23 Jul
2011
23 Jul
'11
11:54 a.m.
Dear Klaus,
Sorry for the late response.
I check the log of the caller and found out:
The message header of 200 OK (with session description) from the server to
the caller , right before the RTP starts, has the
value <sip:10.150.175.210;lr=on;nat=yes> of Record-Route .
Then the caller sent ACK to ip 10.150.175.210.
The ip 10.150.175.210 is the private ip of the amazon server I am using ,
which is unreachable for the caller.
Is it kinda installing Kamailio inside the NAT issue?
What should I do?
Your help is great appreciated.
Thanks,
Coca
2011/7/23 Klaus Darilion <klaus.mailinglists(a)pernau.at>
If it is "your" client, then you should be
able to debug your client to
find out if ACK is sent, and if yes, where the ACK is sent to. I inspected
the last trace and I didn't spotted an error.
regards
klaus
Coca wrote:
--
--------------------------
Room to Fly, Endless Sky!
--Yi Chen
Dear Klaus,
Thank you for your email.
I tried using my UA connecting to the servers provided by other free sip
service provider like Opensips in WAN.
And the same problem didn't happen. My UA did sent ACK after the 200 OK in
the invite process.
It looks like the problem is my Kamailio server 's NAT setting.
I wonder how I can do to verify whether the NAT setting is correctly done.
What kind of method did you use in this issue?
Your help will be great appreciated.
Coca
2011/7/21 Klaus Darilion <klaus.mailinglists(a)pernau.at <mailto:
klaus.mailinglists(a)**pernau.at <klaus.mailinglists(a)pernau.at>>>
IF you take a look at the trace you see, that the caller does not send
ACK after receiving 200 OK. Either it does not send ACK or it sends it
to wrong destination.
Trace at the caller's phone and watch log file of caller.
regards
Klaus
Am 21.07.2011 12:33, schrieb Coca:
receive the
------------------------------**-----------------------------
> and my rtpproxy1.2.1 was installed by apt-get install rtpproxy,
http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**dev<http://l…
--
--------------------------
Room to Fly, Endless Sky!
--Yi Chen
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Dear Klaus,
Since I have Usrloc record made for registration of myUA behind nat
looks like:
Contact:
<sip:1234@192.168.10.50:2305;**transport=TCP;line=**
7e1d8b95f65b25a>;expires=600;**received="sip:27.96.63.122:**
49202;transport=TCP"
I thought my rtpproxy is running.
However , the call can be established even without NAT enable, and
it
also ends unusually after 36s.
Attachment is the ngrep log in my Kamailio server side on 5060 port.
( I have replaced my server ip as xx.xx.xx.xx and the UA name as
myUA)
Any hint will be great appreciated.
Coca
2011/7/21 Klaus Darilion <klaus.mailinglists(a)pernau.at
<mailto:klaus.mailinglists@**pernau.at <klaus.mailinglists(a)pernau.at>>
<mailto:klaus.mailinglists@**pernau.at<klaus.mailinglists@pernau.at>
<mailto:klaus.mailinglists@**pernau.at <klaus.mailinglists(a)pernau.at>
>
This sounds like a NAT problem, where the callee does not ACK request (INVITE-200OK-ACK).
regards
klaus
Am 21.07.2011 10:32, schrieb Coca:
> Hi List,
> I have a Kamailio3.1 server
and RTPProxy running in WAN.
> The calls between UA will
automatically terminated in Callee
UA
side 36s
> Is it my rtpproxy doesn't
working correctly or something
else is
after connected, while no one sends a BYE.
While Kamailio and UA are in LAN at all , everything is just
working well.
wrong?
kamailio-3.1.x-from-git<http://www.kamailio.org/dokuwiki/doku.php/instal…
What can I do to fix it.
Any hint??
BTW,
Kamailio is installed following official guide
http://www.kamailio.org/**dokuwiki/doku.php/install:** The
kamailio.cfg wasn't changed at all except for below:
------------------------------** > 1) adding the following lines:
> #!define WITH_MYSQL
> #!define WITH_AUTH
> #!define WITH_USRLOCDB
> #!define WITH_NAT
> 2)uncommenting the line below
in route[REGISTRAR],
> setbflag(FLB_NATSIPPING);
> 3)change rtpproxy port
corresponding
> modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:22222
<http://127.0.0.1:22222>
<http://127.0.0.1:22222>
------------------------------**-----------------------------
<http://127.0.0.1:22222>")
------------------------------** > with
> 1) /etc/default/rtpproxy changed into:
> # Defaults for rtpproxy
> # The control socket.
>
#CONTROL_SOCK="unix:/var/run/**rtpproxy/rtpproxy.sock"
> # To listen on an UDP socket,
uncomment this line:
>
CONTROL_SOCK=udp:127.0.0.1:**22222 <http://127.0.0.1:22222> <
http://127.0.0.1:22222>
<http://127.0.0.1:22222>
<http://127.0.0.1:22222>
> LISTEN_ADDR=xx.xx.xx.xx
> # Additional options that are passed to the daemon.
> EXTRA_OPTS="-l
${LISTEN_ADDR}"
> 2) and started by
> rtpproxy -l xx.xx.xx.xx -s udp:localhost:22222 -u kamailio
> Your
help will be great appreciated.
> Coca
>
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