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<DIV><FONT face=Arial size=2>thanks Frank.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Calculating the bandwith with these tools shouldn´t
be hard, I assume. The other part of my question is what has me more worried.
That is, are there any instances where depending on the codec (or some
other factors) the RTP stream also goes through * rather than just from A to B.
Any ideas?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
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<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=frank.hoonhout@state.or.us
href="mailto:frank.hoonhout@state.or.us">Frank Hoonhout</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@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Friday, April 25, 2003 12:31
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> RE: [Asterisk-Users] bandwith
calculation</DIV>
<DIV><BR></DIV>
<DIV><SPAN class=296302815-25042003><FONT face=Arial color=#0000ff size=2>I
found these on these.</FONT></SPAN></DIV>
<DIV><SPAN class=296302815-25042003><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=296302815-25042003><FONT face=Arial color=#0000ff size=2>Web
based</FONT></SPAN></DIV>
<DIV><SPAN class=296302815-25042003><FONT face=Arial color=#0000ff size=2><A
href="http://www.packetizer.com/iptel/bandcalc.html">http://www.packetizer.com/iptel/bandcalc.html</A></FONT></SPAN></DIV>
<DIV><SPAN class=296302815-25042003><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=296302815-25042003><FONT face=Arial color=#0000ff
size=2>Window's executable program (my favorite)</FONT></SPAN></DIV>
<DIV><SPAN class=296302815-25042003><FONT face=Arial color=#0000ff size=2><A
href="http://www.networkingfiles.com/Network/clarent.htm">http://www.networkingfiles.com/Network/clarent.htm</A></FONT></SPAN></DIV>
<DIV><SPAN class=296302815-25042003><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=296302815-25042003><FONT face=Arial color=#0000ff
size=2>Frank...</FONT></SPAN></DIV>
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<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left><FONT
face=Tahoma size=2>-----Original Message-----<BR><B>From:</B> <A
href="mailto:asterisk-users-admin@lists.digium.com">asterisk-users-admin@lists.digium.com</A>
[mailto:asterisk-users-admin@lists.digium.com] <B>On Behalf Of </B>Dan
Fernandez<BR><B>Sent:</B> Thursday, April 24, 2003 1:28 PM<BR><B>To:</B>
asterisk-users@lists.digium.com<BR><B>Subject:</B> [Asterisk-Users] bandwith
calculation<BR><BR></FONT></DIV>
<DIV><FONT face=Arial size=2>I would like to know how to calculate
the amount of bandwith I would need to host X number of
calls.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>For example, if user A in San Francisco with an
ATA 186 calls user B in New York with an ATA 186 and Asterisk is being
hosted in a PC in Miami. How much bandwith do I need to have in Miami?
Do I just need bandwith for the setup of the call (ie the SIP part) or are
there any instances where the RTP stream goes through Miami as
well? </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
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