[Bloat] DSLReports Speed Test has latency measurement built-in

Tue Apr 21 10:20:18 EDT 2015

On Tue, 21 Apr 2015, jb wrote:

>> the receiver advertizes a large receive window, so the sender doesn't
> pause > until there is that much data outstanding, or they get a timeout of
> a packet as > a signal to slow down.
>
>> and because you have a gig-E link locally, your machine generates traffic
> \
>> very rapidly, until all that data is 'in flight'. but it's really sitting
> in the buffer of
>> router trying to get through.
>
> Hmm, then I have a quandary because I can easily solve the nasty bumpy
> upload graphs by keeping the advertised receive window on the server capped
> low, however then, paradoxically, there is no more sign of buffer bloat in
> the result, at least for the upload phase.
>
> (The graph under the upload/download graphs for my results shows almost no
> latency increase during the upload phase, now).
>
> Or, I can crank it back open again, serving people with fiber connections
> without having to run heaps of streams in parallel -- and then have people
> complain that the upload result is inefficient, or bumpy, vs what they
> expect.

well, many people expect it to be bumpy (I've heard ISPs explain to customers 
that when a link is full it is bumpy, that's just the way things work)

> And I can't offer an option, because the server receive window (I think)
> cannot be set on a case by case basis. You set it for all TCP and forget it.

I think you are right

> I suspect you guys are going to say the server should be left with a large
> max receive window.. and let people complain to find out what their issue
> is.

what is your customer base? how important is it to provide faster service to teh 
fiber users? Are they transferring ISO images so the difference is significant 
to them? or are they downloading web pages where it's the difference between a 
half second and a quarter second? remember that you are seeing this on the 
upload side.

in the long run, fixing the problem at the client side is the best thing to do, 
but in the meantime, you sometimes have to work around broken customer stuff.

> BTW my setup is wire to billion 7800N, which is a DSL modem and router. I
> believe it is a linux based (judging from the system log) device.

if it's linux based, it would be interesting to learn what sort of settings it 
has. It may be one of the rarer devices that has something in place already to 
do active queue management.

David Lang

> cheers,
> -Justin
>
> On Tue, Apr 21, 2015 at 2:47 PM, David Lang <david at lang.hm> wrote:
>
>> On Tue, 21 Apr 2015, jb wrote:
>>
>>  I've discovered something perhaps you guys can explain it better or shed
>>> some light.
>>> It isn't specifically to do with buffer bloat but it is to do with TCP
>>> tuning.
>>>
>>> Attached is two pictures of my upload to New York speed test server with 1
>>> stream.
>>> It doesn't make any difference if it is 1 stream or 8 streams, the picture
>>> and behaviour remains the same.
>>> I am 200ms from new york so it qualifies as a fairly long (but not very
>>> fat) pipe.
>>>
>>> The nice smooth one is with linux tcp_rmem set to '4096 32768 65535' (on
>>> the server)
>>> The ugly bumpy one is with linux tcp_rmem set to '4096 65535 67108864' (on
>>> the server)
>>>
>>> It actually doesn't matter what that last huge number is, once it goes
>>> much
>>> about 65k, e.g. 128k or 256k or beyond things get bumpy and ugly on the
>>> upload speed.
>>>
>>> Now as I understand this setting, it is the tcp receive window that Linux
>>> advertises, and the last number sets the maximum size it can get to (for
>>> one TCP stream).
>>>
>>> For users with very fast upload speeds, they do not see an ugly bumpy
>>> upload graph, it is smooth and sustained.
>>> But for the majority of users (like me) with uploads less than 5 to
>>> 10mbit,
>>> we frequently see the ugly graph.
>>>
>>> The second tcp_rmem setting is how I have been running the speed test
>>> servers.
>>>
>>> Up to now I thought this was just the distance of the speedtest from the
>>> interface: perhaps the browser was buffering a lot, and didn't feed back
>>> progress but now I realise the bumpy one is actually being influenced by
>>> the server receive window.
>>>
>>> I guess my question is this: Why does ALLOWING a large receive window
>>> appear to encourage problems with upload smoothness??
>>>
>>> This implies that setting the receive window should be done on a
>>> connection
>>> by connection basis: small for slow connections, large, for high speed,
>>> long distance connections.
>>>
>>
>> This is classic bufferbloat
>>
>> the receiver advertizes a large receive window, so the sender doesn't
>> pause until there is that much data outstanding, or they get a timeout of a
>> packet as a signal to slow down.
>>
>> and because you have a gig-E link locally, your machine generates traffic
>> very rapidly, until all that data is 'in flight'. but it's really sitting
>> in the buffer of a router trying to get through.
>>
>> then when a packet times out, the sender slows down a smidge and
>> retransmits it. But the old packet is still sitting in a queue, eating
>> bandwidth. the packets behind it are also going to timeout and be
>> retransmitted before your first retransmitted packet gets through, so you
>> have a large slug of data that's being retransmitted, and the first of the
>> replacement data can't get through until the last of the old (timed out)
>> data is transmitted.
>>
>> then when data starts flowing again, the sender again tries to fill up the
>> window with data in flight.
>>
>>  In addition, if I cap it to 65k, for reasons of smoothness,
>>> that means the bandwidth delay product will keep maximum speed per upload
>>> stream quite low. So a symmetric or gigabit connection is going to need a
>>> ton of parallel streams to see full speed.
>>>
>>> Most puzzling is why would anything special be required on the Client -->
>>> Server side of the equation
>>> but nothing much appears wrong with the Server --> Client side, whether
>>> speeds are very low (GPRS) or very high (gigabit).
>>>
>>
>> but what window sizes are these clients advertising?
>>
>>
>>  Note that also I am not yet sure if smoothness == better throughput. I
>>> have
>>> noticed upload speeds for some people often being under their claimed sync
>>> rate by 10 or 20% but I've no logs that show the bumpy graph is showing
>>> inefficiency. Maybe.
>>>
>>
>> If you were to do a packet capture on the server side, you would see that
>> you have a bunch of packets that are arriving multiple times, but the first
>> time "does't count" because the replacement is already on the way.
>>
>> so your overall throughput is lower for two reasons
>>
>> 1. it's bursty, and there are times when the connection actually is idle
>> (after you have a lot of timed out packets, the sender needs to ramp up
>> it's speed again)
>>
>> 2. you are sending some packets multiple times, consuming more total
>> bandwidth for the same 'goodput' (effective throughput)
>>
>> David Lang
>>
>>
>>  help!
>>>
>>>
>>> On Tue, Apr 21, 2015 at 12:56 PM, Simon Barber <simon at superduper.net>
>>> wrote:
>>>
>>>  One thing users understand is slow web access.  Perhaps translating the
>>>> latency measurement into 'a typical web page will take X seconds longer
>>>> to
>>>> load', or even stating the impact as 'this latency causes a typical web
>>>> page to load slower, as if your connection was only YY% of the measured
>>>> speed.'
>>>>
>>>> Simon
>>>>
>>>> Sent with AquaMail for Android
>>>> http://www.aqua-mail.com
>>>>
>>>>
>>>>
>>>> On April 19, 2015 1:54:19 PM Jonathan Morton <chromatix99 at gmail.com>
>>>> wrote:
>>>>
>>>>>>>> Frequency readouts are probably more accessible to the latter.
>>>>
>>>>>
>>>>>>>>     The frequency domain more accessible to laypersons? I have my
>>>>>>>>
>>>>>>> doubts ;)
>>>>>
>>>>>>
>>>>>>> Gamers, at least, are familiar with “frames per second” and how that
>>>>>>>
>>>>>> corresponds to their monitor’s refresh rate.
>>>>>
>>>>>>
>>>>>>       I am sure they can easily transform back into time domain to get
>>>>>>
>>>>> the frame period ;) .  I am partly kidding, I think your idea is great
>>>>> in
>>>>> that it is a truly positive value which could lend itself to being used
>>>>> in
>>>>> ISP/router manufacturer advertising, and hence might work in the real
>>>>> work;
>>>>> on the other hand I like to keep data as “raw” as possible (not that
>>>>> ^(-1)
>>>>> is a transformation worthy of being called data massage).
>>>>>
>>>>>>
>>>>>>  The desirable range of latencies, when converted to Hz, happens to be
>>>>>>>
>>>>>> roughly the same as the range of desirable frame rates.
>>>>>
>>>>>>
>>>>>>       Just to play devils advocate, the interesting part is time or
>>>>>>
>>>>> saving time so seconds or milliseconds are also intuitively
>>>>> understandable
>>>>> and can be easily added ;)
>>>>>
>>>>> Such readouts are certainly interesting to people like us.  I have no
>>>>> objection to them being reported alongside a frequency readout.  But I
>>>>> think most people are not interested in “time savings” measured in
>>>>> milliseconds; they’re much more aware of the minute- and hour-level time
>>>>> savings associated with greater bandwidth.
>>>>>
>>>>>  - Jonathan Morton
>>>>>
>>>>> _______________________________________________
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>>>>> Bloat at lists.bufferbloat.net
>>>>> https://lists.bufferbloat.net/listinfo/bloat
>>>>>
>>>>>
>>>>
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