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<DIV></FONT><FONT face=Arial size=2>Thanks for your replies.</FONT></DIV>
<OL>
<LI><FONT face=Arial size=2>Our connection mainly for voip, occasionally used
for surfing websites.</FONT></LI>
<LI><FONT face=Arial size=2>We are using codec g711u for local calls through
TE120P, and g729 only if making international calls through our sip provider,
which only allow g723 and g729. How can we get the license for g723? Which
codec would you recommend?</FONT></LI>
<LI><FONT face=Arial size=2>That quality problems we are facing
are jitter, latency and occasionally low volume. What cause these
problems?</FONT></LI>
<LI><FONT face=Arial size=2>No QoS Settings as we are quite new to it. Are we
suppose to give high priority to RTP in our router? What sort of QoS and
traffic shapping would you recommend?</FONT></LI>
<LI><FONT face=Arial size=2>How many users can we expect to use voip(with good
quality) with 512kbps outbound connection?</FONT></LI></OL>
<DIV><FONT face=Arial size=2>Regards,</FONT></DIV>
<DIV><FONT face=Arial size=2>jorain</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<BLOCKQUOTE
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<DIV><FONT face=Arial size=2></FONT><BR><FONT face=Arial size=2>Date: Fri, 7
Dec 2007 10:27:36 -0500<BR>From: "C F" <</FONT><A
href="mailto:shmaltz@gmail.com"><FONT face=Arial
size=2>shmaltz@gmail.com</FONT></A><FONT face=Arial size=2>><BR>Subject:
Re: [asterisk-users] asterisk performance<BR>To: jorain <</FONT><A
href="mailto:jorain@caliber.com.sg"><FONT face=Arial
size=2>jorain@caliber.com.sg</FONT></A><FONT face=Arial size=2>>, "Asterisk
Users Mailing List -<BR>Non-Commercial Discussion" <</FONT><A
href="mailto:asterisk-users@lists.digium.com"><FONT face=Arial
size=2>asterisk-users@lists.digium.com</FONT></A><FONT face=Arial
size=2>><BR>Message-ID:<BR><</FONT><A
href="mailto:81000b5a0712070727s32156f31y4986abc9054144@mail.gmail.com"><FONT
face=Arial
size=2>81000b5a0712070727s32156f31y4986abc9054144@mail.gmail.com</FONT></A><FONT
face=Arial size=2>><BR>Content-Type: text/plain;
charset=ISO-8859-1<BR><BR>by 3rd call do you mean over the internet?<BR>if the
answer is yes, then I wouldn't be surprised. another thing what<BR>codec are
you using?<BR></FONT></DIV>
<DIV><BR><FONT face=Arial size=2>Date: Fri, 7 Dec 2007 17:02:31 +0000<BR>From:
"Giovanni Miano" <</FONT><A href="mailto:giomiano@gmail.com"><FONT
face=Arial size=2>giomiano@gmail.com</FONT></A><FONT face=Arial
size=2>><BR>Subject: Re: [asterisk-users] asterisk performance<BR>To:
"Asterisk Users Mailing List - Non-Commercial Discussion"<BR><</FONT><A
href="mailto:asterisk-users@lists.digium.com"><FONT face=Arial
size=2>asterisk-users@lists.digium.com</FONT></A><FONT face=Arial
size=2>><BR>Message-ID:<BR><</FONT><A
href="mailto:d75be1ca0712070902u6d25ee49w368eda405a32bce8@mail.gmail.com"><FONT
face=Arial
size=2>d75be1ca0712070902u6d25ee49w368eda405a32bce8@mail.gmail.com</FONT></A><FONT
face=Arial size=2>><BR>Content-Type: text/plain;
charset=ISO-8859-1<BR><BR>2007/12/7, C F <</FONT><A
href="mailto:shmaltz@gmail.com"><FONT face=Arial
size=2>shmaltz@gmail.com</FONT></A><FONT face=Arial size=2>>:<BR>> by
3rd call do you mean over the internet?<BR>> if the answer is yes, then I
wouldn't be surprised.<BR><BR>Oh my god!<BR>If it is over internet and you get
crap quality.. you have to be surprised..<BR>It is depends by Latency (Traffic
congestion, Network congestion) and<BR>Packet
loss<BR>---------------------------------------------------------------------------------<BR><BR>jorain,<BR>What
do you mean for "quality problem" ?<BR>Different "quality" problems are
generated by different parameter<BR><BR>braking ? echo? low volume
?<BR><BR>Cheers</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial><FONT size=2><B>From:</B> </FONT></FONT><A
title=mgraves@mstvp.com href="mailto:mgraves@mstvp.com"><FONT face=Arial
size=2>Michael Graves</FONT></A><FONT face=Arial size=2> </FONT></DIV>
<DIV><FONT face=Arial><FONT size=2><B>To:</B> </FONT></FONT><A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com"><FONT face=Arial size=2>Asterisk
Users Mailing List - Non-Commercial Discussion</FONT></A><FONT face=Arial
size=2> ; </FONT><A title=jorain@caliber.com.sg
href="mailto:jorain@caliber.com.sg"><FONT face=Arial
size=2>jorain</FONT></A><FONT face=Arial size=2> </FONT></DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Saturday, December 08, 2007 12:00
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [asterisk-users] asterisk
performance</DIV>
<DIV><FONT face=Arial><BR><FONT size=2></FONT></FONT></DIV>
<DIV><FONT face=Arial size=2 POINTSIZE="11" DEFAULT="SIZE">Your 512k outbound
bandwidth will tend to be the defining factor in call quality here.
<BR><BR>Does your connection only gets used for voip? Or is it shared with
other uses? <BR><BR>Can you use more compressed codecs? G729 will quadruple
you call capacity.<BR><BR>What sort of QoS and traffic shaping do you use?
Note that these are separate matters, and you need
both.<BR><BR>Michael</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>--Original Message Text---<BR><B>From:</B>
jorain<BR><B>Date:</B> Thu, 6 Dec 2007 17:47:18 +0800<BR><BR></FONT><FONT
face=Arial size=2 POINTSIZE="10">Hi all, <BR></FONT><FONT POINTSIZE="11"
DEFAULT="SIZE"><FONT POINTSIZE="10"><BR><FONT face=Arial size=2>We are using
<BR>- a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front
size bus 2MB cache) as the asterisk server <BR>- dell 400sc</FONT><FONT
POINTSIZE="12"><FONT face=Arial size=2>(Intel P4) as a SER server
<BR></FONT><FONT DEFAULT="FACE"><FONT face=Arial size=2 POINTSIZE="10">-
digium isdn card, TE120P at Asterisk server <BR>- Bandwidth: 2Mbps/512kbps
<BR><BR>All SIP Phones are registered to SER server, and SER will route all
outgoing calls to Asterisk server. My problem is the sound quality goes down
if more than 3 concurent calls to PSTN. <BR><BR>Logically i think our system
and bandwidth are more than enough to handle 3 concurent calls, but as the 4th
person use it, the sound become jerky and a bit delay. So how can we improve
the sound quality? <BR><BR><BR>Thanks <BR><BR>Regards, <BR>jorain
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