Re: what is acceptible jitter for voip and videoconferencing?

[Michael Thomas <mike () mtcc com>]

On 9/21/23 3:31 PM, William Herrin wrote:
> On Thu, Sep 21, 2023 at 6:28 AM 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.
> Hi Tom,
>
> Jitter doesn't necessarily cause artifacts in the audio. Modern
> applications implement what's called a "jitter buffer." As the name
> implies, the buffer collects and delays audio for a brief time before
> playing it for the user. This allows time for the packets which have
> been delayed a little longer (jitter) to catch up with the earlier
> ones before they have to be played for the user. Smart implementations
> can adjust the size of the jitter buffer to match the observed
> variation in delay so that sound quality remains the same regardless
> of jitter.
>
> Indeed, on Zoom I barely noticed audio artifacts for a friend who was
> experiencing 800ms jitter. Yes, really, 800ms. We had to quit our
> gaming session because it caused his character actions to be utterly
> spastic, but his audio came through okay.

When I wrote my first implementation of telnet ages ago, i was both 
amused and annoyed about the go-ahead option. Obviously patterned after 
audio meat-space protocols, but I was never convinced it wasn't a 
solution in search of a problem. I wonder if CDMA was really an 
outgrowth of those protocols?

But it's my impression that gaming is by far more affected by latency 
and thus jitter buffers for voice. Don't some ISP's even cater to gamers 
about latency?

Mike