<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=us-ascii">
<meta content="text/html; charset=utf-8">
</head>
<body>
<p dir="auto" style="text-align:left; margin-top:25px; margin-bottom:25px; font-family:sans-serif; font-size:11pt; color:black; background-color:white">
I have this command configured. One thing i found out yesterday is that it apparently happens when all parties are internal also. So no trunk involved.
</p>
<p dir="auto" style="text-align:left; margin-top:25px; margin-bottom:25px; font-family:sans-serif; font-size:11pt; color:black; background-color:white">
Personally I haven't experienced this in my testing. It was 2-3 seconds for audio to establish which i thought was ok. The customet had said they timed it to be 8-9 seconds.</p>
<p dir="auto" style="text-align:left; margin-top:25px; margin-bottom:25px; font-family:sans-serif; font-size:11pt; color:black; background-color:white">
I might have to do a wireshark and see if there are retransmissions for signalling.
<br>
<br>
</p>
<hr tabindex="-1" style="display:inline-block; width:98%">
<div id="divRplyFwdMsg" dir="ltr"><font face="Calibri, sans-serif" color="#000000" style="font-size:11pt"><b>From:</b> Anthony Holloway <avholloway+cisco-voip@gmail.com><br>
<b>Sent:</b> Friday, January 13, 2017 4:50:31 AM<br>
<b>To:</b> Dana Tong; cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] Audio cut-through delays upon transfer</font>
<div> </div>
</div>
<div>
<div dir="ltr">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.
<div><br>
</div>
<div>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:</div>
<div><br>
</div>
<div><font face="monospace">voice service voip</font></div>
<div><font face="monospace"> sip</font></div>
<div><font face="monospace">  mid-call signaling passthru media-change</font></div>
<div><font face="monospace">!</font></div>
<div><br>
</div>
<div>You can read more about it here:</div>
<div><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><br>
</div>
</div>
<br>
<div class="gmail_quote">
<div dir="ltr">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:<br>
</div>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex; border-left:1px #ccc solid; padding-left:1ex">
<div bgcolor="white" lang="EN-GB" class="gmail_msg">
<div class="m_-8811615734268575318WordSection1 gmail_msg">
<p class="MsoNormal gmail_msg"><span class="gmail_msg" style="font-size:11.0pt">Hi all,<u class="gmail_msg"></u><u class="gmail_msg"></u></span></p>
<p class="MsoNormal gmail_msg"><span class="gmail_msg" style="font-size:11.0pt"><u class="gmail_msg"></u> <u class="gmail_msg"></u></span></p>
<p class="MsoNormal gmail_msg"><span class="gmail_msg" 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.<u class="gmail_msg"></u><u class="gmail_msg"></u></span></p>
<p class="MsoNormal gmail_msg"><span class="gmail_msg" style="font-size:9.0pt; font-family:Arial; color:#434343">There doesn’t appear to be any delays in the signalling.
<u class="gmail_msg"></u><u class="gmail_msg"></u></span></p>
<p class="MsoNormal gmail_msg"><span class="gmail_msg" style="font-size:9.0pt; font-family:Arial; color:#434343"><u class="gmail_msg"></u> <u class="gmail_msg"></u></span></p>
<p class="MsoNormal gmail_msg"><span class="gmail_msg" 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.<u class="gmail_msg"></u><u class="gmail_msg"></u></span></p>
<p class="MsoNormal gmail_msg"><span class="gmail_msg" 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. <u class="gmail_msg"></u><u class="gmail_msg"></u></span></p>
<p class="MsoNormal gmail_msg"><span class="gmail_msg" style="font-size:9.0pt; font-family:Arial; color:#434343"><u class="gmail_msg"></u> <u class="gmail_msg"></u></span></p>
<p class="MsoNormal gmail_msg"><span class="gmail_msg" style="font-size:9.0pt; font-family:Arial; color:#434343">Has anyone seen this before at all?<u class="gmail_msg"></u><u class="gmail_msg"></u></span></p>
<p class="MsoNormal gmail_msg"><span class="gmail_msg" style="font-size:9.0pt; font-family:Arial; color:#434343"><u class="gmail_msg"></u> <u class="gmail_msg"></u></span></p>
<p class="MsoNormal gmail_msg">Cheers</p>
<p class="MsoNormal gmail_msg">Dana</p>
<p class="MsoNormal gmail_msg"><u class="gmail_msg"></u> <u class="gmail_msg"></u></p>
</div>
</div>
_______________________________________________<br class="gmail_msg">
cisco-voip mailing list<br class="gmail_msg">
<a href="mailto:cisco-voip@puck.nether.net" class="gmail_msg" target="_blank">cisco-voip@puck.nether.net</a><br class="gmail_msg">
<a href="https://puck.nether.net/mailman/listinfo/cisco-voip" rel="noreferrer" class="gmail_msg" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br class="gmail_msg">
</blockquote>
</div>
</div>
</body>
</html>