NIST used to have an impairment app for injecting latency etc.  Been many years 
since I've used it.

      From: Matthew M. Gamble <[email protected]>
 To: Carlos Alvarez <[email protected]>; "[email protected]" 
<[email protected]> 
 Sent: Wednesday, March 22, 2017 1:27 PM
 Subject: Re: [VoiceOps] Maximum latency
   
#yiv3088174526 #yiv3088174526 -- _filtered #yiv3088174526 {panose-1:2 4 5 3 5 4 
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{}#yiv3088174526 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/    
From: VoiceOps <[email protected]> on behalf of Carlos Alvarez 
<[email protected]>
Date: Wednesday, March 22, 2017 at 2:13 PM
To: "[email protected]" <[email protected]>
Subject: Re: [VoiceOps] Maximum latency    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.    The US 
agents are on MPLS to us, 2-3ms at most.    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.          On Wed, Mar 22, 2017 at 11:08 AM, Ivan 
Kovacevic <[email protected]> wrote: 
  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.   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.    Not sure if this helps…   Best Regards,   Ivan Kovacevic Vice 
President, Client Services Star Telecom |www.startelecom.ca | SIP Based 
Services for Contact Centers | LinkedIn   From: VoiceOps 
[mailto:[email protected]]On Behalf Of Carlos Alvarez
Sent: March 22, 2017 1:59 PM
To: [email protected]
Subject: [VoiceOps] Maximum latency   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.   Error! Filename not specified. 
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