Hi Mikhail,
On Apr 28, 2015, at 13:04 , Mikael Abrahamsson <swm...@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: swm...@swm.pp.se _______________________________________________ Bloat mailing list Bloat@lists.bufferbloat.net https://lists.bufferbloat.net/listinfo/bloat
