<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">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><br><div><div>On Feb 9, 2012, at 8:13 AM, Adam Frankel (afrankel) wrote:</div><br class="Apple-interchange-newline">
  
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    <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>
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      <span style="color:#000000;" class="headerSpan">
        <hr tabindex="0"><font style="font-size:x-small" face="Tahoma"><!--@A@--><b>From:</b>
          Abebe Amare <a class="moz-txt-link-rfc2396E" href="mailto:abucho@gmail.com"><abucho@gmail.com></a><!--@A@--><br>
          <!--@D@--><b>Sent:</b> Thu, Feb 09, 2012 4:52:50 AM<!--@D@--><br>
          <!--@R@--><b>To:</b> Ryan Ratliff <a class="moz-txt-link-rfc2396E" href="mailto:rratliff@cisco.com"><rratliff@cisco.com></a><!--@R@--><br>
          <!--@C@--><b>CC:</b> cisco voip
          <a class="moz-txt-link-rfc2396E" href="mailto:cisco-voip@puck.nether.net"><cisco-voip@puck.nether.net></a><!--@C@--><br>
          <!--@S@--><b>Subject:</b> Re: [cisco-voip] intercluster trunk
          over IPSec VPN<!--@S@--><br>
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      <blockquote style="border: medium none ! important; padding-left:
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        serif;" cite="mid:CANrv87jvJTNsjT8yJ0fwCLiEh_wwZik2UZgcMVL70pBE3a3iUA@mail.gmail.com" 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>
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          On Wed, Feb 8, 2012 at 8:07 PM, Ryan Ratliff <span dir="ltr"><<a moz-do-not-send="true" href="mailto:rratliff@cisco.com">rratliff@cisco.com</a>></span>
          wrote:<br>
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            <div style="word-wrap:break-word">What about a SIP trunk
              with options ping enabled?
              <div><br>
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                      -Ryan</div>
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                      <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 moz-do-not-send="true" href="mailto:Dennis.Heim@cdw.com" target="_blank">Dennis.Heim@cdw.com</a>></span>
                        wrote:<br>
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                            <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 class="webkit-block-placeholder"></div><div><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><br class="webkit-block-placeholder"></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 moz-do-not-send="true" 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 moz-do-not-send="true" 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 class="webkit-block-placeholder"></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 moz-do-not-send="true" href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>
                                  [mailto:<a moz-do-not-send="true" 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 class="webkit-block-placeholder"></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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