Re: what is acceptible jitter for voip and videoconferencing?

Thu, 21 Sep 2023 17:52:31 -0700

That’s because symmetrical latency like you see on a satellite connection isn’t an issue at all for audio, it’s the variation or jitter that may cause issues.

Shane

On Sep 21, 2023, at 5:58 PM, Eric Kuhnke <[email protected]> wrote:


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 <[email protected]> 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 <[email protected]> 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),


but I am even more interested in finding cites to support various viewpoints, including mine, and learning how slas are met to deliver it.

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