Re: [VoiceOps] Maximum latency

[Anthony Orlando via VoiceOps]

Cool. Didn't realize u were running voice traffic while doing that?  What was 
the sound quality like?  

Sent from my iPhone

> On Mar 22, 2017, at 5:49 PM, Alex Balashov  wrote:
> 
> What is it that makes people think you need a special application or
> tool to do these things?
> 
> Here's my default gateway on my home LAN:
> 
> sasha@mouse ~> ping -c 5 172.30.105.1
> PING 172.30.105.1 (172.30.105.1) 56(84) bytes of data.
> 64 bytes from 172.30.105.1: icmp_seq=1 ttl=64 time=0.238 ms
> 64 bytes from 172.30.105.1: icmp_seq=2 ttl=64 time=0.618 ms
> 64 bytes from 172.30.105.1: icmp_seq=3 ttl=64 time=0.457 ms
> 64 bytes from 172.30.105.1: icmp_seq=4 ttl=64 time=0.474 ms
> 64 bytes from 172.30.105.1: icmp_seq=5 ttl=64 time=0.382 ms
> 
> --- 172.30.105.1 ping statistics ---
> 5 packets transmitted, 5 received, 0% packet loss, time 4052ms
> rtt min/avg/max/mdev = 0.238/0.433/0.618/0.126 ms
> 
> And here's my default gateway on my home LAN with 300 ms of latency
> "injected":
> 
> sasha@mouse ~> sudo tc qdisc add dev enp1s0 root netem delay 300ms
> 
> sasha@mouse ~> ping -c 5 172.30.105.1
> PING 172.30.105.1 (172.30.105.1) 56(84) bytes of data.
> 64 bytes from 172.30.105.1: icmp_seq=1 ttl=64 time=300 ms
> 64 bytes from 172.30.105.1: icmp_seq=2 ttl=64 time=300 ms
> 64 bytes from 172.30.105.1: icmp_seq=3 ttl=64 time=300 ms
> 64 bytes from 172.30.105.1: icmp_seq=4 ttl=64 time=301 ms
> 64 bytes from 172.30.105.1: icmp_seq=5 ttl=64 time=300 ms
> 
> --- 172.30.105.1 ping statistics ---
> 5 packets transmitted, 5 received, 0% packet loss, time 4005ms
> rtt min/avg/max/mdev = 300.841/300.902/301.038/0.605 ms
> 
> And here's 300ms latency +/- variability of 40ms:
> 
> sasha@mouse ~> sudo tc qdisc add dev enp1s0 root netem delay 300ms 40ms
> sasha@mouse ~> ping -c 5 172.30.105.1
> PING 172.30.105.1 (172.30.105.1) 56(84) bytes of data.
> 64 bytes from 172.30.105.1: icmp_seq=1 ttl=64 time=299 ms
> 64 bytes from 172.30.105.1: icmp_seq=2 ttl=64 time=316 ms
> 64 bytes from 172.30.105.1: icmp_seq=3 ttl=64 time=324 ms
> 64 bytes from 172.30.105.1: icmp_seq=4 ttl=64 time=321 ms
> 64 bytes from 172.30.105.1: icmp_seq=5 ttl=64 time=326 ms
> 
> --- 172.30.105.1 ping statistics ---
> 5 packets transmitted, 5 received, 0% packet loss, time 4002ms
> rtt min/avg/max/mdev = 299.411/317.749/326.472/9.706 ms
> 
> And here's back to normal:
> 
> [root@mouse ~]# tc qdisc del dev enp1s0 root netem
> [root@mouse ~]# ping 172.30.105.1
> PING 172.30.105.1 (172.30.105.1) 56(84) bytes of data.
> 64 bytes from 172.30.105.1: icmp_seq=1 ttl=64 time=0.342 ms
> ^C
> --- 172.30.105.1 ping statistics ---
> 1 packets transmitted, 1 received, 0% packet loss, time 0ms
> rtt min/avg/max/mdev = 0.342/0.342/0.342/0.000 ms
> 
> And here's "loss insertion" of 5%:
> 
> sasha@mouse ~> sudo tc qdisc add dev enp1s0 root netem loss 5%
> sasha@mouse ~> ping -c 5 172.30.105.1
> PING 172.30.105.1 (172.30.105.1) 56(84) bytes of data.
> 64 bytes from 172.30.105.1: icmp_seq=2 ttl=64 time=0.514 ms
> 64 bytes from 172.30.105.1: icmp_seq=3 ttl=64 time=0.501 ms
> 64 bytes from 172.30.105.1: icmp_seq=4 ttl=64 time=0.546 ms
> 64 bytes from 172.30.105.1: icmp_seq=5 ttl=64 time=0.279 ms
> 
> --- 172.30.105.1 ping statistics ---
> 5 packets transmitted, 4 received, 20% packet loss, time 4056ms
> rtt min/avg/max/mdev = 0.279/0.460/0.546/0.105 ms
> 
> sasha@mouse ~> ping -c 5 172.30.105.1
> PING 172.30.105.1 (172.30.105.1) 56(84) bytes of data.
> 64 bytes from 172.30.105.1: icmp_seq=1 ttl=64 time=0.732 ms
> 64 bytes from 172.30.105.1: icmp_seq=2 ttl=64 time=0.471 ms
> 64 bytes from 172.30.105.1: icmp_seq=3 ttl=64 time=0.474 ms
> 64 bytes from 172.30.105.1: icmp_seq=4 ttl=64 time=0.334 ms
> 64 bytes from 172.30.105.1: icmp_seq=5 ttl=64 time=0.367 ms
> 
> --- 172.30.105.1 ping statistics ---
> 5 packets transmitted, 5 received, 0% packet loss, time 4020ms
> rtt min/avg/max/mdev = 0.334/0.475/0.732/0.141 ms
> 
> No, I'm not that brilliant. I Googled it:
> 
> http://stackoverflow.com/questions/614795/simulate-delayed-and-dropped-packets-on-linux
> 
> App to inject latency? You people kill me sometimes.
> 
> -- Alex
> 
> -- 
> Alex Balashov | Principal | Evariste Systems LLC
> 
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
> ___
> VoiceOps mailing list
> VoiceOps@voiceops.org
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Re: [VoiceOps] Maximum latency

[Alex Balashov]

What is it that makes people think you need a special application or
tool to do these things?

Here's my default gateway on my home LAN:

sasha@mouse ~> ping -c 5 172.30.105.1
PING 172.30.105.1 (172.30.105.1) 56(84) bytes of data.
64 bytes from 172.30.105.1: icmp_seq=1 ttl=64 time=0.238 ms
64 bytes from 172.30.105.1: icmp_seq=2 ttl=64 time=0.618 ms
64 bytes from 172.30.105.1: icmp_seq=3 ttl=64 time=0.457 ms
64 bytes from 172.30.105.1: icmp_seq=4 ttl=64 time=0.474 ms
64 bytes from 172.30.105.1: icmp_seq=5 ttl=64 time=0.382 ms

--- 172.30.105.1 ping statistics ---
5 packets transmitted, 5 received, 0% packet loss, time 4052ms
rtt min/avg/max/mdev = 0.238/0.433/0.618/0.126 ms

And here's my default gateway on my home LAN with 300 ms of latency
"injected":

sasha@mouse ~> sudo tc qdisc add dev enp1s0 root netem delay 300ms

sasha@mouse ~> ping -c 5 172.30.105.1
PING 172.30.105.1 (172.30.105.1) 56(84) bytes of data.
64 bytes from 172.30.105.1: icmp_seq=1 ttl=64 time=300 ms
64 bytes from 172.30.105.1: icmp_seq=2 ttl=64 time=300 ms
64 bytes from 172.30.105.1: icmp_seq=3 ttl=64 time=300 ms
64 bytes from 172.30.105.1: icmp_seq=4 ttl=64 time=301 ms
64 bytes from 172.30.105.1: icmp_seq=5 ttl=64 time=300 ms

--- 172.30.105.1 ping statistics ---
5 packets transmitted, 5 received, 0% packet loss, time 4005ms
rtt min/avg/max/mdev = 300.841/300.902/301.038/0.605 ms

And here's 300ms latency +/- variability of 40ms:

sasha@mouse ~> sudo tc qdisc add dev enp1s0 root netem delay 300ms 40ms
sasha@mouse ~> ping -c 5 172.30.105.1
PING 172.30.105.1 (172.30.105.1) 56(84) bytes of data.
64 bytes from 172.30.105.1: icmp_seq=1 ttl=64 time=299 ms
64 bytes from 172.30.105.1: icmp_seq=2 ttl=64 time=316 ms
64 bytes from 172.30.105.1: icmp_seq=3 ttl=64 time=324 ms
64 bytes from 172.30.105.1: icmp_seq=4 ttl=64 time=321 ms
64 bytes from 172.30.105.1: icmp_seq=5 ttl=64 time=326 ms

--- 172.30.105.1 ping statistics ---
5 packets transmitted, 5 received, 0% packet loss, time 4002ms
rtt min/avg/max/mdev = 299.411/317.749/326.472/9.706 ms

And here's back to normal:

[root@mouse ~]# tc qdisc del dev enp1s0 root netem
[root@mouse ~]# ping 172.30.105.1
PING 172.30.105.1 (172.30.105.1) 56(84) bytes of data.
64 bytes from 172.30.105.1: icmp_seq=1 ttl=64 time=0.342 ms
^C
--- 172.30.105.1 ping statistics ---
1 packets transmitted, 1 received, 0% packet loss, time 0ms
rtt min/avg/max/mdev = 0.342/0.342/0.342/0.000 ms

And here's "loss insertion" of 5%:

sasha@mouse ~> sudo tc qdisc add dev enp1s0 root netem loss 5%
sasha@mouse ~> ping -c 5 172.30.105.1
PING 172.30.105.1 (172.30.105.1) 56(84) bytes of data.
64 bytes from 172.30.105.1: icmp_seq=2 ttl=64 time=0.514 ms
64 bytes from 172.30.105.1: icmp_seq=3 ttl=64 time=0.501 ms
64 bytes from 172.30.105.1: icmp_seq=4 ttl=64 time=0.546 ms
64 bytes from 172.30.105.1: icmp_seq=5 ttl=64 time=0.279 ms

--- 172.30.105.1 ping statistics ---
5 packets transmitted, 4 received, 20% packet loss, time 4056ms
rtt min/avg/max/mdev = 0.279/0.460/0.546/0.105 ms

sasha@mouse ~> ping -c 5 172.30.105.1
PING 172.30.105.1 (172.30.105.1) 56(84) bytes of data.
64 bytes from 172.30.105.1: icmp_seq=1 ttl=64 time=0.732 ms
64 bytes from 172.30.105.1: icmp_seq=2 ttl=64 time=0.471 ms
64 bytes from 172.30.105.1: icmp_seq=3 ttl=64 time=0.474 ms
64 bytes from 172.30.105.1: icmp_seq=4 ttl=64 time=0.334 ms
64 bytes from 172.30.105.1: icmp_seq=5 ttl=64 time=0.367 ms

--- 172.30.105.1 ping statistics ---
5 packets transmitted, 5 received, 0% packet loss, time 4020ms
rtt min/avg/max/mdev = 0.334/0.475/0.732/0.141 ms

No, I'm not that brilliant. I Googled it:

http://stackoverflow.com/questions/614795/simulate-delayed-and-dropped-packets-on-linux

App to inject latency? You people kill me sometimes.

-- Alex

-- 
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
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Re: [VoiceOps] Maximum latency

[Anthony Orlando via VoiceOps]

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

  From: Matthew M. Gamble <mgam...@thoughtfire.ca>
 To: Carlos Alvarez <caalva...@gmail.com>; "voiceops@voiceops.org" 
<voiceops@voiceops.org> 
 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 <voiceops-boun...@voiceops.org> on behalf of Carlos Alvarez 
<caalva...@gmail.com>
Date: Wednesday, March 22, 2017 at 2:13 PM
To: "voiceops@voiceops.org" <voiceops@voiceops.org>
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 <ivan.kovace...@startelecom.ca> 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:voiceops-boun...@voiceops.org]On Behalf Of Carlos Alvarez
Sent: March 22, 2017 1:59 PM
To: voiceops@voiceops.org
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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Re: [VoiceOps] Maximum latency

[Alex Balashov]

My own experience, having lived on the other side of the world for two
years while running my calls through an Atlanta-based PBX, is that high
latency is seldom a problem unless it's consistent. It's variation of
any kind which concerns me. It leads to jitter, audio drop-outs from
retraining jitter buffers, and out-of-order frames.

-- 
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
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Re: [VoiceOps] Maximum latency

[Carlos Alvarez]

"We're gonna need a bigger cable..."

Also, Ryan, good point about routes.  Their provider there says they will
optimize routes, and seems to be doing so.  On our side, they are very
stable.


On Wed, Mar 22, 2017 at 11:25 AM, Patrick Labbett <patrick.labb...@gmail.com
> wrote:

> Coil cable in a box until the desired delay is achieved.  (Reference from
> Flash Boys by Michael Lewis) :)
>
> On Wed, Mar 22, 2017 at 2:22 PM Carlos Alvarez <caalva...@gmail.com>
> wrote:
>
>> 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 <
>> ivan.kovace...@startelecom.ca> 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
>> <http://t.sidekickopen61.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XYg1qMvVnW3LjyBM63K6FlW63JXmj56dLPjf6TyZWx02?t=http%3A%2F%2Fwww.startelecom.ca%2F=6470560385859584=36c7d4bb-13b1-426d-bf96-c8c2ff149bdd>
>> | SIP Based Services for Contact Centers | LinkedIn
>> <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=6470560385859584=36c7d4bb-13b1-426d-bf96-c8c2ff149bdd>
>>
>>
>>
>> *From:* VoiceOps [mailto:voiceops-boun...@voiceops.org] *On Behalf Of *Carlos
>> Alvarez
>> *Sent:* March 22, 2017 1:59 PM
>> *To:* voiceops@voiceops.org
>> *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.
>>
>>
>>
>> [image:
>> http://t.sidekickopen61.com/e1t/o/5/f18dQhb0S7ks8dDMPbW2n0x6l2B9gXrN7sKj6v4LCS0Vf6xNj7dSCJWW64Jplz1pctGFW55kbNc1k1H6H0?si=6470560385859584=36c7d4bb-13b1-426d-bf96-c8c2ff149bdd]
>>
>>
>> ___
>> VoiceOps mailing list
>> VoiceOps@voiceops.org
>> https://puck.nether.net/mailman/listinfo/voiceops
>>
>
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Re: [VoiceOps] Maximum latency

[Patrick Labbett]

Coil cable in a box until the desired delay is achieved.  (Reference from
Flash Boys by Michael Lewis) :)

On Wed, Mar 22, 2017 at 2:22 PM Carlos Alvarez <caalva...@gmail.com> wrote:

> 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 <
> ivan.kovace...@startelecom.ca> 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
> <http://t.sidekickopen61.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XYg1qMvVnW3LjyBM63K6FlW63JXmj56dLPjf6TyZWx02?t=http%3A%2F%2Fwww.startelecom.ca%2F=6470560385859584=36c7d4bb-13b1-426d-bf96-c8c2ff149bdd>
> | SIP Based Services for Contact Centers | LinkedIn
> <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=6470560385859584=36c7d4bb-13b1-426d-bf96-c8c2ff149bdd>
>
>
>
> *From:* VoiceOps [mailto:voiceops-boun...@voiceops.org] *On Behalf Of *Carlos
> Alvarez
> *Sent:* March 22, 2017 1:59 PM
> *To:* voiceops@voiceops.org
> *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.
>
>
>
> [image:
> http://t.sidekickopen61.com/e1t/o/5/f18dQhb0S7ks8dDMPbW2n0x6l2B9gXrN7sKj6v4LCS0Vf6xNj7dSCJWW64Jplz1pctGFW55kbNc1k1H6H0?si=6470560385859584=36c7d4bb-13b1-426d-bf96-c8c2ff149bdd]
>
>
> ___
> VoiceOps mailing list
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> https://puck.nether.net/mailman/listinfo/voiceops
>
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Re: [VoiceOps] Maximum latency

[Matthew M. Gamble]

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 <voiceops-boun...@voiceops.org> on behalf of Carlos Alvarez 
<caalva...@gmail.com>
Date: Wednesday, March 22, 2017 at 2:13 PM
To: "voiceops@voiceops.org" <voiceops@voiceops.org>
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 
<ivan.kovace...@startelecom.ca<mailto:ivan.kovace...@startelecom.ca>> 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<http://t.sidekickopen61.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XYg1qMvVnW3LjyBM63K6FlW63JXmj56dLPjf6TyZWx02?t=http%3A%2F%2Fwww.startelecom.ca%2F=6470560385859584=36c7d4bb-13b1-426d-bf96-c8c2ff149bdd>
 | SIP Based Services for Contact Centers | 
LinkedIn<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=6470560385859584=36c7d4bb-13b1-426d-bf96-c8c2ff149bdd>

From: VoiceOps 
[mailto:voiceops-boun...@voiceops.org<mailto:voiceops-boun...@voiceops.org>] On 
Behalf Of Carlos Alvarez
Sent: March 22, 2017 1:59 PM
To: voiceops@voiceops.org<mailto:voiceops@voiceops.org>
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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Re: [VoiceOps] Maximum latency

[Ivan Kovacevic]

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
<http://t.sidekickopen61.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XYg1qMvVnW3LjyBM63K6FlW63JXmj56dLPjf6TyZWx02?t=http%3A%2F%2Fwww.startelecom.ca%2F=6470560385859584=36c7d4bb-13b1-426d-bf96-c8c2ff149bdd>
| SIP Based Services for Contact Centers | LinkedIn
<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=6470560385859584=36c7d4bb-13b1-426d-bf96-c8c2ff149bdd>



*From:* VoiceOps [mailto:voiceops-boun...@voiceops.org] *On Behalf Of *Carlos
Alvarez
*Sent:* March 22, 2017 1:59 PM
*To:* voiceops@voiceops.org
*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.



[image:
http://t.sidekickopen61.com/e1t/o/5/f18dQhb0S7ks8dDMPbW2n0x6l2B9gXrN7sKj6v4LCS0Vf6xNj7dSCJWW64Jplz1pctGFW55kbNc1k1H6H0?si=6470560385859584=36c7d4bb-13b1-426d-bf96-c8c2ff149bdd]
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Re: [VoiceOps] Maximum latency

[Matthew M. Gamble]

The ITU-T recommendation is that “up to 150 ms mouth-to-ear delay can be 
tolerated by the human ear with virtually no quality loss” and in my 
experience, that’s held true.  They are definitely going to notice the delay.

From: VoiceOps <voiceops-boun...@voiceops.org> on behalf of Carlos Alvarez 
<caalva...@gmail.com>
Date: Wednesday, March 22, 2017 at 1:59 PM
To: "voiceops@voiceops.org" <voiceops@voiceops.org>
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.

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[VoiceOps] Maximum latency

[Carlos Alvarez]

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.
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