<html><body><div style="color:#000; background-color:#fff; font-family:lucida console, sans-serif;font-size:10pt">Hi Lazlo and Flavio,<br><br>today i started from a fresh git install of Opensips 1.8 and use the rtpproxy delivered by opensips. ( http://opensips.org/pub/rtpproxy/) Last time i used the original from http://www.rtpproxy.org/ with the high load of CPU. Now it runs like it should be.<br><br>I´m happy to say - thanks! It works now.<br><br><div><span>Jens<br></span></div><div><br></div> <div style="font-family: lucida console, sans-serif; font-size: 10pt;"> <div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"> <div dir="ltr"> <hr size="1"> <font face="Arial" size="2"> <b><span style="font-weight:bold;">Von:</span></b> Laszlo <voipfreak.net@gmail.com><br> <b><span style="font-weight: bold;">An:</span></b> Jens Sauer <sauer.jens@yahoo.de>; OpenSIPS users mailling list <users@lists.opensips.org>
<br> <b><span style="font-weight: bold;">Gesendet:</span></b> 22:41 Samstag, 22.Juni 2013<br> <b><span style="font-weight: bold;">Betreff:</span></b> Re: [OpenSIPS-Users] NAT - Unable to solve RTP Problem<br> </font> </div> <div class="y_msg_container"><br><div id="yiv7246290632"><div>Hi Jens,</div>
<div>Can you make a ngrep trace?<br>
That way we can see more things about what is going on...</div>
<div>Do you see anything wrong in your syslog during the call?<br></div>
<div>-Laszlo<br>
</div>
<div class="yiv7246290632gmail_quote">2013.06.22. 19:39, "Jens Sauer" <<a rel="nofollow" ymailto="mailto:sauer.jens@yahoo.de" target="_blank" href="mailto:sauer.jens@yahoo.de">sauer.jens@yahoo.de</a>> ezt írta:<br><blockquote class="yiv7246290632gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div><div style="font-size:10pt;font-family:lucida console, sans-serif;"><div><div><div><div style="font-size:10pt;font-family:lucida console, sans-serif;"><div>Hello,</div><div><br></div><div style="font-style:normal;font-size:13.3333px;background-color:transparent;font-family:lucida console, sans-serif;">
since a month i try to solve my rtp problem. I´m quite new to OpenSips but i want it to be running on my Server.</div><div style="font-style:normal;font-size:13.3333px;background-color:transparent;font-family:lucida console, sans-serif;">
<br></div><div style="font-style:normal;font-size:13.3333px;background-color:transparent;font-family:lucida console, sans-serif;">Both Phones (phonerLite) are connected and each one could call the other one, the only
Problem is - no voice!</div><div style="font-style:normal;font-size:13.3333px;background-color:transparent;font-family:lucida console, sans-serif;">Could someone please give me a hint, on using the
right module (and parameter) to solve the problem. The Screenshot shows
my topology. Both Clients use a public stun Server. I tried to use the
rtpproxy but this piece of code use 99% of my CPU - there must be a
different way. <br></div><div style="font-style:normal;font-size:13.3333px;background-color:transparent;font-family:lucida console, sans-serif;"><br></div><div style="font-style:normal;font-size:13.3333px;background-color:transparent;font-family:lucida console, sans-serif;">
Thanks for any help.</div><div style="font-style:normal;font-size:13.3333px;background-color:transparent;font-family:lucida console, sans-serif;"><br></div><div style="font-style:normal;font-size:13.3333px;background-color:transparent;font-family:lucida console, sans-serif;">
Jens</div></div></div></div></div></div></div><br>_______________________________________________<br>
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