=;expires=61;flags=0x0;cflags=0x40;state=0;socket=<udp:178.79.X.X:5060>;methods=0xFFFFFFFF;received=<sip:94.94.X.X:1025>
user_agent=<DICE 1.8.20.1>;reg-id=0
[root@proxy ~]# ngrep -W byline -d eth0 port 5060
interface: eth0 (178.79.X.X/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )
#
T 94.94.X.X:49519 -> 178.79.x.x:5060 [AP]
REGISTER sip:test.domain;transport=tcp;hide SIP/2.0.
Via: SIP/2.0/TCP
94.94.X.X:49519;rport;branch=z9hG4bKPje03c11fd-d742-4502-98d8-69ca456ddd56;alias.
Max-Forwards: 70.
From: <sip:528@test.domain;hide>;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b.
To: <sip:528@test.domain;hide>.
Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1.
CSeq: 30034 REGISTER.
User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17.
Contact: <sip:528@94.94.X.X:1274;transport=TCP;ob>.
Expires: 300.
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS.
Content-Length: 0.
.
#
T 178.79.x.x:5060 -> 94.94.X.X:49519 [AP]
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/TCP
94.94.X.X:49519;rport=49519;branch=z9hG4bKPje03c11fd-d742-4502-98d8-69ca456ddd56;alias.
From: <sip:528@test.domain;hide>;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b.
To: <sip:528@test.domain;hide>;tag=333a0370df4a40d5d5a0c21bb156e2a6.9167.
Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1.
CSeq: 30034 REGISTER.
WWW-Authenticate: Digest realm="test.domain",
nonce="Ugo+VFIKPSgVy8fh9fSy9SDLvT0wO4QV".
Server: kamailio (4.0.2 (x86_64/linux)).
Content-Length: 0.
.
#
U 178.79.x.x:5060 -> 94.94.X.X:1025
....
##
T 94.94.X.X:49519 -> 178.79.x.x:5060 [AP]
REGISTER sip:test.domain;transport=tcp;hide SIP/2.0.
Via: SIP/2.0/TCP
94.94.X.X:49519;rport;branch=z9hG4bKPj68a9c487-ae8c-428a-8c50-dd195a46a6bf;alias.
Max-Forwards: 70.
From: <sip:528@test.domain;hide>;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b.
To: <sip:528@test.domain;hide>.
Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1.
CSeq: 30035 REGISTER.
User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17.
Contact: <sip:528@94.94.X.X:1274;transport=TCP;ob>.
Expires: 300.
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS.
Authorization: Digest username="528", realm="test.domain",
nonce="Ugo+VFIKPSgVy8fh9fSy9SDLvT0wO4QV",
uri="sip:test.domain;transport=tcp;hide",
response="ac1c7311ccb887fc8fb494d8ebf1bd36".
Content-Length: 0.
.
#
T 178.79.x.x:5060 -> 94.94.X.X:49519 [AP]
SIP/2.0 200 OK.
Via: SIP/2.0/TCP
94.94.X.X:49519;rport=49519;branch=z9hG4bKPj68a9c487-ae8c-428a-8c50-dd195a46a6bf;alias.
From: <sip:528@test.domain;hide>;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b.
To: <sip:528@test.domain;hide>;tag=333a0370df4a40d5d5a0c21bb156e2a6.5440.
Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1.
CSeq: 30035 REGISTER.
Contact: <sip:528@94.94.X.X:1274;transport=TCP;ob>;expires=300.
Server: kamailio (4.0.2 (x86_64/linux)).
Content-Length: 0.
.
##
U 94.94.X.X:1025 -> 178.79.x.x:5060
INVITE sip:528@test.domain:5060 SIP/2.0.
Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport.
Max-Forwards: 70.
From: "asterisk" <sip:512@test.domain>;tag=as4531a4ee.
To: <sip:528@test.domain:5060>.
Contact: <sip:512@94.94.X.X:5060>.
Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain.
CSeq: 102 INVITE.
User-Agent: DICE 1.8.20.1.
Date: Tue, 13 Aug 2013 14:04:11 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 262.
.
v=0.
o=root 883039875 883039875 IN IP4 94.94.X.X.
s=Asterisk PBX 11.3.0.
c=IN IP4 94.94.X.X.
t=0 0.
m=audio 18120 RTP/AVP 0 110 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:110 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
#
U 178.79.x.x:5060 -> 94.94.X.X:1025
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport=1025.
From: "asterisk" <sip:512@test.domain>;tag=as4531a4ee.
To: <sip:528@test.domain:5060>;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f.
Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain.
CSeq: 102 INVITE.
Proxy-Authenticate: Digest realm="test.domain",
nonce="Ugo+XVIKPTFBd9tXC0QemfR9bHiZXO6x".
Server: kamailio (4.0.2 (x86_64/linux)).
Content-Length: 0.
.
#
U 94.94.X.X:1025 -> 178.79.x.x:5060
INVITE sip:528@test.domain:5060 SIP/2.0.
Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport.
Max-Forwards: 70.
From: "asterisk" <sip:512@test.domain>;tag=as4531a4ee.
To: <sip:528@test.domain:5060>.
Contact: <sip:512@94.94.X.X:5060>.
Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain.
CSeq: 102 INVITE.
User-Agent: DICE 1.8.20.1.
Date: Tue, 13 Aug 2013 14:04:11 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 262.
.
v=0.
o=root 883039875 883039875 IN IP4 94.94.X.X.
s=Asterisk PBX 11.3.0.
c=IN IP4 94.94.X.X.
t=0 0.
m=audio 18120 RTP/AVP 0 110 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:110 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
#
U 178.79.x.x:5060 -> 94.94.X.X:1025
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport=1025.
From: "asterisk" <sip:512@test.domain>;tag=as4531a4ee.
To: <sip:528@test.domain:5060>;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f.
Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain.
CSeq: 102 INVITE.
Proxy-Authenticate: Digest realm="test.domain",
nonce="Ugo+XVIKPTFBd9tXC0QemfR9bHiZXO6x".
Server: kamailio (4.0.2 (x86_64/linux)).
Content-Length: 0.
.
#
U 94.94.X.X:1025 -> 178.79.x.x:5060
ACK sip:528@test.domain:5060 SIP/2.0.
Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport.
Max-Forwards: 70.
From: "asterisk" <sip:512@test.domain>;tag=as4531a4ee.
To: <sip:528@test.domain:5060>;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f.
Contact: <sip:512@94.94.X.X:5060>.
Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain.
CSeq: 102 ACK.
User-Agent: DICE 1.8.20.1.
Content-Length: 0.
.
#
U 94.94.X.X:1025 -> 178.79.x.x:5060
ACK sip:528@test.domain:5060 SIP/2.0.
Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport.
Max-Forwards: 70.
From: "asterisk" <sip:512@test.domain>;tag=as4531a4ee.
To: <sip:528@test.domain:5060>.
Contact: <sip:512@94.94.X.X:5060>.
Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain.
CSeq: 102 ACK.
User-Agent: DICE 1.8.20.1.
Content-Length: 0.
.
------------------ This is the client behind NAT ---------------------------
REGISTER sip:test.domain;transport=tcp;hide SIP/2.0
Via: SIP/2.0/TCP
192.168.2.90:49519;rport;branch=z9hG4bKPj92d04d45-bd84-45b3-9439-563ebfaebf00;alias
Max-Forwards: 70
From: <sip:528@test.domain;hide>;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4
To: <sip:528@test.domain;hide>
Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1
CSeq: 30028 REGISTER
User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17
Contact: <sip:528@192.168.2.90:5060;transport=TCP;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Content-Length: 0
--end msg--
15:54:02.580 pjsua_acc.c .Acc 0: Registration sent
15:54:02.681 tcpc0x1515a08 !TCP transport 192.168.2.90:49519 is connected to
178.79.x.x:5060
15:54:02.681 pjsua_app.c SIP TCP transport is connected to [178.79.x.x:5060]
15:54:02.785 pjsua_core.c .RX 538 bytes Response msg 401/REGISTER/cseq=30028
(rdata0x1515cf8) from TCP
178.79.x.x:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP
192.168.2.90:49519;rport=49519;branch=z9hG4bKPj92d04d45-bd84-45b3-9439-563ebfaebf00;alias
From: <sip:528@test.domain;hide>;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4
To: <sip:528@test.domain;hide>;tag=333a0370df4a40d5d5a0c21bb156e2a6.9663
Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1
CSeq: 30028 REGISTER
WWW-Authenticate: Digest realm="test.domain",
nonce="Ugo7/FIKOtBneWAQzD/VoJuucFT3cNuF"
Server: kamailio (4.0.2 (x86_64/linux))
Content-Length: 0
--end msg--
15:54:02.785 pjsua_core.c ....TX 849 bytes Request msg REGISTER/cseq=30029
(tdta0x1513000) to TCP 178.79.x.x:5060:
REGISTER sip:test.domain;transport=tcp;hide SIP/2.0
Via: SIP/2.0/TCP
192.168.2.90:49519;rport;branch=z9hG4bKPj03a4a7eb-d086-4add-b84c-ca5c6cf4d05d;alias
Max-Forwards: 70
From: <sip:528@test.domain;hide>;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4
To: <sip:528@test.domain;hide>
Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1
CSeq: 30029 REGISTER
User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17
Contact: <sip:528@192.168.2.90:5060;transport=TCP;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Authorization: Digest username="528", realm="test.domain",
nonce="Ugo7/FIKOtBneWAQzD/VoJuucFT3cNuF",
uri="sip:test.domain;transport=tcp;hide",
response="c3dd687602ec35b2403b3eb142b496f5"
Content-Length: 0
--end msg--
15:54:02.890 pjsua_core.c .RX 495 bytes Response msg 200/REGISTER/cseq=30029
(rdata0x1515cf8) from TCP
178.79.x.x:5060:
SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.2.90:49519;rport=49519;branch=z9hG4bKPj03a4a7eb-d086-4add-b84c-ca5c6cf4d05d;alias
From: <sip:528@test.domain;hide>;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4
To: <sip:528@test.domain;hide>;tag=333a0370df4a40d5d5a0c21bb156e2a6.189b
Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1
CSeq: 30029 REGISTER
Contact: <sip:528@192.168.2.90:5060;transport=TCP;ob>;expires=300
Server: kamailio (4.0.2 (x86_64/linux))
Content-Length: 0
--end msg--
15:54:02.890 pjsua_acc.c ....SIP outbound status for acc 0 is not active
15:54:02.890 pjsua_acc.c ....sip:528@test.domain;transport=tcp;hide:
registration success, status=200 (OK), will
re-register in 300 seconds
---------------------------------------------------
Daniel
On 8/13/13 3:23 PM, Roberto Fichera wrote:
On 08/13/2013 03:15 PM, Roberto Fichera wrote:
On 08/13/2013 02:33 PM, Daniel-Constantin Mierla wrote:
Hello,
On 8/13/13 1:10 PM, Roberto Fichera wrote:
On 08/13/2013 12:03 PM, Daniel-Constantin Mierla wrote:
Hello,
you should grab the ngrep for such call to understand better what happens.
Also, dumping the location records
will be
useful (kamctl ul show).
Also, be sure that tcp connection lifetime is long enough to survive
re-registration. To avoid trying to open
connections behind nat, use set_forward_no_connect() for calls involving nat
traversal.
I'm using the default conf coming from fedora rpm. So, mainly the problem seems
related to kamailio
which doesn't reuse the TCP port used by NATed clients. I've also notice that
the received
field isn't set at all, so this means that the contact will not get aliased at
all.
I would really like to have a look to a working cfg file for TCP NATed clients
that reuse the TCP port.
Even better if the configuration is based on the fedora default rpm.
if received is not set, then means the register was not detected as coming from
behind nat. Is the phone using
stun?
I'm testing with a normal rtpproxy configuration. BTW udp -> udp work perfectly.
Again, put here the ngrep for registration and a call to see if something is
wrong with signaling. There is no help
that we can provide otherwise. The default config works fine for tcp and natted
clients, I use it everywhere for
this
case without issues.
I tried the default cfg enabling both NAT and RTPproxy, but seems that kamailio
doesn't reuse TCP ports.
Anyway, this is a call from UDP (512) -> TCP (526) both behind the same NAT,
from kamailio point of view
I forgot to say that the received field is now present because I've changed the
route[NATDETECT] in the default configuration as
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
-->>> if (nat_uac_test("19") || proto != UDP) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(FLT_NATS);
}
#!endif
return;
}
[root@proxy ~]# kamctl ul show 526
Contact::
<sip:526@94.94.X.X:1238;transport=TCP;ob>;q=;expires=537;flags=0x0;cflags=0x40;state=0;socket=<tcp:178.79.x.x:5060>;methods=0x1FDF;received=<sip:94.94.X.X:61922;transport=TCP>;user_agent=<DICE
Smartphone 1.0/iPhone>;reg-id=0
[root@proxy ~]# kamctl ul show 512
Contact::
<sip:512@94.94.X.X:5060>;q=;expires=32;flags=0x0;cflags=0x40;state=0;socket=<udp:178.79.x.x:5060>;methods=0xFFFFFFFF;received=<sip:94.94.X.X:1025>;user_agent=<DICE
1.8.20.1>;reg-id=0
[root@proxy ~]#
U 94.94.X.X:1025 -> 178.79.x.x:5060
INVITE sip:526@test.domain:5060 SIP/2.0.
Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport.
Max-Forwards: 70.
From: "asterisk" <sip:512@test.domain>;tag=as76007db0.
To: <sip:526@test.domain:5060>.
Contact: <sip:512@94.94.X.X:5060>.
Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain.
CSeq: 102 INVITE.
User-Agent: DICE 1.8.20.1.
Date: Tue, 13 Aug 2013 13:04:30 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 264.
.
v=0.
o=root 1263161426 1263161426 IN IP4 94.94.X.X.
s=Asterisk PBX 11.3.0.
c=IN IP4 94.94.X.X.
t=0 0.
m=audio 10782 RTP/AVP 0 110 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:110 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
#
U 178.79.x.x:5060 -> 94.94.X.X:1025
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025.
From: "asterisk" <sip:512@test.domain>;tag=as76007db0.
To: <sip:526@test.domain:5060>;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00.
Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain.
CSeq: 102 INVITE.
Proxy-Authenticate: Digest realm="test.domain",
nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt".
Server: kamailio (4.0.2 (x86_64/linux)).
Content-Length: 0.
.
#
U 94.94.X.X:1025 -> 178.79.x.x:5060
INVITE sip:526@test.domain:5060 SIP/2.0.
Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport.
Max-Forwards: 70.
From: "asterisk" <sip:512@test.domain>;tag=as76007db0.
To: <sip:526@test.domain:5060>.
Contact: <sip:512@94.94.X.X:5060>.
Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain.
CSeq: 102 INVITE.
User-Agent: DICE 1.8.20.1.
Date: Tue, 13 Aug 2013 13:04:30 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 264.
.
v=0.
o=root 1263161426 1263161426 IN IP4 94.94.X.X.
s=Asterisk PBX 11.3.0.
c=IN IP4 94.94.X.X.
t=0 0.
m=audio 10782 RTP/AVP 0 110 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:110 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
#
U 178.79.x.x:5060 -> 94.94.X.X:1025
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025.
From: "asterisk" <sip:512@test.domain>;tag=as76007db0.
To: <sip:526@test.domain:5060>;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00.
Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain.
CSeq: 102 INVITE.
Proxy-Authenticate: Digest realm="test.domain",
nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt".
Server: kamailio (4.0.2 (x86_64/linux)).
Content-Length: 0.
.
#
U 94.94.X.X:1025 -> 178.79.x.x:5060
ACK sip:526@test.domain:5060 SIP/2.0.
Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport.
Max-Forwards: 70.
From: "asterisk" <sip:512@test.domain>;tag=as76007db0.
To: <sip:526@test.domain:5060>;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00.
Contact: <sip:512@94.94.X.X:5060>.
Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain.
CSeq: 102 ACK.
User-Agent: DICE 1.8.20.1.
Content-Length: 0.
.
#
U 94.94.X.X:1025 -> 178.79.x.x:5060
ACK sip:526@test.domain:5060 SIP/2.0.
Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport.
Max-Forwards: 70.
From: "asterisk" <sip:512@test.domain>;tag=as76007db0.
To: <sip:526@test.domain:5060>.
Contact: <sip:512@94.94.X.X:5060>.
Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain.
CSeq: 102 ACK.
User-Agent: DICE 1.8.20.1.
Content-Length: 0.
.
#
T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP]
ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0.
Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX.
Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025.
Max-Forwards: 16.
From: "asterisk" <sip:512@test.domain>;tag=as76007db0.
To: <sip:526@test.domain:5060>.
Contact: <sip:512@94.94.X.X:1025>.
Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain.
CSeq: 102 ACK.
User-Agent: DICE 1.8.20.1.
Content-Length: 0.
.
#
T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP]
ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0.
Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX.
Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025.
Max-Forwards: 16.
From: "asterisk" <sip:512@test.domain>;tag=as76007db0.
To: <sip:526@test.domain:5060>.
Contact: <sip:512@94.94.X.X:1025>.
Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain.
CSeq: 102 ACK.
User-Agent: DICE 1.8.20.1.
Content-Length: 0.
.
Cheers,
Daniel
Cheers,
Roberto Fichera.
Cheers,
Daniel
On 7/30/13 6:44 PM, Roberto Fichera wrote:
Hi All,
Sorry for cross-posting this email to PJLIB, but maybe there are some things
related.
Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP
client like iPhone using PJSIP
as SIP
library.
Basically once the iPhone side in close the call (TCP->UDP) I'm getting the
error below. Kamailio is running
under
a VPS
without
NATed network so it uses a real public address. Furthermore, note that tcp_main
is answering to a
192.168.2.98 ip
address
which is the iPhone client. This looks really strange to me since it should
answer directly to the
public/port used
for
the registration
and not to a such kind of reserved address. The kamilio configuration is
basically the default with a very few
changes
like NAT, rtpproxy and postgresql backend.
This problems doesn't happen at all when using UDP->UDP calls. But I cannot use
it because as you certain
know UDP
connection under iPhone will not work when the application run in background
mode.
Can someone suggest how to solve this issue or maybe suggest a TCP working
solution for iPhone?
Thanks in advance.
Roberto Fichera.
Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core>
[tcp_main.c:4432]: tcpconn_main_timeout(): connect
192.168.2.98:5060 failed (timeout)
Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command
"9483_9 D
12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388
GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj"
Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting
session 1 on ports 15604/17354
Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in
from callee, 603 in from caller,
957
relayed, 0 dropped
Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in
from callee, 2 in from caller, 7
relayed, 0
dropped
Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports
15604/17354 is cleaned up
Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0
Jul 30 16:21:55 proxy rtpproxy[2262]: "
Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core>
[tcp_main.c:4432]: tcpconn_main_timeout(): connect
192.168.2.98:5060 failed (timeout)
Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command
"9496_16 D
12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj
as74e0c388"
Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request
failed: session
12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags
GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found
Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8
Jul 30 16:22:14 proxy rtpproxy[2262]: "
_______________________________________________
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sr-users@lists.sip-router.org
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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
_______________________________________________
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
