Hi Wes,<br><br>Here is the call statistics from the phone for one call <br><br>Rcvr Codec G729<br>Sender Codec G729<br>Rcvr size 20ms<br>sender size 20ms<br>Rcvr Packets 476<br>sender packets 709<br>Avg Jitter 31<br>
Max Jitter 185<br>Rcvr discarded 1<br>Rcvr lost packets 0<br>Avg MOS LQK 0.0<br>Min MOS LQK 0.0<br>Max MOS LQK 0.0<br>Cumulative Conceal ratio 0.0319<br>Max Conceal ratio 0.0446<br>Conceal sec 5<br>Severly conceal sec 1<br>
<br>Is the jitter too high? <br><br>regards,<br><br>Abebe<br><br><div class="gmail_quote">On Fri, Feb 10, 2012 at 8:32 PM, Lelio Fulgenzi <span dir="ltr"><<a href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div bgcolor="#FFFFFF"><div>I was surprised to find that SLA is not included in the IPBase module of v15, nor in the UC module. You need the Data module. <br>
<br>Sent from my iPhone...<div><br></div><div>"There's no place like 127.0.0.1"</div></div><div><div class="h5"><div><br>On Feb 10, 2012, at 12:20 PM, Dennis Heim <<a href="mailto:Dennis.Heim@cdw.com" target="_blank">Dennis.Heim@cdw.com</a>> wrote:<br>
<br></div><div></div><blockquote type="cite"><div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Maybe a good place for some IP SLA monitoring.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<div>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:#1f497d">Dennis Heim<br>
Senior Engineer (Unified Communications)<br>
CDW Advanced Technology Services<br>
10610 9<sup>th</sup> Place<br>
Bellevue, WA 98004<br>
<br>
425.310.5299 Single Number Reach (WA)<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:#1f497d">317.569.4255 Single Number Reach (IN)<br>
317.569.4201 Fax</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">
<br>
<a href="mailto:dennis.heim@cdw.com" target="_blank"><span style="font-size:10.0pt">dennis.heim@cdw.com</span></a></span><u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:blue"><br>
</span></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><a href="http://www.cdw.com/content/solutions/unified-communications/" target="_blank">cdw.com/content/solutions/unified-communications/</a><u></u><u></u></span></p>
</div>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<div>
<div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>]
<b>On Behalf Of </b>Wes Sisk<br>
<b>Sent:</b> Friday, February 10, 2012 10:36 AM<br>
<b>To:</b> Abebe Amare<br>
<b>Cc:</b> cisco voip<br>
<b>Subject:</b> Re: [cisco-voip] intercluster trunk over IPSec VPN<u></u><u></u></span></p>
</div>
</div>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">most likely still packet throughput issues. packets may be late to the point of discarded. they would not technically be lost in that case.<u></u><u></u></p>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">this would manifest as high jitter. setup the initial all and press the "i" or "?" button twice on the phone to see call statistics. beyond that take a packet capture. wireshark has some decent RTP analysis tools built in.<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">/wes<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
<div>
<div>
<p class="MsoNormal">On Feb 10, 2012, at 6:41 AM, Abebe Amare wrote:<u></u><u></u></p>
</div>
<p class="MsoNormal" style="margin-bottom:12.0pt"><br>
Dears, thank you all for the excellent support<br>
<br>
I managed to keep the VPN tunnel up be sending periodic ping but the problem still persist. Bandwidth is reserved for at least four calls (taking into consideration VPN overhead) on a Packetshaper and the call quality is good mid-conversation. But it is is
clipping the first few seconds. I dont see any packet loss n the CMR records for a test call. What should I be looking for?<br>
<br>
thanks in advance<br>
<br>
Abebe<br>
<br>
<br>
<u></u><u></u></p>
<div>
<p class="MsoNormal">On Thu, Feb 9, 2012 at 5:57 PM, Wes Sisk <<a href="mailto:wsisk@cisco.com" target="_blank">wsisk@cisco.com</a>> wrote:<u></u><u></u></p>
<div>
<p class="MsoNormal">TCP keepalives are only used while a call is active.<u></u><u></u></p>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">When no call is active there is no active h323/h225/h245 signaling, tcp session, or udp. The only exception is when gatekeeper is used. Then gk registration messages are maintained. Those are over UDP between the h323 ep and gk.<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">for a static ICT defined between two CUCM clusters there is no network activity without an active call.<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">For the duration of an active call the tcp keepalive parameter will help.<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">regards,<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">wes<u></u><u></u></p>
</div>
<div>
<div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
<div>
<div>
<p class="MsoNormal">On Feb 9, 2012, at 8:13 AM, Adam Frankel (afrankel) wrote:<u></u><u></u></p>
</div>
<p class="MsoNormal"><u></u> <u></u></p>
<div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt"><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">Options Ping was added in 8.5(1).<br>
<br>
The parameter "Allow TCP KeepAlives For H323 " should take care of this for H323 ICT.
<br>
<br>
-Adam<br>
<br>
<u></u><u></u></span></p>
<div class="MsoNormal" style="text-align:center" align="center"><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">
<hr align="center" size="2" width="100%">
</span></div>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> Abebe Amare
<a href="mailto:abucho@gmail.com" target="_blank"><abucho@gmail.com></a><br>
<b>Sent:</b> Thu, Feb 09, 2012 4:52:50 AM<br>
<b>To:</b> Ryan Ratliff <a href="mailto:rratliff@cisco.com" target="_blank"><rratliff@cisco.com></a><br>
<b>CC:</b> cisco voip <a href="mailto:cisco-voip@puck.nether.net" target="_blank">
<cisco-voip@puck.nether.net></a><br>
<b>Subject:</b> Re: [cisco-voip] intercluster trunk over IPSec VPN<br>
</span><span style="font-size:10.0pt;font-family:"Arial","sans-serif""><br>
<br>
<u></u><u></u></span></p>
<p class="MsoNormal" style="margin-bottom:12.0pt"><span style="font-size:10.0pt">Hi Ryan,<br>
<br>
The CUCM version is 6.1.3.1000-16. Is the SIP options ping parameter available in this version? Where would you enable it if it is available?<br>
<br>
thanks in Advance,<br>
<br>
Abebe<u></u><u></u></span></p>
<div>
<p class="MsoNormal"><span style="font-size:10.0pt">On Wed, Feb 8, 2012 at 8:07 PM, Ryan Ratliff <<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>> wrote:<u></u><u></u></span></p>
<div>
<p class="MsoNormal"><span style="font-size:10.0pt">What about a SIP trunk with options ping enabled?
<u></u><u></u></span></p>
<div>
<p class="MsoNormal"><span style="font-size:10.0pt"><u></u> <u></u></span></p>
<div>
<div>
<p class="MsoNormal"><span style="font-size:13.5pt;font-family:"Helvetica","sans-serif"">-Ryan<u></u><u></u></span></p>
</div>
</div>
<p class="MsoNormal"><span style="font-size:10.0pt"><u></u> <u></u></span></p>
<div>
<div>
<div>
<div>
<p class="MsoNormal"><span style="font-size:10.0pt">On Feb 8, 2012, at 7:05 AM, Abebe Amare wrote:<u></u><u></u></span></p>
</div>
<p class="MsoNormal" style="margin-bottom:12.0pt"><span style="font-size:10.0pt"><br>
Hi Dennis,<br>
<br>
Configuring a persistent L2L tunnel proved to be very elusive. I settled for running a periodic ping scheduled to keep the tunnel running.<br>
<br>
Thanks for your help<br>
<br>
Abebe<u></u><u></u></span></p>
<div>
<p class="MsoNormal"><span style="font-size:10.0pt">On Tue, Feb 7, 2012 at 6:16 PM, Dennis Heim <<a href="mailto:Dennis.Heim@cdw.com" target="_blank">Dennis.Heim@cdw.com</a>> wrote:<u></u><u></u></span></p>
<div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">I think you answered your own question. IPSEC tunnel’s take time to bring up. Maybe you could tweak
some of the VPN negotiating parameters, or create a separate L2 tunnel profile/group just for your voice that is permanent and does not have an inactivity timer.</span><u></u><u></u></p>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><span style="font-size:10.0pt"><u></u><u></u></span></p>
</div>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><span style="font-size:10.0pt"><u></u><u></u></span></p>
</div>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:#1f497d">Dennis Heim<br>
Senior Engineer (Unified Communications)<br>
CDW Advanced Technology Services<br>
10610 9<sup>th</sup> Place<br>
Bellevue, WA 98004<br>
<br>
425.310.5299 Single Number Reach (WA)</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:#1f497d">317.569.4255 Single Number Reach (IN)<br>
317.569.4201 Fax</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><br>
<a href="mailto:dennis.heim@cdw.com" target="_blank"><span style="font-size:10.0pt">dennis.heim@cdw.com</span></a></span><u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:blue"><br>
</span></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><a href="http://www.cdw.com/content/solutions/unified-communications/" target="_blank">cdw.com/content/solutions/unified-communications/</a></span><u></u><u></u></p>
<div>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><span style="font-size:10.0pt"><u></u><u></u></span></p>
</div>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">
<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>]
<b>On Behalf Of </b>Abebe Amare<br>
<b>Sent:</b> Tuesday, February 07, 2012 4:10 AM<br>
<b>To:</b> cisco voip<br>
<b>Subject:</b> [cisco-voip] intercluster trunk over IPSec VPN</span><u></u><u></u></p>
<div>
<div>
<div>
<p class="MsoNormal"><span style="font-size:10.0pt"> <u></u><u></u></span></p>
</div>
<p class="MsoNormal">Dears,<br>
<br>
I have configured an Inter-Cluster trunk from CUCM to another site with CUCME. There is an IPSec L2L VPN terminating at ASA 5500 firewall on both ends<br>
<br>
CUCM --->ASA 5540--->Internet <---ASA 5510<---CUCME<br>
<br>
On the ASA,the IPSec tunnel is terminated after 30 minute of inactivity (default) which is causing a problem. When a phone in one site tries to call another phone in the other site there is a noticeable gap before actual conversation is heard over the phone.
Once conversation starts, there is no delay or break in audio. Has anyone faced this issue?<br>
<br>
best regards,<br>
<br>
Abebe<u></u><u></u></p>
</div>
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</div>
</div>
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<p class="MsoNormal"><span style="font-size:10.0pt"><u></u> <u></u></span></p>
</div>
</div>
<p class="MsoNormal"><span style="font-size:10.0pt">_______________________________________________<br>
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<p class="MsoNormal"><span style="font-size:10.0pt"><u></u> <u></u></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt"><br>
<br>
<br>
<u></u><u></u></span></p>
<pre>_______________________________________________<u></u><u></u></pre>
<pre>cisco-voip mailing list<u></u><u></u></pre>
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<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif""><u></u> <u></u></span></p>
</div>
</div>
<p class="MsoNormal">_______________________________________________<br>
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<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><u></u><u></u></p>
</div>
<p class="MsoNormal"><u></u> <u></u></p>
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<p class="MsoNormal"><u></u> <u></u></p>
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<p class="MsoNormal"><u></u> <u></u></p>
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</div></blockquote><blockquote type="cite"><div><span>_______________________________________________</span><br><span>cisco-voip mailing list</span><br><span><a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a></span><br>
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