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
6 3 2 4;} _filtered #yiv3088174526 {font-family:Calibri;panose-1:2 15 5 2 2 2 4
3 2 4;}#yiv3088174526 #yiv3088174526 p.yiv3088174526MsoNormal, #yiv3088174526
li.yiv3088174526MsoNormal, #yiv3088174526 div.yiv3088174526MsoNormal
{margin:0in;margin-bottom:.0001pt;font-size:12.0pt;}#yiv3088174526 a:link,
#yiv3088174526 span.yiv3088174526MsoHyperlink
{color:blue;text-decoration:underline;}#yiv3088174526 a:visited, #yiv3088174526
span.yiv3088174526MsoHyperlinkFollowed
{color:purple;text-decoration:underline;}#yiv3088174526
span.yiv3088174526EmailStyle17
{font-family:Calibri;color:windowtext;}#yiv3088174526 span.yiv3088174526msoIns
{text-decoration:underline;color:teal;}#yiv3088174526
.yiv3088174526MsoChpDefault {font-size:10.0pt;} _filtered #yiv3088174526
{margin:1.0in 1.0in 1.0in 1.0in;}#yiv3088174526 div.yiv3088174526WordSection1
{}#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.
_______________________________________________
VoiceOps mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________
VoiceOps mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/voiceops