[sr-dev] Call is terminated in callee side after 36s @Kamailio 3.1 @debian sqeeze64
Coca
chanea at gmail.com
Sat Jul 23 17:54:13 CEST 2011
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 at 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:
>
>> 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 at pernau.at <mailto:
>> klaus.mailinglists@**pernau.at <klaus.mailinglists at 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:
>> > Dear Klaus,
>> >
>> > Since I have Usrloc record made for registration of myUA behind nat
>> > looks like:
>> > Contact:
>> >
>> <sip:1234 at 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 at pernau.at
>> <mailto:klaus.mailinglists@**pernau.at <klaus.mailinglists at pernau.at>>
>> > <mailto:klaus.mailinglists@**pernau.at<klaus.mailinglists at pernau.at>
>> <mailto:klaus.mailinglists@**pernau.at <klaus.mailinglists at pernau.at>
>> >>>
>> >
>> > This sounds like a NAT problem, where the callee does not
>> receive the
>> > 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
>> > > after connected, while no one sends a BYE.
>> > >
>> > > While Kamailio and UA are in LAN at all , everything is just
>> > working well.
>> > >
>> > > Is it my rtpproxy doesn't working correctly or something
>> else is
>> > wrong?
>> > > What can I do to fix it.
>> > >
>> > > Any hint??
>> > >
>> > > BTW,
>> > > Kamailio is installed following official guide
>> > >
>> > http://www.kamailio.org/**dokuwiki/doku.php/install:**
>> kamailio-3.1.x-from-git<http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git>
>> > > 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>")
>> > >
>> > >
>> > ------------------------------**
>> ------------------------------**-----------------------------
>> > >
>> > > and my rtpproxy1.2.1 was installed by apt-get install
>> rtpproxy,
>> > > 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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>> >
>> >
>> >
>> >
>> > --
>> > --------------------------
>> > Room to Fly, Endless Sky!
>> >
>> > --Yi Chen
>> >
>> >
>> >
>> > ______________________________**_________________
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>>
>>
>>
>>
>> --
>> --------------------------
>> Room to Fly, Endless Sky!
>>
>> --Yi Chen
>>
>>
>> ------------------------------**------------------------------**
>> ------------
>>
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>>
>
>
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>
--
--------------------------
Room to Fly, Endless Sky!
--Yi Chen
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