[Bloat] Goodput fraction w/ AQM vs bufferbloat

[Fred Baker]

There are a couple of ways to approach this, and they depend on your network model.

In general, if you assume that there is one bottleneck, losses occur in the queue at the bottleneck, and are each retransmitted exactly once (not necessary, but helps), goodput should approximate 100% regardless of the queue depth. Why? Because every packet transits the bottleneck once - if it is dropped at the bottleneck, the retransmission transits the bottleneck. So you are using exactly the capacity of the bottleneck.

the value of a shallow queue is to reduce RTT, not to increase or decrease goodput. cwnd can become too small, however; if it is possible to set cwnd to N without increasing queuing delay, and cwnd is less than N, you're not maximizing throughput. When cwnd grows above N, it merely increases queuing delay, and therefore bufferbloat.

If there are two bottlenecks in series, you have some probability that a packet transits one bottleneck and doesn't transit the other. In that case, there is probably an analytical way to describe the behavior, but it depends on a lot of factors including distributions of competing traffic. There are a number of other possibilities; imagine that you drop a packet, there is a sack, you retransmit it, the ack is lost, and meanwhile there is another loss. You could easily retransmit the retransmission unnecessarily, which reduces goodput. The list of silly possibilities goes on for a while, and we have to assume that each has some probability of happening in the wild.



On May 5, 2011, at 9:01 AM, Jim Gettys wrote:

> On 04/30/2011 03:18 PM, Richard Scheffenegger wrote:
>> I'm curious, has anyone done some simulations to check if the following qualitative statement holds true, and if, what the quantitative effect is:
>> 
>> With bufferbloat, the TCP congestion control reaction is unduely delayed. When it finally happens, the tcp stream is likely facing a "burst loss" event - multiple consecutive packets get dropped. Worse yet, the sender with the lowest RTT across the bottleneck will likely start to retransmit while the (tail-drop) queue is still overflowing.
>> 
>> And a lost retransmission means a major setback in bandwidth (except for Linux with bulk transfers and SACK enabled), as the standard (RFC documented) behaviour asks for a RTO (1sec nominally, 200-500 ms typically) to recover such a lost retransmission...
>> 
>> The second part (more important as an incentive to the ISPs actually), how does the fraction of goodput vs. throughput change, when AQM schemes are deployed, and TCP CC reacts in a timely manner? Small ISPs have to pay for their upstream volume, regardless if that is "real" work (goodput) or unneccessary retransmissions.
>> 
>> When I was at a small cable ISP in switzerland last week, surely enough bufferbloat was readily observable (17ms -> 220ms after 30 sec of a bulk transfer), but at first they had the "not our problem" view, until I started discussing burst loss / retransmissions / goodput vs throughput - with the latest point being a real commercial incentive to them. (They promised to check if AQM would be available in the CPE / CMTS, and put latency bounds in their tenders going forward).
>> 
> I wish I had a good answer to your very good questions.  Simulation would be interesting though real daa is more convincing.
> 
> I haven't looked in detail at all that many traces to try to get a feel for how much bandwidth waste there actually is, and more formal studies like Netalyzr, SamKnows, or the Bismark project would be needed to quantify the loss on the network as a whole.
> 
> I did spend some time last fall with the traces I've taken.  In those, I've typically been seeing 1-3% packet loss in the main TCP transfers.  On the wireless trace I took, I saw 9% loss, but whether that is bufferbloat induced loss or not, I don't know (the data is out there for those who might want to dig).  And as you note, the losses are concentrated in bursts (probably due to the details of Cubic, so I'm told).
> 
> I've had anecdotal reports (and some first hand experience) with much higher loss rates, for example from Nick Weaver at ICSI; but I believe in playing things conservatively with any numbers I quote and I've not gotten consistent results when I've tried, so I just report what's in the packet captures I did take.
> 
> A phenomena that could be occurring is that during congestion avoidance (until TCP loses its cookies entirely and probes for a higher operating point) that TCP is carefully timing it's packets to keep the buffers almost exactly full, so that competing flows (in my case, simple pings) are likely to arrive just when there is no buffer space to accept them and therefore you see higher losses on them than you would on the single flow I've been tracing and getting loss statistics from.
> 
> People who want to look into this further would be a great help.
>                - Jim
> 
> 
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