[Bloat] DSLReports Speed Test has latency measurement built-in

Sebastian Moeller moeller0 at gmx.de
Tue Apr 28 07:49:19 EDT 2015


Hi Mikhail,


On Apr 28, 2015, at 13:04 , Mikael Abrahamsson <swmike at swm.pp.se> wrote:

> On Tue, 28 Apr 2015, David Lang wrote:
> 
>> Voice is actually remarkably tolerant of pure latency. While 60ms of jitter makes a connection almost unusalbe, a few hundred ms of consistant latency isn't a problem. IIRC (from my college days when ATM was the new, hot technology) you have to get up to around a second of latency before pure-consistant latency starts to break things.
> 
> I would say most people start to get trouble when talking to each other when the RTT exceeds around 500-600ms.
> 
> I mostly agree with http://www.cisco.com/c/en/us/support/docs/voice/voice-quality/5125-delay-details.html but RTT of over 500ms is not fun. You basically can't have a heated argument/discussion when the RTT is higher than this :P

	From "Table 4.1 Delay Specifications” of that link we basically have a recapitulation of the ITU-T G.114 source, one-way mouth to ear latency thresholds for acceptable voip performance. The rest of the link discusses additional sources of latency and should allow to come up with a reasonable estimate how much of the latency budget can be spend on the transit. So in my mind an decent thresholds would be (150ms mouth-to-ear-delay - sender-processing - receiver-processing) * 2. Then again I think the discussion turned to relating buffer-bloat inured latency as jitter source, so the thresholds should be framed in a jitter-budget, not pure latency ;).

Best Regards
	Sebastian


> 
> -- 
> Mikael Abrahamsson    email: swmike at swm.pp.se




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