I've been able to track it down with CCM traces and doing some filtering with translator x.<br><br>
<div class="gmail_quote">On Tue, Mar 20, 2012 at 12:26 PM, Wes Sisk <span dir="ltr"><<a href="mailto:wsisk@cisco.com">wsisk@cisco.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PADDING-LEFT:1ex" class="gmail_quote">
<div style="WORD-WRAP:break-word">the only way that comes to mind is checking the remote IP address. from the phone's perspective the remote IP would be the MTP if invoked. Otherwise it would be the gateway.
<div><br>
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<div>
<div>On Mar 20, 2012, at 10:38 AM, Erick Wellnitz wrote:</div><br>
<div>I'm drawing a blank. Am I able to tell what calls are hitting which MTP? <br><br></div>
<div class="gmail_quote">On Mon, Mar 19, 2012 at 2:33 PM, Wes Sisk <span dir="ltr"><<a href="mailto:wsisk@cisco.com" target="_blank">wsisk@cisco.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PADDING-LEFT:1ex" class="gmail_quote">
<div style="WORD-WRAP:break-word">so long as MRGL are correct and there is a capabilities match then yes.
<div><br>
<div>
<div>On Mar 19, 2012, at 3:30 PM, Erick Wellnitz wrote:</div><br>
<div>I'm going to eliminate our CM based software MTPs with n IOS software MTP.</div>
<div> </div>
<div>So, if I configure a trunk with the new MRGL, the trunk will use this to allocate the MTP, correct?<br><br></div>
<div class="gmail_quote">On Mon, Mar 19, 2012 at 1:20 PM, Wes Sisk <span dir="ltr"><<a href="mailto:wsisk@cisco.com" target="_blank">wsisk@cisco.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PADDING-LEFT:1ex" class="gmail_quote">
<div style="WORD-WRAP:break-word">stated alternatively, does any phone colocated with ccm have any issues with vq when using software media resources such as MTP or CFB? stream statistics from the phone (ii, ??, qrt, or cmr records) would help quantify any anomalies in the RTP stream. If your CMR records show consistent poor audio at that remote site then you have a much stronger pointer to the culprit.
<div><br></div>
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<div><br>
<div>
<div>On Mar 19, 2012, at 11:56 AM, Erick Wellnitz wrote:</div><br>
<div>CPU is bleow 20% on all nodes. Network utilization isn't out of the ordinary either.</div>
<div> </div>
<div><br><br> </div>
<div class="gmail_quote">On Mon, Mar 19, 2012 at 10:26 AM, Wes Sisk <span dir="ltr"><<a href="mailto:wsisk@cisco.com" target="_blank">wsisk@cisco.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PADDING-LEFT:1ex" class="gmail_quote">
<div style="WORD-WRAP:break-word">generally you should deplete mtp resources before voice quality impact. software MTP utilizes network I/O and CPU utilization. we've rarely seen VQ issues caused by sw mtp/cfg but it's not unheard of. if you're using sw media resources then verify the network interface for the CUCM server.
<div><br>
<div>
<div>On Mar 16, 2012, at 3:03 PM, Erick Wellnitz wrote:</div><br>
<div>This is unreal. They tell me it's fixed and tested...but two hours later we get more complaints.</div>
<div> </div>
<div>These guys have me starting to doubt myself. CM software MTP wouldn't be causing QoS-like issues, would it? What about if an SIP trunk with "MTP required" checked and we run out of media resources? <br>
<br></div>
<div class="gmail_quote">On Fri, Mar 16, 2012 at 9:48 AM, Erick Wellnitz <span dir="ltr"><<a href="mailto:ewellnitzvoip@gmail.com" target="_blank">ewellnitzvoip@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PADDING-LEFT:1ex" class="gmail_quote">
<div>There was an incorrect QoS setting on a different switch on the remote end. Problem solved.</div>
<div> </div>
<div>I'll save my gripes about groups not talking to each other for another day. ;)<br><br></div>
<div class="gmail_quote">On Thu, Mar 15, 2012 at 4:10 PM, Erick Wellnitz <span dir="ltr"><<a href="mailto:ewellnitzvoip@gmail.com" target="_blank">ewellnitzvoip@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PADDING-LEFT:1ex" class="gmail_quote">
<div>I'm not a LAN/Ethernet QoS guru by any means. Our network group has control over QoS and I'm trying to get to the bottom of an issue we've seen routing calls over a sip trunk to a location using a 1gig Ethernet connection on a DWDM ring. No firewall is present. The configuration doesn't seem quite right to me for some reason.</div>
<div> </div>
<div>Basically what we are doing is using an alternate PSTN gateway for toll free traffic until we get additional capacity added to handle the calls. Users report garbled audio and an 'underwater' quality of the audio. Getting information from the phone on call quality (jitter, etc) may not help because the calls complained about are to an external conference bridge via tol lfree number.</div>
<div> </div>
<div>If this config checks out, I plan to ask the network team if they can capture some data for this particular link.</div>
<div> </div>
<div> </div>
<div>Here is the local end QoS config:</div>
<div> </div>
<div><font color="#004080" size="3" face="Calibri"><font color="#004080" size="3" face="Calibri"><font color="#004080" size="3" face="Calibri">
<p dir="ltr">mls ip cef load-sharing full</p>
<p dir="ltr">mls netflow interface</p>
<p dir="ltr">mls qos map cos-dscp 0 8 16 24 32 46 48 56</p>
<p dir="ltr">mls qos</p>
<p dir="ltr">mls cef error action reset</p>
<div> <br></div></font></font></font></div>
<div><font color="#004080" size="3" face="Calibri"><font color="#004080" size="3" face="Calibri"><font color="#004080" size="3" face="Calibri">
<p dir="ltr">interface GigabitEthernet3/43</p>
<p dir="ltr">description GIG 3/43 </p>
<p dir="ltr">ip address X.X.X.X X.X.X.X</p>
<p dir="ltr">ip flow ingress</p>
<p dir="ltr">ip pim sparse-dense-mode</p>
<p dir="ltr">ip summary-address eigrp 1 X.X.X X X.X.X.X</p>
<p dir="ltr">ip summary-address eigrp 1 X.X.X X X.X.X.X</p>
<p dir="ltr">ip summary-address eigrp 1 X.X.X X X.X.X.X</p>
<p dir="ltr">load-interval 30</p>
<p dir="ltr">wrr-queue bandwidth 30 40 30</p>
<p dir="ltr">wrr-queue queue-limit 40 30 15</p>
<p dir="ltr">wrr-queue threshold 2 60 80 100 100 100 100 100 100</p>
<p dir="ltr">wrr-queue threshold 3 60 80 100 100 100 100 100 100</p>
<p dir="ltr">wrr-queue random-detect min-threshold 1 40 60 80 80 80 80 80 80</p>
<p dir="ltr">wrr-queue random-detect max-threshold 1 70 80 100 100 100 100 100 100</p>
<p dir="ltr">no wrr-queue random-detect 2</p>
<p dir="ltr">no wrr-queue random-detect 3</p>
<p dir="ltr">wrr-queue cos-map 1 3 0</p>
<p dir="ltr">wrr-queue cos-map 2 1 1</p>
<p dir="ltr">wrr-queue cos-map 2 2 2</p>
<p dir="ltr">wrr-queue cos-map 2 3 4</p>
<p dir="ltr">wrr-queue cos-map 3 2 3</p>
<p dir="ltr">wrr-queue cos-map 3 3 6 7</p>
<p dir="ltr">mls qos trust dscp</p>
<div> <br></div>
<p dir="ltr">Here is the remote end:</p><font color="#004080" size="3" face="Calibri"><font color="#004080" size="3" face="Calibri"><font color="#004080" size="3" face="Calibri">
<p dir="ltr">interface GigabitEthernet1/36</p>
<p dir="ltr">description GIGABIT E/NET 1/36 </p>
<p dir="ltr">ip address X.X.X.X X.X.X.X</p>
<p dir="ltr">ip flow ingress</p>
<p dir="ltr">ip pim sparse-dense-mode</p>
<p dir="ltr">load-interval 30</p>
<p dir="ltr">no wrr-queue random-detect 2</p>
<p dir="ltr">no wrr-queue random-detect 3</p>
<p dir="ltr">wrr-queue cos-map 1 3 0</p>
<p dir="ltr">wrr-queue cos-map 2 1 1</p>
<p dir="ltr">wrr-queue cos-map 2 2 2</p>
<p dir="ltr">wrr-queue cos-map 2 3 4</p>
<p dir="ltr">wrr-queue cos-map 3 2 3</p>
<p dir="ltr">wrr-queue cos-map 3 3 6 7</p>
<p dir="ltr">mls qos trust dscp</p></font></font></font></font></font></font></div></blockquote></div><br></blockquote></div><br>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
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</div><br></div><br></div></div></blockquote></div><br></div><br></div></div></div></blockquote></div><br>