Hi Wes, thanks a lot for your detail explanation, it is very educational. <br><br>I should have stated earlier that the Internet connection is a VSAT link. Here is a ping output from the CUCM to the far side CUCME,<br><br>
admin:utils network ping 192.168.100.1<br>PING 192.168.100.1 (192.168.100.1) 56(84) bytes of data.<br>64 bytes from <a href="http://192.168.100.1">192.168.100.1</a>: icmp_seq=0 ttl=254 time=577 ms<br>64 bytes from <a href="http://192.168.100.1">192.168.100.1</a>: icmp_seq=1 ttl=254 time=579 ms<br>
64 bytes from <a href="http://192.168.100.1">192.168.100.1</a>: icmp_seq=2 ttl=254 time=587 ms<br>64 bytes from <a href="http://192.168.100.1">192.168.100.1</a>: icmp_seq=3 ttl=254 time=584 ms<br><br>--- 192.168.100.1 ping statistics ---<br>
4 packets transmitted, 4 received, 0% packet loss, time 3036ms<br>rtt min/avg/max/mdev = 577.892/582.386/587.697/4.048 ms, pipe 2<br><br>I have configured QoS on the gateway routers to put the VPN traffic in LLQ with reserved bandwidth. What other mechanisms do you suggest to improve the voice quality? <br>
<br>best regards,<br><br>Abebe<br><br><div class="gmail_quote">On Mon, Feb 13, 2012 at 6:06 PM, Wes Sisk <span dir="ltr"><<a href="mailto:wsisk@cisco.com">wsisk@cisco.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="word-wrap:break-word">Hi Abebe,<div><br></div><div>Those are pretty bad. These are of particular concern:</div><div><div class="im">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></div><div class="im">Cumulative Conceal ratio 0.0319<br>Max Conceal ratio 0.0446<br>Conceal sec 5<br>Severly conceal sec 1<br>
</div></div><div><br></div><div>I interpret this as:</div><div>G.729 has a lower MOS so we're starting with less audio quality. The phone sent 709 packets but received 476 packets. At 20msec packetization the phone has sent 14.1 seconds of audio but received only 9.5 seconds of audio. The phone cannot begin counting until the first RTP packet arrives so this may account for an additional gap between these metrics and actual user experience. The tx/rx is normally very similar. Either there was a delay in signaling negotiating bidirectional audio or this phone first started receiving RTP packets ~5 seconds into the call. After that Jitter of 31 is not great but would generally still be intelligible audio. Max jitter of 185 is pretty much impossible to recover. Each packet should contain 20msec of audio. At least one packet arrived 185msec late. </div>
<div><br></div><div>Recvr discarded 1 means the phone discarded one rtp packet it received. Discard can happen for several reasons but with max jitter 185 it is very likely the packet was just too late to be used. When a 20msec slice of audio is unavailable the phone attempts to conceal that in the audio stream. This leads to next stats:</div>
<div class="im"><div>Conceal sec 5<br>Severly conceal sec 1<br><br></div></div><div>For 5 seconds the phone used the g.729 packet loss concealment algorithm to try and mask the absence of packets. This is going to be silence, noise, or otherwise unintelligible audio to the user.</div>
<div><br></div><div>It looks like something is indeed inhibiting RTP packet flow toward this phone. A packet capture will show more details but it's not particularly necessary from the phone/application perspective. The underlying packet network needs some improvements to delivery adequate voice quality.</div>
<span class="HOEnZb"><font color="#888888"><div><br></div><div>/wes</div></font></span><div><div class="h5"><div><br><div><div>On Feb 13, 2012, at 9:31 AM, Abebe Amare wrote:</div><br>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" target="_blank">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><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>
</div>
</div>
</div>
</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:10.0pt">_______________________________________________<br>
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</div><p class="MsoNormal"><span style="font-size:10.0pt"><u></u> <u></u></span></p>
</div>
</div>
</div><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>
<pre><a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><u></u><u></u></pre>
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</div><p class="MsoNormal">_______________________________________________<br>
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<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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</div><p class="MsoNormal"><u></u> <u></u></p>
</div><p class="MsoNormal"><u></u> <u></u></p>
</div>
</div>
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