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<FONT 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>
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What sort of QoS and traffic shaping do you use? Note that these are separate matters, and you need both.<BR>
<BR>
Michael<BR>
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--Original Message Text---<BR>
<B>From:</B> jorain<BR>
<B>Date:</B> Thu, 6 Dec 2007 17:47:18 +0800<BR>
<BR>
<FONT SIZE="2" POINTSIZE="10">Hi all, <BR>
<BR>
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 FACE="Times New Roman"><FONT SIZE="3" POINTSIZE="12">(Intel P4) as a SER server <BR>
<FONT FACE="Arial" DEFAULT="FACE"><FONT 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 <BR>
<BR>
<BR>
<BR>
<BR>
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--<br>
Michael Graves<br>
mgraves<at>mstvp.com<br>
o713-861-4005<br>
c713-201-1262<br>
sip:mjgraves@pixelpower.onsip.com<br>
skype mjgraves<br>
fwd 54245</HTML>