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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I am not as knowledgeable as the developers that will respond to
your question but I had the same problem as you. Here is what I did to
combat the delay:<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>First off I started everything from scratch. I reinstalled
Linux and then I reinstalled FreeSWITCH by creating .deb packages.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I then created my own conference profile and set the sample rate
to 4000 and changed the energy level to 20.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I also made sure to test the conference room from phones that
were in completely different areas so there wasn’t a chance for feedback
or really bad echoing problems.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Once I knew the delay was solved I raised the sample rate to
8000. I tested it to make sure it would work properly.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>As Michael stated, this could be your network infrastructure but
I just wanted to let another FreeSWITCH user know what I did to try and stop
the voice delay.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] <b>On Behalf Of </b>Bradley
Brashier<br>
<b>Sent:</b> Tuesday, June 16, 2009 1:52 PM<br>
<b>To:</b> freeswitch-users@lists.freeswitch.org<br>
<b>Subject:</b> [Freeswitch-users] Voice lag in conference<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>I'm creating a conferencing product for use in a system with
theoretically several hundred concurrent calls. I'm using FreeSwitch to create
this product, but am not only new to FreeSwitch, but also the entire telecom
industry as well as Open Source projects in general (I'm a recovering BIOS
guy).<br>
<br>
I've got a bare-bones conference up and running on the server, including a
handshake and a couple of features, and am using the default packages from the
current trunk, but I've noticed that voice lag is a pretty big issue. Common
lag times are several hundred milliseconds, and I've heard as long as a second.
It seems to be at least marginally specific to individual phones -- certain phones
have longer lag than others even on the same call.<br>
<br>
My question is really about what my options are. Is this just a part of SIP? Of
conferencing? Of FreeSwitch? Are there things I can prune or slim down that
will help? Is this a common issue? If it's common, is it expected by the
marketplace? <o:p></o:p></p>
</div>
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