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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:Calibri">I use a FreeBSD box as a bump-in-the-wire to test and introduce additional latency, packet loss, and jitter using ipfw – this is a pretty good overview of how to do it: http://fjoanis.github.io/2013/08/31/Network_Simulation_FreeBSD_DummyNet/<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">VoiceOps <voiceops-bounces@voiceops.org> on behalf of Carlos Alvarez <caalvarez@gmail.com><br>
<b>Date: </b>Wednesday, March 22, 2017 at 2:13 PM<br>
<b>To: </b>"voiceops@voiceops.org" <voiceops@voiceops.org><br>
<b>Subject: </b>Re: [VoiceOps] Maximum latency<o:p></o:p></span></p>
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<p class="MsoNormal">Those are some excellent points, Ivan. They do indeed transfer calls regularly between agents in various US cities, so I assume the foreign ones will also. I'll have to ask.
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<p class="MsoNormal">The US agents are on MPLS to us, 2-3ms at most.<o:p></o:p></p>
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<p class="MsoNormal">Also, my satellite phone seems to have around 700ms delay, which is quite challenging. I should try synthesizing 250ms for them and let them try it. Not sure how, but assume there's some open source out there to do it.<o:p></o:p></p>
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<p class="MsoNormal">On Wed, Mar 22, 2017 at 11:08 AM, Ivan Kovacevic <<a href="mailto:ivan.kovacevic@startelecom.ca" target="_blank">ivan.kovacevic@startelecom.ca</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"><a name="m_5000554820658676621__MailEndCompose"><span lang="EN-CA" style="font-size:11.0pt;font-family:Calibri"> </span></a><span lang="EN-CA"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-CA" style="font-size:11.0pt;font-family:Calibri">The reality is that you can’t get away from it. A private circuit may cut it by 10% and provide more stable Jitter…
but that’s it. So your client and 500,000 other agents in the Philippines and India are in the same boat.
</span><span lang="EN-CA"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-CA" style="font-size:11.0pt;font-family:Calibri"> </span><span lang="EN-CA"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-CA" style="font-size:11.0pt;font-family:Calibri">Where latency gets really nasty and there is some scope to optimize is on call transfers/conferences. Make sure there
is never hairpinning through the off-shore call centre. Or that call transfers or conferences have to double up on the path. It will depend on where the media gateways for the solution reside and whether you have any optimization when multiple call legs are
made through your service. </span><span lang="EN-CA"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-CA" style="font-size:11.0pt;font-family:Calibri"> </span><span lang="EN-CA"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-CA" style="font-size:11.0pt;font-family:Calibri">Not sure if this helps…
</span><span lang="EN-CA"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-CA" style="font-size:11.0pt;font-family:Calibri"> </span><span lang="EN-CA"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-CA" style="font-size:11.0pt;font-family:Calibri;color:#1F497D">Best Regards,</span><span lang="EN-CA"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-CA" style="font-size:11.0pt;font-family:Calibri;color:#1F497D"> </span><span lang="EN-CA"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-CA" style="font-size:11.0pt;font-family:Calibri;color:#1F497D">Ivan Kovacevic</span><span lang="EN-CA"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span style="font-size:11.0pt;font-family:Calibri;color:#1F497D">Vice President, Client Services</span><span lang="EN-CA"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-CA" style="font-size:11.0pt;font-family:Calibri;color:#1F497D">Star Telecom |
</span><span lang="EN-CA"><a href="http://t.sidekickopen61.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XYg1qMvVnW3LjyBM63K6FlW63JXmj56dLPjf6TyZWx02?t=http%3A%2F%2Fwww.startelecom.ca%2F&si=6470560385859584&pi=36c7d4bb-13b1-426d-bf96-c8c2ff149bdd" target="_blank"><span style="font-size:11.0pt;font-family:Calibri">www.startelecom.ca</span></a></span><span lang="EN-CA" style="font-size:11.0pt;font-family:Calibri;color:#1F497D">
| SIP Based Services for Contact Centers | </span><span lang="EN-CA"><a href="http://t.sidekickopen61.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XYg1qMvVnW3LjyBM63K6FlW63JXmj56dLPjf6TyZWx02?t=https%3A%2F%2Fwww.linkedin.com%2Fcompany%2Fstar-telecom-www-startelecom-ca-%3Ftrk%3Dbiz-companies-cym&si=6470560385859584&pi=36c7d4bb-13b1-426d-bf96-c8c2ff149bdd" target="_blank"><span style="font-size:11.0pt;font-family:Calibri">LinkedIn</span></a><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-CA" style="font-size:11.0pt;font-family:Calibri"> </span><span lang="EN-CA"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b><span style="font-size:11.0pt;font-family:Calibri">From:</span></b><span style="font-size:11.0pt;font-family:Calibri"> VoiceOps [mailto:<a href="mailto:voiceops-bounces@voiceops.org" target="_blank">voiceops-bounces@voiceops.org</a>]
<b>On Behalf Of </b>Carlos Alvarez<br>
<b>Sent:</b> March 22, 2017 1:59 PM<br>
<b>To:</b> <a href="mailto:voiceops@voiceops.org" target="_blank">voiceops@voiceops.org</a><br>
<b>Subject:</b> [VoiceOps] Maximum latency</span><span lang="EN-CA"><o:p></o:p></span></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-CA"> <o:p></o:p></span></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-CA">One of our larger customers is about to launch a new call center in Malaysia. The connection there is fast, the trace looks surprisingly clean and short, but
latency is consistently at 239-245ms. We've never knowingly had a connection over 130ms. Does anyone have experiences, good or bad, with latency approaching a quarter second? The jitter level seems fine, so I believe they'll just have a delay but decent
call quality.<o:p></o:p></span></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-CA"> <o:p></o:p></span></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b>Error! Filename not specified.</b><span lang="EN-CA"><o:p></o:p></span></p>
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