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: [email protected]
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