<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2800.1528" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT face=Arial size=2>We had phones in Perth hooked up to an Asterisk box
in Melbourne, and the call was fine - so I know it can be done.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>PaulH</FONT></DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=rdekema@gmail.com href="mailto:rdekema@gmail.com">Rusty Dekema</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">Asterisk Users Mailing List -
Non-Commercial Discussion</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Tuesday, January 10, 2006 11:02
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Asterisk-Users] MTU and
Voice Delay (latency??)</DIV>
<DIV><BR></DIV>How far (physically) is the Asterisk server location from the
location of the phones? Have you tried pinging the Asterisk server from the
network to which the phones are connected? <BR><BR>As a rule of thumb, If the
two sites are within 2500 miles of each other and the network connection
between them is working properly, the round trip time for a 64 byte ping
should be less than 100 ms, the round trip time should not vary from one ping
to another by more than 2-5 ms (typical), and there should be virtually no
dropped packets (well under 0.1%). <BR><BR>If your network does not meet these
standards, then it may well be the cause of your problems. In that case, if
you e-mail me a traceroute from the phone location to the Asterisk location as
well as the output of a ping from the phone location to the Asterisk location
(preferably including at least 100 repetitions), I will take a look at it and
let you know what I think. <BR><BR>If your network seems fine by the above
standards, then you/we will have to pursue other Asterisk/SIP/RTP-related
avenues of troubleshooting. <BR><BR>Regards,<BR>Rusty<BR><BR><BR><BR>
<DIV><SPAN class=gmail_quote>On 1/9/06, <B class=gmail_sendername>Geoff
Manning</B> <<A href="mailto:gmanning@zoom.com">gmanning@zoom.com</A>>
wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">Our
users are experiencing some unacceptable delay when trying to have
a<BR>conversation. The delay is so noticeable that they keep stepping on
each<BR>others words and resort to calling the customers via cell
phone.<BR><BR>Here is the setup<BR><BR>SDSL Connection (PPPoA)<BR>Speedtouch
610 SDSL Modem<BR>3Com 2224PWR Plus Switch (phones on separate VLAN)<BR>8
Cisco 796 Phones<BR><BR>All connecting to a remote Asterisk
Server.<BR><BR>We found that the MTU for the SDSL modem was set to 1500 and
I have since <BR>changed it to 1458 which is the ISP's recommended
setting.<BR><BR>Can this MTU difference cause the delay my users are
experiencing? All the<BR>voice packets would become fragmented so it sounds
logical.<BR><BR>And simply changing the MTU on the modem, will that fix it,
I can't find a <BR>way to change it at the Cisco phone
level.<BR><BR>Thanks,<BR>Geoff
Manning<BR><BR>_______________________________________________<BR>--Bandwidth
and Colocation provided by <A href="http://Easynews.com">Easynews.com</A>
--<BR><BR>Asterisk-Users mailing list<BR>To UNSUBSCRIBE or update options
visit:<BR> <A
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR></BLOCKQUOTE></DIV><BR>
<P>
<HR>
<P></P>_______________________________________________<BR>--Bandwidth and
Colocation provided by Easynews.com --<BR><BR>Asterisk-Users mailing
list<BR>To UNSUBSCRIBE or update options visit:<BR>
http://lists.digium.com/mailman/listinfo/asterisk-users<BR></BLOCKQUOTE></BODY></HTML>