On Sat, 23 Aug 2014, Michael Welzl wrote:
On 21. aug. 2014, at 10:30, David Lang <da...@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 <j...@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.
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?
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?
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.
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.
If people are retrying when they really don't need to, that cuts down on the
avialable airtime.
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.
David Lang
what am I getting wrong? this seems to be just the conversation I was hoping
to have ( so thanks!) - I'd like to figure out if there's a fault in my
logic.
Cheers,
Michael
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