[Bloat] Measuring latency-under-load consistently

[richard]

On Sat, 2011-03-12 at 23:57 +0200, Jonathan Morton wrote:
> On 12 Mar, 2011, at 6:04 am, richard wrote:
> 
> > OK - you make a good case for a new measure as my understanding of
> > jitter is latency related and typically measured at the link level (udp)
> > rather than at the application level.
> > 
> > I infer then that this will do things like impact the CPU load and disk
> > load, and might for example introduce "ringing" or harmonics into such
> > sub systems if/when applications end up "in sync" due to being "less
> > smooth" in their data output to the lower level IP levels.
> 
> I'm not sure how significant those effects would be, compared to simple
>  data starvation at the client.  Most Web servers operate with all the
>  frequently-accessed data in RAM (via disk cache) and serve many
>  clients at once or in quick succession, whose network paths don't have
>  the same bottlenecks in general.
> 
> It was my understanding that UDP-bsed protocols tended to tolerate

my bad - meant ICMP (as in ping)

>  packet loss through redundancy and graceful degradation rather than
>  retransmission, though there are always exceptions to the rule.  So a
>  video streaming server would be transmitting smoothly, with the client
>  giving feedback on how much data had been received and how much packet
>  loss it was experiencing.  Even if that status information is
>  considerably delayed, I don't see why load spikes at the server should
>  occur.

some of the video servers I deal with are using TCP (Windows Media for
example unless you configure it otherwise) but in general you're right,
the general rule with concurrent (as opposed to multicast) unicast
streams is that the server clocks the outbound and cares little about
whether the packets actually get there - that's the receiver's problem.

> 
> A fileserver, on the other hand, would not care very much.  Even if the
>  TCP window has grown to a megabyte, it takes longer to seek disk heads
>  than to read that much off the platter, so these lumps would be
>  absorbed by the normal readahead and elevator algorithms anyway. 
>  However, large TCP windows do consume RAM in both server and client,
>  and with a sufficient number of simultaneous clients, that could
>  theoretically cause trouble.  Constraining TCP windows to near the
>  actual BDP is more efficient all around.

Yes - have had RAM exhaustion problems on busy servers with large video
files - major headache.

> 
> > It will be affected by session drops due to timeouts as well as the need
> > to "fill the pipe" on a reconnect in such applications as streaming
> > video (my area) so that a key frame can come into the system and restart
> > the interrupted video play.
> 
> In the event of a TCP session drop, I think I will consider it a test
>  failure and give zero scores across the board.  Sufficient delay or
>  packet loss to cause that indicates a pretty badly broken network.

Yup - that's what the problem is all right
> 
> With that said, I can think of a case where it is likely to happen. 
>  Remember that I was seeing 30 seconds of buffering on a 500kbps 3G
>  link...  now what happens if the link drops to GPRS speeds?  There
>  would be over a megabyte of data queued up behind a 50Kbps link (at
>  best).  No wonder stuff basically didn't work when that happened.
> 
>  - Jonathan
> 

Your insights into this are great - thanks :)

richard

-- 
Richard C. Pitt                 Pacific Data Capture
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