[Cerowrt-devel] [Make-wifi-fast] [tsvwg] Comments on draft-szigeti-tsvwg-ieee-802-11e
david at lang.hm
Fri Aug 7 18:31:02 EDT 2015
On Fri, 7 Aug 2015, dpreed at reed.com wrote:
> On Friday, August 7, 2015 4:03pm, "David Lang" <david at lang.hm> said:
>> Wifi is the only place I know of where the transmit bit rate is going to vary
>> depending on the next hop address.
> This is an interesting core issue. The question is whether additional
> queueing helps or hurts this, and whether the MAC protocol of WiFi deals well
> or poorly with this issue. It is clear that this is a peculiarly WiFi'ish
> It's not clear that the best transmit rate remains stable for very long, or
> even how to predict the "best rate" for the next station since the next
> station is one you may not have transmitted to for a long time, so your "best
> rate" information is old.
I wasn't even talking about the stability of the data rate to one destination. I
was talking about the fact that you may have a 1.3Gb connection to system A (a
desktop with a -ac 3x3 radio) and a 1Mb connection to machine B (an IoT 802.11b
trying to do BQL across 3+ orders of magnatude in speed isn't going to work
wihtout taking the speed into account.
Even if all you do is estimate with the last known speed, you will do better
than ignorming the speed entirely.
If the wifi can 'return' data to the queue when the transmission fails, it can
then fetch less data when it 're-transmits' the data at a lower speed.
> Queueing makes information about the channel older,
> by binding it too early. Sending longer frames means retransmitting longer
> frames when they don't get through, rather than agilely picking a better rate
> after a few bits.
As I understand wifi, once a transmission starts, it must continue at that same
data rate, it can't change mid-transmission (and tehre would be no way of
getting feedback in the middle of a transmission to know that it would need to
> The MAC protocol really should give the receiver some opportunity to control
> the rate of the next packet it gets (which it can do because it can measure
> the channel from the transmitter to itself, by listening to prior
> transmissions). Or at least to signal channel changes that might require a
> new signalling rate.
> This suggests that a transmitter might want to "warn" a receiver that some
> packets will be coming its way, so the receiver can preemptively change the
> desired rate. Thus, perhaps an RTS-CTS like mechanism can be embedded in the
> MAC protocol, which requires that the device "look ahead" at the packets it
> might be sending.
the recipient will receive a signal at any data rate, you don't have to tell it
ahead of time what rate is going to be sent. If it's being sent with a known
encoding, it will be decoded.
The sender picks the rate based on a number of things
1. what the other end said they could do based on the mode that they are
connected with (b vs g vs n vs bonded n vs ac vs 2x2 ac etc)
2. what has worked in the past. (with failed transmissions resulting in dropping
there may be other data like last known signal strength in the mix as well.
> On the other hand, that only works if the transmitter deliberately congests
> itself so that it has a queue built up to look at.
no, the table of associated devices keeps track of things like the last known
signal strength, connection mode, etc. no congestion needed.
> The tradeoffs are not obvious here at all. On the other hand, one could do
> something much simpler - just have the transmitter slow down to the worst-case
> rate required by any receiving system.
that's 1Mb/sec. This is the rate used for things like SSID broadcasts.
Once a system connects, you know from the connection handshake what speeds could
work. no need to limit yourself the the minimum that they all can know at that
> As the number of stations in range gets larger, though, it seems unlikely that
> "batching" multiple packets to the same destination is a good idea at all -
> because to achieve that, one must have n_destinations * batch_size chunks of
> data queued in the system as a whole, and that gets quite large. I suspect it
> would be better to find a lower level way to just keep the packets going out
> as fast as they arrive, so no clog occurs, and to slow down the stuff at the
> source as quickly as possible.
no, no, no
you are falling into the hardware designer trap that we just talked about :-)
you don't wait for the buffers to fill and always send full buffers, you
oppertunisticaly send data up to the max size.
you do want to send multiple packets if you have them waiting. Because if you
can send 10 packets to machine A and 10 packets to machine B in the time that it
would take to send one packet to A, one packet to B, a second packet to A and a
second packet to B, you have a substantial win for both A and B at the cost of
very little latency for either.
If there is so little traffic that sending the packets out one at a time doesn't
generate any congeston, then good, do that . but when you max out the
airtime, getting more data through in the same amount of airtime by sending
larger batches is a win
 if you are trying to share the same channel with others, this may be a
problem as it uses more airtime to send the same amount of data than always
batching. But this is a case of less than optimal network design ;-)
> [one should also dive into the reason for maintaining variable rates -
> multipath to a particular destination may require longer symbols for decoding
> without ISI. And when multipath is involved, you may have to retransmit at a
> slower rate. There's usually not much "noise" at the receiver compared to the
> multipath environment. (one of the reasons why mesh can be a lot better is
> that shorter distances have much less multipath effect, so you can get higher
> symbol rates by going multi-hop, and of course higher symbol rates compensate
> for more airtime occupied by a packet due to repeating).]
distance, interference, noise, etc are all variable in wifi. As a result, you
need to adapt.
The problem is that the adaptation is sometimes doing the wrong thing.
simlifying things a bit:
If your data doesn't get through at rate A, is the right thing to drop to rate
A/2 and re-transmit?
If the reason it didn't go through is that the signal is too weak for the rateA
encoding, then yes.
If the reason it didn't go through is that your transmission was stepped on by
something you can't hear (and can't hear you), but the recipient can here, then
slowing down means that you take twice the airtime to get the message through,
and you now have twice the chance of being stepped on again. Repeat and you
quickly get to everyone broadcasting at low rates and nothing getting through.
This is the key reason that dense wifi networks 'fall off the cliff' when they
hit saturation, the backoff that is entirely correct for a weak-signal, low
usage situations is entirely wrong in dense environments.
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