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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:Calibri;mso-fareast-language:EN-US">I just checked this, and it has been enabled already.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:Calibri;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:Calibri;mso-fareast-language:EN-US">Cheers<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:Calibri;mso-fareast-language:EN-US">Dana<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:Calibri;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span style="font-family:Calibri;color:black">From: </span>
</b><span style="font-family:Calibri;color:black">"Pavan Dave (pavdave)" <pavdave@cisco.com><br>
<b>Date: </b>Friday, 13 January 2017 at 5:35 am<br>
<b>To: </b>Anthony Holloway <avholloway+cisco-voip@gmail.com>, Dana Tong <dana.tong@yellit.com.au>, "cisco-voip@puck.nether.net" <cisco-voip@puck.nether.net><br>
<b>Subject: </b>RE: [cisco-voip] Audio cut-through delays upon transfer<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:Calibri">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.</span><o:p></o:p></p>
<p class="pntnotetable" style="text-indent:36.0pt"><b>Note </b><img width="1" height="2" id="Picture_x005f_x0020_1" src="cid:image002.png@01D26D79.D59AD450" alt="ttp://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"><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)</span><o:p></o:p></p>
<p class="pntnotetable">Regards,<o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:Calibri">Pavan Davé</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:Calibri">Contact Center TAC Engineer</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:Calibri">Cisco Systems – Richardson, TX, USA</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:Calibri"> </span><o:p></o:p></p>
<p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:Calibri">From:</span></b><span style="font-size:11.0pt;font-family:Calibri"> 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</span><o:p></o:p></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 style="font-size:11.0pt">Hi all,</span></span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span style="font-size:11.0pt"> </span></span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span style="font-size:9.0pt;font-family:Arial;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><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span style="font-size:9.0pt;font-family:Arial;color:#434343">There doesn’t appear to be any delays in the signalling.
</span></span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span style="font-size:9.0pt;font-family:Arial;color:#434343"> </span></span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span style="font-size:9.0pt;font-family:Arial;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><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span style="font-size:9.0pt;font-family:Arial;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><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span style="font-size:9.0pt;font-family:Arial;color:#434343"> </span></span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span style="font-size:9.0pt;font-family:Arial;color:#434343">Has anyone seen this before at all?</span></span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span class="gmailmsg"><span style="font-size:9.0pt;font-family:Arial;color:#434343"> </span></span><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Cheers<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Dana<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
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<p class="MsoNormal">_______________________________________________<br>
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