[Cerowrt-devel] Ubiquiti QOS

Sun May 25 13:00:53 PDT 2014

Not that it is directly relevant, but there is no essential reason to require 50 ms. of buffering.  That might be true of some particular QOS-related router algorithm.  50 ms. is about all one can tolerate in any router between source and destination for today's networks - an upper-bound rather than a minimum.

The optimum buffer state for throughput is 1-2 packets worth - in other words, if we have an MTU of 1500, 1500 - 3000 bytes. Only the bottleneck buffer (the input queue to the lowest speed link along the path) should have this much actually buffered. Buffering more than this increases end-to-end latency beyond its optimal state.  Increased end-to-end latency reduces the effectiveness of control loops, creating more congestion.

The rationale for having 50 ms. of buffering is probably to avoid disruption of bursty mixed flows where the bursts might persist for 50 ms. and then die. One reason for this is that source nodes run operating systems that tend to release packets in bursts. That's a whole other discussion - in an ideal world, source nodes would avoid bursty packet releases by letting the control by the receiver window be "tight" timing-wise.  That is, to transmit a packet immediately at the instant an ACK arrives increasing the window.  This would pace the flow - current OS's tend (due to scheduling mismatches) to send bursts of packets, "catching up" on sending that could have been spaced out and done earlier if the feedback from the receiver's window advancing were heeded.

That is, endpoint network stacks (TCP implementations) can worsen congestion by "dallying".  The ideal end-to-end flows occupying a congested router would have their packets paced so that the packets end up being sent in the least bursty manner that an application can support.  The effect of this pacing is to move the "backlog" for each flow quickly into the source node for that flow, which then provides back pressure on the application driving the flow, which ultimately is necessary to stanch congestion.  The ideal congestion control mechanism slows the sender part of the application to a pace that can go through the network without contributing to buffering.

Current network stacks (including Linux's) don't achieve that goal - their pushback on application sources is minimal - instead they accumulate buffering internal to the network implementation.  This contributes to end-to-end latency as well.  But if you think about it, this is almost as bad as switch-level bufferbloat in terms of degrading user experience.  The reason I say "almost" is that there are tools, rarely used in practice, that allow an application to specify that buffering should not build up in the network stack (in the kernel or wherever it is).  But the default is not to use those APIs, and to buffer way too much.

Remember, the network send stack can act similarly to a congested switch (it is a switch among all the user applications running on that node).  IF there is a heavy file transfer, the file transfer's buffering acts to increase latency for all other networked communications on that machine.

Traditionally this problem has been thought of only as a within-node fairness issue, but in fact it has a big effect on the switches in between source and destination due to the lack of dispersed pacing of the packets at the source - in other words, the current design does nothing to stem the "burst groups" from a single source mentioned above.

So we do need the source nodes to implement less "bursty" sending stacks.  This is especially true for multiplexed source nodes, such as web servers implementing thousands of flows.

A combination of codel-style switch-level buffer management and the stack at the sender being implemented to spread packets in a particular TCP flow out over time would improve things a lot.  To achieve best throughput, the optimal way to spread packets out on an end-to-end basis is to update the receive window (sending ACK) at the receive end as quickly as possible, and to respond to the updated receive window as quickly as possible when it increases.

Just like the "bufferbloat" issue, the problem is caused by applications like streaming video, file transfers and big web pages that the application programmer sees as not having a latency requirement within the flow, so the application programmer does not have an incentive to control pacing.  Thus the operating system has got to push back on the applications' flow somehow, so that the flow ends up paced once it enters the Internet itself.  So there's no real problem caused by large buffering in the network stack at the endpoint, as long as the stack's delivery to the Internet is paced by some mechanism, e.g. tight management of receive window control on an end-to-end basis.

I don't think this can be fixed by cerowrt, so this is out of place here.  It's partially ameliorated by cerowrt, if it aggressively drops packets from flows that burst without pacing. fq_codel does this, if the buffer size it aims for is small - but the problem is that the OS stacks don't respond by pacing... they tend to respond by bursting, not because TCP doesn't provide the mechanisms for pacing, but because the OS stack doesn't transmit as soon as it is allowed to - thus building up a burst unnecessarily.

Bursts on a flow are thus bad in general.  They make congestion happen when it need not.

On Sunday, May 25, 2014 11:42am, "Mikael Abrahamsson" <swmike at swm.pp.se> said:

> On Sun, 25 May 2014, Dane Medic wrote:
> 
> > Is it true that devices with less than 64 MB can't handle QOS? ->
> > https://lists.chambana.net/pipermail/commotion-dev/2014-May/001816.html
> 
> At gig speeds you need around 50ms worth of buffering. 1 gigabit/s =
> 125 megabyte/s meaning for 50ms you need 6.25 megabyte of buffer.
> 
> I also don't see why performance and memory size would be relevant, I'd
> say forwarding performance has more to do with CPU speed than anything
> else.
> 
> --
> Mikael Abrahamsson    email: swmike at swm.pp.se
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