This discussion focuses on interactive audio/data streams. One-way streams
are treated differently, since delay is less important.
Media transport typically uses IPv4/6 networks and can often be captured at
one of the endpoints or somewhere in the network path. E.g., I captured a
Skype call
frequency measurement
>
> Subject: [time-nuts] an interesting timing problem
> Message-ID:
> Content-Type: text/plain; charset=utf-8; format=flowed
>
> Given that there's a lot more people spending time zooming, webexing,
> teaming, skype, facetime, etc. these days, I'm curious
On 5/6/20 7:33 AM, Chris Howard wrote:
At my current job we were looking into delay timings of video systems.
We were doing end-to-end measurement by putting a time display in front
of a monitor
and have the camera show both the time display and the monitor.
It looks a bit like the old
Good day
Easy way of testing such is to make a skype or team viewer call and then
get the other side to synchronize the computer that side as simultaneously
as possible with you by click on change date and time settings and then
Internet time. It for a few seconds are absolutely perfect but
At my current job we were looking into delay timings of video systems.
We were doing end-to-end measurement by putting a time display in front
of a monitor
and have the camera show both the time display and the monitor.
It looks a bit like the old infinite mirror.
If you arrange things right
Given that there's a lot more people spending time zooming, webexing,
teaming, skype, facetime, etc. these days, I'm curious if anyone has
figured out to *quantify* the issues of lag, desynchronization, etc.
How would one go about instrumenting it (without access to the source
code or servers