<div dir="ltr">Whatever you are talking to on the other end is negotiating 30 ms and actually sending 20. Sounds like the sipura/cisco bug.<div><br></div><div style>Go in the UI for the device and find the packet time param which is set to 30 and change it to 20.</div>
<div style><br></div><div style>From our wiki: <a href="http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo">http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo</a></div><div style><br></div><div style><h2 style="color:rgb(0,0,0);background-image:none;font-weight:normal;margin:0px 0px 0.6em;padding-top:0.5em;padding-bottom:0.17em;border-bottom-width:1px;border-bottom-style:solid;border-bottom-color:rgb(170,170,170);font-size:19px;font-family:sans-serif;line-height:19.046875px">
<span class="">SPA3102 DEVICE CONFIGURATION:</span></h2><p style="margin:0.4em 0px 0.5em;line-height:19.046875px;color:rgb(0,0,0);font-family:sans-serif;font-size:13px">NOTE: There is a bug in the default configuration of the SPA-3102 and other Linksys devices that sets the RTP Packet Size to .030, this should be set to .020 to avoid problems.</p>
<p style="margin:0.4em 0px 0.5em;line-height:19.046875px;color:rgb(0,0,0);font-family:sans-serif;font-size:13px"><b>admin -> advanced -> Voice -> PSTN</b></p><p style="margin:0.4em 0px 0.5em;line-height:19.046875px;color:rgb(0,0,0);font-family:sans-serif;font-size:13px">
<b><br></b></p><p style="margin:0.4em 0px 0.5em;line-height:19.046875px;color:rgb(0,0,0);font-family:sans-serif;font-size:13px"><b><br></b></p></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Mar 20, 2013 at 8:02 PM, Garey Arrington <span dir="ltr"><<a href="mailto:gareyarrington@yahoo.com" target="_blank">gareyarrington@yahoo.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div style="font-size:12pt;font-family:times new roman,new york,times,serif"><div><span style="font-family:'Courier New',courier,monaco,monospace,sans-serif;font-size:13px">When making inbound and outbound calls I am getting slightly less than acceptable audio quality after a few seconds. The machine is a 64 bit Centos 5.5 vm running on Xen with kernel version: 2.6.18-194.el5</span></div>
<div><span style="font-family:'Courier New',courier,monaco,monospace,sans-serif;font-size:13px"><br></span></div><div style="font-style:normal;background-color:transparent"><span style="font-family:'Courier New',courier,monaco,monospace,sans-serif;font-size:13px">I am getting warnings like with each call:</span></div>
<div style="font-style:normal;background-color:transparent"><span style="font-family:'Courier New',courier,monaco,monospace,sans-serif;font-size:small"><br></span></div><div style="font-size:13px;font-style:normal;background-color:transparent;font-family:'Courier New',courier,monaco,monospace,sans-serif">
<div style="font-style:normal;background-color:transparent"><span style="font-family:'Courier New',courier,monaco,monospace,sans-serif;font-size:13px">2013-03-21 04:22:54.655968 [WARNING] mod_sofia.c:1274 Asynchronous PTIME not supported, changing our end from 30 to 20</span></div>
<div style="font-style:normal"><span style="font-family:'Courier New',courier,monaco,monospace,sans-serif;font-size:13px">and</span></div><div><div><span style="font-family:'Courier New',courier,monaco,monospace,sans-serif;font-size:small">2013-03-21 04:22:54.056083 [WARNING] switch_time.c:578 Increasing global timer resolution to 10ms to handle interval 30</span></div>
<div style="font-style:normal"><span style="font-family:'Courier New',courier,monaco,monospace,sans-serif;font-size:small"><br></span></div><div style="font-style:normal"><span style="font-family:'Courier New',courier,monaco,monospace,sans-serif;font-size:small"><br>
</span></div><div style="font-style:normal"><span style="font-family:'Courier New',courier,monaco,monospace,sans-serif;font-size:small">I have tried the fix that I have consistently found for this issue which is to add this to external.xml:</span></div>
<div style="font-style:normal"><pre><span style="font-family:'Courier New',courier,monaco,monospace,sans-serif;font-size:small"><param name="rtp-autofix-timing" value="false"/></span></pre>
<pre><span style="font-family:'Courier New',courier,monaco,monospace,sans-serif;font-size:13px"><br></span></pre><pre><span style="font-family:'Courier New',courier,monaco,monospace,sans-serif;font-size:13px">But this makes the audio quality so bad you can hardly make out anything.</span></pre>
<pre><span style="font-family:'Courier New',courier,monaco,monospace,sans-serif;font-size:13px"><br></span></pre><pre><span style="font-family:'Courier New',courier,monaco,monospace,sans-serif;font-size:13px">Can someone point me in a better direction for a solution to this problem?</span></pre>
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