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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>One thing to add…<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I am testing now by saturating the link
and making an internal call to another IP phone at another remote site that of
course goes over the same T1 (plus another T1) and I do not see any jitter or
rxlost problems. I had noticed that I never saw any problem to our other
office so I wanted to test it now. Max jitter is only 29 and no rxlost in 12
minutes so far with the T1 saturated plus the stream has to go over another T1
as well.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Could this be an issue with the 6608? I
remember coming across a bug about the sample size changing from 10ms to 20ms
using g729 on a 6608 resulting in poor quality. That bug fit perfectly, but it
was for ccm 3.0.10 I think.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Anyway, this is getting stranger by the
minute… ;) <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b><span
style='font-weight:bold'>On Behalf Of </span></b>Keith Klevenski<br>
<b><span style='font-weight:bold'>Sent:</span></b> Tuesday, October 24, 2006
4:43 PM<br>
<b><span style='font-weight:bold'>To:</span></b> cisco-voip@puck.nether.net<br>
<b><span style='font-weight:bold'>Subject:</span></b> [cisco-voip] Strict
Priority Queuing over T1 for Voice</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Hi all,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I’ve been battling a problem that seemed to have
surfaced recently as we have added more users to the office. We have an
office of about 35 users now with a point to point T1 (HDLC) between this
office and the datacenter where the CallManager’s (running 4.1.3) and
PSTN gateway (6608 blade) are. Basically this:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>PSTN<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> |<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>6608/6509 (datacenter) (12.1(11b)E14 on msfc and 6.4(10) on
sup)<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> | trunk<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>7200 (datacenter) 12.3(6a)<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> |<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>T1<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> |<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>2811 (remote office) 12.3(11)T7<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> | trunk<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>3750 (remote office)<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> |<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Phones<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>We have strict priority queuing configured on the T1
interfaces (and the FastEthernet interfaces although I don’t think this
is necessary) with DSCP EF traffic in the high queue and signaling (AF31/21/22/23
and CS3) the medium queue and other traffic in the normal and low queues.
<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>The problem is when the T1 is saturated we start seeing high
jitter (1-2 seconds) and rxlost packets on the phones and voice quality of
course suffers. The thing is I see NO dropped packets on any interface
from the PSTN to the switch port the phone is connected to other than the call
statistics on the phone. This confuses me greatly unless the phone itself
is dropping them… it would seem that if the phone determines there are
missing packets (rxlost) they would show up on some interface somewhere but
they do not. Sniffer traces at the phone show the missing packets.
Sniffer traces at the trunk at the remote office show missing packets, but
traces on the trunk in the datacenter (from the 7200 to 6509) show all packets
accounted for. Seems like the T1 is the issue. There are 24 line
code violations on the T1 in the last 24 hours, but no slips. The problem
only seems to occur when there is high traffic on the T1. We also noticed
no other traffic is marked EF and the RTP stream is EF at the datacenter and at
the office so that isn’t being remarked or anything. But there are
no input queue drops on the 2811 (which I wouldn’t expect since the RTP
stream should be CEF switched anyway). If the input interface cannot
handle the amount of traffic and the traffic is all CEF switched where would
you see input drops? There are a handful of buffer failures, but nothing
that would account for the number of complaints I’ve been getting lately.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I’ve connected my phone directly to the 2811 and
bypassed the 3750 with the same results.<br>
<br>
I also noticed that the rxsize fluctuates between 10ms and 20ms also.
Shouldn’t that always be 20ms? I thought that was odd.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I guess my main question is when using strict priority
queuing shouldn’t the voice packets ALWAYS get sent first no matter
what?? I want to believe they are since there are NO drops in the high
priority queue outbound from the datacenter. What is the best way to do
QoS for voice over a T1 when you only care about the voice? Seems like
strict priority queuing is the best way to assure voice traffic is sent
first. We don’t really care about starving out any other
traffic. If packets are being dropped shouldn’t I see them somewhere
other than on the phone?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Any input on this problem is welcome!<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Keith</span></font><o:p></o:p></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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