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<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left><FONT
face=Tahoma>-----Original Message-----<BR><B>From:</B>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of
</B>BSUMRALLL@aol.com<BR><B>Sent:</B> Sunday, July 24, 2005 9:11
PM<BR><B>To:</B> asterisk-users@lists.digium.com<BR><B>Subject:</B> Re:
[Asterisk-Users] super high bandwidth codec<BR><BR></FONT></DIV><FONT
id=role_document>
<DIV>It has nothing to do with bandwidth.</DIV>
<DIV>It has everything to do with your routing gear!<SPAN
class=839145704-25072005> </SPAN></DIV>
<DIV><SPAN class=839145704-25072005></SPAN> </DIV></BLOCKQUOTE>
<DIV dir=ltr><SPAN class=839145704-25072005>This is completely incorrect. Skype
uses a codec that uses far more bandwidth than traditional telephony provides,
which is why it's audio can have more range than even the best quality
phone call. In theory, there is nothing preventing an all VOIP network from
using such a codec, but as a practical matter, at least part of most phone calls
are via traditional phone gear and/or networks, you don't see it widely
deployed.</SPAN><SPAN
class=839145704-25072005> </SPAN></DIV></FONT></BODY></HTML>
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