<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">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>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><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 class="HOEnZb"><font color="#888888"><div>James<br></div><div><br>
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