On Thu, 7 May 2015, Sebastian Moeller wrote:
Is this 40ms sort of set in stone? If so we have a new indicator
for bad buffer-bloat if inured latency > 40 ms link is unsuitable for
decent voip (using old equipment). Is the newer voip stuff that telcos
roll out currently any smarter?
The 40ms is fairly typical for what I encountered 10 years ago. To deploy
them there was a requirement to have QoS (basically low-latency queuing on
Cisco) for DSCP EF traffic, otherwise things didn't work on the 0.5-2
megabit/s connections that were common back then.
I'd say anything people are trying to deploy now for use on the Internet
without QoS, 40ms just won't work and has never really worked. You need
adaptive PDV-buffers and they need to be able to handle hundreds of ms of
PDV.
If you look at this old document from Cisco (10 years old):
http://www.ciscopress.com/articles/article.asp?p=357102
"Voice (Bearer Traffic)
The following list summarizes the key QoS requirements and recommendations
for voice (bearer traffic):
Voice traffic should be marked to DSCP EF per the QoS Baseline and RFC
3246.
Loss should be no more than 1 percent.
One-way latency (mouth to ear) should be no more than 150 ms.
Average one-way jitter should be targeted at less than 30 ms.
A range of 21 to 320 kbps of guaranteed priority bandwidth is required per
call (depending on the sampling rate, the VoIP codec, and Layer 2 media
overhead).
Voice quality directly is affected by all three QoS quality factors: loss,
latency, and jitter."
This requirement kind of reflects the requirements of the VoIP systems of
the day with 40ms PDV buffer. There is also a section a page down or so
about "Jitter buffers" where there is a recommendation to have adaptive
jitter buffers, which I didn't encounter back then but I really hope is a
lot more common today.
--
Mikael Abrahamsson email: [email protected]
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