Re: what is acceptible jitter for voip and videoconferencing?

[Saku Ytti <saku () ytti fi>]

On Wed, 20 Sept 2023 at 03:15, Dave Taht <dave.taht () gmail com> wrote:

> I go back many, many years as to baseline numbers for managing voip networks, including things like CISCO LLQ, 
> diffserv, fqm prioritizing vlans, and running
> voip networks entirely separately... I worked on codecs, such as oslec, and early sip stacks, but that was over 20 
> years ago.

I don't believe LLQ has utility in hardware based routers, packets
stay inside hardware based routers single digit microseconds with
nanoseconds of jitter. For software based devices, I'm sure the
situation is different.
Practical example, tier1 network running 3 vendors, with no LLQ can go
across the globe with lower jitter (microseconds) than I can ping my
M1 laptop 127.0.0.1, because I have to do context switches, the
network does not. This is in the BE queue measured in real operation
under long periods, without any engineering effort to try to achieve
low jitter.

> The thing is, I have been unable to find much research (as yet) as to why my number exists. Over here I am taking a 
> poll as to what number is most correct (10ms, 30ms, 100ms, 200ms),

I know there are academic papers as well as vendor graphs showing the
impact of jitter on quality.  Here is one:
https://scholarworks.gsu.edu/cgi/viewcontent.cgi?article=1043&context=cs_theses
- this appears to roughly say '20ms' G711 is fine. But I'm sure this
is actually very complex to answer well, and I'm sure choice of codec
greatly impacts the answer, like whatsapp uses Opus, skype uses Silk
(maybe teams too?).  And there are many more rare/exotic codecs
optimised for very specific scenarios, like massive packet loss.

-- 
  ++ytti