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<DIV id=idOWAReplyText87063 dir=ltr>
<DIV dir=ltr><FONT face=Arial color=#000000 size=2>Sorry that should read "will
be <EM>exaggerated</EM> beause of the larger payload
size."</FONT></DIV></DIV>
<DIV dir=ltr><BR>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> cisco-voip-bounces@puck.nether.net on
behalf of Joe Pollere (US)<BR><B>Sent:</B> Fri 11/10/2006 4:07 PM<BR><B>To:</B>
Scott ODonnell; cisco-voip@puck.nether.net<BR><B>Subject:</B> Re: [cisco-voip]
Optimizing RTP on the WAN.<BR></FONT><BR></DIV>
<DIV dir=ltr>
<DIV id=idOWAReplyText31838 dir=ltr>
<DIV dir=ltr><FONT face=Arial color=#000000 size=2>You could change the default
sampling rate from 20 ms to 30 ms but the bandwidth savings is really not that
significant. However If you are experiencing any dropped packets it will be
exasperated beause of the larger payload size. </FONT></DIV>
<DIV dir=ltr><FONT face=Arial size=2></FONT> </DIV>
<DIV dir=ltr><FONT face=Arial size=2>See this for reference:</FONT></DIV>
<DIV dir=ltr><FONT face=Arial color=#000000 size=2></FONT> </DIV>
<DIV dir=ltr><FONT face=Arial color=#000000 size=2><A
href="http://www.informit.com/articles/article.asp?p=357102&seqNum=1&rl=1">http://www.informit.com/articles/article.asp?p=357102&seqNum=1&rl=1</A></FONT></DIV>
<DIV dir=ltr> </DIV>
<DIV dir=ltr><snip></DIV></DIV>
<P dir=ltr>Table 2-1 details the bandwidth per VoIP flow (both G.711 and G.729)
at a default packetization rate of 50 packets per second (pps) and at a custom
packetization rate of 33 pps. This does not include Layer 2 overhead and does
not take into account any possible compression schemes, such as Compressed
Real-Time Transport Protocol (cRTP, discussed in detail in Chapter 7,
"Link-Specific Tools"). </P>
<P dir=ltr>For example, assume a G.711 VoIP codec at the default packetization
rate (50 pps). A new VoIP packet is generated every 20 ms (1 second / 50 pps).
The payload of each VoIP packet is 160 bytes; with the IP, UDP, and RTP headers
(20 + 8 + 12 bytes, respectively) included, this packet become 200 bytes in
length. Converting bits to bytes requires multiplying by 8 and yields 1600 bps
per packet. When multiplied by the total number of packets per second (50 pps),
this arrives at the Layer 3 bandwidth requirement for uncompressed G.711 VoIP:
80 kbps. This example calculation corresponds to the first row of Table 2-1.</P>
<H4 dir=ltr>Table 2-1 Voice Bandwidth (Without Layer 2 Overhead)</H4>
<DIV dir=ltr>
<TABLE cellSpacing=2 cellPadding=2 border=2>
<TR vAlign=top>
<TD vAlign=top width=74>
<P><B><FONT size=-1>Bandwidth Consumption</FONT></B></P></TD>
<TD vAlign=top width=71>
<P><B><FONT size=-1>Packetization Interval</FONT></B></P></TD>
<TD vAlign=top width=71>
<P><B><FONT size=-1>Voice Payload in Bytes </FONT></B></P></TD>
<TD vAlign=top width=71>
<P><B><FONT size=-1>Packets Per Second</FONT></B></P></TD>
<TD vAlign=top width=71>
<P><B><FONT size=-1>Bandwidth Per Conversation</FONT></B></P></TD></TR>
<TR vAlign=top>
<TD vAlign=top width=74>
<P><FONT size=-1><TT>G.711</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>20 ms</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>160</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>50</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>80 kbps</TT></FONT></P></TD></TR>
<TR vAlign=top>
<TD vAlign=top width=74>
<P><FONT size=-1><TT>G.711</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>30 ms</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>240</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>33</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>74 kbps</TT></FONT></P></TD></TR>
<TR vAlign=top>
<TD vAlign=top width=74>
<P><FONT size=-1><TT>G.729A</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>20 ms</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>20</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>50</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>24 kbps</TT></FONT></P></TD></TR>
<TR vAlign=top>
<TD vAlign=top width=74>
<P><FONT size=-1><TT>G.729A</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>30 ms</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>30</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>33</TT></FONT></P></TD>
<TD vAlign=top width=71>
<P><FONT size=-1><TT>19 kbps</TT></FONT></P></TD></TR></TABLE></DIV>
<DIV dir=ltr><BR></DIV>
<DIV class=note dir=ltr>
<P class=normaltitle><B>NOTE</B></P>
<P>The Service Parameters menu in Cisco CallManager Administration can be used
to adjust the packet rate. It is possible to configure the sampling rate above
30 ms, but this usually results in poor voice quality.</P>
<P><snip></P>
<P> </P></DIV>
<DIV dir=ltr>HTH's </DIV>
<DIV dir=ltr> </DIV>
<DIV dir=ltr>Joe<BR></DIV>
<DIV dir=ltr>
<HR tabIndex=-1>
</DIV>
<DIV dir=ltr><FONT face=Tahoma size=2><B>From:</B>
cisco-voip-bounces@puck.nether.net on behalf of Scott ODonnell<BR><B>Sent:</B>
Fri 11/10/2006 3:55 PM<BR><B>To:</B>
cisco-voip@puck.nether.net<BR><B>Subject:</B> [cisco-voip] Optimizing RTP on the
WAN.<BR></FONT><BR></DIV>
<DIV>
<DIV>I have a customer that is getting clobbered high WAN utilitization due
to RTP streams.</DIV>
<DIV>While I recognize that at some point you just need more bandwidth are there
any other ways to reduce the bandwidth RTP requires beyond codec selection and
CRTP?</DIV>
<DIV> </DIV>
<DIV>I vaguely remember back in the old "toll bypass" days, you could adjust the
number of samples per packet or the sample rate and it could make a real
difference in the bandwidth.</DIV>
<DIV> </DIV>
<DIV>Any knobs like that in CallManager?</DIV>
<DIV> </DIV>
<DIV>Scott</DIV>
<DIV> </DIV></DIV></DIV></BODY></HTML>
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