[sr-dev] Call is terminated in callee side after 36s @Kamailio 3.1 @debian sqeeze64

Klaus Darilion klaus.mailinglists at pernau.at
Thu Jul 21 13:58:11 CEST 2011


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 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
>     > 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>")
>     >
>     >
>     -----------------------------------------------------------------------------------------
>     >
>     > 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>
>     >
>     > 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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>     > sr-dev at lists.sip-router.org <mailto:sr-dev at lists.sip-router.org>
>     > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
> 
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> 
> 
> 
> 
> -- 
> --------------------------
> Room to Fly, Endless Sky!
> 
>                   --Yi Chen
> 
> 
> 
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