<div dir="ltr">Hey Anthony,<div><br></div><div>We are trying using PCMU only and will run more tests.  But one thing we&#39;re confused on about opus is that we haven&#39;t changed anything since we initially set this up weeks ago.  So assuming we didn&#39;t change anything, and this is all **locally running** code, how would OPUS be causing a problem here?</div>

<div><br></div><div>Thanks for your help!</div><div class="gmail_extra"><br clear="all"><div><div dir="ltr"><div><br></div><div>James<br></div><div><br><div><br></div></div></div></div>
<br><br><div class="gmail_quote">On Thu, Nov 14, 2013 at 11:47 AM, Anthony Minessale <span dir="ltr">&lt;<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div dir="ltr">HINT If you are using webrtc with FS especially in production you must stay current on latest master.<div>1.4 is only there to test the packaging at this stage.</div><div><br></div><div>Try not using opus.  There seem to be some oddities with it lately.  Try using isac or pcmu.</div>


<div>Also try setting up a call to delay_echo or to something like a conference so you can compare.</div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote"><div><div class="h5">On Thu, Nov 14, 2013 at 1:40 PM, James Mortensen <span dir="ltr">&lt;<a href="mailto:james.mortensen@synclio.com" target="_blank">james.mortensen@synclio.com</a>&gt;</span> wrote:<br>


</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><div dir="ltr">Hello,<div><br></div><div>I have two FreeSWITCH 1.4b servers running on Amazon EC2.  Starting yesterday, the call quality suddenly got so bad that callers from the PSTN can&#39;t hear the other person talking.</div>




<div><br></div><div>The setup is Chrome WebRTC -&gt; FreeSWITCH -&gt; PSTN (Level 3/Bandwidth.com)</div><div><br></div><div>We&#39;ve cleared all users off the server and are trying to find out the cause. We&#39;re not dropping packets on the server or the client, yet there&#39;s problems with the audio. <br>




</div><div><br></div><div>The audio issues occur with both servers, regardless of how many active calls there are.  We don&#39;t have a problem with Chrome to Chrome peer to peer calls, so it seems like it&#39;s a problem inside FreeSWITCH.</div>




<div><br></div><div>Here&#39;s what we&#39;ve tried:</div><div><ul><li>Checking for dropped packets</li><li>Restarting FreeSWITCH</li><li>Moving all users to another server so we test only with 1 active call.</li><li>Trying calls with both Level 3 as well as Bandwidth.com trunk providers.</li>




</ul><div>Does anyone have some ideas on what we can try next?</div></div><div><br></div><div><div><div dir="ltr"><div><br></div><div>Thank you!</div><span><font color="#888888"><div>James<br></div><div><br>
</div></font></span></div></div></div></div>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>


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<br></blockquote></div><br></div></div>