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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Without rtpproxy or mediaproxy, the both SIP clients have to be
reached from Internet, or it has to have the public IP.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>But in your case, I don’t think you can have both client on
Internet.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Tung<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
sr-users-bounces@lists.sip-router.org
[mailto:sr-users-bounces@lists.sip-router.org] <b>On Behalf Of </b>truong ngoc
THANH<br>
<b>Sent:</b> Tuesday, August 24, 2010 5:24 PM<br>
<b>To:</b> Alex Balashov<br>
<b>Cc:</b> kamailio<br>
<b>Subject:</b> Re: [SR-Users] help to configure RTP stream with NAT.<o:p></o:p></span></p>
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<p class=MsoNormal style='margin-bottom:12.0pt'>Dear Alex Balashov,<br>
thanks for helping<br>
i try to disable force_rtp_proxy() in kamailio.cfg.but when i make call, no
stream transfer. the call can make but can not hear anything .<o:p></o:p></p>
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<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal>TRUONG NGOC THANH<br>
Telecommunications Engineer<br>
Tel: 0984 480 646<br>
Y!M: ngoc217thanh<o:p></o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Alex Balashov
<abalashov@evaristesys.com><br>
<b>To:</b> sr-users@lists.sip-router.org<br>
<b>Sent:</b> Tue, August 24, 2010 4:58:27 PM<br>
<b>Subject:</b> Re: [SR-Users] help to configure RTP stream with NAT.<br>
</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'><br>
On 08/24/2010 05:41 AM, truong ngoc THANH wrote:<br>
<br>
> hi all,<br>
> i have using RTP proxy, and i see that RTP stream is handled by RTP<br>
> proxy. so how to configure in kamailio or which module make RTP stream<br>
> direct from sip client to another one ?<br>
> please suggest if anyone know.<br>
<br>
On calls where you do not want rtpproxy to relay media, just don't use it
(don't call force_rtp_proxy())?<br>
<br>
-- Alex Balashov - Principal<br>
Evariste Systems LLC<br>
1170 Peachtree Street<br>
12th Floor, Suite 1200<br>
Atlanta, GA 30309<br>
Tel: +1-678-954-0670<br>
Fax: +1-404-961-1892<br>
Web: <a href="http://www.evaristesys.com/" target="_blank">http://www.evaristesys.com/</a><br>
<br>
_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users"
target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><o:p></o:p></span></p>
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