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<DIV dir=ltr align=left><SPAN class=596460301-01022005><FONT face=Arial
color=#0000ff size=2>Maybe you're transcoding on the server with cpu intensive
codecs? That would be the first thing I'd look at. Try using G.711
(ulaw) on both SIP phones and remove reinvite=no and canreinvite=no from
your phone declarations in sip.conf.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=596460301-01022005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=596460301-01022005><FONT face=Arial
color=#0000ff size=2>Hope that helps.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=596460301-01022005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=596460301-01022005><FONT face=Arial
color=#0000ff size=2>Guills</FONT></SPAN></DIV><BR>
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style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> Nic le Roux [mailto:nicl@i-procc.za.net]
<BR><B>Sent:</B> Monday, January 31, 2005 7:01 AM<BR><B>To:</B>
asterisk-users@lists.digium.com<BR><B>Subject:</B> [Asterisk-Users] Audio
Quality over LAN very bad<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><SPAN class=341515314-31012005><FONT face=Arial size=2>Hi
All,</FONT></SPAN></DIV>
<DIV><SPAN class=341515314-31012005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=341515314-31012005><FONT face=Arial size=2>I'm running
Asterisk on the following</FONT></SPAN></DIV>
<DIV><SPAN class=341515314-31012005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=341515314-31012005><FONT face=Arial
size=2>vendor_id : GenuineIntel<BR>model
name : Celeron (Coppermine)<BR>cpu
MHz : 668.202<BR>cache
size : 128 KB</FONT></SPAN></DIV><SPAN
class=341515314-31012005>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><SPAN class=341515314-31012005></SPAN><FONT face=Arial size=2>w<SPAN
class=341515314-31012005>ith 192 MB Ram</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=341515314-31012005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=341515314-31012005>Audio coming from
Asterisk (the demo ) is excellent when using a SIP phone on the LAN to
Asterisk,</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN class=341515314-31012005>and when dialling
in from outside via ISDN to Asterisk.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=341515314-31012005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=341515314-31012005>However, when
connecting from SIP phone to SIP phone (across LAN) and dialling from
externally to SIP which is on the local LAN</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN class=341515314-31012005>it is very choppy
and one can barely make out the other party.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN class=341515314-31012005>I'm using an Eicon
Diva 2-m card and 100mb network all round.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=341515314-31012005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=341515314-31012005>What could be the
cause as I believe bandwidth is ruled out.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=341515314-31012005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=341515314-31012005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=341515314-31012005>Thanks and
regards</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=341515314-31012005>Nic</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=341515314-31012005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=341515314-31012005></SPAN> </DIV>
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