[Bloat] sigcomm wifi

Sat Aug 23 15:26:59 EDT 2014

[removing Lars and Jim from direct cc, don't want to spam them - I don't know if they're sooo interested in this thread?]

On 23. aug. 2014, at 01:50, David Lang <david at lang.hm> wrote:

> On Sat, 23 Aug 2014, Michael Welzl wrote:
> 
>> On 21. aug. 2014, at 10:30, David Lang <david at lang.hm> wrote:
>> 
>>> On Thu, 21 Aug 2014, Michael Welzl wrote:
>>> 
>>>> On 21. aug. 2014, at 08:52, Eggert, Lars wrote:
>>>> 
>>>>> On 2014-8-21, at 0:05, Jim Gettys <jg at freedesktop.org> wrote:
>>>>>> And what kinds of AP's?  All the 1G guarantees you is that your bottleneck is in the wifi hop, and they can suffer as badly as anything else (particularly consumer home routers).
>>>>>> The reason why 802.11 works ok at IETF and NANOG is that:
>>>>>> o) they use Cisco enterprise AP's, which are not badly over buffered.
>>>> 
>>>> I'd like to better understand this particular bloat problem:
>>>> 
>>>> 100s of senders try to send at the same time. They can't all do that, so their cards retry a fixed number of times (10 or something, I don't remember, probably configurable), for which they need to have a buffer.
>>>> 
>>>> Say, the buffer is too big. Say, we make it smaller. Then an 802.11 sender trying to get its time slot in a crowded network will have to drop a packet, requiring the TCP sender to retransmit the packet instead. The TCP sender will think it's congestion (not entirely wrong) and reduce its window (not entirely wrong either). How appropriate TCP's cwnd reduction is probably depends on how "true" the notion of congestion is ... i.e. if I can buffer only one packet and just don't get to send it, or it gets a CRC error ("collides" in the air), then that can be seen as a pure matter of luck. Then I provoke a sender reaction that's like the old story of TCP mis-interpreting random losses as a sign of congestion. I think in most practical systems this old story is now a myth because wireless equipment will try to buffer data for a relatively long time instead of exhibiting sporadic random drops to upper layers. That is, in principle, a good thing - but buffering too much has of c!
> ourse all the problems that we know.. Not an easy trade-off at all I think.
>>> 
>>> in this case the loss is a direct sign of congestion.
>> 
>> "this case" - I talk about different buffer lengths. E.g., take the minimal buffer that would just function, and set retransmissions to 0. Then, a packet loss is a pretty random matter - just because you and I contended, doesn't mean that the net is truly "overloaded" ?  So my point is that the buffer creates a continuum from "random loss" to "actual congestion" - we want loss to mean "actual congestion", but how large should it be to meaningfully convey that?
>> 
>> 
>>> remember that TCP was developed back in the days of 10base2 networks where everyone on the network was sharing a wire and it was very possible for multiple senders to start transmitting on the wire at the same time, just like with radio.
>> 
>> cable or wireless: is one such occurrence "congestion"?
>> i.e. is halving the cwnd really the right response to that sort of "congestion"? (contention, really)
> 
> possibly not, but in practice it may be 'good enough'
> 
> but to make it work well, you probably want to play games with how much you back off, and how quickly you retry if you don't get a response.
> 
> The fact that the radio link can have it's own ack for the packet can actually be an improvement over doing it at the TCP level as it only need to ack/retry for that hop, and if that hop was good, there's far less of a need to retry if the server is just slow.

Yep... I remember a neat paper from colleagues at Trento University that piggybacked TCP's ACKs on link layer ACKs, thereby avoiding the collisions between TCP's ACKs and other data packets - really nice. Not sure if it wasn't just simulations, though.

> so if we try and do the retries in the OS stack, it will need to know the difference between "failed to get out the first hop due to collision" and "got out the first hop, waiting for the server across the globe to respond" with different timeouts/retries for them.
> 
>>> A large part of the problem with high-density wifi is that it just wasn't designed for that sort of environment, and there are a lot of things that it does that work great for low-density, weak signal environments, but just make the problem worse for high-density environements
>>> 
>>> batching packets together
>>> slowing down the transmit speed if you aren't getting through
>> 
>> well... this *should* only happen when there's an actual physical signal quality degradation, not just collisions. at least minstrel does quite a good job at ensuring that, most of the time.
> 
> "should" :-)
> 
> but can the firmware really tell the difference between quality degredation due to interference and collisions with other transmitters?

Well, with heuristics it can, sort of. As a simple example from one older mechanism, consider: multiple consecutive losses are *less* likely from random collisions than from link noise. That sort of thing. Minstrel worked best our tests, using tables of rates that worked well / didn't work well in the past:
http://heim.ifi.uio.no/michawe/research/publications/wowmom2012.pdf

>>> retries of packets that the OS has given up on (including the user has closed the app that sent them)
>>> 
>>> Ideally we want the wifi layer to be just like the wired layer, buffer only what's needed to get it on the air without 'dead air' (where the driver is waiting for the OS to give it more data), at that point, we can do the retries from the OS as appropriate.
>>> 
>>>> I have two questions: 1) is my characterization roughly correct? 2) have people investigated the downsides (negative effect on TCP) of buffering *too little* in wireless equipment? (I suspect so?)  Finding where "too little" begins could give us a better idea of what the ideal buffer length should really be.
>>> 
>>> too little buffering will reduce the throughput as a result of unused airtime.
>> 
>> so that's a function of, at least: 1) incoming traffic rate; 2) no. retries * ( f(MAC behavior; number of other senders trying) ).
> 
> incoming to the AP you mean?

incoming to whoever is sending and would be retrying - mostly the AP, yes.

> It also matters if you are worrying about aggregate throughput of a lot of users, or per-connection throughput for a single user.
> 
> From a sender's point of view, if it takes 100 time units to send a packet, and 1-5 time units to queue the next packet for transmission, you loose a few percentage of your possible airtime and there's very little concern.
> 
> but if it takes 10 time units to send the packet and 1-5 time units to queue the next packet, you have just lost a lot of potential bandwidth.
> 
> But from the point of view of the aggregate, these gaps just give someone else a chance to transmit and have very little effect on the amount of traffic arriving at the AP.
> 
> I was viewing things from the point of view of the app on the laptop.

Yes... I agree, and that's the more common + more reasonable way to think about it. I tend to think upstream, which of course is far less common, but maybe even more problematic. Actually I suspect the following: things get seriously bad when a lot of senders are sending upstream together; this isn't really happening much in practice - BUT when we have a very very large number of hosts connected in a conference style situation, all the HTTP GETs and SMTP messages and whatnot *do* create lots of collisions, a situation that isn't really too common (and maybe not envisioned / parametrized for), and that's why things often get so bad. (At least one of the reasons.)

>>> But at the low data rates involved, the system would have to be extremely busy to be a significant amount of time if even one packet at a time is buffered.
>> 
>> 
>> 
>>> You are also conflating the effect of the driver/hardware buffering with it doing retries.
>> 
>> because of the "function" i wrote above: the more you retry, the more you need to buffer when traffic continuously arrives because you're stuck trying to send a frame again.
> 
> huh, I'm missing something here, retrying sends would require you to buffer more when sending.

aren't you the saying the same thing as I ?  Sorry else, I might have expressed it confusingly somehow

> If people are retrying when they really don't need to, that cuts down on the avialable airtime.

Yes

> But if you have continual transmissions taking place, so you have a hard time getting a chance to send your traffic, then you really do have congestion and should be dropping packets to let the sender know that it shouldn't try to generate as much.

Yes; but the complexity that I was pointing at (but maybe it's a simple parameter, more like a 0 or 1 situation in practice?) lies in the word "continual". How long do you try before you decide that the sending TCP should really think it *is* congestion?  To really optimize the behavior, that would have to depend on the RTT, which you can't easily know.

Cheers,
Michael

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