On Tue, 28 Apr 2015, Sebastian Moeller wrote:

> 	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 ;).

Yes, it's all about mouth-to-ear and then back again. I have historically 
been involved a few times in analyzing end-to-end latency when customer 
complaints came in about delay, it seemed that customers started 
complaining around 450-550 ms RTT (mouth-network-ear-mouth-network-ear).

This usually was a result of multiple PDV (Packet Delay Variation, a.k.a 
jitter) buffers due media conversions on the voice path, for instance when 
there was VoIP-TDM-VoIP-ATM-VoIP and potentially even more conversions due 
to VoIP/PSTN/Mobile interaction.

So this is one reason I am interested in the bufferbloat movement, because 
with less bufferbloat then one can get away with smaller PDV buffers, 
which means less end-to-end delay for realtime applications.

-- 
Mikael Abrahamsson    email: swmike@swm.pp.se