Re: what is acceptible jitter for voip and videoconferencing?

[Eric Kuhnke <eric.kuhnke () gmail com>]

Artifacts in audio are a product of packet loss or jitter resulting in
codec issues issues leading to human subject perceptible audio anomalies,
not so much latency by itself. Two way voice is remarkably NOT terrible on
a 495ms RTT satellite based two-way geostationary connection as long as
there is little or no packet loss.

On Thu, Sep 21, 2023 at 12:47 PM Tom Beecher <beecher () beecher cc> wrote:

> My understanding has always been that 30ms was set based on human
> perceptibility. 30ms was the average point at which the average person
> could start to detect artifacts in the audio.
>
> On Tue, Sep 19, 2023 at 8:13 PM Dave Taht <dave.taht () gmail com> wrote:
>
> > Dear nanog-ers:
> >
> > 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.
> >
> > 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),
> >
> > https://www.linkedin.com/feed/update/urn:li:ugcPost:7110029608753713152/
> >
> > but I am even more interested in finding cites to support various
> > viewpoints, including mine, and learning how slas are met to deliver it.
> >
> > --
> > Oct 30:
> > https://netdevconf.info/0x17/news/the-maestro-and-the-music-bof.html
> > Dave Täht CSO, LibreQos