<div>Dear Klaus,</div><div><br></div><div>Sorry for the late response. </div><div>I check the log of the caller and found out: </div><div><br></div><div>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 .</div>
<div>Then the caller sent ACK to ip 10.150.175.210.</div><div>The ip 10.150.175.210 is the private ip of the amazon server I am using , which is unreachable for the caller.</div><div><br></div><div>Is it kinda installing Kamailio inside the NAT issue?</div>
<div>What should I do?</div><div><br></div><div>Your help is great appreciated.</div><div><br></div><div>Thanks,</div><div><br></div><div>Coca</div><br><br><div class="gmail_quote">2011/7/23 Klaus Darilion <span dir="ltr"><<a href="mailto:klaus.mailinglists@pernau.at">klaus.mailinglists@pernau.at</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">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.<br>
<br>
regards<br>
klaus<br>
<br>
Coca wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="im">
Dear Klaus,<br>
<br>
Thank you for your email.<br>
I tried using my UA connecting to the servers provided by other free sip service provider like Opensips in WAN.<br>
And the same problem didn't happen. My UA did sent ACK after the 200 OK in the invite process.<br>
<br>
It looks like the problem is my Kamailio server 's NAT setting.<br>
<br>
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?<br>
<br>
Your help will be great appreciated.<br>
<br>
Coca<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br></div>
2011/7/21 Klaus Darilion <<a href="mailto:klaus.mailinglists@pernau.at" target="_blank">klaus.mailinglists@pernau.at</a> <mailto:<a href="mailto:klaus.mailinglists@pernau.at" target="_blank">klaus.mailinglists@<u></u>pernau.at</a>>><div>
<div></div><div class="h5"><br>
<br>
IF you take a look at the trace you see, that the caller does not send<br>
ACK after receiving 200 OK. Either it does not send ACK or it sends it<br>
to wrong destination.<br>
<br>
Trace at the caller's phone and watch log file of caller.<br>
<br>
regards<br>
Klaus<br>
<br>
Am 21.07.2011 12:33, schrieb Coca:<br>
> Dear Klaus,<br>
><br>
> Since I have Usrloc record made for registration of myUA behind nat<br>
> looks like:<br>
> Contact:<br>
><br>
<<a href="http://sip:1234@192.168.10.50:2305" target="_blank">sip:1234@192.168.10.50:2305</a>;<u></u>transport=TCP;line=<u></u>7e1d8b95f65b25a>;expires=600;<u></u>received="sip:<a href="http://27.96.63.122" target="_blank">27.96.63.122</a>:<u></u>49202;transport=TCP"<br>
><br>
> I thought my rtpproxy is running.<br>
> However , the call can be established even without NAT enable, and it<br>
> also ends unusually after 36s.<br>
><br>
> Attachment is the ngrep log in my Kamailio server side on 5060 port.<br>
> ( I have replaced my server ip as xx.xx.xx.xx and the UA name as<br>
myUA)<br>
><br>
> Any hint will be great appreciated.<br>
><br>
> Coca<br>
><br>
><br>
><br>
><br>
><br>
> 2011/7/21 Klaus Darilion <<a href="mailto:klaus.mailinglists@pernau.at" target="_blank">klaus.mailinglists@pernau.at</a><br>
<mailto:<a href="mailto:klaus.mailinglists@pernau.at" target="_blank">klaus.mailinglists@<u></u>pernau.at</a>><br>
> <mailto:<a href="mailto:klaus.mailinglists@pernau.at" target="_blank">klaus.mailinglists@<u></u>pernau.at</a><br>
<mailto:<a href="mailto:klaus.mailinglists@pernau.at" target="_blank">klaus.mailinglists@<u></u>pernau.at</a>>>><br>
><br>
> This sounds like a NAT problem, where the callee does not<br>
receive the<br>
> ACK request (INVITE-200OK-ACK).<br>
><br>
> regards<br>
> klaus<br>
><br>
> Am 21.07.2011 10:32, schrieb Coca:<br>
> > Hi List,<br>
> ><br>
> > I have a Kamailio3.1 server and RTPProxy running in WAN.<br>
> ><br>
> > The calls between UA will automatically terminated in Callee UA<br>
> side 36s<br>
> > after connected, while no one sends a BYE.<br>
> ><br>
> > While Kamailio and UA are in LAN at all , everything is just<br>
> working well.<br>
> ><br>
> > Is it my rtpproxy doesn't working correctly or something<br>
else is<br>
> wrong?<br>
> > What can I do to fix it.<br>
> ><br>
> > Any hint??<br>
> ><br>
> > BTW,<br>
> > Kamailio is installed following official guide<br>
> ><br>
> <a href="http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git" target="_blank">http://www.kamailio.org/<u></u>dokuwiki/doku.php/install:<u></u>kamailio-3.1.x-from-git</a><br>
> > The kamailio.cfg wasn't changed at all except for below:<br>
> ><br>
> ------------------------------<u></u>------------------------------<u></u>-----------------------------<br>
> > 1) adding the following lines:<br>
> > #!define WITH_MYSQL<br>
> > #!define WITH_AUTH<br>
> > #!define WITH_USRLOCDB<br>
> > #!define WITH_NAT<br>
> ><br>
> > 2)uncommenting the line below in route[REGISTRAR],<br>
> > setbflag(FLB_NATSIPPING);<br>
> ><br>
> > 3)change rtpproxy port corresponding<br>
> > modparam("rtpproxy", "rtpproxy_sock", "udp:<a href="http://127.0.0.1:22222" target="_blank">127.0.0.1:22222</a><br>
<<a href="http://127.0.0.1:22222" target="_blank">http://127.0.0.1:22222</a>><br>
> <<a href="http://127.0.0.1:22222" target="_blank">http://127.0.0.1:22222</a>><br>
> > <<a href="http://127.0.0.1:22222" target="_blank">http://127.0.0.1:22222</a>>")<br>
> ><br>
> ><br>
> ------------------------------<u></u>------------------------------<u></u>-----------------------------<br>
> ><br>
> > and my rtpproxy1.2.1 was installed by apt-get install rtpproxy,<br>
> > with<br>
> > 1) /etc/default/rtpproxy changed into:<br>
> ><br>
> > # Defaults for rtpproxy<br>
> ><br>
> ><br>
> > # The control socket.<br>
> ><br>
> > #CONTROL_SOCK="unix:/var/run/<u></u>rtpproxy/rtpproxy.sock"<br>
> ><br>
> > # To listen on an UDP socket, uncomment this line:<br>
> ><br>
> > CONTROL_SOCK=udp:<a href="http://127.0.0.1:22222" target="_blank">127.0.0.1:<u></u>22222</a> <<a href="http://127.0.0.1:22222" target="_blank">http://127.0.0.1:22222</a>><br>
<<a href="http://127.0.0.1:22222" target="_blank">http://127.0.0.1:22222</a>><br>
> <<a href="http://127.0.0.1:22222" target="_blank">http://127.0.0.1:22222</a>><br>
> ><br>
> > LISTEN_ADDR=xx.xx.xx.xx<br>
> ><br>
> ><br>
> > # Additional options that are passed to the daemon.<br>
> ><br>
> > EXTRA_OPTS="-l ${LISTEN_ADDR}"<br>
> ><br>
> > 2) and started by<br>
> > rtpproxy -l xx.xx.xx.xx -s udp:localhost:22222 -u kamailio<br>
> ><br>
> ><br>
> ><br>
> > Your help will be great appreciated.<br>
> ><br>
> > Coca<br>
> ><br>
> ><br>
> ><br>
> > ______________________________<u></u>_________________<br>
> > sr-dev mailing list<br>
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> Room to Fly, Endless Sky!<br>
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</div></div></blockquote><div><div></div><div class="h5">
<br>
<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>--------------------------<br>Room to Fly, Endless Sky!<br><br> --Yi Chen<br>