<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">most likely still packet throughput issues. packets may be late to the point of discarded. they would not technically be lost in that case.<div><br></div><div>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.</div><div><br></div><div>/wes</div><div><br><div><div>On Feb 10, 2012, at 6:41 AM, Abebe Amare wrote:</div><br class="Apple-interchange-newline">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><br><div class="gmail_quote">On Thu, Feb 9, 2012 at 5:57 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">TCP keepalives are only used while a call is active.<div><br></div><div>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.</div>
<div><br></div><div>for a static ICT defined between two CUCM clusters there is no network activity without an active call.</div><div><br></div><div>For the duration of an active call the tcp keepalive parameter will help.</div>
<div><br></div><div>regards,</div><div>wes</div><div><div class="h5"><div><br><div><div>On Feb 9, 2012, at 8:13 AM, Adam Frankel (afrankel) wrote:</div><br>
<div style="font-family:Arial;font-size:13px" bgcolor="#FFFFFF" text="#000000">
<div style="font-family:Arial;font-size:13px"><font face="Arial">Options
Ping was added in 8.5(1).<br>
<br>
The parameter "</font>Allow TCP KeepAlives For H323 " should
take care of this for H323 ICT. <br>
<font face="Arial"><br>
-Adam<br>
</font><br>
<br>
<span style="">
<hr><font style="font-size:x-small" face="Tahoma"><b>From:</b>
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>
</font><br>
</span>
<blockquote style="border:medium none!important;padding-left:0px!important;padding-right:0px!important;margin-left:0px!important;margin-right:0px!important;font-family:serif" type="cite">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<br>
<br>
<div class="gmail_quote">
On Wed, Feb 8, 2012 at 8:07 PM, Ryan Ratliff <span dir="ltr"><<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@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">What about a SIP trunk
with options ping enabled?
<div><br>
<div>
<span style="text-indent:0px;letter-spacing:normal;font-variant:normal;text-align:auto;font-style:normal;font-weight:normal;line-height:normal;border-collapse:separate;text-transform:none;font-size:medium;white-space:normal;font-family:Helvetica;word-spacing:0px">
<div>
-Ryan</div>
</span>
</div>
<br>
<div>
<div>
<div>
<div>On Feb 8, 2012, at 7:05 AM, Abebe Amare
wrote:</div>
<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<br>
<br>
<div class="gmail_quote">
On Tue, Feb 7, 2012 at 6:16 PM, Dennis Heim <span dir="ltr"><<a href="mailto:Dennis.Heim@cdw.com" target="_blank">Dennis.Heim@cdw.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div link="blue" vlink="purple" lang="EN-US">
<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></p><div><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><br></div><div><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><br>
</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></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;color:blue">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"><span style="color:blue">cdw.com/content/solutions/unified-communications/</span></a></span></p>
<div><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><br></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></p>
<div>
<div><div> <br></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</p>
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