<div dir="ltr">Hey Anthony,<div><br></div><div>We are trying using PCMU only and will run more tests. But one thing we're confused on about opus is that we haven't changed anything since we initially set this up weeks ago. So assuming we didn'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"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></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"><<a href="mailto:james.mortensen@synclio.com" target="_blank">james.mortensen@synclio.com</a>></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't hear the other person talking.</div>
<div><br></div><div>The setup is Chrome WebRTC -> FreeSWITCH -> PSTN (Level 3/Bandwidth.com)</div><div><br></div><div>We've cleared all users off the server and are trying to find out the cause. We're not dropping packets on the server or the client, yet there'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't have a problem with Chrome to Chrome peer to peer calls, so it seems like it's a problem inside FreeSWITCH.</div>
<div><br></div><div>Here's what we'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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