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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">To piggy back on this, if you are going to be in situations where calls are going on and off hold and transferring, setting duplex streaming to “True” on CUCM will also help.<o:p></o:p></span></p>
<p class="pntnotetable" style="text-indent:.5in"><b>Note </b><img width="1" height="2" style="width:.0119in;height:.0238in" id="Picture_x0020_1" src="cid:image002.png@01D26CD8.CAE97940" alt="http://www.cisco.com/c/dam/en/us/td/i/templates/blank.gif">To prevent
the SDP mode from being set to inactive in a multiple-hold scenario, set the Duplex Streaming Enabled clusterwide service parameter (<b>System > Service Parameters</b>) to
<em>True</em>. <o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"><a href="http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_5_1/ccmsys/accm-851-cm/a08trnk.html#wp1149990">http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_5_1/ccmsys/accm-851-cm/a08trnk.html#wp1149990</a>
(Step 10 under Table 41-1)<o:p></o:p></span></p>
<p class="pntnotetable">Regards,<span style="font-size:11.0pt;font-family:"Calibri",sans-serif"><o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">Pavan Davé<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">Contact Center TAC Engineer<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">Cisco Systems – Richardson, TX, USA<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"><o:p> </o:p></span></p>
<p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">From:</span></b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"> cisco-voip [mailto:cisco-voip-bounces@puck.nether.net]
<b>On Behalf Of </b>Anthony Holloway<br>
<b>Sent:</b> Thursday, January 12, 2017 12:51 PM<br>
<b>To:</b> Dana Tong <dana.tong@yellit.com.au>; cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] Audio cut-through delays upon transfer<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">This sounds like the old too-many-messages-are-required-to-make-the-transfer-happen scenario. Ok, I just made up that name, but basically the SIP messages are just busy doing whatever SIP messages do, to make the transfer happen. From
stopping media, to invoke MOH, to tearing down MOH, to transferring, to establishing media, it's just a lot, and takes a long time.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">You can cut down on the amount of messages in a few ways, but the one I see used most often is the command on the CUBE:<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><span style="font-family:"Courier New"">voice service voip</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-family:"Courier New""> sip</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-family:"Courier New""> mid-call signaling passthru media-change</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-family:"Courier New"">!</span><o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">You can read more about it here:<o:p></o:p></p>
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<p class="MsoNormal"><a href="http://www.cisco.com/en/US/docs/ios-xml/ios/voice/cube_proto/configuration/15-2mt/cube-midcall-reinvite.html">http://www.cisco.com/en/US/docs/ios-xml/ios/voice/cube_proto/configuration/15-2mt/cube-midcall-reinvite.html</a><o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">On Wed, Jan 11, 2017 at 10:00 PM Dana Tong <<a href="mailto:dana.tong@yellit.com.au">dana.tong@yellit.com.au</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span lang="EN-GB" style="font-size:11.0pt">Hi all,</span></span><span lang="EN-GB"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span lang="EN-GB" style="font-size:11.0pt"> </span></span><span lang="EN-GB"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span lang="EN-GB" style="font-size:9.0pt;font-family:"Arial",sans-serif;color:#434343">I have a customer who is reporting a (one-way) audio delay after transferring
calls with Cisco Jabber. They are almost all soft-phone users.</span></span><span lang="EN-GB"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span lang="EN-GB" style="font-size:9.0pt;font-family:"Arial",sans-serif;color:#434343">There doesn’t appear to be any delays in the signalling.
</span></span><span lang="EN-GB"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span lang="EN-GB" style="font-size:9.0pt;font-family:"Arial",sans-serif;color:#434343"> </span></span><span lang="EN-GB"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span lang="EN-GB" style="font-size:9.0pt;font-family:"Arial",sans-serif;color:#434343">I’ve just asked them to clarify if this is for all calls or just external
calls. PSTN access is via CUBE which is configured for early offer.</span></span><span lang="EN-GB"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span lang="EN-GB" style="font-size:9.0pt;font-family:"Arial",sans-serif;color:#434343">When I tested I couldn’t see a problem but the customer insists that
there is a delay for media being transmitted to the person who receives the transferred call.
</span></span><span lang="EN-GB"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span lang="EN-GB" style="font-size:9.0pt;font-family:"Arial",sans-serif;color:#434343"> </span></span><span lang="EN-GB"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span lang="EN-GB" style="font-size:9.0pt;font-family:"Arial",sans-serif;color:#434343">Has anyone seen this before at all?</span></span><span lang="EN-GB"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span lang="EN-GB" style="font-size:9.0pt;font-family:"Arial",sans-serif;color:#434343"> </span></span><span lang="EN-GB"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-GB">Cheers<o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-GB">Dana<o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-GB"> <o:p></o:p></span></p>
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<p class="MsoNormal">_______________________________________________<br>
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