seanadams;176002 Wrote: 
> I think you are asking the wrong question here because you are confused
> about how TCP and buffering work. Case in point, it took me a moment to
> figure out that by m/s you meant milliseconds and not meters per
> second!
> 
> Anyway, a 50ms delay is absolutely mice nuts. On wireless we routinely
> deal with that sort of variance in the normal flow of packets, not to
> mention a few percent of packet loss. Even if you could adjust the
> buffer size, I am not sure what that would help you to illustrate.
> 
> Is there an alternative failure mode (for example a competing product
> which does not recover as quickly), where you would like to show that
> streaming does not perform as well? If you could describe what that
> failure looks like, maybe we could suggest a test to illustrate it.
> 
> If you want to show the difference between, say, a 50ms outage and a
> 250ms outage, then VOIP is probably the best thing to demonstrate. VOIP
> potocols, by design, sacrifice reliability in order to achieve real-time
> streaming with low latency. Squeezebox is the opposite - it plays from
> stored data which can be read into its memory as fast as the network
> will deliver it, even in a bursty manner, so the stream can continue
> even with heavy packet loss or a complete outage of several seconds.
> You couldn't make Squeezebox behave like VOIP just by reducing the
> buffer size, because TCP is *totally* different!

Sean I am sorry if i did not explain myself correctly, below is my
simplistic view. Please correct me if I am wrong. 

TCP is partially responsible to ensure  the data it receives from the
SS  is accurate and in the proper order and pass's it up to the next
layer. Lets say it drops the data into the "music bucket" If an errored
packet arrives at the SB it is discarded an another copy of a particular
packet is requested  from the SS .  The upper layers of the SB's 
software have the task  (among other things of course)to stream out
music leaking from the bottom of the bucket . Because the music has to
be delivered in a uninterupted fashion a certain  amount of buffer
space, or bucket content storing capability is built in to the SB to 
allow TCP time  to do it's thing during poor network conditions. 

If the bucket size  is to small or does not exist , and the network was
constantly congested or in a poor state (dropping packets) it would not
matter if TCP or UDP was used, the user would experience interuption in
the music. The goal is to never let the bucket go empty. TCP will ensure
what goes into the bucket is accurate, however the rate it fills the
bucket is dependant on the network, the SS and other things.

I would imagine you guy's played around with the buffer size, and
examined the characteristics of the various network medium etc. at some
point in time?

What I was hoping to do, (it appears I can't, but thats ok) is decrease
the SB buffer size, to a point that it cannot tolerate any packet drops.
Yes TCP will make sure what goes into the bucket is still accurate. 

Our network equipment uses very quick failure recovery mechanisms, and
has very deep queue's and sophisticated schedulers. By dialing back the
SB's buffering capabilities, I am putting the emphasis on our network
equipment to deliver the content with little or no  packet drops, and
very little latency. My plan was to congest the network with best
effort traffic, prioritise the music traffic and introduce failures
into the network, knowing that SS/SB current setup was tweaked in such
a way that it cannot tolerate interuption in packet flow. I can do this
all day long with a traffic generator, but seeing real life applications
such as music and video is sexy and sexy sells.

In a perfect world i would demonstrate the SB/SS working as intended in
stock fashion, then i could dial back the buffer size  to a point of
lets say 75-100 milli-seconds, congest and introduce network failures
and let the customer observe the results.

Thanks for your time


-- 
sfraser
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