[55766] in North American Network Operators' Group
RE: VoIP QOS best practices
daemon@ATHENA.MIT.EDU (Spencer.Wood@dot.state.oh.us)
Mon Feb 10 15:12:26 2003
In-Reply-To: <2697FBAC6B1EF643B30E7B16B6E6958F0E9C88@master.oneunified.net>
To: "Ray Burkholder" <ray@oneunified.net>
Cc: nanog@nanog.org
From: Spencer.Wood@dot.state.oh.us
Date: Mon, 10 Feb 2003 14:57:34 -0500
Errors-To: owner-nanog-outgoing@merit.edu
This is a multipart message in MIME format.
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Also note that those sizes are for the voice part of the payload
only....It does not take into account any payload/packet overhead...
We use G.711 quite a bit on our network, and are traffic flows are right
around 80k...
Spencer
************************************************************
Spencer Wood, Network Manager
Ohio Department Of Transportation
1320 Arthur E. Adams Drive
Columbus, Ohio 43221
E-Mail: Spencer.Wood@dot.state.oh.us
Phone: 614.644.5422/Fax: 614.887.4021/Pager: 866.591.9954
*************************************************************
"Ray Burkholder" <ray@oneunified.net>
Sent by: owner-nanog@merit.edu
02/10/2003 02:21 PM
To: "Charles Youse" <cyouse@register.com>, "Alec H. Peterson"
<ahp@hilander.com>
cc: <nanog@nanog.org>
Subject: RE: VoIP QOS best practices
G.711 gives you the 64kbps quality you get on a channel in a PRI line.
No compression is performed.
G.729 is a well accepted codec that performs compression, and with ip
packet overhead, uses about 16 to 24 kbps (can't remember which). It
gives voice quality very close to G.711.
G.723 has a noticeable voice quality change, and is in the 6 to 8 kbps
range.
The optimal is G.729 for quality vs bandwidth issues.
There are some other considerations involved but these are the main
ones.
Ray Burkholder
> -----Original Message-----
> From: Charles Youse [mailto:cyouse@register.com]
> Sent: February 10, 2003 14:42
> To: Alec H. Peterson
> Cc: nanog@nanog.org
> Subject: RE: VoIP QOS best practices
>
>
>
> Speaking of codecs, what are the primary variables one uses
> when choosing a codec? I imagine this is some function of
> how much bandwidth you want to use versus how much CPU to
> encode the voice stream.
>
> C.
>
> -----Original Message-----
> From: Alec H. Peterson [mailto:ahp@hilander.com]
> Sent: Monday, February 10, 2003 1:40 PM
> To: Bill Woodcock; Charles Youse
> Cc: nanog@nanog.org
> Subject: RE: VoIP QOS best practices
>
>
> --On Monday, February 10, 2003 10:19 -0800 Bill Woodcock
> <woody@pch.net>
> wrote:
>
> >
> > It works fine on 64k connections, okay on many 9600bps
> connections. T1 is
> > way more than is necessary.
>
> I'd say that largely depends on which codec you are using and
> how many
> simultaneous calls you will have going.
>
> Alec
>
> --
> Alec H. Peterson -- ahp@hilander.com
> Chief Technology Officer
> Catbird Networks, http://www.catbird.com
>
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<br><font size=2 face="sans-serif">Also note that those sizes are for the
voice part of the payload only....It does not take into account any payload/packet
overhead...</font>
<br>
<br><font size=2 face="sans-serif">We use G.711 quite a bit on our network,
and are traffic flows are right around 80k...</font>
<br>
<br><font size=2 face="sans-serif">Spencer</font>
<br><font size=2 face="sans-serif"><br>
</font><font size=2 face="Courier New">************************************************************<br>
Spencer Wood, Network Manager<br>
Ohio Department Of Transportation<br>
1320 Arthur E. Adams Drive<br>
Columbus, Ohio 43221</font><font size=3> </font>
<p><font size=2 face="Courier New">E-Mail: </font><a href=mailto:Spencer.Wood@dot.state.oh.us><font size=2 color=blue face="Courier New"><u>Spencer.Wood@dot.state.oh.us</u></font></a><font size=2 face="Courier New"><br>
Phone: 614.644.5422/Fax: 614.887.4021/Pager: 866.591.9954</font><font size=3>
</font><font size=2 face="Courier New"><br>
*************************************************************</font><font size=3>
</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td>
<td><font size=1 face="sans-serif"><b>"Ray Burkholder" <ray@oneunified.net></b></font>
<br><font size=1 face="sans-serif">Sent by: owner-nanog@merit.edu</font>
<p><font size=1 face="sans-serif">02/10/2003 02:21 PM</font>
<td><font size=1 face="Arial"> </font>
<br><font size=1 face="sans-serif"> To:
"Charles Youse" <cyouse@register.com>,
"Alec H. Peterson" <ahp@hilander.com></font>
<br><font size=1 face="sans-serif"> cc:
<nanog@nanog.org></font>
<br><font size=1 face="sans-serif"> Subject:
RE: VoIP QOS best practices</font></table>
<br>
<br>
<br><font size=2><tt><br>
G.711 gives you the 64kbps quality you get on a channel in a PRI line.<br>
No compression is performed.<br>
<br>
G.729 is a well accepted codec that performs compression, and with ip<br>
packet overhead, uses about 16 to 24 kbps (can't remember which). It<br>
gives voice quality very close to G.711.<br>
<br>
G.723 has a noticeable voice quality change, and is in the 6 to 8 kbps<br>
range.<br>
<br>
The optimal is G.729 for quality vs bandwidth issues. <br>
<br>
There are some other considerations involved but these are the main<br>
ones.<br>
<br>
Ray Burkholder<br>
<br>
<br>
> -----Original Message-----<br>
> From: Charles Youse [mailto:cyouse@register.com] <br>
> Sent: February 10, 2003 14:42<br>
> To: Alec H. Peterson<br>
> Cc: nanog@nanog.org<br>
> Subject: RE: VoIP QOS best practices<br>
> <br>
> <br>
> <br>
> Speaking of codecs, what are the primary variables one uses <br>
> when choosing a codec? I imagine this is some function of <br>
> how much bandwidth you want to use versus how much CPU to <br>
> encode the voice stream.<br>
> <br>
> C.<br>
> <br>
> -----Original Message-----<br>
> From: Alec H. Peterson [mailto:ahp@hilander.com]<br>
> Sent: Monday, February 10, 2003 1:40 PM<br>
> To: Bill Woodcock; Charles Youse<br>
> Cc: nanog@nanog.org<br>
> Subject: RE: VoIP QOS best practices<br>
> <br>
> <br>
> --On Monday, February 10, 2003 10:19 -0800 Bill Woodcock <br>
> <woody@pch.net> <br>
> wrote:<br>
> <br>
> ><br>
> > It works fine on 64k connections, okay on many 9600bps <br>
> connections. T1 is<br>
> > way more than is necessary.<br>
> <br>
> I'd say that largely depends on which codec you are using and <br>
> how many <br>
> simultaneous calls you will have going.<br>
> <br>
> Alec<br>
> <br>
> --<br>
> Alec H. Peterson -- ahp@hilander.com<br>
> Chief Technology Officer<br>
> Catbird Networks, http://www.catbird.com<br>
> <br>
</tt></font>
<br>
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