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Date: Fri, 1 May 2020 10:56:31 +0200
From: "Carles Gomez Montenegro" <carlesgo@entel.upc.edu>
To: "Yuchung Cheng" <ycheng=40google.com@dmarc.ietf.org>
Cc: jon.crowcroft@cl.cam.ac.uk, "tcpm@ietf.org Extensions" <tcpm@ietf.org>, "Neal Cardwell" <ncardwell@google.com>
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Subject: Re: [tcpm] question regarding draft-gomez-tcpm-delack-suppr-reqs
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Dear Yuchung,

Thanks for your questions and comments.

Please find below my inline responses:

> thanks for the talk!
>
> https://tools.ietf.org/html/draft-gomez-tcpm-delack-suppr-reqs-01#section-3.1
> "Note that, while Appropriate Byte Counting (ABC) [RFC3465] might be used
>    to address this problem, it remains as an experimental mechanism, not
>    fully included in RFC 5681, which specifies standard TCP congestion
>    control.  "
>
> It's not clear what "fully included" means here. AFAICT, RFC5681
> includes the essence of ABC explicitly.
>
> https://tools.ietf.org/html/rfc5681#section-3.1
>
> " During slow start, a TCP increments cwnd by at most SMSS bytes for
>    each ACK received that cumulatively acknowledges new data.  Slow
>    start ends when cwnd exceeds ssthresh (or, optionally, when it
>    reaches it, as noted above) or when congestion is observed.  While
>    traditionally TCP implementations have increased cwnd by precisely
>    SMSS bytes upon receipt of an ACK covering new data, we RECOMMEND
>    that TCP implementations increase cwnd, per:
>
>       cwnd += min (N, SMSS)                      (2)
>
>    where N is the number of previously unacknowledged bytes acknowledged
>    in the incoming ACK.  This adjustment is part of Appropriate Byte
>    Counting [RFC3465] and provides robustness against misbehaving
>    receivers that may attempt to induce a sender to artificially inflate
>    cwnd using a mechanism known as "ACK Division" [SCWA99].  ACK
>    Division consists of a receiver sending multiple ACKs for a single
>    TCP data segment, each acknowledging only a portion of its data.  A
>    TCP that increments cwnd by SMSS for each such ACK will
>    inappropriately inflate the amount of data injected into the network."

Note that the problem pointed out is that, as stated in the RFC 5681 text
above:

   "During slow start, a TCP increments cwnd by at most SMSS bytes for
    each ACK received that cumulatively acknowledges new data"

It may be desirable to support, during slow start, a cwnd increase by more
than SMSS per each ACK acknowledging new data.

In fact, couple of paragraphs later, RFC 5681 explicitly indicates the
following:

   "We note that [RFC3465] allows for cwnd increases of more than SMSS
   bytes for incoming acknowledgments during slow start on an
   experimental basis; however, such behavior is not allowed as part of
   the standard."

That is what we meant by "not fully included in RFC 5681" in our document.

> As a side note, Linux stack has implemented the principle of abc in
> its congestion control (including slow start)

Thanks for the information.

Yes, in fact, I second the WG members who stated on the chat, during the
interim, that it is surprising that ABC remains as an experimental
document.

I realize that, at an implementation level, the problem pointed out above
is being addressed in many platforms.

> regarding delayed ACK harming slow-start or latency, I am not too sure
> either:
>
> 1) The ACK is delayed but not the data
>
> 2) The delay of the ACK can indeed delay cwnd growth during slowstart.
>
> But the ACK is delayed because the packet is likely a sub-MSS data,
> which is again likely caused by application limit (i.e. the
> application has sent all its data like a HTTP response). In this case
> the delay of the cwnd growth does not hurt the performance since the
> application has no data left to send.
>
>    The only penalty occurs when the application has more data to send
> but cwnd is still gated by the arrival of the delayed ACK. However, in
> many request - response protocol, either the (non-delayed) ACKs
> triggered by prior full-MSS data inflight or the delayed ACKs would
> have returned to grow the cwnd to sending immediately.
>
> I can construct scenarios where delayed ACK significantly hurts
> application. But I don't think the scenario is too common. I will
> suggest more tests to quantify the effect of delayed ACK effect on
> slow-start.

Yes, I tend to agree.

In fact, the two main problems due to Delayed ACKs that we identified so
far in the area of slow start are:

- The fact that ABC might not always be supported (due to its
"Experimental" status).

- Precluding sender behaviors such as chirping.

Once again, thanks for your valuable feedback!

Cheers,

Carles


> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>



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Date: Fri, 1 May 2020 11:51:19 +0200
From: "Carles Gomez Montenegro" <carlesgo@entel.upc.edu>
To: "Scheffenegger, Richard" <rs.ietf@gmx.at>
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Subject: Re: [tcpm] On Sender Control of Delayed Acks in TCP
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Hi Richard,

Thanks a lot for your detailed explanation, pointers and comments.

Please find below my inline responses:

> Hi,
>
>
> as many here know, I'm very much interested in transactional data
> exchange across TCP sessions, rather than the more common architectural
> approach of assuming unlimited data to transmit unidirectional.
>
> One of the issues in that space is, that these application limited
> sessions can very frequently run out of data, and expirience idle periods.
>
> Eliciting an ACK under certain circumstances, for a timely continuation
> of the data transmission / growth of the congestion window is a known
> method to reduce network delay.
>
> E.g. Linux has been using the CWR flag for the purpose of sending out an
> immediate ACK by the receiver, since there is an increased chance of a
> very small cwnd when the sender had just reduced it's congestion window.
>
> https://lore.kernel.org/patchwork/patch/970486/
>
> and we also found latency improvements doing this when running
> ECN-enabled sessions
>
> https://reviews.freebsd.org/D22670
>
> There are also numerous advisories, suggesting to disable delayed ACK
> completely when latency and/or timely bandwidth recovery under
> adversarial conditions is a major driver (e.g.
> https://www.vmware.com/content/dam/digitalmarketing/vmware/en/pdf/techpaper/performance/esxi7-nfs-read-perf.pdf
> for one that was just published very recently).

This is, all, very useful feedback!

We plan to document the "ACK Pull" approach described above in our draft,
and the associated rationale.

Also, in our draft we currently mention that completely disabling delayed
ACKs might not be a good solution, but as you mentioned, there may be
environments where it may make sense to do so. We will add a note on this
as well.

> To summarize: While I agree that under steady load conditions, there are
> probably some benefits to be had by TCP itself thinning out ACKs, and
> address certain environment (e.g. asymetric link bandwidth), there are
> also conditions, where a timely ACK is crucial in maintaining low
> latency and timely bandwidth recovery. Short of having a mechansim do
> explicitly do this, at least for the latter issue, TLP (or TLP-like)
> methods are the only ones to currently address that - but the duplicate
> unconditional transmission of a (small) data packet does have additional
> cost associated with it.

Thanks for the assessment of the need for ACK thinning and ACK Pull
(depending on the considered scenario).

Cheers,

Carles


> Best regards,
>     Richard
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>



From nobody Fri May  1 03:23:38 2020
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Date: Fri, 1 May 2020 12:22:42 +0200
From: "Carles Gomez Montenegro" <carlesgo@entel.upc.edu>
To: "Neal Cardwell" <ncardwell@google.com>
Cc: "Scheffenegger, Richard" <rs.ietf@gmx.at>, "tcpm@ietf.org" <tcpm@ietf.org>, "Gorry Fairhurst" <gorry@erg.abdn.ac.uk>, "Bob Briscoe" <ietf@bobbriscoe.net>, jon.crowcroft@cl.cam.ac.uk
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Subject: Re: [tcpm] On Sender Control of Delayed Acks in TCP
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Hi Neil,

Thanks a lot for your comments!

Please find below my inline responses:

> On Wed, Apr 29, 2020 at 1:03 PM Scheffenegger, Richard <rs.ietf@gmx.at>
> wrote:
>> Eliciting an ACK under certain circumstances, for a timely continuation
>> of the data transmission / growth of the congestion window is a known
>> method to reduce network delay.
>>
>> E.g. Linux has been using the CWR flag for the purpose of sending out an
>> immediate ACK by the receiver, since there is an increased chance of a
>> very small cwnd when the sender had just reduced it's congestion window.
>>
>> https://lore.kernel.org/patchwork/patch/970486/
>>
>> and we also found latency improvements doing this when running
>> ECN-enabled sessions
>>
>> https://reviews.freebsd.org/D22670
>
> Yes, agreed.
>
> IMHO an explicit ACK-pull mechanism would be very nice for the cwnd<=1
> case.

(While this question is about the solution space, I'm anyway curious...)
In the context of datacenter networks, would you have any suggestion or
preference regarding the solutions that have been mentioned so far?

Or perhaps any particular requirement for a potential solution?

> In datacenter networks the BDPs tend to be so small, and the numbers
> of flows can be so large, that it's not unusual for a flow's fair
> share of the BDP to be one packet or less. If the flows want to
> maintain low queuing delays, then ideally they would be able to reduce
> their cwnd to 1 packet (or effectively even lower, with pacing). But
> the current behavior of TCP delayed ACKs often causes terrible delays
> with a cwnd of 1.

This is very useful input to expand the current section 3.2 of our
requirements draft, which focuses on high bit rate environments.

> The Linux TCP approach of sending an immediate ACK upon receiving a
> packet with CWR set (see Richard's link above) helps a lot with this
> case. But that's a non-standard heuristic, and doesn't help for
> senders that have a congestion control that does not use ECN CWR
> signals (e.g. Accurate ECN). Unless I'm missing something, it seems
> like endpoints using Accurate ECN are going to run into this same
> latency issue. To avoid this, I guess they may want to either use a
> cwnd >= 2, or make use of some new explicit ACK-pull mechanism.

Agreed.

Once again, thanks for your comments!

Carles



> best,
> neal
>



From nobody Fri May  1 05:20:03 2020
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From: Neal Cardwell <ncardwell@google.com>
Date: Fri, 1 May 2020 08:19:38 -0400
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To: Carles Gomez Montenegro <carlesgo@entel.upc.edu>
Cc: "Scheffenegger, Richard" <rs.ietf@gmx.at>, "tcpm@ietf.org" <tcpm@ietf.org>, Gorry Fairhurst <gorry@erg.abdn.ac.uk>, Bob Briscoe <ietf@bobbriscoe.net>, jon.crowcroft@cl.cam.ac.uk
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Subject: Re: [tcpm] On Sender Control of Delayed Acks in TCP
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On Fri, May 1, 2020 at 6:23 AM Carles Gomez Montenegro
<carlesgo@entel.upc.edu> wrote:
>
> Hi Neil,
>
> Thanks a lot for your comments!
>
> Please find below my inline responses:
>
> > On Wed, Apr 29, 2020 at 1:03 PM Scheffenegger, Richard <rs.ietf@gmx.at>
> > wrote:
> >> Eliciting an ACK under certain circumstances, for a timely continuation
> >> of the data transmission / growth of the congestion window is a known
> >> method to reduce network delay.
> >>
> >> E.g. Linux has been using the CWR flag for the purpose of sending out an
> >> immediate ACK by the receiver, since there is an increased chance of a
> >> very small cwnd when the sender had just reduced it's congestion window.
> >>
> >> https://lore.kernel.org/patchwork/patch/970486/
> >>
> >> and we also found latency improvements doing this when running
> >> ECN-enabled sessions
> >>
> >> https://reviews.freebsd.org/D22670
> >
> > Yes, agreed.
> >
> > IMHO an explicit ACK-pull mechanism would be very nice for the cwnd<=1
> > case.
>
> (While this question is about the solution space, I'm anyway curious...)
> In the context of datacenter networks, would you have any suggestion or
> preference regarding the solutions that have been mentioned so far?
>
> Or perhaps any particular requirement for a potential solution?

Among the solutions outlined in
  https://tools.ietf.org/html/draft-gomez-tcpm-delack-suppr-reqs-00
my first preference would be the AKP option, since that approach
avoids burning a precious flag bit.

The AKP option may be stripped by some middleboxes, but (a) as a
performance optimization, that should be acceptable, and (b) for the
datacenter case (where cwnd=1 is a motivating use case) this should
not be a concern.

IMHO a flag bit makes sense for a small signal that a sender might
want to send frequently or at a high rate, but senders should not be
trying to force immediate ACKs frequently.

best,
neal


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Subject: Re: [tcpm] A longer follow-up on my comment on draft-gomez-tcpm-delack-suppr-reqs
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Gorry,

On 30/04/2020 09:36, Gorry Fairhurst wrote:
> See some more explanation in line.
>
> On 29/04/2020 22:58, Bob Briscoe wrote:
>> Gorry,
>>
>> On 29/04/2020 18:54, Gorry Fairhurst wrote:
>>
>>> - DAASS is a simple fix, that appears to help a lot when there is 
>>> data waiting for the arrival of the delayed ACK. 
>>
>> Any heuristic at the receiver to detect the end of a pattern of 
>> start-up such as slow-start, ossifies that pattern into the Internet. 
>> Sender control of the DelACK ratio removes that ossification.
>>
> I initially thought the same, DAASS tried to fix multiple problems. 
> However, as regards DAASS  reducing the waiti for the ACK to a rounds 
> of initial cwnd growth, I think the key understanding is the sender 
> use of byte accounting (for stretch-ACKs) not individual ACKs, (that 
> might imply a need for some burst-mitigation for stretch ACKs, because 
> you reduce ACK clocking).
>
> I like what I understand Chrome's QUIC does:  In the first 100 
> received packets, send an ACK for each packet (could be each pair of 
> packets).  This rapidly grows a cwnd of about 140 packets (assuming 
> exponetial growth). Once the cwnd is >100 then an ACK covering 10 
> packets/segments becomes a small proportion of the cwnd (the 
> additional delay from per-packet ACKs is small). The receiver can 
> moves to a larger default ACK ratio.

[BB] If it's likely that we need to move beyond slow-start behaviour in 
future, building heuristics around slow-start is the worst possible step 
we could take. That's the ossification I'm talking about.

When we implemented paced chirping, it was really hard not to confuse 
the heuristics in Linux receivers. Linux receivers are built to detect 
the end of today's slow-start behaviour. Our sender wasn't using 
anything like slow start behaviour, which triggered their heuristics 
much too early.

Magically increasing the delack ratio after 100 packets is another 
behaviour we must not encourage.

Sender control of the DelAck ratio can be really simple. Then we have a 
simple solution that will last for ever.
We must not dig the heuristics and magic numbers hole any deeper.


>
>>
>>> ABC is important also (well what I think people understand by ABC - 
>>> rather than the RFC on this).
>>
>> Yes.
>>
>>> * High bit rate environments and short data segments
>>>
>>> If you really don’t want delayed ACKs, the protocol using TCP ought 
>>> to disable nagle. Or you would perhaps use DAASS... I’m not sure 
>>> which problem you speak about?
>>>
>>>
>>> * Beyond classic ACK transmission behavior
>>> I do agree that Delayed ACKs preclude using sender behaviors 
>>> intended to quickly and non-intrusively probe for available capacity 
>>> during slow start. This is important to me, but I am very wary about 
>>> the idea of sending more ACKs to do this, and we need to be careful 
>>> to gain sufficient information about what the receiver has seen - 
>>> not just to get a bunch of separate (each cumulative) ACKs. This 
>>> really doesn’t give much fidelity to the sender. I’d be happy to 
>>> talk more about what might be better!
>>
>> Indeed, ACK frames in QUIC that give the arrival times of data 
>> packets are an alternative - as long as they are not stretched beyond 
>> the time at which they would have been useful.
>>
> Interesting, I think.
>> But we are more limited for space in TCP. As long as a TCP server is 
>> serving bulk and short flows, it can put more effort into receiving 
>> more ACKs during flow startup, and save its processing power by using 
>> less frequent ACKs for the majority of its data - transmitted in 
>> congestion avoidance.
>>
>>
> Also seems true.
>>> * IoT scenarios
>>> I can see the argument of an ACK energy cost for the IoT device. At 
>>> least, I would assume IoT devices can be tuned, and the apps they 
>>> talk to can be tuned. Appropriate guidance helps!
>>
>> Tuning is for a scenario, not for widely differing scenarios. The 
>> message of this draft is that a connection sometimes needs to be able 
>> to adapt its ACK ratio at run-time. Not every IoT device is always 
>> deployed for just one scenario. Neither is its peer. To me this says 
>> adaptation mustn't be complex (and doesn't have to be).
>>
> Heterogeniety of the network segments is tricky. Some optimisations 
> could be handled by link-specific methods - header compression perhaps 
> is an example, other things need to be adapted end-to-end. However, 
> end-to-end transport has to work for all types of path (links) - and 
> often does not know the path.

[BB] Sender control of the receiver's DelAck ratio would be always 
better and never worse in these cases. It might not solve the lack of 
path knowledge. But wherever path knowledge can be gathered, sender 
control of the receiver's DelAck ratio would then be useful/necessary.

Then, any sender can implement "link-specific methods" unilaterally (or 
not if it chooses). Sndr-rcvr co-ordination is an important piece that's 
missing.

You seem to be generating confusion. There is still good reason to add a 
generally useful missing capability (sender control of receiver DelACK 
ratio), whether or not other capabilities might sometimes be missing in 
certain odd scenarios.

Do one thing and do it well.


>>>
>>> * Bursty Apps
>>> I think you can add varying workloads as something important. By 
>>> which I mean apps that do transactional stuff, or are controlled by 
>>> applications where the data transmission need varies.When we looked 
>>> at CWV and the ideas that the applications control the traffic 
>>> patterns in one group of applications, we also noted that these 
>>> applications change the way they use the network. That’s important 
>>> for a timely restart to growing the cwnd (etc). Such applications 
>>> can also be very sensitive to network delay.  These are probably 
>>> also of interest!
>>
>> Yes.
>>
>>>
>>> So....  the reason for hesitancy overall is I see this as tricky 
>>> space to get correct.
>>
>> But trickiness is the point of starting this requirements document, 
>> isn't it?
>> Please can you articulate better why you seem negative about this 
>> exercise?
>>>
>>> There are many places where fewer ACKs are good - if you have per 
>>> ACK interrupt costs - if the capacity consumed by ACKs is important 
>>> - the cost of sending in the link technology is high - etc. This is 
>>> complicated for TCP (and much less so for QUIC - because QUIC has a 
>>> notion of pacing; QUIC’s loss recovery is different; and QUIC’s ACKs 
>>> are not easily thinned, at least at present). Any TCP method has to 
>>> live with networks that experience pain from more ACKs, but also may 
>>> deploy (various) mitigations.
>>
>> Yes. But reading between the lines I hear you say "Don't start this 
>> exercise".
>> Pls explicitly articulate your implicit message.
>>
> The slide asked about adopting this work - my words were against doing 
> that at that time with that revision of this document.
>
> I miss seeing presentations with results and showing the thinking. I  
> wonder if we need a specific agenda slot to focus on such topics. I 
> expect we'll learn a lot from some data from a variety of 
> problem-spaces/use-cases, especially if the timeslot is sufficient to 
> include thinking and experience from people outside of TCPM. (That's 
> another thought - and probably should be another thread).

[BB] I certainly agree with the need to base this on experience.
However, all we have at the moment is experience of protocols /without/ 
sender control of the DelAck behaviour of the receiver. That's useful 
experience, but only to demonstrate the /problem/.

It would be good to hear some empirical experience of protocols /with/ 
sender control.


>
>>>
>>> There are also cases where you’d like more information than a 
>>> delayed ACK provides (e.g. chirping and similar probing methods) and 
>>> there are application pathologies that would love an ACK at the end 
>>> of a packet burst (be that 1 or 100s of packets). Any method has to 
>>> live with a wide variety of paths and applications.
>>>
>>> Happy to discuss more. And I suspect other people may also respond 
>>> on list. We have lots of data for QUIC and TCP - but most of our 
>>> data focusses on testbeds or on actual broadband satellite services.
>>

[BB] As above. Data about the existing Internet without the proposed 
capability is useful to characterize the problem. But need to be aware 
of its limitations at characterizing the solution-space proposed here.


Bob


>> Thanks for elaborating.
>>
>>
>>
>> Bob
> Gorry
>>
>>
>>>
>>> Gorry
>>>
>>>
>>>
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

-- 
________________________________________________________________
Bob Briscoe                               http://bobbriscoe.net/


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From: Yuchung Cheng <ycheng@google.com>
Date: Fri, 1 May 2020 09:48:57 -0700
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To: Carles Gomez Montenegro <carlesgo@entel.upc.edu>
Cc: jon.crowcroft@cl.cam.ac.uk, "tcpm@ietf.org Extensions" <tcpm@ietf.org>,  Neal Cardwell <ncardwell@google.com>
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Subject: Re: [tcpm] question regarding draft-gomez-tcpm-delack-suppr-reqs
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On Fri, May 1, 2020 at 1:56 AM Carles Gomez Montenegro
<carlesgo@entel.upc.edu> wrote:
>
> Dear Yuchung,
>
> Thanks for your questions and comments.
>
> Please find below my inline responses:
>
> > thanks for the talk!
> >
> > https://tools.ietf.org/html/draft-gomez-tcpm-delack-suppr-reqs-01#section-3.1
> > "Note that, while Appropriate Byte Counting (ABC) [RFC3465] might be used
> >    to address this problem, it remains as an experimental mechanism, not
> >    fully included in RFC 5681, which specifies standard TCP congestion
> >    control.  "
> >
> > It's not clear what "fully included" means here. AFAICT, RFC5681
> > includes the essence of ABC explicitly.
> >
> > https://tools.ietf.org/html/rfc5681#section-3.1
> >
> > " During slow start, a TCP increments cwnd by at most SMSS bytes for
> >    each ACK received that cumulatively acknowledges new data.  Slow
> >    start ends when cwnd exceeds ssthresh (or, optionally, when it
> >    reaches it, as noted above) or when congestion is observed.  While
> >    traditionally TCP implementations have increased cwnd by precisely
> >    SMSS bytes upon receipt of an ACK covering new data, we RECOMMEND
> >    that TCP implementations increase cwnd, per:
> >
> >       cwnd += min (N, SMSS)                      (2)
> >
> >    where N is the number of previously unacknowledged bytes acknowledged
> >    in the incoming ACK.  This adjustment is part of Appropriate Byte
> >    Counting [RFC3465] and provides robustness against misbehaving
> >    receivers that may attempt to induce a sender to artificially inflate
> >    cwnd using a mechanism known as "ACK Division" [SCWA99].  ACK
> >    Division consists of a receiver sending multiple ACKs for a single
> >    TCP data segment, each acknowledging only a portion of its data.  A
> >    TCP that increments cwnd by SMSS for each such ACK will
> >    inappropriately inflate the amount of data injected into the network."
>
> Note that the problem pointed out is that, as stated in the RFC 5681 text
> above:
>
>    "During slow start, a TCP increments cwnd by at most SMSS bytes for
>     each ACK received that cumulatively acknowledges new data"
>
> It may be desirable to support, during slow start, a cwnd increase by more
> than SMSS per each ACK acknowledging new data.
I agree. We in fact implemented Linux to increase cwnd by the exact
amount sequence (in unit of packets) so slow-start increase function is
regardless of ack-thinning effect but purely based on amount
delivered. This has an important effect on
Wireless as the other ack talk mentions. It also helps in case of ACK
losses or reordering.




>
> In fact, couple of paragraphs later, RFC 5681 explicitly indicates the
> following:
>
>    "We note that [RFC3465] allows for cwnd increases of more than SMSS
>    bytes for incoming acknowledgments during slow start on an
>    experimental basis; however, such behavior is not allowed as part of
>    the standard."
>
> That is what we meant by "not fully included in RFC 5681" in our document.
>
> > As a side note, Linux stack has implemented the principle of abc in
> > its congestion control (including slow start)
>
> Thanks for the information.
>
> Yes, in fact, I second the WG members who stated on the chat, during the
> interim, that it is surprising that ABC remains as an experimental
> document.
>
> I realize that, at an implementation level, the problem pointed out above
> is being addressed in many platforms.
>
> > regarding delayed ACK harming slow-start or latency, I am not too sure
> > either:
> >
> > 1) The ACK is delayed but not the data
> >
> > 2) The delay of the ACK can indeed delay cwnd growth during slowstart.
> >
> > But the ACK is delayed because the packet is likely a sub-MSS data,
> > which is again likely caused by application limit (i.e. the
> > application has sent all its data like a HTTP response). In this case
> > the delay of the cwnd growth does not hurt the performance since the
> > application has no data left to send.
> >
> >    The only penalty occurs when the application has more data to send
> > but cwnd is still gated by the arrival of the delayed ACK. However, in
> > many request - response protocol, either the (non-delayed) ACKs
> > triggered by prior full-MSS data inflight or the delayed ACKs would
> > have returned to grow the cwnd to sending immediately.
> >
> > I can construct scenarios where delayed ACK significantly hurts
> > application. But I don't think the scenario is too common. I will
> > suggest more tests to quantify the effect of delayed ACK effect on
> > slow-start.
>
> Yes, I tend to agree.
>
> In fact, the two main problems due to Delayed ACKs that we identified so
> far in the area of slow start are:
>
> - The fact that ABC might not always be supported (due to its
> "Experimental" status).
>
> - Precluding sender behaviors such as chirping.
>
> Once again, thanks for your valuable feedback!
>
> Cheers,
>
> Carles
>
>
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
> >
>
>


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From: Mark Allman <mallman@icsi.berkeley.edu>
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-rto-consider-11.txt
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>         Title           : Requirements for Time-Based Loss Detection
>         Author          : Mark Allman
> 	Filename        : draft-ietf-tcpm-rto-consider-11.txt
> 	Pages           : 10
> 	Date            : 2020-04-30

I believe this version addresses the wording issue in requirement
(3) brought up during WGLC.  Per yesterday's meeting I think that
means we can forward the document on to the IESG.

allman

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Neal,

On 01/05/2020 13:19, Neal Cardwell wrote:
> On Fri, May 1, 2020 at 6:23 AM Carles Gomez Montenegro
> <carlesgo@entel.upc.edu> wrote:
>> Hi Neil,
>>
>> Thanks a lot for your comments!
>>
>> Please find below my inline responses:
>>
>>> On Wed, Apr 29, 2020 at 1:03 PM Scheffenegger, Richard <rs.ietf@gmx.at>
>>> wrote:
>>>> Eliciting an ACK under certain circumstances, for a timely continuation
>>>> of the data transmission / growth of the congestion window is a known
>>>> method to reduce network delay.
>>>>
>>>> E.g. Linux has been using the CWR flag for the purpose of sending out an
>>>> immediate ACK by the receiver, since there is an increased chance of a
>>>> very small cwnd when the sender had just reduced it's congestion window.
>>>>
>>>> https://lore.kernel.org/patchwork/patch/970486/
>>>>
>>>> and we also found latency improvements doing this when running
>>>> ECN-enabled sessions
>>>>
>>>> https://reviews.freebsd.org/D22670
>>> Yes, agreed.
>>>
>>> IMHO an explicit ACK-pull mechanism would be very nice for the cwnd<=1
>>> case.
>> (While this question is about the solution space, I'm anyway curious...)
>> In the context of datacenter networks, would you have any suggestion or
>> preference regarding the solutions that have been mentioned so far?
>>
>> Or perhaps any particular requirement for a potential solution?
> Among the solutions outlined in
>    https://tools.ietf.org/html/draft-gomez-tcpm-delack-suppr-reqs-00
> my first preference would be the AKP option, since that approach
> avoids burning a precious flag bit.

[BB] Eh? AKP does burn a flag bit (and you say it does below). Quoting 
the draft:

    TCP ACK Pull defines the AKP flag as bit number 6 of the 13th byte of
    the TCP header.


(BTW, Carles, in this quoted sentence, you ought to call it "the TCP ACK 
Pull /proposal/" otherwise sounds like it's an official part of TCP 
already.)

>
> The AKP option may be stripped by some middleboxes, but (a) as a
> performance optimization, that should be acceptable, and (b) for the
> datacenter case (where cwnd=1 is a motivating use case) this should
> not be a concern.
>
> IMHO a flag bit makes sense for a small signal that a sender might
> want to send frequently or at a high rate, but senders should not be
> trying to force immediate ACKs frequently.

[BB] For me, the problem with the AKP approach is that it's /only/ 
really good in environments where you've got a small window. When you 
have a large window and you can cope with a certain ACK ratio (say 
1:16), you would have to set AKP on every 16th packet, which doesn't 
really express what the sender wants (and if the receiver had just sent 
an ACK of its own accord just before it got one of these regular AKP 
requests, should it send another?). The sender in this case really wants 
to say "I don't particularly mind which packets you ACK, but there's no 
need to do it more often than 1:16".

I think there's some mileage in a TCP Option that aggregates a few 
behaviours together under one new TCP Option. I reckon 3 bits would be 
enough for a good delayed ACK option, but a whole TCP Option for that 
could be overkill. There was going to be a talk about a low latency TCP 
Option in netdev0x14 before it was postponed (sorry I don't remember who 
by). That might be a good excuse to collect together a few related 
functions. It could also improve the deployment incentives.


Bob



>
> best,
> neal

-- 
________________________________________________________________
Bob Briscoe                               http://bobbriscoe.net/


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    Neal,<br>
    <br>
    <div class="moz-cite-prefix">On 01/05/2020 13:19, Neal Cardwell
      wrote:<br>
    </div>
    <blockquote type="cite"
cite="mid:CADVnQy=wPUx62y7VNqjSPP+snKX4vVvK5q=qqYb1j+0nGrtezQ@mail.gmail.com">
      <pre class="moz-quote-pre" wrap="">On Fri, May 1, 2020 at 6:23 AM Carles Gomez Montenegro
<a class="moz-txt-link-rfc2396E" href="mailto:carlesgo@entel.upc.edu">&lt;carlesgo@entel.upc.edu&gt;</a> wrote:
</pre>
      <blockquote type="cite">
        <pre class="moz-quote-pre" wrap="">
Hi Neil,

Thanks a lot for your comments!

Please find below my inline responses:

</pre>
        <blockquote type="cite">
          <pre class="moz-quote-pre" wrap="">On Wed, Apr 29, 2020 at 1:03 PM Scheffenegger, Richard <a class="moz-txt-link-rfc2396E" href="mailto:rs.ietf@gmx.at">&lt;rs.ietf@gmx.at&gt;</a>
wrote:
</pre>
          <blockquote type="cite">
            <pre class="moz-quote-pre" wrap="">Eliciting an ACK under certain circumstances, for a timely continuation
of the data transmission / growth of the congestion window is a known
method to reduce network delay.

E.g. Linux has been using the CWR flag for the purpose of sending out an
immediate ACK by the receiver, since there is an increased chance of a
very small cwnd when the sender had just reduced it's congestion window.

<a class="moz-txt-link-freetext" href="https://lore.kernel.org/patchwork/patch/970486/">https://lore.kernel.org/patchwork/patch/970486/</a>

and we also found latency improvements doing this when running
ECN-enabled sessions

<a class="moz-txt-link-freetext" href="https://reviews.freebsd.org/D22670">https://reviews.freebsd.org/D22670</a>
</pre>
          </blockquote>
          <pre class="moz-quote-pre" wrap="">
Yes, agreed.

IMHO an explicit ACK-pull mechanism would be very nice for the cwnd&lt;=1
case.
</pre>
        </blockquote>
        <pre class="moz-quote-pre" wrap="">
(While this question is about the solution space, I'm anyway curious...)
In the context of datacenter networks, would you have any suggestion or
preference regarding the solutions that have been mentioned so far?

Or perhaps any particular requirement for a potential solution?
</pre>
      </blockquote>
      <pre class="moz-quote-pre" wrap="">
Among the solutions outlined in
  <a class="moz-txt-link-freetext" href="https://tools.ietf.org/html/draft-gomez-tcpm-delack-suppr-reqs-00">https://tools.ietf.org/html/draft-gomez-tcpm-delack-suppr-reqs-00</a>
my first preference would be the AKP option, since that approach
avoids burning a precious flag bit.</pre>
    </blockquote>
    <br>
    [BB] Eh? AKP does burn a flag bit (and you say it does below).
    Quoting the draft:<br>
    <pre class="newpage">   TCP ACK Pull defines the AKP flag as bit number 6 of the 13th byte of
   the TCP header.</pre>
    <br>
    (BTW, Carles, in this quoted sentence, you ought to call it "the TCP
    ACK Pull /proposal/" otherwise sounds like it's an official part of
    TCP already.)<br>
    <br>
    <blockquote type="cite"
cite="mid:CADVnQy=wPUx62y7VNqjSPP+snKX4vVvK5q=qqYb1j+0nGrtezQ@mail.gmail.com">
      <pre class="moz-quote-pre" wrap="">

The AKP option may be stripped by some middleboxes, but (a) as a
performance optimization, that should be acceptable, and (b) for the
datacenter case (where cwnd=1 is a motivating use case) this should
not be a concern.

IMHO a flag bit makes sense for a small signal that a sender might
want to send frequently or at a high rate, but senders should not be
trying to force immediate ACKs frequently.</pre>
    </blockquote>
    <br>
    [BB] For me, the problem with the AKP approach is that it's /only/
    really good in environments where you've got a small window. When
    you have a large window and you can cope with a certain ACK ratio
    (say 1:16), you would have to set AKP on every 16th packet, which
    doesn't really express what the sender wants (and if the receiver
    had just sent an ACK of its own accord just before it got one of
    these regular AKP requests, should it send another?). The sender in
    this case really wants to say "I don't particularly mind which
    packets you ACK, but there's no need to do it more often than 1:16".<br>
    <br>
    I think there's some mileage in a TCP Option that aggregates a few
    behaviours together under one new TCP Option. I reckon 3 bits would
    be enough for a good delayed ACK option, but a whole TCP Option for
    that could be overkill. There was going to be a talk about a low
    latency TCP Option in netdev0x14 before it was postponed (sorry I
    don't remember who by). That might be a good excuse to collect
    together a few related functions. It could also improve the
    deployment incentives.<br>
    <br>
    <br>
    Bob<br>
    <br>
    <br>
    <br>
    <blockquote type="cite"
cite="mid:CADVnQy=wPUx62y7VNqjSPP+snKX4vVvK5q=qqYb1j+0nGrtezQ@mail.gmail.com">
      <pre class="moz-quote-pre" wrap="">

best,
neal
</pre>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
________________________________________________________________
Bob Briscoe                               <a class="moz-txt-link-freetext" href="http://bobbriscoe.net/">http://bobbriscoe.net/</a></pre>
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-rto-consider-11.txt
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Great! I'll start my AD review while Yoshi starts his shepherd writeup.

Thanks Mark!

On Fri, May 1, 2020 at 10:56 AM Mark Allman <mallman@icsi.berkeley.edu>
wrote:

>
> >         Title           : Requirements for Time-Based Loss Detection
> >         Author          : Mark Allman
> >       Filename        : draft-ietf-tcpm-rto-consider-11.txt
> >       Pages           : 10
> >       Date            : 2020-04-30
>
> I believe this version addresses the wording issue in requirement
> (3) brought up during WGLC.  Per yesterday's meeting I think that
> means we can forward the document on to the IESG.
>
> allman
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<div dir=3D"ltr"><div>Great! I&#39;ll start my AD review while Yoshi starts=
 his shepherd writeup.</div><div><br></div><div>Thanks Mark!<br></div></div=
><br><div class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Fr=
i, May 1, 2020 at 10:56 AM Mark Allman &lt;<a href=3D"mailto:mallman@icsi.b=
erkeley.edu">mallman@icsi.berkeley.edu</a>&gt; wrote:<br></div><blockquote =
class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px sol=
id rgb(204,204,204);padding-left:1ex"><br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0: Requirements for Time-Based Loss Detection<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0Author=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 : Mark Allman<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-=
ietf-tcpm-rto-consider-11.txt<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0: 10<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 : 2020-04-30<br>
<br>
I believe this version addresses the wording issue in requirement<br>
(3) brought up during WGLC.=C2=A0 Per yesterday&#39;s meeting I think that<=
br>
means we can forward the document on to the IESG.<br>
<br>
allman<br>
_______________________________________________<br>
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<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
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Date: Sat, 2 May 2020 10:34:15 +0200
From: "Carles Gomez Montenegro" <carlesgo@entel.upc.edu>
To: "Yuchung Cheng" <ycheng@google.com>
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Subject: Re: [tcpm] question regarding draft-gomez-tcpm-delack-suppr-reqs
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Hi Yuchung,

<snip>

>> Note that the problem pointed out is that, as stated in the RFC 5681
>> text
>> above:
>>
>>    "During slow start, a TCP increments cwnd by at most SMSS bytes for
>>     each ACK received that cumulatively acknowledges new data"
>>
>> It may be desirable to support, during slow start, a cwnd increase by
>> more
>> than SMSS per each ACK acknowledging new data.
> I agree. We in fact implemented Linux to increase cwnd by the exact
> amount sequence (in unit of packets) so slow-start increase function is
> regardless of ack-thinning effect but purely based on amount
> delivered. This has an important effect on
> Wireless as the other ack talk mentions. It also helps in case of ACK
> losses or reordering.

Agreed!

By the way, in the next draft update, we will add that, despite its
"Experimental" status, ABC is supported in many implementations.

<snip>

Thanks,

Carles


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Date: Sat, 2 May 2020 10:49:02 +0200
From: "Carles Gomez Montenegro" <carlesgo@entel.upc.edu>
To: "Neal Cardwell" <ncardwell@google.com>
Cc: "Scheffenegger, Richard" <rs.ietf@gmx.at>, "tcpm@ietf.org" <tcpm@ietf.org>, "Gorry Fairhurst" <gorry@erg.abdn.ac.uk>, "Bob Briscoe" <ietf@bobbriscoe.net>, jon.crowcroft@cl.cam.ac.uk
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Subject: Re: [tcpm] On Sender Control of Delayed Acks in TCP
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Hi Neal,

>> > On Wed, Apr 29, 2020 at 1:03 PM Scheffenegger, Richard
>> <rs.ietf@gmx.at>
>> > wrote:
>> >> Eliciting an ACK under certain circumstances, for a timely
>> continuation
>> >> of the data transmission / growth of the congestion window is a known
>> >> method to reduce network delay.
>> >>
>> >> E.g. Linux has been using the CWR flag for the purpose of sending out
>> an
>> >> immediate ACK by the receiver, since there is an increased chance of
>> a
>> >> very small cwnd when the sender had just reduced it's congestion
>> window.
>> >>
>> >> https://lore.kernel.org/patchwork/patch/970486/
>> >>
>> >> and we also found latency improvements doing this when running
>> >> ECN-enabled sessions
>> >>
>> >> https://reviews.freebsd.org/D22670
>> >
>> > Yes, agreed.
>> >
>> > IMHO an explicit ACK-pull mechanism would be very nice for the cwnd<=1
>> > case.
>>
>> (While this question is about the solution space, I'm anyway curious...)
>> In the context of datacenter networks, would you have any suggestion or
>> preference regarding the solutions that have been mentioned so far?
>>
>> Or perhaps any particular requirement for a potential solution?
>
> Among the solutions outlined in
>   https://tools.ietf.org/html/draft-gomez-tcpm-delack-suppr-reqs-00
> my first preference would be the AKP option, since that approach
> avoids burning a precious flag bit.
>
> The AKP option may be stripped by some middleboxes, but (a) as a
> performance optimization, that should be acceptable, and (b) for the
> datacenter case (where cwnd=1 is a motivating use case) this should
> not be a concern.
>
> IMHO a flag bit makes sense for a small signal that a sender might
> want to send frequently or at a high rate, but senders should not be
> trying to force immediate ACKs frequently.

Thanks a lot for the very useful feedback.

In the next draft update, we will expand the discussion of potential
solutions accordingly.

Cheers,

Carles


> best,
> neal


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Date: Sat, 2 May 2020 11:39:30 +0200
From: "Carles Gomez Montenegro" <carlesgo@entel.upc.edu>
To: "Bob Briscoe" <ietf@bobbriscoe.net>
Cc: "Neal Cardwell" <ncardwell@google.com>, "Scheffenegger, Richard" <rs.ietf@gmx.at>, "tcpm@ietf.org" <tcpm@ietf.org>, "Gorry Fairhurst" <gorry@erg.abdn.ac.uk>, jon.crowcroft@cl.cam.ac.uk
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Hi Bob,

(Chiming in...)

> Neal,
>
> On 01/05/2020 13:19, Neal Cardwell wrote:
>> On Fri, May 1, 2020 at 6:23 AM Carles Gomez Montenegro
>> <carlesgo@entel.upc.edu> wrote:
>>> Hi Neil,
>>>
>>> Thanks a lot for your comments!
>>>
>>> Please find below my inline responses:
>>>
>>>> On Wed, Apr 29, 2020 at 1:03 PM Scheffenegger, Richard
>>>> <rs.ietf@gmx.at>
>>>> wrote:
>>>>> Eliciting an ACK under certain circumstances, for a timely
>>>>> continuation
>>>>> of the data transmission / growth of the congestion window is a known
>>>>> method to reduce network delay.
>>>>>
>>>>> E.g. Linux has been using the CWR flag for the purpose of sending out
>>>>> an
>>>>> immediate ACK by the receiver, since there is an increased chance of
>>>>> a
>>>>> very small cwnd when the sender had just reduced it's congestion
>>>>> window.
>>>>>
>>>>> https://lore.kernel.org/patchwork/patch/970486/
>>>>>
>>>>> and we also found latency improvements doing this when running
>>>>> ECN-enabled sessions
>>>>>
>>>>> https://reviews.freebsd.org/D22670
>>>> Yes, agreed.
>>>>
>>>> IMHO an explicit ACK-pull mechanism would be very nice for the cwnd<=1
>>>> case.
>>> (While this question is about the solution space, I'm anyway
>>> curious...)
>>> In the context of datacenter networks, would you have any suggestion or
>>> preference regarding the solutions that have been mentioned so far?
>>>
>>> Or perhaps any particular requirement for a potential solution?
>> Among the solutions outlined in
>>    https://tools.ietf.org/html/draft-gomez-tcpm-delack-suppr-reqs-00
>> my first preference would be the AKP option, since that approach
>> avoids burning a precious flag bit.
>
> [BB] Eh? AKP does burn a flag bit (and you say it does below). Quoting
> the draft:
>
>     TCP ACK Pull defines the AKP flag as bit number 6 of the 13th byte of
>     the TCP header.

[CG] I think that Neal referred to section 5.4 of the requirements draft  
 ("A new 'ACK Pull' TCP option").

> (BTW, Carles, in this quoted sentence, you ought to call it "the TCP ACK
> Pull /proposal/" otherwise sounds like it's an official part of TCP
> already.)

[CG] Well noted, thanks!  (Nevertheless, if that other document went
ahead, I guess that "proposal" would need to be eventually removed.)

>> The AKP option may be stripped by some middleboxes, but (a) as a
>> performance optimization, that should be acceptable, and (b) for the
>> datacenter case (where cwnd=1 is a motivating use case) this should
>> not be a concern.
>>
>> IMHO a flag bit makes sense for a small signal that a sender might
>> want to send frequently or at a high rate, but senders should not be
>> trying to force immediate ACKs frequently.
>
> [BB] For me, the problem with the AKP approach is that it's /only/
> really good in environments where you've got a small window. When you
> have a large window and you can cope with a certain ACK ratio (say
> 1:16), you would have to set AKP on every 16th packet, which doesn't
> really express what the sender wants (and if the receiver had just sent
> an ACK of its own accord just before it got one of these regular AKP
> requests, should it send another?). The sender in this case really wants
> to say "I don't particularly mind which packets you ACK, but there's no
> need to do it more often than 1:16".
>
> I think there's some mileage in a TCP Option that aggregates a few
> behaviours together under one new TCP Option. I reckon 3 bits would be
> enough for a good delayed ACK option, but a whole TCP Option for that
> could be overkill. There was going to be a talk about a low latency TCP
> Option in netdev0x14 before it was postponed (sorry I don't remember who
> by). That might be a good excuse to collect together a few related
> functions. It could also improve the deployment incentives.

[CG] Thanks for the comments!

[CG] It seems that there are two different areas where avoiding the
typical Delayed ACKs approach may be of particular interest: i) scenarios
where very small window sizes are used (cwnd=1), and ii) scenarios where
large window sizes, and high ACK ratios, can be used. We will reflect
these considerations in the next draft update.

Cheers,

Carles


> Bob
>
>
>
>>
>> best,
>> neal
>
> --
> ________________________________________________________________
> Bob Briscoe                               http://bobbriscoe.net/


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From: Neal Cardwell <ncardwell@google.com>
Date: Sat, 2 May 2020 10:10:49 -0400
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To: Carles Gomez Montenegro <carlesgo@entel.upc.edu>
Cc: Bob Briscoe <ietf@bobbriscoe.net>, "Scheffenegger, Richard" <rs.ietf@gmx.at>, "tcpm@ietf.org" <tcpm@ietf.org>,  Gorry Fairhurst <gorry@erg.abdn.ac.uk>, jon.crowcroft@cl.cam.ac.uk
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Subject: Re: [tcpm] On Sender Control of Delayed Acks in TCP
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On Sat, May 2, 2020 at 5:40 AM Carles Gomez Montenegro
<carlesgo@entel.upc.edu> wrote:
>
> Hi Bob,
>
> (Chiming in...)
>
> > Neal,
> >
> > On 01/05/2020 13:19, Neal Cardwell wrote:
> >> On Fri, May 1, 2020 at 6:23 AM Carles Gomez Montenegro
> >> <carlesgo@entel.upc.edu> wrote:
> >>> Hi Neil,
> >>>
> >>> Thanks a lot for your comments!
> >>>
> >>> Please find below my inline responses:
> >>>
> >>>> On Wed, Apr 29, 2020 at 1:03 PM Scheffenegger, Richard
> >>>> <rs.ietf@gmx.at>
> >>>> wrote:
> >>>>> Eliciting an ACK under certain circumstances, for a timely
> >>>>> continuation
> >>>>> of the data transmission / growth of the congestion window is a known
> >>>>> method to reduce network delay.
> >>>>>
> >>>>> E.g. Linux has been using the CWR flag for the purpose of sending out
> >>>>> an
> >>>>> immediate ACK by the receiver, since there is an increased chance of
> >>>>> a
> >>>>> very small cwnd when the sender had just reduced it's congestion
> >>>>> window.
> >>>>>
> >>>>> https://lore.kernel.org/patchwork/patch/970486/
> >>>>>
> >>>>> and we also found latency improvements doing this when running
> >>>>> ECN-enabled sessions
> >>>>>
> >>>>> https://reviews.freebsd.org/D22670
> >>>> Yes, agreed.
> >>>>
> >>>> IMHO an explicit ACK-pull mechanism would be very nice for the cwnd<=1
> >>>> case.
> >>> (While this question is about the solution space, I'm anyway
> >>> curious...)
> >>> In the context of datacenter networks, would you have any suggestion or
> >>> preference regarding the solutions that have been mentioned so far?
> >>>
> >>> Or perhaps any particular requirement for a potential solution?
> >> Among the solutions outlined in
> >>    https://tools.ietf.org/html/draft-gomez-tcpm-delack-suppr-reqs-00
> >> my first preference would be the AKP option, since that approach
> >> avoids burning a precious flag bit.
> >
> > [BB] Eh? AKP does burn a flag bit (and you say it does below). Quoting
> > the draft:
> >
> >     TCP ACK Pull defines the AKP flag as bit number 6 of the 13th byte of
> >     the TCP header.
>
> [CG] I think that Neal referred to section 5.4 of the requirements draft
>  ("A new 'ACK Pull' TCP option").

Yes, as Carles says, when I mentioned "AKP option" I was referring to
the alternative discussed in section 5.4, "A new 'ACK Pull' TCP
option", which does not burn a flag bit:

  https://tools.ietf.org/html/draft-gomez-tcpm-delack-suppr-reqs-00#section-5.4

When I mentioned "flag bit" I was referring to section 5.3, "TCP ACK
Pull (AKP) flag", which does burn a flag bit:

  https://tools.ietf.org/html/draft-gomez-tcpm-delack-suppr-reqs-00#section-5.3

> > (BTW, Carles, in this quoted sentence, you ought to call it "the TCP ACK
> > Pull /proposal/" otherwise sounds like it's an official part of TCP
> > already.)
>
> [CG] Well noted, thanks!  (Nevertheless, if that other document went
> ahead, I guess that "proposal" would need to be eventually removed.)
>
> >> The AKP option may be stripped by some middleboxes, but (a) as a
> >> performance optimization, that should be acceptable, and (b) for the
> >> datacenter case (where cwnd=1 is a motivating use case) this should
> >> not be a concern.
> >>
> >> IMHO a flag bit makes sense for a small signal that a sender might
> >> want to send frequently or at a high rate, but senders should not be
> >> trying to force immediate ACKs frequently.
> >
> > [BB] For me, the problem with the AKP approach is that it's /only/
> > really good in environments where you've got a small window. When you
> > have a large window and you can cope with a certain ACK ratio (say
> > 1:16), you would have to set AKP on every 16th packet, which doesn't
> > really express what the sender wants (and if the receiver had just sent
> > an ACK of its own accord just before it got one of these regular AKP
> > requests, should it send another?). The sender in this case really wants
> > to say "I don't particularly mind which packets you ACK, but there's no
> > need to do it more often than 1:16".
> >
> > I think there's some mileage in a TCP Option that aggregates a few
> > behaviours together under one new TCP Option. I reckon 3 bits would be
> > enough for a good delayed ACK option, but a whole TCP Option for that
> > could be overkill. There was going to be a talk about a low latency TCP
> > Option in netdev0x14 before it was postponed (sorry I don't remember who
> > by). That might be a good excuse to collect together a few related
> > functions. It could also improve the deployment incentives.
>
> [CG] Thanks for the comments!
>
> [CG] It seems that there are two different areas where avoiding the
> typical Delayed ACKs approach may be of particular interest: i) scenarios
> where very small window sizes are used (cwnd=1), and ii) scenarios where
> large window sizes, and high ACK ratios, can be used. We will reflect
> these considerations in the next draft update.

Yes, I wholeheartedly agree with Bob's point that it's also extremely
useful for TCP senders to be able to say things like: "I don't
particularly mind which packets you ACK, but there's no need to do it
more often than 1:16". QUIC has a mechanism like this, and it seems to
allow significant CPU savings, even allowing QUIC to run faster than
TCP in some cases, AFAICT due to the reduced overhead for handling
ACKs:
  https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency
  https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00

So I agree with the usefulness of the two-use-case taxonomy Carles has outlined:

1: "please ACK now"; because sender's cwnd is low

2: "ACK at frequency K"; because sender's cwnd is high

It seems those could both be addressed cleanly with one or two TCP
option kinds. A single TCP option kind might suffice if the ACK
frequency of 0 or 1 is taken to mean "please ACK now".

best,
neal


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Date: Sat, 2 May 2020 16:26:21 +0200
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To: Neal Cardwell <ncardwell=40google.com@dmarc.ietf.org>
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Subject: Re: [tcpm] On Sender Control of Delayed Acks in TCP
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> On 2. May 2020, at 16:10, Neal Cardwell =
<ncardwell=3D40google.com@dmarc.ietf.org> wrote:
>=20
> On Sat, May 2, 2020 at 5:40 AM Carles Gomez Montenegro
> <carlesgo@entel.upc.edu> wrote:
>>=20
>> Hi Bob,
>>=20
>> (Chiming in...)
>>=20
>>> Neal,
>>>=20
>>> On 01/05/2020 13:19, Neal Cardwell wrote:
>>>> On Fri, May 1, 2020 at 6:23 AM Carles Gomez Montenegro
>>>> <carlesgo@entel.upc.edu> wrote:
>>>>> Hi Neil,
>>>>>=20
>>>>> Thanks a lot for your comments!
>>>>>=20
>>>>> Please find below my inline responses:
>>>>>=20
>>>>>> On Wed, Apr 29, 2020 at 1:03 PM Scheffenegger, Richard
>>>>>> <rs.ietf@gmx.at>
>>>>>> wrote:
>>>>>>> Eliciting an ACK under certain circumstances, for a timely
>>>>>>> continuation
>>>>>>> of the data transmission / growth of the congestion window is a =
known
>>>>>>> method to reduce network delay.
>>>>>>>=20
>>>>>>> E.g. Linux has been using the CWR flag for the purpose of =
sending out
>>>>>>> an
>>>>>>> immediate ACK by the receiver, since there is an increased =
chance of
>>>>>>> a
>>>>>>> very small cwnd when the sender had just reduced it's congestion
>>>>>>> window.
>>>>>>>=20
>>>>>>> https://lore.kernel.org/patchwork/patch/970486/
>>>>>>>=20
>>>>>>> and we also found latency improvements doing this when running
>>>>>>> ECN-enabled sessions
>>>>>>>=20
>>>>>>> https://reviews.freebsd.org/D22670
>>>>>> Yes, agreed.
>>>>>>=20
>>>>>> IMHO an explicit ACK-pull mechanism would be very nice for the =
cwnd<=3D1
>>>>>> case.
>>>>> (While this question is about the solution space, I'm anyway
>>>>> curious...)
>>>>> In the context of datacenter networks, would you have any =
suggestion or
>>>>> preference regarding the solutions that have been mentioned so =
far?
>>>>>=20
>>>>> Or perhaps any particular requirement for a potential solution?
>>>> Among the solutions outlined in
>>>>   https://tools.ietf.org/html/draft-gomez-tcpm-delack-suppr-reqs-00
>>>> my first preference would be the AKP option, since that approach
>>>> avoids burning a precious flag bit.
>>>=20
>>> [BB] Eh? AKP does burn a flag bit (and you say it does below). =
Quoting
>>> the draft:
>>>=20
>>>    TCP ACK Pull defines the AKP flag as bit number 6 of the 13th =
byte of
>>>    the TCP header.
>>=20
>> [CG] I think that Neal referred to section 5.4 of the requirements =
draft
>> ("A new 'ACK Pull' TCP option").
>=20
> Yes, as Carles says, when I mentioned "AKP option" I was referring to
> the alternative discussed in section 5.4, "A new 'ACK Pull' TCP
> option", which does not burn a flag bit:
>=20
>  =
https://tools.ietf.org/html/draft-gomez-tcpm-delack-suppr-reqs-00#section-=
5.4
>=20
> When I mentioned "flag bit" I was referring to section 5.3, "TCP ACK
> Pull (AKP) flag", which does burn a flag bit:
>=20
>  =
https://tools.ietf.org/html/draft-gomez-tcpm-delack-suppr-reqs-00#section-=
5.3
>=20
>>> (BTW, Carles, in this quoted sentence, you ought to call it "the TCP =
ACK
>>> Pull /proposal/" otherwise sounds like it's an official part of TCP
>>> already.)
>>=20
>> [CG] Well noted, thanks!  (Nevertheless, if that other document went
>> ahead, I guess that "proposal" would need to be eventually removed.)
>>=20
>>>> The AKP option may be stripped by some middleboxes, but (a) as a
>>>> performance optimization, that should be acceptable, and (b) for =
the
>>>> datacenter case (where cwnd=3D1 is a motivating use case) this =
should
>>>> not be a concern.
>>>>=20
>>>> IMHO a flag bit makes sense for a small signal that a sender might
>>>> want to send frequently or at a high rate, but senders should not =
be
>>>> trying to force immediate ACKs frequently.
>>>=20
>>> [BB] For me, the problem with the AKP approach is that it's /only/
>>> really good in environments where you've got a small window. When =
you
>>> have a large window and you can cope with a certain ACK ratio (say
>>> 1:16), you would have to set AKP on every 16th packet, which doesn't
>>> really express what the sender wants (and if the receiver had just =
sent
>>> an ACK of its own accord just before it got one of these regular AKP
>>> requests, should it send another?). The sender in this case really =
wants
>>> to say "I don't particularly mind which packets you ACK, but there's =
no
>>> need to do it more often than 1:16".
>>>=20
>>> I think there's some mileage in a TCP Option that aggregates a few
>>> behaviours together under one new TCP Option. I reckon 3 bits would =
be
>>> enough for a good delayed ACK option, but a whole TCP Option for =
that
>>> could be overkill. There was going to be a talk about a low latency =
TCP
>>> Option in netdev0x14 before it was postponed (sorry I don't remember =
who
>>> by). That might be a good excuse to collect together a few related
>>> functions. It could also improve the deployment incentives.
>>=20
>> [CG] Thanks for the comments!
>>=20
>> [CG] It seems that there are two different areas where avoiding the
>> typical Delayed ACKs approach may be of particular interest: i) =
scenarios
>> where very small window sizes are used (cwnd=3D1), and ii) scenarios =
where
>> large window sizes, and high ACK ratios, can be used. We will reflect
>> these considerations in the next draft update.
>=20
> Yes, I wholeheartedly agree with Bob's point that it's also extremely
> useful for TCP senders to be able to say things like: "I don't
> particularly mind which packets you ACK, but there's no need to do it
> more often than 1:16". QUIC has a mechanism like this, and it seems to
> allow significant CPU savings, even allowing QUIC to run faster than
> TCP in some cases, AFAICT due to the reduced overhead for handling
> ACKs:
>  =
https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency=

>  https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00
>=20
> So I agree with the usefulness of the two-use-case taxonomy Carles has =
outlined:
>=20
> 1: "please ACK now"; because sender's cwnd is low
>=20
> 2: "ACK at frequency K"; because sender's cwnd is high
>=20
> It seems those could both be addressed cleanly with one or two TCP
> option kinds. A single TCP option kind might suffice if the ACK
> frequency of 0 or 1 is taken to mean "please ACK now".
I might be useful if the sender can trigger an ACK for this particular =
packet,
but leave the rate unchanged. And change the rate from now on.
If using an option, you could encode "send an ack for this packet, but =
don't
change the ack rate" by using 0, and encode "change the ack rate to 1 =
for n"
by using n, n > 0.

Best regards
Michael
>=20
> best,
> neal
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: Neal Cardwell <ncardwell@google.com>
Date: Sat, 2 May 2020 10:34:40 -0400
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To: Michael Tuexen <michael.tuexen@lurchi.franken.de>
Cc: Carles Gomez Montenegro <carlesgo@entel.upc.edu>, jon.crowcroft@cl.cam.ac.uk, "tcpm@ietf.org" <tcpm@ietf.org>
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Subject: Re: [tcpm] On Sender Control of Delayed Acks in TCP
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On Sat, May 2, 2020 at 10:26 AM Michael Tuexen
<michael.tuexen@lurchi.franken.de> wrote:
>
> > On 2. May 2020, at 16:10, Neal Cardwell <ncardwell=40google.com@dmarc.ietf.org> wrote:
> >
> > On Sat, May 2, 2020 at 5:40 AM Carles Gomez Montenegro
...
> >> [CG] It seems that there are two different areas where avoiding the
> >> typical Delayed ACKs approach may be of particular interest: i) scenarios
> >> where very small window sizes are used (cwnd=1), and ii) scenarios where
> >> large window sizes, and high ACK ratios, can be used. We will reflect
> >> these considerations in the next draft update.
> >
> > Yes, I wholeheartedly agree with Bob's point that it's also extremely
> > useful for TCP senders to be able to say things like: "I don't
> > particularly mind which packets you ACK, but there's no need to do it
> > more often than 1:16". QUIC has a mechanism like this, and it seems to
> > allow significant CPU savings, even allowing QUIC to run faster than
> > TCP in some cases, AFAICT due to the reduced overhead for handling
> > ACKs:
> >  https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency
> >  https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00
> >
> > So I agree with the usefulness of the two-use-case taxonomy Carles has outlined:
> >
> > 1: "please ACK now"; because sender's cwnd is low
> >
> > 2: "ACK at frequency K"; because sender's cwnd is high
> >
> > It seems those could both be addressed cleanly with one or two TCP
> > option kinds. A single TCP option kind might suffice if the ACK
> > frequency of 0 or 1 is taken to mean "please ACK now".
> I might be useful if the sender can trigger an ACK for this particular packet,
> but leave the rate unchanged. And change the rate from now on.
> If using an option, you could encode "send an ack for this packet, but don't
> change the ack rate" by using 0, and encode "change the ack rate to 1 for n"
> by using n, n > 0.

Yes, that sounds like a very nice idea.

best,
neal


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From: Matt Mathis <mattmathis@google.com>
Date: Sat, 2 May 2020 07:55:25 -0700
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To: Neal Cardwell <ncardwell=40google.com@dmarc.ietf.org>
Cc: Michael Tuexen <michael.tuexen@lurchi.franken.de>,  Jon Crowcroft <jon.crowcroft@cl.cam.ac.uk>, "tcpm@ietf.org" <tcpm@ietf.org>
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Subject: Re: [tcpm] On Sender Control of Delayed Acks in TCP
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Another probably small opportunity would be to make the RTT estimators less
noisy, by having the receiver always ACK the first (or last) TSO segment.
 The RTT is currently fuzzed by both del ACK and TCP strides.

Thanks,
--MM--
The best way to predict the future is to create it.  - Alan Kay

We must not tolerate intolerance;
       however our response must be carefully measured:
            too strong would be hypocritical and risks spiraling out of
control;
            too weak risks being mistaken for tacit approval.


On Sat, May 2, 2020 at 7:35 AM Neal Cardwell <ncardwell=
40google.com@dmarc.ietf.org> wrote:

> On Sat, May 2, 2020 at 10:26 AM Michael Tuexen
> <michael.tuexen@lurchi.franken.de> wrote:
> >
> > > On 2. May 2020, at 16:10, Neal Cardwell <ncardwell=
> 40google.com@dmarc.ietf.org> wrote:
> > >
> > > On Sat, May 2, 2020 at 5:40 AM Carles Gomez Montenegro
> ....
> > >> [CG] It seems that there are two different areas where avoiding the
> > >> typical Delayed ACKs approach may be of particular interest: i)
> scenarios
> > >> where very small window sizes are used (cwnd=1), and ii) scenarios
> where
> > >> large window sizes, and high ACK ratios, can be used. We will reflect
> > >> these considerations in the next draft update.
> > >
> > > Yes, I wholeheartedly agree with Bob's point that it's also extremely
> > > useful for TCP senders to be able to say things like: "I don't
> > > particularly mind which packets you ACK, but there's no need to do it
> > > more often than 1:16". QUIC has a mechanism like this, and it seems to
> > > allow significant CPU savings, even allowing QUIC to run faster than
> > > TCP in some cases, AFAICT due to the reduced overhead for handling
> > > ACKs:
> > >
> https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency
> > >  https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00
> > >
> > > So I agree with the usefulness of the two-use-case taxonomy Carles has
> outlined:
> > >
> > > 1: "please ACK now"; because sender's cwnd is low
> > >
> > > 2: "ACK at frequency K"; because sender's cwnd is high
> > >
> > > It seems those could both be addressed cleanly with one or two TCP
> > > option kinds. A single TCP option kind might suffice if the ACK
> > > frequency of 0 or 1 is taken to mean "please ACK now".
> > I might be useful if the sender can trigger an ACK for this particular
> packet,
> > but leave the rate unchanged. And change the rate from now on.
> > If using an option, you could encode "send an ack for this packet, but
> don't
> > change the ack rate" by using 0, and encode "change the ack rate to 1
> for n"
> > by using n, n > 0.
>
> Yes, that sounds like a very nice idea.
>
> best,
> neal
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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<div dir=3D"ltr">Another probably small opportunity would be to make the RT=
T estimators less noisy, by having the receiver always ACK the first (or la=
st) TSO segment.=C2=A0 =C2=A0The RTT is currently fuzzed by both del ACK an=
d TCP strides.<div><br clear=3D"all"><div><div dir=3D"ltr" class=3D"gmail_s=
ignature" data-smartmail=3D"gmail_signature"><div dir=3D"ltr"><div><div dir=
=3D"ltr"><div><div dir=3D"ltr"><div>Thanks,</div>--MM--<br>The best way to =
predict the future is to create it. =C2=A0- Alan Kay<br><br>We must not tol=
erate intolerance;</div><div dir=3D"ltr">=C2=A0 =C2=A0 =C2=A0 =C2=A0however=
 our response must be carefully measured:=C2=A0</div><div>=C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 too strong would be hypocritical and risks spirali=
ng out of control;</div><div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 too =
weak risks being mistaken for tacit approval.</div></div></div></div></div>=
</div></div><br></div></div><br><div class=3D"gmail_quote"><div dir=3D"ltr"=
 class=3D"gmail_attr">On Sat, May 2, 2020 at 7:35 AM Neal Cardwell &lt;ncar=
dwell=3D<a href=3D"mailto:40google.com@dmarc.ietf.org">40google.com@dmarc.i=
etf.org</a>&gt; wrote:<br></div><blockquote class=3D"gmail_quote" style=3D"=
margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-lef=
t:1ex">On Sat, May 2, 2020 at 10:26 AM Michael Tuexen<br>
&lt;<a href=3D"mailto:michael.tuexen@lurchi.franken.de" target=3D"_blank">m=
ichael.tuexen@lurchi.franken.de</a>&gt; wrote:<br>
&gt;<br>
&gt; &gt; On 2. May 2020, at 16:10, Neal Cardwell &lt;ncardwell=3D<a href=
=3D"mailto:40google.com@dmarc.ietf.org" target=3D"_blank">40google.com@dmar=
c.ietf.org</a>&gt; wrote:<br>
&gt; &gt;<br>
&gt; &gt; On Sat, May 2, 2020 at 5:40 AM Carles Gomez Montenegro<br>
....<br>
&gt; &gt;&gt; [CG] It seems that there are two different areas where avoidi=
ng the<br>
&gt; &gt;&gt; typical Delayed ACKs approach may be of particular interest: =
i) scenarios<br>
&gt; &gt;&gt; where very small window sizes are used (cwnd=3D1), and ii) sc=
enarios where<br>
&gt; &gt;&gt; large window sizes, and high ACK ratios, can be used. We will=
 reflect<br>
&gt; &gt;&gt; these considerations in the next draft update.<br>
&gt; &gt;<br>
&gt; &gt; Yes, I wholeheartedly agree with Bob&#39;s point that it&#39;s al=
so extremely<br>
&gt; &gt; useful for TCP senders to be able to say things like: &quot;I don=
&#39;t<br>
&gt; &gt; particularly mind which packets you ACK, but there&#39;s no need =
to do it<br>
&gt; &gt; more often than 1:16&quot;. QUIC has a mechanism like this, and i=
t seems to<br>
&gt; &gt; allow significant CPU savings, even allowing QUIC to run faster t=
han<br>
&gt; &gt; TCP in some cases, AFAICT due to the reduced overhead for handlin=
g<br>
&gt; &gt; ACKs:<br>
&gt; &gt;=C2=A0 <a href=3D"https://www.fastly.com/blog/measuring-quic-vs-tc=
p-computational-efficiency" rel=3D"noreferrer" target=3D"_blank">https://ww=
w.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency</a><br>
&gt; &gt;=C2=A0 <a href=3D"https://tools.ietf.org/html/draft-iyengar-quic-d=
elayed-ack-00" rel=3D"noreferrer" target=3D"_blank">https://tools.ietf.org/=
html/draft-iyengar-quic-delayed-ack-00</a><br>
&gt; &gt;<br>
&gt; &gt; So I agree with the usefulness of the two-use-case taxonomy Carle=
s has outlined:<br>
&gt; &gt;<br>
&gt; &gt; 1: &quot;please ACK now&quot;; because sender&#39;s cwnd is low<b=
r>
&gt; &gt;<br>
&gt; &gt; 2: &quot;ACK at frequency K&quot;; because sender&#39;s cwnd is h=
igh<br>
&gt; &gt;<br>
&gt; &gt; It seems those could both be addressed cleanly with one or two TC=
P<br>
&gt; &gt; option kinds. A single TCP option kind might suffice if the ACK<b=
r>
&gt; &gt; frequency of 0 or 1 is taken to mean &quot;please ACK now&quot;.<=
br>
&gt; I might be useful if the sender can trigger an ACK for this particular=
 packet,<br>
&gt; but leave the rate unchanged. And change the rate from now on.<br>
&gt; If using an option, you could encode &quot;send an ack for this packet=
, but don&#39;t<br>
&gt; change the ack rate&quot; by using 0, and encode &quot;change the ack =
rate to 1 for n&quot;<br>
&gt; by using n, n &gt; 0.<br>
<br>
Yes, that sounds like a very nice idea.<br>
<br>
best,<br>
neal<br>
<br>
_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
</blockquote></div>

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To: Carles Gomez Montenegro <carlesgo@entel.upc.edu>
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Subject: Re: [tcpm] question regarding draft-gomez-tcpm-delack-suppr-reqs
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Hi Carles,

>> I agree. We in fact implemented Linux to increase cwnd by the exact
>> amount sequence (in unit of packets) so slow-start increase function =
is
>> regardless of ack-thinning effect but purely based on amount
>> delivered. This has an important effect on
>> Wireless as the other ack talk mentions. It also helps in case of ACK
>> losses or reordering.
>=20
> Agreed!
>=20
> By the way, in the next draft update, we will add that, despite its
> "Experimental" status, ABC is supported in many implementations.

Some time ago, I had brought up this same problem with ABC (RFC 3465) =
that increasing by at most one MSS during slow start makes the growth =
some what non-exponential in cases where ACK ratio < 1 (for example, =
receivers doing GRO or ACK compression, ACK losses etc). To allow an =
exponential growth, we should certainly increase congestion window by =
the amount of bytes ACKed (what Linux does).


Vidhi=20

> On May 2, 2020, at 1:34 AM, Carles Gomez Montenegro =
<carlesgo@entel.upc.edu> wrote:
>=20
> Hi Yuchung,
>=20
> <snip>
>=20
>>> Note that the problem pointed out is that, as stated in the RFC 5681
>>> text
>>> above:
>>>=20
>>>   "During slow start, a TCP increments cwnd by at most SMSS bytes =
for
>>>    each ACK received that cumulatively acknowledges new data"
>>>=20
>>> It may be desirable to support, during slow start, a cwnd increase =
by
>>> more
>>> than SMSS per each ACK acknowledging new data.
>> I agree. We in fact implemented Linux to increase cwnd by the exact
>> amount sequence (in unit of packets) so slow-start increase function =
is
>> regardless of ack-thinning effect but purely based on amount
>> delivered. This has an important effect on
>> Wireless as the other ack talk mentions. It also helps in case of ACK
>> losses or reordering.
>=20
> Agreed!
>=20
> By the way, in the next draft update, we will add that, despite its
> "Experimental" status, ABC is supported in many implementations.
>=20
> <snip>
>=20
> Thanks,
>=20
> Carles
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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To: Matt Mathis <mattmathis=40google.com@dmarc.ietf.org>, Neal Cardwell <ncardwell=40google.com@dmarc.ietf.org>
Cc: Michael Tuexen <michael.tuexen@lurchi.franken.de>, "tcpm@ietf.org" <tcpm@ietf.org>, Jon Crowcroft <jon.crowcroft@cl.cam.ac.uk>
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Subject: Re: [tcpm] On Sender Control of Delayed Acks in TCP
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On 02/05/2020 15:55, Matt Mathis wrote:
> Another probably small opportunity would be to make the RTT estimators 
> less noisy, by having the receiver always ACK the first (or last) TSO 
> segment.   The RTT is currently fuzzed by both del ACK and TCP strides.
>
I agree this might help: The TCP spec lets receivers do this right now, 
if they wish to?
> Thanks,
> --MM--
> The best way to predict the future is to create it.  - Alan Kay
>
> We must not tolerate intolerance;
>        however our response must be carefully measured:
>             too strong would be hypocritical and risks spiraling out 
> of control;
>   too weak risks being mistaken for tacit approval.
>
>

> On Sat, May 2, 2020 at 7:35 AM Neal Cardwell 
> <ncardwell=40google.com@dmarc.ietf.org 
> <mailto:40google.com@dmarc.ietf.org>> wrote:
>
>     On Sat, May 2, 2020 at 10:26 AM Michael Tuexen
>     <michael.tuexen@lurchi.franken.de
>     <mailto:michael.tuexen@lurchi.franken.de>> wrote:
>     >
>     > > On 2. May 2020, at 16:10, Neal Cardwell
>     <ncardwell=40google.com@dmarc.ietf.org
>     <mailto:40google.com@dmarc.ietf.org>> wrote:
>     > >
>     > > On Sat, May 2, 2020 at 5:40 AM Carles Gomez Montenegro
>     .....
>     > >> [CG] It seems that there are two different areas where
>     avoiding the
>     > >> typical Delayed ACKs approach may be of particular interest:
>     i) scenarios
>     > >> where very small window sizes are used (cwnd=1), and ii)
>     scenarios where
>     > >> large window sizes, and high ACK ratios, can be used. We will
>     reflect
>     > >> these considerations in the next draft update.
>     > >
>     > > Yes, I wholeheartedly agree with Bob's point that it's also
>     extremely
>     > > useful for TCP senders to be able to say things like: "I don't
>     > > particularly mind which packets you ACK, but there's no need
>     to do it
>     > > more often than 1:16". QUIC has a mechanism like this, and it
>     seems to
>     > > allow significant CPU savings, even allowing QUIC to run
>     faster than
>     > > TCP in some cases, AFAICT due to the reduced overhead for handling
>     > > ACKs:
>     > >
>     https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency
>     > > https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00
>     > >
>
QUIC doesn't *YET* have a mechanism to adjust the ACK Ratio, timing, or 
ACK content, it's likely an extension to the spec in the future. Anyway, 
I don't think TCP is the same as QUIC - TCP's notion of pacing is 
different; TCP's design differs, and there are many flavours and apps 
that use TCP in different ways.

Changing the ACK policy and usage also brings lots of trade-offs: some 
may be good in some use-cases, but also can work out really bad if the 
design doesn't match the application/network path.

>     > > So I agree with the usefulness of the two-use-case taxonomy
>     Carles has outlined:
>     > >
>     > > 1: "please ACK now"; because sender's cwnd is low
>     > >
>     > > 2: "ACK at frequency K"; because sender's cwnd is high
>     > >
>     > > It seems those could both be addressed cleanly with one or two TCP
>     > > option kinds. A single TCP option kind might suffice if the ACK
>     > > frequency of 0 or 1 is taken to mean "please ACK now".
>     > I might be useful if the sender can trigger an ACK for this
>     particular packet,
>     > but leave the rate unchanged. And change the rate from now on.
>     > If using an option, you could encode "send an ack for this
>     packet, but don't
>     > change the ack rate" by using 0, and encode "change the ack rate
>     to 1 for n"
>     > by using n, n > 0.
>
>     Yes, that sounds like a very nice idea.
>
>     best,
>     neal
>
>     _______________________________________________
>     tcpm mailing list
>     tcpm@ietf.org <mailto:tcpm@ietf.org>
>     https://www.ietf.org/mailman/listinfo/tcpm
>
>
>
I like that also. An option calling for an ACK-now, seems a little 
reminiscent of the way some stacks used "PUSH". I can immediately see 
some good things about eliciting that immediate response for a 
particular sequence number.

Gorry

> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

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    <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
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  <body>
    <p><br>
    </p>
    <div class="moz-cite-prefix">On 02/05/2020 15:55, Matt Mathis wrote:<br>
    </div>
    <blockquote type="cite"
cite="mid:CAH56bmBNxknNvgOFajYJ-FuVXNU=Ra43UKyp-AEgD4m_p070pQ@mail.gmail.com">
      <meta http-equiv="content-type" content="text/html; charset=UTF-8">
      <div dir="ltr">Another probably small opportunity would be to make
        the RTT estimators less noisy, by having the receiver always ACK
        the first (or last) TSO segment.   The RTT is currently fuzzed
        by both del ACK and TCP strides.
        <div><br clear="all">
        </div>
      </div>
    </blockquote>
    I agree this might help: The TCP spec lets receivers do this right
    now, if they wish to? <br>
    <blockquote type="cite"
cite="mid:CAH56bmBNxknNvgOFajYJ-FuVXNU=Ra43UKyp-AEgD4m_p070pQ@mail.gmail.com">
      <div dir="ltr">
        <div>
          <div>
            <div dir="ltr" class="gmail_signature"
              data-smartmail="gmail_signature">
              <div dir="ltr">
                <div>
                  <div dir="ltr">
                    <div>
                      <div dir="ltr">
                        <div>Thanks,</div>
                        --MM--<br>
                        The best way to predict the future is to create
                        it.  - Alan Kay<br>
                        <br>
                        We must not tolerate intolerance;</div>
                      <div dir="ltr">       however our response must be
                        carefully measured: </div>
                      <div>            too strong would be hypocritical
                        and risks spiraling out of control;</div>
                    </div>
                  </div>
                </div>
              </div>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <blockquote type="cite"
cite="mid:CAH56bmBNxknNvgOFajYJ-FuVXNU=Ra43UKyp-AEgD4m_p070pQ@mail.gmail.com">
      <div dir="ltr">
        <div>
          <div>
            <div dir="ltr" class="gmail_signature"
              data-smartmail="gmail_signature">
              <div dir="ltr">
                <div>
                  <div dir="ltr">
                    <div>
                      <div>   too weak risks being mistaken for tacit
                        approval.</div>
                    </div>
                  </div>
                </div>
              </div>
            </div>
          </div>
          <br>
        </div>
      </div>
      <br>
    </blockquote>
    <br>
    <blockquote type="cite"
cite="mid:CAH56bmBNxknNvgOFajYJ-FuVXNU=Ra43UKyp-AEgD4m_p070pQ@mail.gmail.com">
      <div class="gmail_quote">
        <div dir="ltr" class="gmail_attr">On Sat, May 2, 2020 at 7:35 AM
          Neal Cardwell &lt;ncardwell=<a
            href="mailto:40google.com@dmarc.ietf.org"
            moz-do-not-send="true">40google.com@dmarc.ietf.org</a>&gt;
          wrote:<br>
        </div>
        <blockquote class="gmail_quote" style="margin:0px 0px 0px
          0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">On
          Sat, May 2, 2020 at 10:26 AM Michael Tuexen<br>
          &lt;<a href="mailto:michael.tuexen@lurchi.franken.de"
            target="_blank" moz-do-not-send="true">michael.tuexen@lurchi.franken.de</a>&gt;
          wrote:<br>
          &gt;<br>
          &gt; &gt; On 2. May 2020, at 16:10, Neal Cardwell
          &lt;ncardwell=<a href="mailto:40google.com@dmarc.ietf.org"
            target="_blank" moz-do-not-send="true">40google.com@dmarc.ietf.org</a>&gt;
          wrote:<br>
          &gt; &gt;<br>
          &gt; &gt; On Sat, May 2, 2020 at 5:40 AM Carles Gomez
          Montenegro<br>
          .....<br>
          &gt; &gt;&gt; [CG] It seems that there are two different areas
          where avoiding the<br>
          &gt; &gt;&gt; typical Delayed ACKs approach may be of
          particular interest: i) scenarios<br>
          &gt; &gt;&gt; where very small window sizes are used (cwnd=1),
          and ii) scenarios where<br>
          &gt; &gt;&gt; large window sizes, and high ACK ratios, can be
          used. We will reflect<br>
          &gt; &gt;&gt; these considerations in the next draft update.<br>
          &gt; &gt;<br>
          &gt; &gt; Yes, I wholeheartedly agree with Bob's point that
          it's also extremely<br>
          &gt; &gt; useful for TCP senders to be able to say things
          like: "I don't<br>
          &gt; &gt; particularly mind which packets you ACK, but there's
          no need to do it<br>
          &gt; &gt; more often than 1:16". QUIC has a mechanism like
          this, and it seems to<br>
          &gt; &gt; allow significant CPU savings, even allowing QUIC to
          run faster than<br>
          &gt; &gt; TCP in some cases, AFAICT due to the reduced
          overhead for handling<br>
          &gt; &gt; ACKs:<br>
          &gt; &gt;  <a
href="https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency"
            rel="noreferrer" target="_blank" moz-do-not-send="true">https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency</a><br>
          &gt; &gt;  <a
            href="https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00"
            rel="noreferrer" target="_blank" moz-do-not-send="true">https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00</a><br>
          &gt; &gt;<br>
        </blockquote>
      </div>
    </blockquote>
    <p>QUIC doesn't *YET* have a mechanism to adjust the ACK Ratio,
      timing, or ACK content, it's likely an extension to the spec in
      the future. Anyway, I don't think TCP is the same as QUIC - TCP's
      notion of pacing is different; TCP's design differs, and there are
      many flavours and apps that use TCP in different ways. </p>
    <p>Changing the ACK policy and usage also brings lots of trade-offs:
      some may be good in some use-cases, but also can work out really
      bad if the design doesn't match the application/network path.</p>
    <blockquote type="cite"
cite="mid:CAH56bmBNxknNvgOFajYJ-FuVXNU=Ra43UKyp-AEgD4m_p070pQ@mail.gmail.com">
      <div class="gmail_quote">
        <blockquote class="gmail_quote" style="margin:0px 0px 0px
          0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
          &gt; &gt; So I agree with the usefulness of the two-use-case
          taxonomy Carles has outlined:<br>
          &gt; &gt;<br>
          &gt; &gt; 1: "please ACK now"; because sender's cwnd is low<br>
          &gt; &gt;<br>
          &gt; &gt; 2: "ACK at frequency K"; because sender's cwnd is
          high<br>
          &gt; &gt;<br>
          &gt; &gt; It seems those could both be addressed cleanly with
          one or two TCP<br>
          &gt; &gt; option kinds. A single TCP option kind might suffice
          if the ACK<br>
          &gt; &gt; frequency of 0 or 1 is taken to mean "please ACK
          now".<br>
          &gt; I might be useful if the sender can trigger an ACK for
          this particular packet,<br>
          &gt; but leave the rate unchanged. And change the rate from
          now on.<br>
          &gt; If using an option, you could encode "send an ack for
          this packet, but don't<br>
          &gt; change the ack rate" by using 0, and encode "change the
          ack rate to 1 for n"<br>
          &gt; by using n, n &gt; 0.<br>
          <br>
          Yes, that sounds like a very nice idea.<br>
          <br>
          best,<br>
          neal<br>
          <br>
          _______________________________________________<br>
          tcpm mailing list<br>
          <a href="mailto:tcpm@ietf.org" target="_blank"
            moz-do-not-send="true">tcpm@ietf.org</a><br>
          <a href="https://www.ietf.org/mailman/listinfo/tcpm"
            rel="noreferrer" target="_blank" moz-do-not-send="true">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
        </blockquote>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"><br>
      </fieldset>
    </blockquote>
    <p>I like that also. An option calling for an ACK-now, seems a
      little reminiscent of the way some stacks used "PUSH". I can
      immediately see some good things about eliciting that immediate
      response for a particular sequence number.</p>
    <p>Gorry<br>
    </p>
    <blockquote type="cite"
cite="mid:CAH56bmBNxknNvgOFajYJ-FuVXNU=Ra43UKyp-AEgD4m_p070pQ@mail.gmail.com">
      <pre class="moz-quote-pre" wrap="">_______________________________________________
tcpm mailing list
<a class="moz-txt-link-abbreviated" href="mailto:tcpm@ietf.org">tcpm@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/tcpm">https://www.ietf.org/mailman/listinfo/tcpm</a>
</pre>
    </blockquote>
  </body>
</html>

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From: Ian Swett <ianswett@google.com>
Date: Mon, 4 May 2020 09:14:38 -0400
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To: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
Cc: Matt Mathis <mattmathis=40google.com@dmarc.ietf.org>,  Neal Cardwell <ncardwell=40google.com@dmarc.ietf.org>,  Michael Tuexen <michael.tuexen@lurchi.franken.de>, "tcpm@ietf.org" <tcpm@ietf.org>,  Jon Crowcroft <jon.crowcroft@cl.cam.ac.uk>
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Subject: Re: [tcpm] On Sender Control of Delayed Acks in TCP
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On Mon, May 4, 2020 at 8:16 AM Gorry Fairhurst <gorry@erg.abdn.ac.uk> wrote:

>
> On 02/05/2020 15:55, Matt Mathis wrote:
>
> Another probably small opportunity would be to make the RTT estimators
> less noisy, by having the receiver always ACK the first (or last) TSO
> segment.   The RTT is currently fuzzed by both del ACK and TCP strides.
>
> I agree this might help: The TCP spec lets receivers do this right now, if
> they wish to?
>
> Thanks,
> --MM--
> The best way to predict the future is to create it.  - Alan Kay
>
> We must not tolerate intolerance;
>        however our response must be carefully measured:
>             too strong would be hypocritical and risks spiraling out of
> control;
>
>   too weak risks being mistaken for tacit approval.
>
>
>
> On Sat, May 2, 2020 at 7:35 AM Neal Cardwell <ncardwell=
> 40google.com@dmarc.ietf.org> wrote:
>
>> On Sat, May 2, 2020 at 10:26 AM Michael Tuexen
>> <michael.tuexen@lurchi.franken.de> wrote:
>> >
>> > > On 2. May 2020, at 16:10, Neal Cardwell <ncardwell=
>> 40google.com@dmarc.ietf.org> wrote:
>> > >
>> > > On Sat, May 2, 2020 at 5:40 AM Carles Gomez Montenegro
>> .....
>> > >> [CG] It seems that there are two different areas where avoiding the
>> > >> typical Delayed ACKs approach may be of particular interest: i)
>> scenarios
>> > >> where very small window sizes are used (cwnd=1), and ii) scenarios
>> where
>> > >> large window sizes, and high ACK ratios, can be used. We will reflect
>> > >> these considerations in the next draft update.
>> > >
>> > > Yes, I wholeheartedly agree with Bob's point that it's also extremely
>> > > useful for TCP senders to be able to say things like: "I don't
>> > > particularly mind which packets you ACK, but there's no need to do it
>> > > more often than 1:16". QUIC has a mechanism like this, and it seems to
>> > > allow significant CPU savings, even allowing QUIC to run faster than
>> > > TCP in some cases, AFAICT due to the reduced overhead for handling
>> > > ACKs:
>> > >
>> https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency
>> > >  https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00
>> > >
>>
> QUIC doesn't *YET* have a mechanism to adjust the ACK Ratio, timing, or
> ACK content, it's likely an extension to the spec in the future. Anyway, I
> don't think TCP is the same as QUIC - TCP's notion of pacing is different;
> TCP's design differs, and there are many flavours and apps that use TCP in
> different ways.
>

I'm curious about some of these comments.  I agree TCP and QUIC are
different(otherwise why standardize QUIC), but TCP's notion of pacing is no
different from QUIC's in my opinion.  Both have various ways to implement
pacing, and neither has any spec specifying exactly how it should be done.
For TCP, the closest I can find is RFC3449
<https://tools.ietf.org/html/rfc3449#section-4.6>.  You seem to be implying
that the set of apps which use TCP will always be wider than QUIC.  If
anything, I suspect the opposite is true, since QUIC can enable
applications where 100% reliability is undesirable.

> Changing the ACK policy and usage also brings lots of trade-offs: some may
> be good in some use-cases, but also can work out really bad if the design
> doesn't match the application/network path.
>
I completely agree.

> > > So I agree with the usefulness of the two-use-case taxonomy Carles has
>> outlined:
>> > >
>> > > 1: "please ACK now"; because sender's cwnd is low
>> > >
>> > > 2: "ACK at frequency K"; because sender's cwnd is high
>> > >
>> > > It seems those could both be addressed cleanly with one or two TCP
>> > > option kinds. A single TCP option kind might suffice if the ACK
>> > > frequency of 0 or 1 is taken to mean "please ACK now".
>> > I might be useful if the sender can trigger an ACK for this particular
>> packet,
>> > but leave the rate unchanged. And change the rate from now on.
>> > If using an option, you could encode "send an ack for this packet, but
>> don't
>> > change the ack rate" by using 0, and encode "change the ack rate to 1
>> for n"
>> > by using n, n > 0.
>>
>> Yes, that sounds like a very nice idea.
>>
>> best,
>> neal
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>
>
>
> I like that also. An option calling for an ACK-now, seems a little
> reminiscent of the way some stacks used "PUSH". I can immediately see some
> good things about eliciting that immediate response for a particular
> sequence number.
>
> Gorry
>
> _______________________________________________
> tcpm mailing listtcpm@ietf.orghttps://www.ietf.org/mailman/listinfo/tcpm
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div dir=3D"ltr"><br></div><br><div class=3D"gmail_quote">=
<div dir=3D"ltr" class=3D"gmail_attr">On Mon, May 4, 2020 at 8:16 AM Gorry =
Fairhurst &lt;<a href=3D"mailto:gorry@erg.abdn.ac.uk">gorry@erg.abdn.ac.uk<=
/a>&gt; wrote:<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0=
px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
 =20
   =20
 =20
  <div>
    <p><br>
    </p>
    <div>On 02/05/2020 15:55, Matt Mathis wrote:<br>
    </div>
    <blockquote type=3D"cite">
     =20
      <div dir=3D"ltr">Another probably small opportunity would be to make
        the RTT estimators less noisy, by having the receiver always ACK
        the first (or last) TSO segment.=C2=A0 =C2=A0The RTT is currently f=
uzzed
        by both del ACK and TCP strides.
        <div><br clear=3D"all">
        </div>
      </div>
    </blockquote>
    I agree this might help: The TCP spec lets receivers do this right
    now, if they wish to? <br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div>
          <div>
            <div dir=3D"ltr">
              <div dir=3D"ltr">
                <div>
                  <div dir=3D"ltr">
                    <div>
                      <div dir=3D"ltr">
                        <div>Thanks,</div>
                        --MM--<br>
                        The best way to predict the future is to create
                        it. =C2=A0- Alan Kay<br>
                        <br>
                        We must not tolerate intolerance;</div>
                      <div dir=3D"ltr">=C2=A0 =C2=A0 =C2=A0 =C2=A0however o=
ur response must be
                        carefully measured:=C2=A0</div>
                      <div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 too st=
rong would be hypocritical
                        and risks spiraling out of control;</div>
                    </div>
                  </div>
                </div>
              </div>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div>
          <div>
            <div dir=3D"ltr">
              <div dir=3D"ltr">
                <div>
                  <div dir=3D"ltr">
                    <div>
                      <div> =C2=A0 too weak risks being mistaken for tacit
                        approval.</div>
                    </div>
                  </div>
                </div>
              </div>
            </div>
          </div>
          <br>
        </div>
      </div>
      <br>
    </blockquote>
    <br>
    <blockquote type=3D"cite">
      <div class=3D"gmail_quote">
        <div dir=3D"ltr" class=3D"gmail_attr">On Sat, May 2, 2020 at 7:35 A=
M
          Neal Cardwell &lt;ncardwell=3D<a href=3D"mailto:40google.com@dmar=
c.ietf.org" target=3D"_blank">40google.com@dmarc.ietf.org</a>&gt;
          wrote:<br>
        </div>
        <blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex=
;border-left:1px solid rgb(204,204,204);padding-left:1ex">On
          Sat, May 2, 2020 at 10:26 AM Michael Tuexen<br>
          &lt;<a href=3D"mailto:michael.tuexen@lurchi.franken.de" target=3D=
"_blank">michael.tuexen@lurchi.franken.de</a>&gt;
          wrote:<br>
          &gt;<br>
          &gt; &gt; On 2. May 2020, at 16:10, Neal Cardwell
          &lt;ncardwell=3D<a href=3D"mailto:40google.com@dmarc.ietf.org" ta=
rget=3D"_blank">40google.com@dmarc.ietf.org</a>&gt;
          wrote:<br>
          &gt; &gt;<br>
          &gt; &gt; On Sat, May 2, 2020 at 5:40 AM Carles Gomez
          Montenegro<br>
          .....<br>
          &gt; &gt;&gt; [CG] It seems that there are two different areas
          where avoiding the<br>
          &gt; &gt;&gt; typical Delayed ACKs approach may be of
          particular interest: i) scenarios<br>
          &gt; &gt;&gt; where very small window sizes are used (cwnd=3D1),
          and ii) scenarios where<br>
          &gt; &gt;&gt; large window sizes, and high ACK ratios, can be
          used. We will reflect<br>
          &gt; &gt;&gt; these considerations in the next draft update.<br>
          &gt; &gt;<br>
          &gt; &gt; Yes, I wholeheartedly agree with Bob&#39;s point that
          it&#39;s also extremely<br>
          &gt; &gt; useful for TCP senders to be able to say things
          like: &quot;I don&#39;t<br>
          &gt; &gt; particularly mind which packets you ACK, but there&#39;=
s
          no need to do it<br>
          &gt; &gt; more often than 1:16&quot;. QUIC has a mechanism like
          this, and it seems to<br>
          &gt; &gt; allow significant CPU savings, even allowing QUIC to
          run faster than<br>
          &gt; &gt; TCP in some cases, AFAICT due to the reduced
          overhead for handling<br>
          &gt; &gt; ACKs:<br>
          &gt; &gt;=C2=A0 <a href=3D"https://www.fastly.com/blog/measuring-=
quic-vs-tcp-computational-efficiency" rel=3D"noreferrer" target=3D"_blank">=
https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency<=
/a><br>
          &gt; &gt;=C2=A0 <a href=3D"https://tools.ietf.org/html/draft-iyen=
gar-quic-delayed-ack-00" rel=3D"noreferrer" target=3D"_blank">https://tools=
.ietf.org/html/draft-iyengar-quic-delayed-ack-00</a><br>
          &gt; &gt;<br>
        </blockquote>
      </div>
    </blockquote>
    <p>QUIC doesn&#39;t *YET* have a mechanism to adjust the ACK Ratio,
      timing, or ACK content, it&#39;s likely an extension to the spec in
      the future. Anyway, I don&#39;t think TCP is the same as QUIC - TCP&#=
39;s
      notion of pacing is different; TCP&#39;s design differs, and there ar=
e
      many flavours and apps that use TCP in different ways.</p></div></blo=
ckquote><div><br></div><div>I&#39;m curious about some of these comments.=
=C2=A0 I agree TCP and QUIC are different(otherwise why standardize QUIC), =
but TCP&#39;s notion of pacing is no different from QUIC&#39;s in my opinio=
n.=C2=A0 Both have various ways to implement pacing, and neither has any sp=
ec specifying exactly how it should be done.=C2=A0 For TCP, the closest I c=
an find is <a href=3D"https://tools.ietf.org/html/rfc3449#section-4.6">RFC3=
449</a>.=C2=A0 You seem to be implying that the set of apps which use TCP w=
ill always be wider than QUIC.=C2=A0 If anything, I suspect the opposite is=
 true, since QUIC can enable applications where 100% reliability is undesir=
able.</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.=
8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div><p> </p>
    <p>Changing the ACK policy and usage also brings lots of trade-offs:
      some may be good in some use-cases, but also can work out really
      bad if the design doesn&#39;t match the application/network path.</p>=
</div></blockquote><div>I completely agree.</div><blockquote class=3D"gmail=
_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204=
,204);padding-left:1ex"><div>
    <blockquote type=3D"cite">
      <div class=3D"gmail_quote">
        <blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex=
;border-left:1px solid rgb(204,204,204);padding-left:1ex">
          &gt; &gt; So I agree with the usefulness of the two-use-case
          taxonomy Carles has outlined:<br>
          &gt; &gt;<br>
          &gt; &gt; 1: &quot;please ACK now&quot;; because sender&#39;s cwn=
d is low<br>
          &gt; &gt;<br>
          &gt; &gt; 2: &quot;ACK at frequency K&quot;; because sender&#39;s=
 cwnd is
          high<br>
          &gt; &gt;<br>
          &gt; &gt; It seems those could both be addressed cleanly with
          one or two TCP<br>
          &gt; &gt; option kinds. A single TCP option kind might suffice
          if the ACK<br>
          &gt; &gt; frequency of 0 or 1 is taken to mean &quot;please ACK
          now&quot;.<br>
          &gt; I might be useful if the sender can trigger an ACK for
          this particular packet,<br>
          &gt; but leave the rate unchanged. And change the rate from
          now on.<br>
          &gt; If using an option, you could encode &quot;send an ack for
          this packet, but don&#39;t<br>
          &gt; change the ack rate&quot; by using 0, and encode &quot;chang=
e the
          ack rate to 1 for n&quot;<br>
          &gt; by using n, n &gt; 0.<br>
          <br>
          Yes, that sounds like a very nice idea.<br>
          <br>
          best,<br>
          neal<br>
          <br>
          _______________________________________________<br>
          tcpm mailing list<br>
          <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org<=
/a><br>
          <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"nor=
eferrer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/tcpm</a><b=
r>
        </blockquote>
      </div>
      <br>
      <fieldset><br>
      </fieldset>
    </blockquote>
    <p>I like that also. An option calling for an ACK-now, seems a
      little reminiscent of the way some stacks used &quot;PUSH&quot;. I ca=
n
      immediately see some good things about eliciting that immediate
      response for a particular sequence number.</p>
    <p>Gorry<br>
    </p>
    <blockquote type=3D"cite">
      <pre>_______________________________________________
tcpm mailing list
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tcpm</a>
</pre>
    </blockquote>
  </div>

_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
</blockquote></div></div>

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To: Ian Swett <ianswett@google.com>
Cc: Matt Mathis <mattmathis=40google.com@dmarc.ietf.org>, Neal Cardwell <ncardwell=40google.com@dmarc.ietf.org>, Michael Tuexen <michael.tuexen@lurchi.franken.de>, "tcpm@ietf.org" <tcpm@ietf.org>, Jon Crowcroft <jon.crowcroft@cl.cam.ac.uk>
References: <683902e8-a2af-cfb7-ffd0-c5c5742e5bd5@gmx.at> <CADVnQykY3OqXy=RcEfa-OpfK2x=W5_FTrdrx7PvKuqgEt92uNw@mail.gmail.com> <7d145f1203f6344b92f6f8aa11a78239.squirrel@webmail.entel.upc.edu> <CADVnQy=wPUx62y7VNqjSPP+snKX4vVvK5q=qqYb1j+0nGrtezQ@mail.gmail.com> <c6da08f4-03d7-da46-26b1-168f5953329f@bobbriscoe.net> <909de4172e46712f543f723d0ae2d638.squirrel@webmail.entel.upc.edu> <CADVnQynE3EMh-9qX7TkxifNvaKke7=PpWW2nB3Z6t1q7CYQXbg@mail.gmail.com> <9B5D12AC-248F-4E61-B6C5-1DA9529C7A42@lurchi.franken.de> <CADVnQy=gGzsvG2F935bM7wqGbbSZi+A5E=vxkpz=Bwc+ab2rkg@mail.gmail.com> <CAH56bmBNxknNvgOFajYJ-FuVXNU=Ra43UKyp-AEgD4m_p070pQ@mail.gmail.com> <54809559-05ef-b104-6cee-e61c8231dbae@erg.abdn.ac.uk> <CAKcm_gPtcsNPL5Og-fFveQgV0in+NbVVHJCTu+37a42_xA-Nyw@mail.gmail.com>
From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Date: Mon, 4 May 2020 14:38:43 +0100
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Subject: Re: [tcpm] On Sender Control of Delayed Acks in TCP
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On 04/05/2020 14:14, Ian Swett wrote:
>
>
> On Mon, May 4, 2020 at 8:16 AM Gorry Fairhurst <gorry@erg.abdn.ac.uk 
> <mailto:gorry@erg.abdn.ac.uk>> wrote:
>
>
>     On 02/05/2020 15:55, Matt Mathis wrote:
>>     Another probably small opportunity would be to make the RTT
>>     estimators less noisy, by having the receiver always ACK the
>>     first (or last) TSO segment.   The RTT is currently fuzzed by
>>     both del ACK and TCP strides.
>>
>     I agree this might help: The TCP spec lets receivers do this right
>     now, if they wish to?
>>     Thanks,
>>     --MM--
>>     The best way to predict the future is to create it.  - Alan Kay
>>
>>     We must not tolerate intolerance;
>>            however our response must be carefully measured:
>>                 too strong would be hypocritical and risks spiraling
>>     out of control;
>>       too weak risks being mistaken for tacit approval.
>>
>>
>
>>     On Sat, May 2, 2020 at 7:35 AM Neal Cardwell
>>     <ncardwell=40google.com@dmarc.ietf.org
>>     <mailto:40google.com@dmarc.ietf.org>> wrote:
>>
>>         On Sat, May 2, 2020 at 10:26 AM Michael Tuexen
>>         <michael.tuexen@lurchi.franken.de
>>         <mailto:michael.tuexen@lurchi.franken.de>> wrote:
>>         >
>>         > > On 2. May 2020, at 16:10, Neal Cardwell
>>         <ncardwell=40google.com@dmarc.ietf.org
>>         <mailto:40google.com@dmarc.ietf.org>> wrote:
>>         > >
>>         > > On Sat, May 2, 2020 at 5:40 AM Carles Gomez Montenegro
>>         .....
>>         > >> [CG] It seems that there are two different areas where
>>         avoiding the
>>         > >> typical Delayed ACKs approach may be of particular
>>         interest: i) scenarios
>>         > >> where very small window sizes are used (cwnd=1), and ii)
>>         scenarios where
>>         > >> large window sizes, and high ACK ratios, can be used. We
>>         will reflect
>>         > >> these considerations in the next draft update.
>>         > >
>>         > > Yes, I wholeheartedly agree with Bob's point that it's
>>         also extremely
>>         > > useful for TCP senders to be able to say things like: "I
>>         don't
>>         > > particularly mind which packets you ACK, but there's no
>>         need to do it
>>         > > more often than 1:16". QUIC has a mechanism like this,
>>         and it seems to
>>         > > allow significant CPU savings, even allowing QUIC to run
>>         faster than
>>         > > TCP in some cases, AFAICT due to the reduced overhead for
>>         handling
>>         > > ACKs:
>>         > >
>>         https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency
>>         > > https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00
>>         > >
>>
>     QUIC doesn't *YET* have a mechanism to adjust the ACK Ratio,
>     timing, or ACK content, it's likely an extension to the spec in
>     the future. Anyway, I don't think TCP is the same as QUIC - TCP's
>     notion of pacing is different; TCP's design differs, and there are
>     many flavours and apps that use TCP in different ways.
>
>
> I'm curious about some of these comments.  I agree TCP and QUIC are 
> different(otherwise why standardize QUIC), but TCP's notion of pacing 
> is no different from QUIC's in my opinion.  Both have various ways to 
> implement pacing, and neither has any spec specifying exactly how it 
> should be done.  For TCP, the closest I can find is RFC3449 
> <https://tools.ietf.org/html/rfc3449#section-4.6>.  You seem to be 
> implying that the set of apps which use TCP will always be wider than 
> QUIC.
QUIC seems to anticipate the need for pacing in its deisgn assumptions - 
I may be wrong? TCP has various recommendations that to me suggest 
pacing is a good idea (I would agree), but I expect lots of stacks do 
not implement burst mitigation.
> If anything, I suspect the opposite is true, since QUIC can enable 
> applications where 100% reliability is undesirable.

My comment was meant to say there are already a very wide range of 
protocols using TCP. From routing protocols, DNS, to other core 
protocols, to streaming video, to web, bulk transfer, IoT, transaction 
processing, etc. I was hoping to suggest that already many of these 
stacks have been optimised, and we are living with many decades of 
legacy and diversity.

On the other hand, I agree QUIC has applicability to datagrams for 
instance where reliability is not required for data, so I do need to 
learn more about how/whether these will handle CC and pacing, etc.

>     Changing the ACK policy and usage also brings lots of trade-offs:
>     some may be good in some use-cases, but also can work out really
>     bad if the design doesn't match the application/network path.
>
> I completely agree.
>
>>         > > So I agree with the usefulness of the two-use-case
>>         taxonomy Carles has outlined:
>>         > >
>>         > > 1: "please ACK now"; because sender's cwnd is low
>>         > >
>>         > > 2: "ACK at frequency K"; because sender's cwnd is high
>>         > >
>>         > > It seems those could both be addressed cleanly with one
>>         or two TCP
>>         > > option kinds. A single TCP option kind might suffice if
>>         the ACK
>>         > > frequency of 0 or 1 is taken to mean "please ACK now".
>>         > I might be useful if the sender can trigger an ACK for this
>>         particular packet,
>>         > but leave the rate unchanged. And change the rate from now on.
>>         > If using an option, you could encode "send an ack for this
>>         packet, but don't
>>         > change the ack rate" by using 0, and encode "change the ack
>>         rate to 1 for n"
>>         > by using n, n > 0.
>>
>>         Yes, that sounds like a very nice idea.
>>
>>         best,
>>         neal
>>
>>         _______________________________________________
>>         tcpm mailing list
>>         tcpm@ietf.org <mailto:tcpm@ietf.org>
>>         https://www.ietf.org/mailman/listinfo/tcpm
>>
>>
>>
>     I like that also. An option calling for an ACK-now, seems a little
>     reminiscent of the way some stacks used "PUSH". I can immediately
>     see some good things about eliciting that immediate response for a
>     particular sequence number.
>
>     Gorry
>
>>     _______________________________________________
>>     tcpm mailing list
>>     tcpm@ietf.org  <mailto:tcpm@ietf.org>
>>     https://www.ietf.org/mailman/listinfo/tcpm
>     _______________________________________________
>     tcpm mailing list
>     tcpm@ietf.org <mailto:tcpm@ietf.org>
>     https://www.ietf.org/mailman/listinfo/tcpm
>
-- 
G. Fairhurst, School of Engineering


Gorry


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    <p><br>
    </p>
    <div class="moz-cite-prefix">On 04/05/2020 14:14, Ian Swett wrote:<br>
    </div>
    <blockquote type="cite"
cite="mid:CAKcm_gPtcsNPL5Og-fFveQgV0in+NbVVHJCTu+37a42_xA-Nyw@mail.gmail.com">
      <meta http-equiv="content-type" content="text/html; charset=UTF-8">
      <div dir="ltr">
        <div dir="ltr"><br>
        </div>
        <br>
        <div class="gmail_quote">
          <div dir="ltr" class="gmail_attr">On Mon, May 4, 2020 at 8:16
            AM Gorry Fairhurst &lt;<a href="mailto:gorry@erg.abdn.ac.uk"
              moz-do-not-send="true">gorry@erg.abdn.ac.uk</a>&gt; wrote:<br>
          </div>
          <blockquote class="gmail_quote" style="margin:0px 0px 0px
            0.8ex;border-left:1px solid
            rgb(204,204,204);padding-left:1ex">
            <div>
              <p><br>
              </p>
              <div>On 02/05/2020 15:55, Matt Mathis wrote:<br>
              </div>
              <blockquote type="cite">
                <div dir="ltr">Another probably small opportunity would
                  be to make the RTT estimators less noisy, by having
                  the receiver always ACK the first (or last) TSO
                  segment.   The RTT is currently fuzzed by both del ACK
                  and TCP strides.
                  <div><br clear="all">
                  </div>
                </div>
              </blockquote>
              I agree this might help: The TCP spec lets receivers do
              this right now, if they wish to? <br>
              <blockquote type="cite">
                <div dir="ltr">
                  <div>
                    <div>
                      <div dir="ltr">
                        <div dir="ltr">
                          <div>
                            <div dir="ltr">
                              <div>
                                <div dir="ltr">
                                  <div>Thanks,</div>
                                  --MM--<br>
                                  The best way to predict the future is
                                  to create it.  - Alan Kay<br>
                                  <br>
                                  We must not tolerate intolerance;</div>
                                <div dir="ltr">       however our
                                  response must be carefully measured: </div>
                                <div>            too strong would be
                                  hypocritical and risks spiraling out
                                  of control;</div>
                              </div>
                            </div>
                          </div>
                        </div>
                      </div>
                    </div>
                  </div>
                </div>
              </blockquote>
              <blockquote type="cite">
                <div dir="ltr">
                  <div>
                    <div>
                      <div dir="ltr">
                        <div dir="ltr">
                          <div>
                            <div dir="ltr">
                              <div>
                                <div>   too weak risks being mistaken
                                  for tacit approval.</div>
                              </div>
                            </div>
                          </div>
                        </div>
                      </div>
                    </div>
                    <br>
                  </div>
                </div>
                <br>
              </blockquote>
              <br>
              <blockquote type="cite">
                <div class="gmail_quote">
                  <div dir="ltr" class="gmail_attr">On Sat, May 2, 2020
                    at 7:35 AM Neal Cardwell &lt;ncardwell=<a
                      href="mailto:40google.com@dmarc.ietf.org"
                      target="_blank" moz-do-not-send="true">40google.com@dmarc.ietf.org</a>&gt;
                    wrote:<br>
                  </div>
                  <blockquote class="gmail_quote" style="margin:0px 0px
                    0px 0.8ex;border-left:1px solid
                    rgb(204,204,204);padding-left:1ex">On Sat, May 2,
                    2020 at 10:26 AM Michael Tuexen<br>
                    &lt;<a
                      href="mailto:michael.tuexen@lurchi.franken.de"
                      target="_blank" moz-do-not-send="true">michael.tuexen@lurchi.franken.de</a>&gt;
                    wrote:<br>
                    &gt;<br>
                    &gt; &gt; On 2. May 2020, at 16:10, Neal Cardwell
                    &lt;ncardwell=<a
                      href="mailto:40google.com@dmarc.ietf.org"
                      target="_blank" moz-do-not-send="true">40google.com@dmarc.ietf.org</a>&gt;
                    wrote:<br>
                    &gt; &gt;<br>
                    &gt; &gt; On Sat, May 2, 2020 at 5:40 AM Carles
                    Gomez Montenegro<br>
                    .....<br>
                    &gt; &gt;&gt; [CG] It seems that there are two
                    different areas where avoiding the<br>
                    &gt; &gt;&gt; typical Delayed ACKs approach may be
                    of particular interest: i) scenarios<br>
                    &gt; &gt;&gt; where very small window sizes are used
                    (cwnd=1), and ii) scenarios where<br>
                    &gt; &gt;&gt; large window sizes, and high ACK
                    ratios, can be used. We will reflect<br>
                    &gt; &gt;&gt; these considerations in the next draft
                    update.<br>
                    &gt; &gt;<br>
                    &gt; &gt; Yes, I wholeheartedly agree with Bob's
                    point that it's also extremely<br>
                    &gt; &gt; useful for TCP senders to be able to say
                    things like: "I don't<br>
                    &gt; &gt; particularly mind which packets you ACK,
                    but there's no need to do it<br>
                    &gt; &gt; more often than 1:16". QUIC has a
                    mechanism like this, and it seems to<br>
                    &gt; &gt; allow significant CPU savings, even
                    allowing QUIC to run faster than<br>
                    &gt; &gt; TCP in some cases, AFAICT due to the
                    reduced overhead for handling<br>
                    &gt; &gt; ACKs:<br>
                    &gt; &gt;  <a
href="https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency"
                      rel="noreferrer" target="_blank"
                      moz-do-not-send="true">https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency</a><br>
                    &gt; &gt;  <a
                      href="https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00"
                      rel="noreferrer" target="_blank"
                      moz-do-not-send="true">https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00</a><br>
                    &gt; &gt;<br>
                  </blockquote>
                </div>
              </blockquote>
              <p>QUIC doesn't *YET* have a mechanism to adjust the ACK
                Ratio, timing, or ACK content, it's likely an extension
                to the spec in the future. Anyway, I don't think TCP is
                the same as QUIC - TCP's notion of pacing is different;
                TCP's design differs, and there are many flavours and
                apps that use TCP in different ways.</p>
            </div>
          </blockquote>
          <div><br>
          </div>
          <div>I'm curious about some of these comments.  I agree TCP
            and QUIC are different(otherwise why standardize QUIC), but
            TCP's notion of pacing is no different from QUIC's in my
            opinion.  Both have various ways to implement pacing, and
            neither has any spec specifying exactly how it should be
            done.  For TCP, the closest I can find is <a
              href="https://tools.ietf.org/html/rfc3449#section-4.6"
              moz-do-not-send="true">RFC3449</a>.  You seem to be
            implying that the set of apps which use TCP will always be
            wider than QUIC. <br>
          </div>
        </div>
      </div>
    </blockquote>
    QUIC seems to anticipate the need for pacing in its deisgn
    assumptions - I may be wrong? TCP has various recommendations that
    to me suggest pacing is a good idea (I would agree), but I expect
    lots of stacks do not implement burst mitigation.<br>
    <blockquote type="cite"
cite="mid:CAKcm_gPtcsNPL5Og-fFveQgV0in+NbVVHJCTu+37a42_xA-Nyw@mail.gmail.com">
      <div dir="ltr">
        <div class="gmail_quote">
          <div> If anything, I suspect the opposite is true, since QUIC
            can enable applications where 100% reliability is
            undesirable.</div>
        </div>
      </div>
    </blockquote>
    <p>My comment was meant to say there are already a very wide range
      of protocols using TCP. From routing protocols, DNS, to other core
      protocols, to streaming video, to web, bulk transfer, IoT,
      transaction processing, etc. I was hoping to suggest that already
      many of these stacks have been optimised, and we are living with
      many decades of legacy and diversity. <br>
    </p>
    <p>On the other hand, I agree QUIC has applicability to datagrams
      for instance where reliability is not required for data, so I do
      need to learn more about how/whether these will handle CC and
      pacing, etc.  <br>
    </p>
    <blockquote type="cite"
cite="mid:CAKcm_gPtcsNPL5Og-fFveQgV0in+NbVVHJCTu+37a42_xA-Nyw@mail.gmail.com">
      <div dir="ltr">
        <div class="gmail_quote">
          <blockquote class="gmail_quote" style="margin:0px 0px 0px
            0.8ex;border-left:1px solid
            rgb(204,204,204);padding-left:1ex">
            <div>
              <p> </p>
              <p>Changing the ACK policy and usage also brings lots of
                trade-offs: some may be good in some use-cases, but also
                can work out really bad if the design doesn't match the
                application/network path.</p>
            </div>
          </blockquote>
          <div>I completely agree.</div>
          <blockquote class="gmail_quote" style="margin:0px 0px 0px
            0.8ex;border-left:1px solid
            rgb(204,204,204);padding-left:1ex">
            <div>
              <blockquote type="cite">
                <div class="gmail_quote">
                  <blockquote class="gmail_quote" style="margin:0px 0px
                    0px 0.8ex;border-left:1px solid
                    rgb(204,204,204);padding-left:1ex"> &gt; &gt; So I
                    agree with the usefulness of the two-use-case
                    taxonomy Carles has outlined:<br>
                    &gt; &gt;<br>
                    &gt; &gt; 1: "please ACK now"; because sender's cwnd
                    is low<br>
                    &gt; &gt;<br>
                    &gt; &gt; 2: "ACK at frequency K"; because sender's
                    cwnd is high<br>
                    &gt; &gt;<br>
                    &gt; &gt; It seems those could both be addressed
                    cleanly with one or two TCP<br>
                    &gt; &gt; option kinds. A single TCP option kind
                    might suffice if the ACK<br>
                    &gt; &gt; frequency of 0 or 1 is taken to mean
                    "please ACK now".<br>
                    &gt; I might be useful if the sender can trigger an
                    ACK for this particular packet,<br>
                    &gt; but leave the rate unchanged. And change the
                    rate from now on.<br>
                    &gt; If using an option, you could encode "send an
                    ack for this packet, but don't<br>
                    &gt; change the ack rate" by using 0, and encode
                    "change the ack rate to 1 for n"<br>
                    &gt; by using n, n &gt; 0.<br>
                    <br>
                    Yes, that sounds like a very nice idea.<br>
                    <br>
                    best,<br>
                    neal<br>
                    <br>
                    _______________________________________________<br>
                    tcpm mailing list<br>
                    <a href="mailto:tcpm@ietf.org" target="_blank"
                      moz-do-not-send="true">tcpm@ietf.org</a><br>
                    <a href="https://www.ietf.org/mailman/listinfo/tcpm"
                      rel="noreferrer" target="_blank"
                      moz-do-not-send="true">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
                  </blockquote>
                </div>
                <br>
                <fieldset><br>
                </fieldset>
              </blockquote>
              <p>I like that also. An option calling for an ACK-now,
                seems a little reminiscent of the way some stacks used
                "PUSH". I can immediately see some good things about
                eliciting that immediate response for a particular
                sequence number.</p>
              <p>Gorry<br>
              </p>
              <blockquote type="cite">
                <pre>_______________________________________________
tcpm mailing list
<a href="mailto:tcpm@ietf.org" target="_blank" moz-do-not-send="true">tcpm@ietf.org</a>
<a href="https://www.ietf.org/mailman/listinfo/tcpm" target="_blank" moz-do-not-send="true">https://www.ietf.org/mailman/listinfo/tcpm</a>
</pre>
              </blockquote>
            </div>
            _______________________________________________<br>
            tcpm mailing list<br>
            <a href="mailto:tcpm@ietf.org" target="_blank"
              moz-do-not-send="true">tcpm@ietf.org</a><br>
            <a href="https://www.ietf.org/mailman/listinfo/tcpm"
              rel="noreferrer" target="_blank" moz-do-not-send="true">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
          </blockquote>
        </div>
      </div>
    </blockquote>
    <pre class="moz-signature" cols="72">-- 
G. Fairhurst, School of Engineering


Gorry

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the TCP Maintenance and Minor Extensions WG of the IETF.

        Title           : Requirements for Time-Based Loss Detection
        Author          : Mark Allman
	Filename        : draft-ietf-tcpm-rto-consider-12.txt
	Pages           : 10
	Date            : 2020-05-04

Abstract:
    Many protocols must detect packet loss for various reasons (e.g., to
    ensure reliability using retransmissions or to understand the level
    of congestion along a network path).  While many mechanisms have
    been designed to detect loss, protocols ultimately can only count on
    the passage of time without delivery confirmation to declare a
    packet "lost".  Each implementation of a time-based loss detection
    mechanism represents a balance between correctness and timeliness
    and therefore no implementation suits all situations.  This document
    provides high-level requirements for time-based loss detectors
    appropriate for general use in the Internet.  Within the
    requirements, implementations have latitude to define particulars
    that best address each situation.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tcpm-rto-consider/

There are also htmlized versions available at:
https://tools.ietf.org/html/draft-ietf-tcpm-rto-consider-12
https://datatracker.ietf.org/doc/html/draft-ietf-tcpm-rto-consider-12

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-rto-consider-12


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/



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From: Ian Swett <ianswett@google.com>
Date: Mon, 4 May 2020 10:03:55 -0400
Message-ID: <CAKcm_gN_44XCkgxuJLXGp1jMG4_rj=X48ro7LBEw-txPCHum8A@mail.gmail.com>
To: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
Cc: Matt Mathis <mattmathis=40google.com@dmarc.ietf.org>,  Neal Cardwell <ncardwell=40google.com@dmarc.ietf.org>,  Michael Tuexen <michael.tuexen@lurchi.franken.de>, "tcpm@ietf.org" <tcpm@ietf.org>,  Jon Crowcroft <jon.crowcroft@cl.cam.ac.uk>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/tcpm/QBRx1tRxPR6igHrHojpxhd5arBk>
Subject: Re: [tcpm] On Sender Control of Delayed Acks in TCP
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Thanks for the clarification.  In implementation, I expect the status of
pacing to be extremely similar for TCP and QUIC.  Many stacks and
applications that do it for TCP will do it for QUIC, and those which don't
pace TCP probably won't bother with QUIC.

On Mon, May 4, 2020 at 9:38 AM Gorry Fairhurst <gorry@erg.abdn.ac.uk> wrote:

>
> On 04/05/2020 14:14, Ian Swett wrote:
>
>
>
> On Mon, May 4, 2020 at 8:16 AM Gorry Fairhurst <gorry@erg.abdn.ac.uk>
> wrote:
>
>>
>> On 02/05/2020 15:55, Matt Mathis wrote:
>>
>> Another probably small opportunity would be to make the RTT estimators
>> less noisy, by having the receiver always ACK the first (or last) TSO
>> segment.   The RTT is currently fuzzed by both del ACK and TCP strides.
>>
>> I agree this might help: The TCP spec lets receivers do this right now,
>> if they wish to?
>>
>> Thanks,
>> --MM--
>> The best way to predict the future is to create it.  - Alan Kay
>>
>> We must not tolerate intolerance;
>>        however our response must be carefully measured:
>>             too strong would be hypocritical and risks spiraling out of
>> control;
>>
>>   too weak risks being mistaken for tacit approval.
>>
>>
>>
>> On Sat, May 2, 2020 at 7:35 AM Neal Cardwell <ncardwell=
>> 40google.com@dmarc.ietf.org> wrote:
>>
>>> On Sat, May 2, 2020 at 10:26 AM Michael Tuexen
>>> <michael.tuexen@lurchi.franken.de> wrote:
>>> >
>>> > > On 2. May 2020, at 16:10, Neal Cardwell <ncardwell=
>>> 40google.com@dmarc.ietf.org> wrote:
>>> > >
>>> > > On Sat, May 2, 2020 at 5:40 AM Carles Gomez Montenegro
>>> .....
>>> > >> [CG] It seems that there are two different areas where avoiding the
>>> > >> typical Delayed ACKs approach may be of particular interest: i)
>>> scenarios
>>> > >> where very small window sizes are used (cwnd=1), and ii) scenarios
>>> where
>>> > >> large window sizes, and high ACK ratios, can be used. We will
>>> reflect
>>> > >> these considerations in the next draft update.
>>> > >
>>> > > Yes, I wholeheartedly agree with Bob's point that it's also extremely
>>> > > useful for TCP senders to be able to say things like: "I don't
>>> > > particularly mind which packets you ACK, but there's no need to do it
>>> > > more often than 1:16". QUIC has a mechanism like this, and it seems
>>> to
>>> > > allow significant CPU savings, even allowing QUIC to run faster than
>>> > > TCP in some cases, AFAICT due to the reduced overhead for handling
>>> > > ACKs:
>>> > >
>>> https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency
>>> > >  https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00
>>> > >
>>>
>> QUIC doesn't *YET* have a mechanism to adjust the ACK Ratio, timing, or
>> ACK content, it's likely an extension to the spec in the future. Anyway, I
>> don't think TCP is the same as QUIC - TCP's notion of pacing is different;
>> TCP's design differs, and there are many flavours and apps that use TCP in
>> different ways.
>>
>
> I'm curious about some of these comments.  I agree TCP and QUIC are
> different(otherwise why standardize QUIC), but TCP's notion of pacing is no
> different from QUIC's in my opinion.  Both have various ways to implement
> pacing, and neither has any spec specifying exactly how it should be done.
> For TCP, the closest I can find is RFC3449
> <https://tools.ietf.org/html/rfc3449#section-4.6>.  You seem to be
> implying that the set of apps which use TCP will always be wider than QUIC.
>
> QUIC seems to anticipate the need for pacing in its deisgn assumptions - I
> may be wrong? TCP has various recommendations that to me suggest pacing is
> a good idea (I would agree), but I expect lots of stacks do not implement
> burst mitigation.
>
> If anything, I suspect the opposite is true, since QUIC can enable
> applications where 100% reliability is undesirable.
>
> My comment was meant to say there are already a very wide range of
> protocols using TCP. From routing protocols, DNS, to other core protocols,
> to streaming video, to web, bulk transfer, IoT, transaction processing,
> etc. I was hoping to suggest that already many of these stacks have been
> optimised, and we are living with many decades of legacy and diversity.
>
> On the other hand, I agree QUIC has applicability to datagrams for
> instance where reliability is not required for data, so I do need to learn
> more about how/whether these will handle CC and pacing, etc.
>
> Changing the ACK policy and usage also brings lots of trade-offs: some may
>> be good in some use-cases, but also can work out really bad if the design
>> doesn't match the application/network path.
>>
> I completely agree.
>
>> > > So I agree with the usefulness of the two-use-case taxonomy Carles
>>> has outlined:
>>> > >
>>> > > 1: "please ACK now"; because sender's cwnd is low
>>> > >
>>> > > 2: "ACK at frequency K"; because sender's cwnd is high
>>> > >
>>> > > It seems those could both be addressed cleanly with one or two TCP
>>> > > option kinds. A single TCP option kind might suffice if the ACK
>>> > > frequency of 0 or 1 is taken to mean "please ACK now".
>>> > I might be useful if the sender can trigger an ACK for this particular
>>> packet,
>>> > but leave the rate unchanged. And change the rate from now on.
>>> > If using an option, you could encode "send an ack for this packet, but
>>> don't
>>> > change the ack rate" by using 0, and encode "change the ack rate to 1
>>> for n"
>>> > by using n, n > 0.
>>>
>>> Yes, that sounds like a very nice idea.
>>>
>>> best,
>>> neal
>>>
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>>>
>>
>>
>> I like that also. An option calling for an ACK-now, seems a little
>> reminiscent of the way some stacks used "PUSH". I can immediately see some
>> good things about eliciting that immediate response for a particular
>> sequence number.
>>
>> Gorry
>>
>> _______________________________________________
>> tcpm mailing listtcpm@ietf.orghttps://www.ietf.org/mailman/listinfo/tcpm
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>
> --
> G. Fairhurst, School of Engineering
>
>
> Gorry
>
>
>

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Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Thanks for the clarification.=C2=A0 In implementation, I e=
xpect the status of pacing to be extremely similar for TCP and QUIC.=C2=A0 =
Many stacks and applications that do it for TCP will do it for QUIC, and th=
ose which don&#39;t pace TCP probably won&#39;t bother with QUIC.</div><br>=
<div class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">On Mon, Ma=
y 4, 2020 at 9:38 AM Gorry Fairhurst &lt;<a href=3D"mailto:gorry@erg.abdn.a=
c.uk">gorry@erg.abdn.ac.uk</a>&gt; wrote:<br></div><blockquote class=3D"gma=
il_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,2=
04,204);padding-left:1ex">
 =20
   =20
 =20
  <div>
    <p><br>
    </p>
    <div>On 04/05/2020 14:14, Ian Swett wrote:<br>
    </div>
    <blockquote type=3D"cite">
     =20
      <div dir=3D"ltr">
        <div dir=3D"ltr"><br>
        </div>
        <br>
        <div class=3D"gmail_quote">
          <div dir=3D"ltr" class=3D"gmail_attr">On Mon, May 4, 2020 at 8:16
            AM Gorry Fairhurst &lt;<a href=3D"mailto:gorry@erg.abdn.ac.uk" =
target=3D"_blank">gorry@erg.abdn.ac.uk</a>&gt; wrote:<br>
          </div>
          <blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8=
ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
            <div>
              <p><br>
              </p>
              <div>On 02/05/2020 15:55, Matt Mathis wrote:<br>
              </div>
              <blockquote type=3D"cite">
                <div dir=3D"ltr">Another probably small opportunity would
                  be to make the RTT estimators less noisy, by having
                  the receiver always ACK the first (or last) TSO
                  segment.=C2=A0 =C2=A0The RTT is currently fuzzed by both =
del ACK
                  and TCP strides.
                  <div><br clear=3D"all">
                  </div>
                </div>
              </blockquote>
              I agree this might help: The TCP spec lets receivers do
              this right now, if they wish to? <br>
              <blockquote type=3D"cite">
                <div dir=3D"ltr">
                  <div>
                    <div>
                      <div dir=3D"ltr">
                        <div dir=3D"ltr">
                          <div>
                            <div dir=3D"ltr">
                              <div>
                                <div dir=3D"ltr">
                                  <div>Thanks,</div>
                                  --MM--<br>
                                  The best way to predict the future is
                                  to create it. =C2=A0- Alan Kay<br>
                                  <br>
                                  We must not tolerate intolerance;</div>
                                <div dir=3D"ltr">=C2=A0 =C2=A0 =C2=A0 =C2=
=A0however our
                                  response must be carefully measured:=C2=
=A0</div>
                                <div>=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 too strong would be
                                  hypocritical and risks spiraling out
                                  of control;</div>
                              </div>
                            </div>
                          </div>
                        </div>
                      </div>
                    </div>
                  </div>
                </div>
              </blockquote>
              <blockquote type=3D"cite">
                <div dir=3D"ltr">
                  <div>
                    <div>
                      <div dir=3D"ltr">
                        <div dir=3D"ltr">
                          <div>
                            <div dir=3D"ltr">
                              <div>
                                <div> =C2=A0 too weak risks being mistaken
                                  for tacit approval.</div>
                              </div>
                            </div>
                          </div>
                        </div>
                      </div>
                    </div>
                    <br>
                  </div>
                </div>
                <br>
              </blockquote>
              <br>
              <blockquote type=3D"cite">
                <div class=3D"gmail_quote">
                  <div dir=3D"ltr" class=3D"gmail_attr">On Sat, May 2, 2020
                    at 7:35 AM Neal Cardwell &lt;ncardwell=3D<a href=3D"mai=
lto:40google.com@dmarc.ietf.org" target=3D"_blank">40google.com@dmarc.ietf.=
org</a>&gt;
                    wrote:<br>
                  </div>
                  <blockquote class=3D"gmail_quote" style=3D"margin:0px 0px=
 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">On Sat,=
 May 2,
                    2020 at 10:26 AM Michael Tuexen<br>
                    &lt;<a href=3D"mailto:michael.tuexen@lurchi.franken.de"=
 target=3D"_blank">michael.tuexen@lurchi.franken.de</a>&gt;
                    wrote:<br>
                    &gt;<br>
                    &gt; &gt; On 2. May 2020, at 16:10, Neal Cardwell
                    &lt;ncardwell=3D<a href=3D"mailto:40google.com@dmarc.ie=
tf.org" target=3D"_blank">40google.com@dmarc.ietf.org</a>&gt;
                    wrote:<br>
                    &gt; &gt;<br>
                    &gt; &gt; On Sat, May 2, 2020 at 5:40 AM Carles
                    Gomez Montenegro<br>
                    .....<br>
                    &gt; &gt;&gt; [CG] It seems that there are two
                    different areas where avoiding the<br>
                    &gt; &gt;&gt; typical Delayed ACKs approach may be
                    of particular interest: i) scenarios<br>
                    &gt; &gt;&gt; where very small window sizes are used
                    (cwnd=3D1), and ii) scenarios where<br>
                    &gt; &gt;&gt; large window sizes, and high ACK
                    ratios, can be used. We will reflect<br>
                    &gt; &gt;&gt; these considerations in the next draft
                    update.<br>
                    &gt; &gt;<br>
                    &gt; &gt; Yes, I wholeheartedly agree with Bob&#39;s
                    point that it&#39;s also extremely<br>
                    &gt; &gt; useful for TCP senders to be able to say
                    things like: &quot;I don&#39;t<br>
                    &gt; &gt; particularly mind which packets you ACK,
                    but there&#39;s no need to do it<br>
                    &gt; &gt; more often than 1:16&quot;. QUIC has a
                    mechanism like this, and it seems to<br>
                    &gt; &gt; allow significant CPU savings, even
                    allowing QUIC to run faster than<br>
                    &gt; &gt; TCP in some cases, AFAICT due to the
                    reduced overhead for handling<br>
                    &gt; &gt; ACKs:<br>
                    &gt; &gt;=C2=A0 <a href=3D"https://www.fastly.com/blog/=
measuring-quic-vs-tcp-computational-efficiency" rel=3D"noreferrer" target=
=3D"_blank">https://www.fastly.com/blog/measuring-quic-vs-tcp-computational=
-efficiency</a><br>
                    &gt; &gt;=C2=A0 <a href=3D"https://tools.ietf.org/html/=
draft-iyengar-quic-delayed-ack-00" rel=3D"noreferrer" target=3D"_blank">htt=
ps://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00</a><br>
                    &gt; &gt;<br>
                  </blockquote>
                </div>
              </blockquote>
              <p>QUIC doesn&#39;t *YET* have a mechanism to adjust the ACK
                Ratio, timing, or ACK content, it&#39;s likely an extension
                to the spec in the future. Anyway, I don&#39;t think TCP is
                the same as QUIC - TCP&#39;s notion of pacing is different;
                TCP&#39;s design differs, and there are many flavours and
                apps that use TCP in different ways.</p>
            </div>
          </blockquote>
          <div><br>
          </div>
          <div>I&#39;m curious about some of these comments.=C2=A0 I agree =
TCP
            and QUIC are different(otherwise why standardize QUIC), but
            TCP&#39;s notion of pacing is no different from QUIC&#39;s in m=
y
            opinion.=C2=A0 Both have various ways to implement pacing, and
            neither has any spec specifying exactly how it should be
            done.=C2=A0 For TCP, the closest I can find is <a href=3D"https=
://tools.ietf.org/html/rfc3449#section-4.6" target=3D"_blank">RFC3449</a>.=
=C2=A0 You seem to be
            implying that the set of apps which use TCP will always be
            wider than QUIC. <br>
          </div>
        </div>
      </div>
    </blockquote>
    QUIC seems to anticipate the need for pacing in its deisgn
    assumptions - I may be wrong? TCP has various recommendations that
    to me suggest pacing is a good idea (I would agree), but I expect
    lots of stacks do not implement burst mitigation.<br>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div class=3D"gmail_quote">
          <div> If anything, I suspect the opposite is true, since QUIC
            can enable applications where 100% reliability is
            undesirable.</div>
        </div>
      </div>
    </blockquote>
    <p>My comment was meant to say there are already a very wide range
      of protocols using TCP. From routing protocols, DNS, to other core
      protocols, to streaming video, to web, bulk transfer, IoT,
      transaction processing, etc. I was hoping to suggest that already
      many of these stacks have been optimised, and we are living with
      many decades of legacy and diversity. <br>
    </p>
    <p>On the other hand, I agree QUIC has applicability to datagrams
      for instance where reliability is not required for data, so I do
      need to learn more about how/whether these will handle CC and
      pacing, etc.=C2=A0 <br>
    </p>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">
        <div class=3D"gmail_quote">
          <blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8=
ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
            <div>
              <p> </p>
              <p>Changing the ACK policy and usage also brings lots of
                trade-offs: some may be good in some use-cases, but also
                can work out really bad if the design doesn&#39;t match the
                application/network path.</p>
            </div>
          </blockquote>
          <div>I completely agree.</div>
          <blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8=
ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
            <div>
              <blockquote type=3D"cite">
                <div class=3D"gmail_quote">
                  <blockquote class=3D"gmail_quote" style=3D"margin:0px 0px=
 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"> &gt; &=
gt; So I
                    agree with the usefulness of the two-use-case
                    taxonomy Carles has outlined:<br>
                    &gt; &gt;<br>
                    &gt; &gt; 1: &quot;please ACK now&quot;; because sender=
&#39;s cwnd
                    is low<br>
                    &gt; &gt;<br>
                    &gt; &gt; 2: &quot;ACK at frequency K&quot;; because se=
nder&#39;s
                    cwnd is high<br>
                    &gt; &gt;<br>
                    &gt; &gt; It seems those could both be addressed
                    cleanly with one or two TCP<br>
                    &gt; &gt; option kinds. A single TCP option kind
                    might suffice if the ACK<br>
                    &gt; &gt; frequency of 0 or 1 is taken to mean
                    &quot;please ACK now&quot;.<br>
                    &gt; I might be useful if the sender can trigger an
                    ACK for this particular packet,<br>
                    &gt; but leave the rate unchanged. And change the
                    rate from now on.<br>
                    &gt; If using an option, you could encode &quot;send an
                    ack for this packet, but don&#39;t<br>
                    &gt; change the ack rate&quot; by using 0, and encode
                    &quot;change the ack rate to 1 for n&quot;<br>
                    &gt; by using n, n &gt; 0.<br>
                    <br>
                    Yes, that sounds like a very nice idea.<br>
                    <br>
                    best,<br>
                    neal<br>
                    <br>
                    _______________________________________________<br>
                    tcpm mailing list<br>
                    <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm=
@ietf.org</a><br>
                    <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" =
rel=3D"noreferrer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/=
tcpm</a><br>
                  </blockquote>
                </div>
                <br>
                <fieldset><br>
                </fieldset>
              </blockquote>
              <p>I like that also. An option calling for an ACK-now,
                seems a little reminiscent of the way some stacks used
                &quot;PUSH&quot;. I can immediately see some good things ab=
out
                eliciting that immediate response for a particular
                sequence number.</p>
              <p>Gorry<br>
              </p>
              <blockquote type=3D"cite">
                <pre>_______________________________________________
tcpm mailing list
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tcpm</a>
</pre>
              </blockquote>
            </div>
            _______________________________________________<br>
            tcpm mailing list<br>
            <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.or=
g</a><br>
            <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"n=
oreferrer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/tcpm</a>=
<br>
          </blockquote>
        </div>
      </div>
    </blockquote>
    <pre cols=3D"72">--=20
G. Fairhurst, School of Engineering


Gorry

</pre>
  </div>

</blockquote></div>

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To: Ian Swett <ianswett@google.com>
Cc: Matt Mathis <mattmathis=40google.com@dmarc.ietf.org>, Neal Cardwell <ncardwell=40google.com@dmarc.ietf.org>, Michael Tuexen <michael.tuexen@lurchi.franken.de>, "tcpm@ietf.org" <tcpm@ietf.org>, Jon Crowcroft <jon.crowcroft@cl.cam.ac.uk>
References: <683902e8-a2af-cfb7-ffd0-c5c5742e5bd5@gmx.at> <CADVnQykY3OqXy=RcEfa-OpfK2x=W5_FTrdrx7PvKuqgEt92uNw@mail.gmail.com> <7d145f1203f6344b92f6f8aa11a78239.squirrel@webmail.entel.upc.edu> <CADVnQy=wPUx62y7VNqjSPP+snKX4vVvK5q=qqYb1j+0nGrtezQ@mail.gmail.com> <c6da08f4-03d7-da46-26b1-168f5953329f@bobbriscoe.net> <909de4172e46712f543f723d0ae2d638.squirrel@webmail.entel.upc.edu> <CADVnQynE3EMh-9qX7TkxifNvaKke7=PpWW2nB3Z6t1q7CYQXbg@mail.gmail.com> <9B5D12AC-248F-4E61-B6C5-1DA9529C7A42@lurchi.franken.de> <CADVnQy=gGzsvG2F935bM7wqGbbSZi+A5E=vxkpz=Bwc+ab2rkg@mail.gmail.com> <CAH56bmBNxknNvgOFajYJ-FuVXNU=Ra43UKyp-AEgD4m_p070pQ@mail.gmail.com> <54809559-05ef-b104-6cee-e61c8231dbae@erg.abdn.ac.uk> <CAKcm_gPtcsNPL5Og-fFveQgV0in+NbVVHJCTu+37a42_xA-Nyw@mail.gmail.com> <89b3c210-ac29-462e-daa1-373eeb4addf4@erg.abdn.ac.uk> <CAKcm_gN_44XCkgxuJLXGp1jMG4_rj=X48ro7LBEw-txPCHum8A@mail.gmail.com>
From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: Re: [tcpm] On Sender Control of Delayed Acks in TCP
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On 04/05/2020 15:03, Ian Swett wrote:
> Thanks for the clarification.  In implementation, I expect the status 
> of pacing to be extremely similar for TCP and QUIC.  Many stacks and 
> applications that do it for TCP will do it for QUIC, and those which 
> don't pace TCP probably won't bother with QUIC.

Probably so, time will tell.

Gorry

>
> On Mon, May 4, 2020 at 9:38 AM Gorry Fairhurst <gorry@erg.abdn.ac.uk 
> <mailto:gorry@erg.abdn.ac.uk>> wrote:
>
>
>     On 04/05/2020 14:14, Ian Swett wrote:
>>
>>
>>     On Mon, May 4, 2020 at 8:16 AM Gorry Fairhurst
>>     <gorry@erg.abdn.ac.uk <mailto:gorry@erg.abdn.ac.uk>> wrote:
>>
>>
>>         On 02/05/2020 15:55, Matt Mathis wrote:
>>>         Another probably small opportunity would be to make the RTT
>>>         estimators less noisy, by having the receiver always ACK the
>>>         first (or last) TSO segment.  The RTT is currently fuzzed by
>>>         both del ACK and TCP strides.
>>>
>>         I agree this might help: The TCP spec lets receivers do this
>>         right now, if they wish to?
>>>         Thanks,
>>>         --MM--
>>>         The best way to predict the future is to create it.  - Alan Kay
>>>
>>>         We must not tolerate intolerance;
>>>                however our response must be carefully measured:
>>>                     too strong would be hypocritical and risks
>>>         spiraling out of control;
>>>           too weak risks being mistaken for tacit approval.
>>>
>>>
>>
>>>         On Sat, May 2, 2020 at 7:35 AM Neal Cardwell
>>>         <ncardwell=40google.com@dmarc.ietf.org
>>>         <mailto:40google.com@dmarc.ietf.org>> wrote:
>>>
>>>             On Sat, May 2, 2020 at 10:26 AM Michael Tuexen
>>>             <michael.tuexen@lurchi.franken.de
>>>             <mailto:michael.tuexen@lurchi.franken.de>> wrote:
>>>             >
>>>             > > On 2. May 2020, at 16:10, Neal Cardwell
>>>             <ncardwell=40google.com@dmarc.ietf.org
>>>             <mailto:40google.com@dmarc.ietf.org>> wrote:
>>>             > >
>>>             > > On Sat, May 2, 2020 at 5:40 AM Carles Gomez Montenegro
>>>             .....
>>>             > >> [CG] It seems that there are two different areas
>>>             where avoiding the
>>>             > >> typical Delayed ACKs approach may be of particular
>>>             interest: i) scenarios
>>>             > >> where very small window sizes are used (cwnd=1),
>>>             and ii) scenarios where
>>>             > >> large window sizes, and high ACK ratios, can be
>>>             used. We will reflect
>>>             > >> these considerations in the next draft update.
>>>             > >
>>>             > > Yes, I wholeheartedly agree with Bob's point that
>>>             it's also extremely
>>>             > > useful for TCP senders to be able to say things
>>>             like: "I don't
>>>             > > particularly mind which packets you ACK, but there's
>>>             no need to do it
>>>             > > more often than 1:16". QUIC has a mechanism like
>>>             this, and it seems to
>>>             > > allow significant CPU savings, even allowing QUIC to
>>>             run faster than
>>>             > > TCP in some cases, AFAICT due to the reduced
>>>             overhead for handling
>>>             > > ACKs:
>>>             > >
>>>             https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency
>>>             > >
>>>             https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00
>>>             > >
>>>
>>         QUIC doesn't *YET* have a mechanism to adjust the ACK Ratio,
>>         timing, or ACK content, it's likely an extension to the spec
>>         in the future. Anyway, I don't think TCP is the same as QUIC
>>         - TCP's notion of pacing is different; TCP's design differs,
>>         and there are many flavours and apps that use TCP in
>>         different ways.
>>
>>
>>     I'm curious about some of these comments.  I agree TCP and QUIC
>>     are different(otherwise why standardize QUIC), but TCP's notion
>>     of pacing is no different from QUIC's in my opinion.  Both have
>>     various ways to implement pacing, and neither has any spec
>>     specifying exactly how it should be done. For TCP, the closest I
>>     can find is RFC3449
>>     <https://tools.ietf.org/html/rfc3449#section-4.6>. You seem to be
>>     implying that the set of apps which use TCP will always be wider
>>     than QUIC.
>     QUIC seems to anticipate the need for pacing in its deisgn
>     assumptions - I may be wrong? TCP has various recommendations that
>     to me suggest pacing is a good idea (I would agree), but I expect
>     lots of stacks do not implement burst mitigation.
>>     If anything, I suspect the opposite is true, since QUIC can
>>     enable applications where 100% reliability is undesirable.
>
>     My comment was meant to say there are already a very wide range of
>     protocols using TCP. From routing protocols, DNS, to other core
>     protocols, to streaming video, to web, bulk transfer, IoT,
>     transaction processing, etc. I was hoping to suggest that already
>     many of these stacks have been optimised, and we are living with
>     many decades of legacy and diversity.
>
>     On the other hand, I agree QUIC has applicability to datagrams for
>     instance where reliability is not required for data, so I do need
>     to learn more about how/whether these will handle CC and pacing, etc.
>
>>         Changing the ACK policy and usage also brings lots of
>>         trade-offs: some may be good in some use-cases, but also can
>>         work out really bad if the design doesn't match the
>>         application/network path.
>>
>>     I completely agree.
>>
>>>             > > So I agree with the usefulness of the two-use-case
>>>             taxonomy Carles has outlined:
>>>             > >
>>>             > > 1: "please ACK now"; because sender's cwnd is low
>>>             > >
>>>             > > 2: "ACK at frequency K"; because sender's cwnd is high
>>>             > >
>>>             > > It seems those could both be addressed cleanly with
>>>             one or two TCP
>>>             > > option kinds. A single TCP option kind might suffice
>>>             if the ACK
>>>             > > frequency of 0 or 1 is taken to mean "please ACK now".
>>>             > I might be useful if the sender can trigger an ACK for
>>>             this particular packet,
>>>             > but leave the rate unchanged. And change the rate from
>>>             now on.
>>>             > If using an option, you could encode "send an ack for
>>>             this packet, but don't
>>>             > change the ack rate" by using 0, and encode "change
>>>             the ack rate to 1 for n"
>>>             > by using n, n > 0.
>>>
>>>             Yes, that sounds like a very nice idea.
>>>
>>>             best,
>>>             neal
>>>
>>>             _______________________________________________
>>>             tcpm mailing list
>>>             tcpm@ietf.org <mailto:tcpm@ietf.org>
>>>             https://www.ietf.org/mailman/listinfo/tcpm
>>>
>>>
>>>
>>         I like that also. An option calling for an ACK-now, seems a
>>         little reminiscent of the way some stacks used "PUSH". I can
>>         immediately see some good things about eliciting that
>>         immediate response for a particular sequence number.
>>
>>         Gorry
>>
>>>         _______________________________________________
>>>         tcpm mailing list
>>>         tcpm@ietf.org  <mailto:tcpm@ietf.org>
>>>         https://www.ietf.org/mailman/listinfo/tcpm
>>         _______________________________________________
>>         tcpm mailing list
>>         tcpm@ietf.org <mailto:tcpm@ietf.org>
>>         https://www.ietf.org/mailman/listinfo/tcpm
>>
>     -- 
>     G. Fairhurst, School of Engineering
>
>
>     Gorry
>
-- 
G. Fairhurst, School of Engineering


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    <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
  </head>
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    <p><br>
    </p>
    <div class="moz-cite-prefix">On 04/05/2020 15:03, Ian Swett wrote:<br>
    </div>
    <blockquote type="cite"
cite="mid:CAKcm_gN_44XCkgxuJLXGp1jMG4_rj=X48ro7LBEw-txPCHum8A@mail.gmail.com">
      <meta http-equiv="content-type" content="text/html; charset=UTF-8">
      <div dir="ltr">Thanks for the clarification.  In implementation, I
        expect the status of pacing to be extremely similar for TCP and
        QUIC.  Many stacks and applications that do it for TCP will do
        it for QUIC, and those which don't pace TCP probably won't
        bother with QUIC.</div>
    </blockquote>
    <p>Probably so, time will tell.</p>
    <p>Gorry<br>
    </p>
    <blockquote type="cite"
cite="mid:CAKcm_gN_44XCkgxuJLXGp1jMG4_rj=X48ro7LBEw-txPCHum8A@mail.gmail.com"><br>
      <div class="gmail_quote">
        <div dir="ltr" class="gmail_attr">On Mon, May 4, 2020 at 9:38 AM
          Gorry Fairhurst &lt;<a href="mailto:gorry@erg.abdn.ac.uk"
            moz-do-not-send="true">gorry@erg.abdn.ac.uk</a>&gt; wrote:<br>
        </div>
        <blockquote class="gmail_quote" style="margin:0px 0px 0px
          0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
          <div>
            <p><br>
            </p>
            <div>On 04/05/2020 14:14, Ian Swett wrote:<br>
            </div>
            <blockquote type="cite">
              <div dir="ltr">
                <div dir="ltr"><br>
                </div>
                <br>
                <div class="gmail_quote">
                  <div dir="ltr" class="gmail_attr">On Mon, May 4, 2020
                    at 8:16 AM Gorry Fairhurst &lt;<a
                      href="mailto:gorry@erg.abdn.ac.uk" target="_blank"
                      moz-do-not-send="true">gorry@erg.abdn.ac.uk</a>&gt;
                    wrote:<br>
                  </div>
                  <blockquote class="gmail_quote" style="margin:0px 0px
                    0px 0.8ex;border-left:1px solid
                    rgb(204,204,204);padding-left:1ex">
                    <div>
                      <p><br>
                      </p>
                      <div>On 02/05/2020 15:55, Matt Mathis wrote:<br>
                      </div>
                      <blockquote type="cite">
                        <div dir="ltr">Another probably small
                          opportunity would be to make the RTT
                          estimators less noisy, by having the receiver
                          always ACK the first (or last) TSO segment. 
                           The RTT is currently fuzzed by both del ACK
                          and TCP strides.
                          <div><br clear="all">
                          </div>
                        </div>
                      </blockquote>
                      I agree this might help: The TCP spec lets
                      receivers do this right now, if they wish to? <br>
                      <blockquote type="cite">
                        <div dir="ltr">
                          <div>
                            <div>
                              <div dir="ltr">
                                <div dir="ltr">
                                  <div>
                                    <div dir="ltr">
                                      <div>
                                        <div dir="ltr">
                                          <div>Thanks,</div>
                                          --MM--<br>
                                          The best way to predict the
                                          future is to create it.  -
                                          Alan Kay<br>
                                          <br>
                                          We must not tolerate
                                          intolerance;</div>
                                        <div dir="ltr">       however
                                          our response must be carefully
                                          measured: </div>
                                        <div>            too strong
                                          would be hypocritical and
                                          risks spiraling out of
                                          control;</div>
                                      </div>
                                    </div>
                                  </div>
                                </div>
                              </div>
                            </div>
                          </div>
                        </div>
                      </blockquote>
                      <blockquote type="cite">
                        <div dir="ltr">
                          <div>
                            <div>
                              <div dir="ltr">
                                <div dir="ltr">
                                  <div>
                                    <div dir="ltr">
                                      <div>
                                        <div>   too weak risks being
                                          mistaken for tacit approval.</div>
                                      </div>
                                    </div>
                                  </div>
                                </div>
                              </div>
                            </div>
                            <br>
                          </div>
                        </div>
                        <br>
                      </blockquote>
                      <br>
                      <blockquote type="cite">
                        <div class="gmail_quote">
                          <div dir="ltr" class="gmail_attr">On Sat, May
                            2, 2020 at 7:35 AM Neal Cardwell
                            &lt;ncardwell=<a
                              href="mailto:40google.com@dmarc.ietf.org"
                              target="_blank" moz-do-not-send="true">40google.com@dmarc.ietf.org</a>&gt;
                            wrote:<br>
                          </div>
                          <blockquote class="gmail_quote"
                            style="margin:0px 0px 0px
                            0.8ex;border-left:1px solid
                            rgb(204,204,204);padding-left:1ex">On Sat,
                            May 2, 2020 at 10:26 AM Michael Tuexen<br>
                            &lt;<a
                              href="mailto:michael.tuexen@lurchi.franken.de"
                              target="_blank" moz-do-not-send="true">michael.tuexen@lurchi.franken.de</a>&gt;
                            wrote:<br>
                            &gt;<br>
                            &gt; &gt; On 2. May 2020, at 16:10, Neal
                            Cardwell &lt;ncardwell=<a
                              href="mailto:40google.com@dmarc.ietf.org"
                              target="_blank" moz-do-not-send="true">40google.com@dmarc.ietf.org</a>&gt;
                            wrote:<br>
                            &gt; &gt;<br>
                            &gt; &gt; On Sat, May 2, 2020 at 5:40 AM
                            Carles Gomez Montenegro<br>
                            .....<br>
                            &gt; &gt;&gt; [CG] It seems that there are
                            two different areas where avoiding the<br>
                            &gt; &gt;&gt; typical Delayed ACKs approach
                            may be of particular interest: i) scenarios<br>
                            &gt; &gt;&gt; where very small window sizes
                            are used (cwnd=1), and ii) scenarios where<br>
                            &gt; &gt;&gt; large window sizes, and high
                            ACK ratios, can be used. We will reflect<br>
                            &gt; &gt;&gt; these considerations in the
                            next draft update.<br>
                            &gt; &gt;<br>
                            &gt; &gt; Yes, I wholeheartedly agree with
                            Bob's point that it's also extremely<br>
                            &gt; &gt; useful for TCP senders to be able
                            to say things like: "I don't<br>
                            &gt; &gt; particularly mind which packets
                            you ACK, but there's no need to do it<br>
                            &gt; &gt; more often than 1:16". QUIC has a
                            mechanism like this, and it seems to<br>
                            &gt; &gt; allow significant CPU savings,
                            even allowing QUIC to run faster than<br>
                            &gt; &gt; TCP in some cases, AFAICT due to
                            the reduced overhead for handling<br>
                            &gt; &gt; ACKs:<br>
                            &gt; &gt;  <a
href="https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency"
                              rel="noreferrer" target="_blank"
                              moz-do-not-send="true">https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency</a><br>
                            &gt; &gt;  <a
                              href="https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00"
                              rel="noreferrer" target="_blank"
                              moz-do-not-send="true">https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00</a><br>
                            &gt; &gt;<br>
                          </blockquote>
                        </div>
                      </blockquote>
                      <p>QUIC doesn't *YET* have a mechanism to adjust
                        the ACK Ratio, timing, or ACK content, it's
                        likely an extension to the spec in the future.
                        Anyway, I don't think TCP is the same as QUIC -
                        TCP's notion of pacing is different; TCP's
                        design differs, and there are many flavours and
                        apps that use TCP in different ways.</p>
                    </div>
                  </blockquote>
                  <div><br>
                  </div>
                  <div>I'm curious about some of these comments.  I
                    agree TCP and QUIC are different(otherwise why
                    standardize QUIC), but TCP's notion of pacing is no
                    different from QUIC's in my opinion.  Both have
                    various ways to implement pacing, and neither has
                    any spec specifying exactly how it should be done. 
                    For TCP, the closest I can find is <a
                      href="https://tools.ietf.org/html/rfc3449#section-4.6"
                      target="_blank" moz-do-not-send="true">RFC3449</a>. 
                    You seem to be implying that the set of apps which
                    use TCP will always be wider than QUIC. <br>
                  </div>
                </div>
              </div>
            </blockquote>
            QUIC seems to anticipate the need for pacing in its deisgn
            assumptions - I may be wrong? TCP has various
            recommendations that to me suggest pacing is a good idea (I
            would agree), but I expect lots of stacks do not implement
            burst mitigation.<br>
            <blockquote type="cite">
              <div dir="ltr">
                <div class="gmail_quote">
                  <div> If anything, I suspect the opposite is true,
                    since QUIC can enable applications where 100%
                    reliability is undesirable.</div>
                </div>
              </div>
            </blockquote>
            <p>My comment was meant to say there are already a very wide
              range of protocols using TCP. From routing protocols, DNS,
              to other core protocols, to streaming video, to web, bulk
              transfer, IoT, transaction processing, etc. I was hoping
              to suggest that already many of these stacks have been
              optimised, and we are living with many decades of legacy
              and diversity. <br>
            </p>
            <p>On the other hand, I agree QUIC has applicability to
              datagrams for instance where reliability is not required
              for data, so I do need to learn more about how/whether
              these will handle CC and pacing, etc.  <br>
            </p>
            <blockquote type="cite">
              <div dir="ltr">
                <div class="gmail_quote">
                  <blockquote class="gmail_quote" style="margin:0px 0px
                    0px 0.8ex;border-left:1px solid
                    rgb(204,204,204);padding-left:1ex">
                    <div>
                      <p> </p>
                      <p>Changing the ACK policy and usage also brings
                        lots of trade-offs: some may be good in some
                        use-cases, but also can work out really bad if
                        the design doesn't match the application/network
                        path.</p>
                    </div>
                  </blockquote>
                  <div>I completely agree.</div>
                  <blockquote class="gmail_quote" style="margin:0px 0px
                    0px 0.8ex;border-left:1px solid
                    rgb(204,204,204);padding-left:1ex">
                    <div>
                      <blockquote type="cite">
                        <div class="gmail_quote">
                          <blockquote class="gmail_quote"
                            style="margin:0px 0px 0px
                            0.8ex;border-left:1px solid
                            rgb(204,204,204);padding-left:1ex"> &gt;
                            &gt; So I agree with the usefulness of the
                            two-use-case taxonomy Carles has outlined:<br>
                            &gt; &gt;<br>
                            &gt; &gt; 1: "please ACK now"; because
                            sender's cwnd is low<br>
                            &gt; &gt;<br>
                            &gt; &gt; 2: "ACK at frequency K"; because
                            sender's cwnd is high<br>
                            &gt; &gt;<br>
                            &gt; &gt; It seems those could both be
                            addressed cleanly with one or two TCP<br>
                            &gt; &gt; option kinds. A single TCP option
                            kind might suffice if the ACK<br>
                            &gt; &gt; frequency of 0 or 1 is taken to
                            mean "please ACK now".<br>
                            &gt; I might be useful if the sender can
                            trigger an ACK for this particular packet,<br>
                            &gt; but leave the rate unchanged. And
                            change the rate from now on.<br>
                            &gt; If using an option, you could encode
                            "send an ack for this packet, but don't<br>
                            &gt; change the ack rate" by using 0, and
                            encode "change the ack rate to 1 for n"<br>
                            &gt; by using n, n &gt; 0.<br>
                            <br>
                            Yes, that sounds like a very nice idea.<br>
                            <br>
                            best,<br>
                            neal<br>
                            <br>
_______________________________________________<br>
                            tcpm mailing list<br>
                            <a href="mailto:tcpm@ietf.org"
                              target="_blank" moz-do-not-send="true">tcpm@ietf.org</a><br>
                            <a
                              href="https://www.ietf.org/mailman/listinfo/tcpm"
                              rel="noreferrer" target="_blank"
                              moz-do-not-send="true">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
                          </blockquote>
                        </div>
                        <br>
                        <fieldset><br>
                        </fieldset>
                      </blockquote>
                      <p>I like that also. An option calling for an
                        ACK-now, seems a little reminiscent of the way
                        some stacks used "PUSH". I can immediately see
                        some good things about eliciting that immediate
                        response for a particular sequence number.</p>
                      <p>Gorry<br>
                      </p>
                      <blockquote type="cite">
                        <pre>_______________________________________________
tcpm mailing list
<a href="mailto:tcpm@ietf.org" target="_blank" moz-do-not-send="true">tcpm@ietf.org</a>
<a href="https://www.ietf.org/mailman/listinfo/tcpm" target="_blank" moz-do-not-send="true">https://www.ietf.org/mailman/listinfo/tcpm</a>
</pre>
                      </blockquote>
                    </div>
                    _______________________________________________<br>
                    tcpm mailing list<br>
                    <a href="mailto:tcpm@ietf.org" target="_blank"
                      moz-do-not-send="true">tcpm@ietf.org</a><br>
                    <a href="https://www.ietf.org/mailman/listinfo/tcpm"
                      rel="noreferrer" target="_blank"
                      moz-do-not-send="true">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
                  </blockquote>
                </div>
              </div>
            </blockquote>
            <pre cols="72">-- 
G. Fairhurst, School of Engineering


Gorry

</pre>
          </div>
        </blockquote>
      </div>
    </blockquote>
    <pre class="moz-signature" cols="72">-- 
G. Fairhurst, School of Engineering</pre>
  </body>
</html>

--------------81527E3B048AEE1C140DF93B--


From nobody Mon May  4 10:37:03 2020
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From: "Scharf, Michael" <Michael.Scharf@hs-esslingen.de>
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Date: Mon, 4 May 2020 17:36:53 +0000
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Hi all,

The material of the past interim meeting can be found at: https://datatrack=
er.ietf.org/meeting/interim-2020-tcpm-01/session/tcpm

I have uploaded draft minutes based on the excellent Etherpad notes from Ri=
chard Richard Scheffenegger (thanks!): https://datatracker.ietf.org/meeting=
/interim-2020-tcpm-01/materials/minutes-interim-2020-tcpm-01-202004291600

Please let the chairs know if anything needs to be fixed therein.

Thanks

Michael


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On Sat, May 2, 2020 at 8:21 PM Vidhi Goel <vidhi_goel@apple.com> wrote:
>
> Hi Carles,
>
> >> I agree. We in fact implemented Linux to increase cwnd by the exact
> >> amount sequence (in unit of packets) so slow-start increase function i=
s
> >> regardless of ack-thinning effect but purely based on amount
> >> delivered. This has an important effect on
> >> Wireless as the other ack talk mentions. It also helps in case of ACK
> >> losses or reordering.
> >
> > Agreed!
> >
> > By the way, in the next draft update, we will add that, despite its
> > "Experimental" status, ABC is supported in many implementations.
>
> Some time ago, I had brought up this same problem with ABC (RFC 3465) tha=
t increasing by at most one MSS during slow start makes the growth some wha=
t non-exponential in cases where ACK ratio < 1 (for example, receivers doin=
g GRO or ACK compression, ACK losses etc). To allow an exponential growth, =
we should certainly increase congestion window by the amount of bytes ACKed=
 (what Linux does).

+1
I do want to mention that there's a legit concern of always
increasing based on #bytes: slow start (or generally congestion
control) can be more bursty with stretched-ACKs. IMHO idle-restart
bursts are just as bad and probably even more frequent with many TCP
senders disable slow-start after-idle and don't bother to pace. The
best form to de-burst is through (good) pacers like Linux's fq/pacing.
In other words congestion control and burst control should not be
juggled together via only congestion window changes.



>
>
> Vidhi
>
> > On May 2, 2020, at 1:34 AM, Carles Gomez Montenegro <carlesgo@entel.upc=
.edu> wrote:
> >
> > Hi Yuchung,
> >
> > <snip>
> >
> >>> Note that the problem pointed out is that, as stated in the RFC 5681
> >>> text
> >>> above:
> >>>
> >>>   "During slow start, a TCP increments cwnd by at most SMSS bytes for
> >>>    each ACK received that cumulatively acknowledges new data"
> >>>
> >>> It may be desirable to support, during slow start, a cwnd increase by
> >>> more
> >>> than SMSS per each ACK acknowledging new data.
> >> I agree. We in fact implemented Linux to increase cwnd by the exact
> >> amount sequence (in unit of packets) so slow-start increase function i=
s
> >> regardless of ack-thinning effect but purely based on amount
> >> delivered. This has an important effect on
> >> Wireless as the other ack talk mentions. It also helps in case of ACK
> >> losses or reordering.
> >
> > Agreed!
> >
> > By the way, in the next draft update, we will add that, despite its
> > "Experimental" status, ABC is supported in many implementations.
> >
> > <snip>
> >
> > Thanks,
> >
> > Carles
> >
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
>


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From: juhamatk@gmail.com
Date: Tue, 5 May 2020 10:25:12 +0300
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To: Joseph Touch <touch@strayalpha.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/tcpm/XxrnRDPyZpfcmsNai9D_V0vkiA8>
Subject: Re: [tcpm] RFC 5925 SNE algorithm concern
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Hello,

And there seems to be a minor parenthesis mismatch in the pseudo-code
of the improved algorithm I sent earlier. Here is a new fixed version
below for the record.

Improved algorithm suggestion pseudo-code:
     PRE =3D 0;
     // set the flag when the SEG.SEQ first rolls over
     if ((RCV.SNE_FLAG =3D=3D 0)
        && (RCV.PREV_SEQ > 0x7fffffff) && (SEG.SEQ < 0x7fffffff)) {
          RCV.SNE =3D RCV.SNE + 1;
          RCV.SNE_FLAG =3D 1;
     }
     // decide which SNE to use after incremented,
     // pre-increment is used for retransmits
     if ((RCV.SNE_FLAG =3D=3D 1) && (SEG.SEQ > 0x7fffffff)) {
        SNE =3D RCV.SNE - 1; // use the pre-increment value
        PRE =3D 1;
     } else {
        SNE =3D RCV.SNE; // use the current value
     }
     // reset the flag in the *middle* of the window,
     // include SEG.SEQ equals half so that we do not miss
     // that edge case
     if (PRE =3D=3D 0) {
        if(RCV.PREV_SEQ < 0x7fffffff && SEG.SEQ >=3D 0x7fffffff) {
           RCV.SNE_FLAG =3D 0;
        }
        // save the current SEQ for the next time through the code,
        // if we are not retransmitting
        RCV.PREV_SEQ =3D SEG.SEQ;
     }

BR,
--
 Juhamatti


On Thu, 30 Apr 2020 at 19:43, <juhamatk@gmail.com> wrote:
>
>
> Hello Joe,
>
> Thanks for your reply. While I talk about retransmit, I refer actually to=
 the other side retransmitting. So this is the receive side pseudo code. Do=
es that change the situation?
>
> Thanks for the clarifications,
> --
>  Juhamatti
>
> to 30. huhtikuuta 2020 klo 17.31 Joseph Touch <touch@strayalpha.com> kirj=
oitti:
>>
>> Hi, Juhamatti,
>>
>> See embedded below:
>>
>> > On Apr 30, 2020, at 12:31 AM, juhamatk@gmail.com wrote:
>> >
>> > Hello all,
>> >
>> > While working with RFC 5925, I have been checking the details of
>> > example SNE algorithm. I have a slight concern with the example
>> > algorithm proposed by the RFC.
>> >
>> > The idea of SNE is to simulate 64-bit sequence number with 32-bit
>> > sequences to prevent replay attacks. RFC 5925 proposes the following
>> > algorithm to implement 64-bit sequence numbering:
>> >
>> > Original algorithm pseudo-code in the RFC 5925 + errata:
>> >     /* set the flag when the SEG.SEQ first rolls over */
>> >     if ((RCV.SNE_FLAG =3D=3D 0)
>> >        && (RCV.PREV_SEQ > 0x7fffffff) && (SEG.SEQ < 0x7fffffff)) {
>> >           RCV.SNE =3D RCV.SNE + 1;
>> >           RCV.SNE_FLAG =3D 1;
>> >     }
>> >     /* decide which SNE to use after incremented */
>> >     if ((RCV.SNE_FLAG =3D=3D 1) && (SEG.SEQ > 0x7fffffff)) {
>> >        SNE =3D RCV.SNE - 1; # use the pre-increment value
>> >     } else {
>> >        SNE =3D RCV.SNE; # use the current value
>> >     }
>> >     /* reset the flag in the *middle* of the window */
>> >     if ((RCV.PREV_SEQ < 0x7fffffff) && (SEG.SEQ > 0x7fffffff)) {
>> >        RCV.SNE_FLAG =3D 0;
>> >     }
>> >     /* save the current SEQ for the next time through the code */
>> >     RCV.PREV_SEQ =3D SEG.SEQ;
>> >
>> > When looking into this in detail, to me it looks like there might be
>> > an issue when 32-bit sequence wraps back to the beginning and a
>> > retransmission occurs with the high sequence range.
>>
>> Retransmission should always use the value cached with that segment, so =
there should not be an issue.
>>
>> > Then SNE is
>> > increased unexpectedly and does not follow proper sequencing. A test
>> > case below:
>> >
>> > State:
>> > PREV_SEQ =3D 0xffffffe
>> > RCV.SNE =3D 0
>> > RCV.SNE_FLAG =3D 0
>> >
>> > New frame:
>> > SEG.SEQ =3D 0x0
>> > RCV.SNE =3D> 1
>> > RCV.SNE_FLAG =3D> 1
>> > PREV_SEQ =3D> 0
>> >
>> > Retransmit:
>> > SEG.SEQ =3D 0xfffffffd
>> > RCV.SNE_FLAG =3D> 0 - incorrectly updated
>> > PREV_SEQ =3D> 0xfffffffd - incorrectly updated
>>
>> You=E2=80=99re using the pseudocode that creates new SNEs with a previou=
sly transmitted value. It=E2=80=99s acting like you=E2=80=99ve wrapped too =
quickly.
>>
>> >
>> > New frame:
>> > SEG.SEQ =3D 0x2
>> > RCV.SNE =3D> 2 - incorrect value, should be 1 based on sequencing
>> > RCV.SNE_FLAG =3D> 1
>> > PREV_SEQ =3D> 0x2
>> >
>> > I could not figure out any reason why the above sequence would not be
>> > valid.
>>
>> Because retransmission should be using the values cached when the origin=
al transmission occurred.
>>
>> I don=E2=80=99t think the new code is needed - at least from this exampl=
e.
>>
>> Joe
>>
>> > Even though it is mentioned in the RFC that the algorithm is
>> > just an example, the sequencing mismatch is unfortunate as mismatching
>> > SNEs on different sides will bring the authenticated TCP session down
>> > and cause data loss through timeout. Thus, for interoperability, it is
>> > crucial that correct sequencing is used on both ends.
>> >
>> > If above is really a real problem, then perhaps an improved algorithm
>> > could be as follows:
>> >
>> > Improved algorithm suggestion pseudo-code:
>> >     PRE =3D 0;
>> >     // set the flag when the SEG.SEQ first rolls over
>> >     if ((RCV.SNE_FLAG =3D=3D 0)
>> >        && (RCV.PREV_SEQ > 0x7fffffff) && (SEG.SEQ < 0x7fffffff)) {
>> >          RCV.SNE =3D RCV.SNE + 1;
>> >          RCV.SNE_FLAG =3D 1;
>> >     }
>> >     // decide which SNE to use after incremented,
>> >     // pre-increment is used for retransmits
>> >     if ((RCV.SNE_FLAG =3D=3D 1) && (SEG.SEQ > 0x7fffffff)) {
>> >        SNE =3D RCV.SNE - 1; # use the pre-increment value
>> >        PRE =3D 1;
>> >     } else {
>> >        SNE =3D RCV.SNE; # use the current value
>> >     }
>> >     // reset the flag in the *middle* of the window,
>> >     // include SEG.SEQ equals half so that we do not miss
>> >     // that edge case
>> >     if (PRE =3D=3D 0) {
>> >        if(RCV.SNE_FLAG =3D=3D 1 &&
>> >           RCV.PREV_SEQ < 0x7fffffff) && (SEG.SEQ >=3D 0x7fffffff)) {
>> >           RCV.SNE_FLAG =3D 0;
>> >        }
>> >        // save the current SEQ for the next time through the code,
>> >        // if we are not retransmitting
>> >        RCV.PREV_SEQ =3D SEG.SEQ;
>> >     }
>> >
>> > This passes the problematic case (and bunch of others I have tested).
>> > This also includes a small additional fix in the middle window
>> > handling where SEG.SEQ was missing one edge value (SEG.SEQ >
>> > 0x7fffffff =3D> SEG.SEQ >=3D 0x7fffffff). I hope I have not missed
>> > anything else, but it is certainly possible, so comments are welcome.
>> >
>> > So, what do you think, is this a real concern? If it is a real issue,
>> > what would be the best way to handle this, perhaps an errata with an
>> > improved algorithm after throughout review of the new algorithm?
>> >
>> > Thank you,
>> > --
>> > Juhamatti
>> >
>> > _______________________________________________
>> > tcpm mailing list
>> > tcpm@ietf.org
>> > https://www.ietf.org/mailman/listinfo/tcpm
>>


From nobody Tue May  5 08:31:45 2020
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Subject: [tcpm] Fwd: Reminder: Survey on planning for possible online IETF meetings
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> Begin forwarded message:
>=20
> From: Alissa Cooper <alissa@cooperw.in>
> Subject: Fwd: Reminder: Survey on planning for possible online IETF =
meetings
> Date: 5. May 2020 at 13:48:03 CEST
> To: IETF WG Chairs <wgchairs@ietf.org>
>=20
> Please circulate this to your working group lists. The survey data =
will be very important as we plan future IETF meetings.
Done.

Best regards
Michael
>=20
> Thanks,
> Alissa
>=20
>> Begin forwarded message:
>>=20
>> From: IETF Executive Director <exec-director@ietf.org>
>> Subject: Reminder: Survey on planning for possible online IETF =
meetings
>> Date: May 4, 2020 at 3:03:35 AM EDT
>> To: "IETF Announcement List" <ietf-announce@ietf.org>
>> Reply-To: ietf108planning@ietf.org
>>=20
>> This is a reminder that we need the IETF community to help us plan =
for the possibility that one or more upcoming IETF meetings in 2020 and =
possibly 2021 may not be able to go ahead in person.  You can help us =
with this by filling out the following survey:=20
>>=20
>> 	https://www.surveymonkey.com/r/5328FFJ
>>=20
>> So far we have 114 responses and we would ideally like 500 or more.
>>=20
>> The survey contains the following pages and will take 15-20 minutes =
to complete:
>>=20
>> 1. Welcome
>> 2. Online IETF 107 and the subsequent virtual interims
>> 3. Replacing a cancelled in-person meeting
>> 4. Online meeting format and timezone
>> 5. Replicating humming
>> 6. Replicating the hallway environment
>> 7. Fees
>> 8. Thanks and anything else
>>=20
>> We run the survey in anonymous mode which means that we only see data =
that you explicitly provide.
>>=20
>> Thank you in advance for your help.
>>=20
>> --=20
>> Alissa Cooper, IETF Chair
>> Jay Daley, IETF Executive Director
>> Colin Perkins, IRTF Chair
>>=20
>> _______________________________________________
>> IETF-Announce mailing list
>> IETF-Announce@ietf.org
>> https://www.ietf.org/mailman/listinfo/ietf-announce
>=20


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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-rto-consider-11.txt
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From nobody Wed May  6 12:07:38 2020
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Subject: Re: [tcpm] RFC 5925 SNE algorithm concern
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--Apple-Mail=_F8BAE5A5-1401-4BC7-9935-3D86928C0329
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	charset=utf-8

Hi, all,

Juhamatti pointed out a flaw in the pseudocode for receive-side SNE =
computation in RFC 5925.

The good news (in a sense) is that the code was not normative;=20

   The implementation of SNEs is not specified in this document, but one
   possible way is described here that can be used for either RCV.SNE,
   SND.SNE, or both.

The corrections that Juhamatti posted don=E2=80=99t address the key =
issue, which is that there are several possible interpretations of any =
received SEQ. There is no simple way to fix this except to handle all =
the cases (which the code below does - I believe minimally):

	- it is in-order
		- moving the receive window forward
		- incrementing the SNE if it wraps the number space
	- it is out-of-order
		- which might use either the current SNE or SNE-1, =
depending on whether it is pre-wrap of the number space

Because the number space wraps, all received SEQ could be interpreted as =
in-order or out-of-order; the only way to know the difference is the =
distance between the two =E2=80=94 which cannot be larger than 0x7FFF =
FFFF (half the number space minus 1).

The following is code that covers the cases and passes all current =
tests. Given this is not as obvious as it might seem and might be more =
useful than for just RFC 5925 and the SNE alg in 5925 is =E2=80=9Cnot =
specified=E2=80=9D anyway, I propose to submit this as a stand-alone =
doc.

So here are the ways forward:

	1) have that doc =E2=80=9CUPDATE=E2=80=9D RFC 5925
		this is the cleanest because it would more definitely be =
caught by implementers
		but it requires the new alg be standards-track, which =
seems odd (though we could say what SNE is as a standard
		and still leave its implementation as informative rather =
than normative)

	2) publish that doc as informational
		then we could issue an errata on 5925 to point to the =
new RFC

	3) issue the new alg as an errata to 5925 only

Any thoughts?

Joe

=E2=80=94=E2=80=94

PS - here=E2=80=99s the alg:


#DEFINE HALF (0x80000000)

SNE =3D RCV_SNE;                               // use current SNE (as =
default)
if (distance(SEG_SEQ,RCV_PREV_SEQ) < HALF) { // both in same SNE range?
  if ((SEG_SEQ >=3D HALF)=20
      && (RCV_PREV_SEQ < HALF)) {            // jumps fwd over N/2?
    RCV_SNE_FLAG =3D 0;                        // reset wrap increment =
flag
  }
  RCV_PREV_SEQ =3D MAX(SEG_SEQ,RCV_PREV_SEQ);  // move prev forward if =
needed
} else {                                     // both in diff SNE ranges
  if (SEG_SEQ < HALF) {                      // jumps forward over zero?
    RCV_PREV_SEQ =3D SEG_SEQ;                  // update prev
    if (RCV_SNE_FLAG =3D=3D 0) {.                // first jump over =
zero? (wrap)
      RCV_SNE_FLAG =3D 1;                      // set flag so we =
increment once
      RCV_SNE =3D RCV_SNE + 1;                 // increment window
      SNE =3D RCV_SNE;                         // use updated SNE value
    }
  } else {                                   // jump backward over zero
    SNE =3D RCV_SNE - 1;                       // use pre-rollover SNE =
value
  }
}=

--Apple-Mail=_F8BAE5A5-1401-4BC7-9935-3D86928C0329
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	charset=utf-8

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; line-break: after-white-space;" class=3D"">Hi, =
all,<div class=3D""><br class=3D""></div><div class=3D"">Juhamatti =
pointed out a flaw in the pseudocode for receive-side SNE computation in =
RFC 5925.</div><div class=3D""><br class=3D""></div><div class=3D"">The =
good news (in a sense) is that the code was not =
normative;&nbsp;</div><div class=3D""><br class=3D""></div><div =
class=3D""><pre class=3D"newpage" style=3D"font-size: =
13.333333015441895px; margin-top: 0px; margin-bottom: 0px; break-before: =
page;">   The implementation of SNEs is not specified in this document, =
but one
   possible way is described here that can be used for either RCV.SNE,
   SND.SNE, or both.</pre></div><div class=3D""><div><br =
class=3D""></div><div>The corrections that Juhamatti posted don=E2=80=99t =
address the key issue, which is that there are several possible =
interpretations of any received SEQ. There is no simple way to fix this =
except to handle all the cases (which the code below does - I believe =
minimally):</div><div><br class=3D""></div><div><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>- it is =
in-order</div><div><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">		</span>- moving the receive =
window forward</div><div><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">		</span>-&nbsp;incrementing the =
SNE if it wraps the number space</div><div><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>- it is =
out-of-order</div><div><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">		</span>- which might use either =
the current SNE or SNE-1, depending on whether it is pre-wrap of the =
number space</div><div><br class=3D""></div><div>Because the number =
space wraps, all received SEQ could be interpreted as in-order or =
out-of-order; the only way to know the difference is the distance =
between the two =E2=80=94 which cannot be larger than 0x7FFF FFFF (half =
the number space minus 1).</div><div><br class=3D""></div><div>The =
following is code that covers the cases and passes all current tests. =
Given this is not as obvious as it might seem and might be more useful =
than for just RFC 5925 and the SNE alg in 5925 is =E2=80=9Cnot =
specified=E2=80=9D anyway, I propose to submit this as a stand-alone =
doc.</div><div><br class=3D""></div><div>So here are the ways =
forward:</div><div><br class=3D""></div><div><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>1) have =
that doc =E2=80=9CUPDATE=E2=80=9D RFC 5925</div><div><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">		=
</span>this is the cleanest because it would more definitely be caught =
by implementers</div><div><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">		</span>but it requires the new =
alg be standards-track, which seems odd (though we could say what SNE is =
as a standard</div><div><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">		</span>and still leave its =
implementation as informative rather than normative)</div><div><br =
class=3D""></div><div><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>2) publish that doc as =
informational</div><div><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">		</span>then we could issue an =
errata on 5925 to point to the new RFC</div><div><br =
class=3D""></div><div><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>3) issue the new alg as an errata =
to 5925 only</div><div><br class=3D""></div><div>Any =
thoughts?</div><div><br class=3D""></div><div>Joe</div><div><br =
class=3D""></div><div>=E2=80=94=E2=80=94</div><div><br =
class=3D""></div><div>PS - here=E2=80=99s the alg:</div><div><br =
class=3D""></div><div><br class=3D""></div></div><div><span =
style=3D"color: rgba(0, 0, 0, 0.85098); font-family: Menlo; =
background-color: rgb(255, 255, 255);" class=3D"">#DEFINE HALF =
(0x80000000)</span></div><div><span style=3D"color: rgba(0, 0, 0, =
0.85098); font-family: Menlo; background-color: rgb(255, 255, 255);" =
class=3D""><br class=3D""></span></div><div><span style=3D"color: =
rgba(0, 0, 0, 0.85098); font-family: Menlo; background-color: rgb(255, =
255, 255);" class=3D"">SNE =3D RCV_SNE; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&nbsp;</span><span style=3D"font-family: Menlo; background-color: =
rgb(255, 255, 255); color: rgb(93, 108, 121);" class=3D"">// use current =
SNE (as default)</span></div><div style=3D"margin: 0px; font-stretch: =
normal; line-height: normal; font-family: Menlo; color: rgba(0, 0, 0, =
0.85098); background-color: rgb(255, 255, 255);" class=3D""><span =
style=3D"color: #9b2393" class=3D"">if</span> =
(distance(SEG_SEQ,RCV_PREV_SEQ) &lt; <span style=3D"color: #1c00cf" =
class=3D"">HALF</span>) {&nbsp;<span style=3D"color: #5d6c79" =
class=3D"">// both in same SNE range?</span></div><div style=3D"margin: =
0px; font-stretch: normal; line-height: normal; font-family: Menlo; =
background-color: rgb(255, 255, 255);" class=3D"">&nbsp; if<span =
style=3D"color: rgba(0, 0, 0, 0.85098);" class=3D""> ((SEG_SEQ &gt;=3D =
</span><span style=3D"color: rgb(28, 0, 207);" class=3D"">HALF</span><span=
 style=3D"color: rgba(0, 0, 0, 0.85098);" class=3D"">)</span><span =
style=3D"color: rgba(0, 0, 0, 0.85098);" =
class=3D"">&nbsp;</span></div><div style=3D"margin: 0px; font-stretch: =
normal; line-height: normal; font-family: Menlo; background-color: =
rgb(255, 255, 255);" class=3D""><span style=3D"color: rgba(0, 0, 0, =
0.85098);" class=3D"">&nbsp; &nbsp; &nbsp; &amp;&amp; (RCV_PREV_SEQ &lt; =
</span><span style=3D"color: rgb(28, 0, 207);" class=3D"">HALF</span><span=
 style=3D"color: rgba(0, 0, 0, 0.85098);" class=3D"">)) { &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp;</span><span style=3D"color: rgb(93, 108, =
121);" class=3D"">// jumps fwd over N/2?</span></div><div style=3D"margin:=
 0px; font-stretch: normal; line-height: normal; font-family: Menlo; =
color: rgba(0, 0, 0, 0.85098); background-color: rgb(255, 255, 255);" =
class=3D"">&nbsp; &nbsp; RCV_SNE_FLAG =3D <span style=3D"color: #1c00cf" =
class=3D"">0</span>; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<span style=3D"color: #5d6c79" =
class=3D"">// reset wrap increment flag</span></div><div style=3D"margin: =
0px; font-stretch: normal; line-height: normal; font-family: Menlo; =
color: rgba(0, 0, 0, 0.85098); background-color: rgb(255, 255, 255);" =
class=3D"">&nbsp; }</div><div style=3D"margin: 0px; font-stretch: =
normal; line-height: normal; font-family: Menlo; color: rgba(0, 0, 0, =
0.85098); background-color: rgb(255, 255, 255);" class=3D"">&nbsp; =
RCV_PREV_SEQ =3D MAX(SEG_SEQ,RCV_PREV_SEQ); &nbsp;<span style=3D"color: =
#5d6c79" class=3D"">// move prev forward if needed</span></div><div =
style=3D"margin: 0px; font-stretch: normal; line-height: normal; =
font-family: Menlo; color: rgba(0, 0, 0, 0.85098); background-color: =
rgb(255, 255, 255);" class=3D"">} <span style=3D"color: #9b2393" =
class=3D"">else</span> {&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;<span style=3D"color: #5d6c79" class=3D"">// both in diff =
SNE ranges</span></div><div style=3D"margin: 0px; font-stretch: normal; =
line-height: normal; font-family: Menlo; background-color: rgb(255, 255, =
255);" class=3D"">&nbsp; if<span style=3D"color: rgba(0, 0, 0, =
0.85098);" class=3D""> (SEG_SEQ &lt; </span><span style=3D"color: =
rgb(28, 0, 207);" class=3D"">HALF</span><span style=3D"color: rgba(0, 0, =
0, 0.85098);" class=3D"">) { &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;</span><span style=3D"color: rgb(93, =
108, 121);" class=3D"">// jumps forward over zero?</span></div><div =
style=3D"margin: 0px; font-stretch: normal; line-height: normal; =
font-family: Menlo; color: rgba(0, 0, 0, 0.85098); background-color: =
rgb(255, 255, 255);" class=3D"">&nbsp; &nbsp; RCV_PREV_SEQ =3D SEG_SEQ; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<span =
style=3D"color: #5d6c79" class=3D"">// update prev</span></div><div =
style=3D"margin: 0px; font-stretch: normal; line-height: normal; =
font-family: Menlo; background-color: rgb(255, 255, 255);" =
class=3D"">&nbsp; &nbsp; if<span style=3D"color: rgba(0, 0, 0, =
0.85098);" class=3D""> (RCV_SNE_FLAG =3D=3D </span><span style=3D"color: =
rgb(28, 0, 207);" class=3D"">0</span><span style=3D"color: rgba(0, 0, 0, =
0.85098);" class=3D"">) {. &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;/</span><span style=3D"color: rgb(93, 108, 121);" =
class=3D"">/ first jump over zero? (wrap)</span></div><div =
style=3D"margin: 0px; font-stretch: normal; line-height: normal; =
font-family: Menlo; color: rgba(0, 0, 0, 0.85098); background-color: =
rgb(255, 255, 255);" class=3D"">&nbsp; &nbsp; &nbsp; RCV_SNE_FLAG =3D =
<span style=3D"color: #1c00cf" class=3D"">1</span>; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<span =
style=3D"color: #5d6c79" class=3D"">// set flag so we increment =
once</span></div><div style=3D"margin: 0px; font-stretch: normal; =
line-height: normal; font-family: Menlo; color: rgba(0, 0, 0, 0.85098); =
background-color: rgb(255, 255, 255);" class=3D"">&nbsp; &nbsp; &nbsp; =
RCV_SNE =3D RCV_SNE + <span style=3D"color: #1c00cf" class=3D"">1</span>; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&nbsp;<span =
style=3D"color: #5d6c79" class=3D"">// increment window</span></div><div =
style=3D"margin: 0px; font-stretch: normal; line-height: normal; =
font-family: Menlo; color: rgba(0, 0, 0, 0.85098); background-color: =
rgb(255, 255, 255);" class=3D"">&nbsp; &nbsp; &nbsp; SNE =3D RCV_SNE; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;&nbsp;<span style=3D"color: #5d6c79" class=3D"">// use =
updated SNE value</span></div><div style=3D"margin: 0px; font-stretch: =
normal; line-height: normal; font-family: Menlo; color: rgba(0, 0, 0, =
0.85098); background-color: rgb(255, 255, 255);" class=3D"">&nbsp; =
&nbsp; }</div><div style=3D"margin: 0px; font-stretch: normal; =
line-height: normal; font-family: Menlo; color: rgba(0, 0, 0, 0.85098); =
background-color: rgb(255, 255, 255);" class=3D"">&nbsp; } <span =
style=3D"color: #9b2393" class=3D"">else</span> { &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp;&nbsp;<span style=3D"color: #5d6c79" =
class=3D"">// jump backward over zero</span></div><div style=3D"margin: =
0px; font-stretch: normal; line-height: normal; font-family: Menlo; =
color: rgba(0, 0, 0, 0.85098); background-color: rgb(255, 255, 255);" =
class=3D"">&nbsp; &nbsp; SNE =3D RCV_SNE - <span style=3D"color: =
#1c00cf" class=3D"">1</span>; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&nbsp;<span style=3D"color: #5d6c79" =
class=3D"">// use pre-rollover SNE value</span></div><div style=3D"margin:=
 0px; font-stretch: normal; line-height: normal; font-family: Menlo; =
color: rgba(0, 0, 0, 0.85098); background-color: rgb(255, 255, 255);" =
class=3D"">&nbsp; }</div><div style=3D"margin: 0px; font-stretch: =
normal; line-height: normal; font-family: Menlo; color: rgba(0, 0, 0, =
0.85098); background-color: rgb(255, 255, 255);" =
class=3D"">}</div></body></html>=

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Subject: Re: [tcpm] RFC 5925 SNE algorithm concern
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PS -=20

#define distance(x,y)   (((x)<(y))?((y)-(x)):((x)-(y)))

The idea is that if we treat the two values as if they=E2=80=99re in the =
same SNE, would they be valid (i.e., jump less than half). If so, =
they=E2=80=99re in the same SNE. If not, then they must be in different =
SNEs (and then their distance would be less than half).

Joe

> On May 6, 2020, at 12:07 PM, Joseph Touch <touch@strayalpha.com> =
wrote:
>=20
> Hi, all,
>=20
> Juhamatti pointed out a flaw in the pseudocode for receive-side SNE =
computation in RFC 5925.
>=20
> The good news (in a sense) is that the code was not normative;=20
>=20
>    The implementation of SNEs is not specified in this document, but =
one
>    possible way is described here that can be used for either RCV.SNE,
>    SND.SNE, or both.
>=20
> The corrections that Juhamatti posted don=E2=80=99t address the key =
issue, which is that there are several possible interpretations of any =
received SEQ. There is no simple way to fix this except to handle all =
the cases (which the code below does - I believe minimally):
>=20
> 	- it is in-order
> 		- moving the receive window forward
> 		- incrementing the SNE if it wraps the number space
> 	- it is out-of-order
> 		- which might use either the current SNE or SNE-1, =
depending on whether it is pre-wrap of the number space
>=20
> Because the number space wraps, all received SEQ could be interpreted =
as in-order or out-of-order; the only way to know the difference is the =
distance between the two =E2=80=94 which cannot be larger than 0x7FFF =
FFFF (half the number space minus 1).
>=20
> The following is code that covers the cases and passes all current =
tests. Given this is not as obvious as it might seem and might be more =
useful than for just RFC 5925 and the SNE alg in 5925 is =E2=80=9Cnot =
specified=E2=80=9D anyway, I propose to submit this as a stand-alone =
doc.
>=20
> So here are the ways forward:
>=20
> 	1) have that doc =E2=80=9CUPDATE=E2=80=9D RFC 5925
> 		this is the cleanest because it would more definitely be =
caught by implementers
> 		but it requires the new alg be standards-track, which =
seems odd (though we could say what SNE is as a standard
> 		and still leave its implementation as informative rather =
than normative)
>=20
> 	2) publish that doc as informational
> 		then we could issue an errata on 5925 to point to the =
new RFC
>=20
> 	3) issue the new alg as an errata to 5925 only
>=20
> Any thoughts?
>=20
> Joe
>=20
> =E2=80=94=E2=80=94
>=20
> PS - here=E2=80=99s the alg:
>=20
>=20
> #DEFINE HALF (0x80000000)
>=20
> SNE =3D RCV_SNE;                               // use current SNE (as =
default)
> if (distance(SEG_SEQ,RCV_PREV_SEQ) < HALF) { // both in same SNE =
range?
>   if ((SEG_SEQ >=3D HALF)=20
>       && (RCV_PREV_SEQ < HALF)) {            // jumps fwd over N/2?
>     RCV_SNE_FLAG =3D 0;                        // reset wrap increment =
flag
>   }
>   RCV_PREV_SEQ =3D MAX(SEG_SEQ,RCV_PREV_SEQ);  // move prev forward if =
needed
> } else {                                     // both in diff SNE =
ranges
>   if (SEG_SEQ < HALF) {                      // jumps forward over =
zero?
>     RCV_PREV_SEQ =3D SEG_SEQ;                  // update prev
>     if (RCV_SNE_FLAG =3D=3D 0) {.                // first jump over =
zero? (wrap)
>       RCV_SNE_FLAG =3D 1;                      // set flag so we =
increment once
>       RCV_SNE =3D RCV_SNE + 1;                 // increment window
>       SNE =3D RCV_SNE;                         // use updated SNE =
value
>     }
>   } else {                                   // jump backward over =
zero
>     SNE =3D RCV_SNE - 1;                       // use pre-rollover SNE =
value
>   }
> }
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; line-break: after-white-space;" class=3D"">PS =
-&nbsp;<div class=3D""><br class=3D""></div><div class=3D""><div><div =
style=3D"margin: 0px; font-stretch: normal; font-size: 18px; =
line-height: normal; font-family: Menlo; color: rgb(100, 56, 32); =
background-color: rgb(255, 255, 255);" class=3D"">#define distance(x,y) =
&nbsp; (((x)&lt;(y))?((y)-(x)):((x)-(y)))</div><div class=3D""><br =
class=3D""></div><div class=3D"">The idea is that if we treat the two =
values as if they=E2=80=99re in the same SNE, would they be valid (i.e., =
jump less than half). If so, they=E2=80=99re in the same SNE. If not, =
then they must be in different SNEs (and then their distance would be =
less than half).</div><div class=3D""><br class=3D""></div><div =
class=3D"">Joe</div><div class=3D""><br class=3D""></div><blockquote =
type=3D"cite" class=3D""><div class=3D"">On May 6, 2020, at 12:07 PM, =
Joseph Touch &lt;<a href=3D"mailto:touch@strayalpha.com" =
class=3D"">touch@strayalpha.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><meta =
http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dutf-8" =
class=3D""><div style=3D"word-wrap: break-word; -webkit-nbsp-mode: =
space; line-break: after-white-space;" class=3D"">Hi, all,<div =
class=3D""><br class=3D""></div><div class=3D"">Juhamatti pointed out a =
flaw in the pseudocode for receive-side SNE computation in RFC =
5925.</div><div class=3D""><br class=3D""></div><div class=3D"">The good =
news (in a sense) is that the code was not normative;&nbsp;</div><div =
class=3D""><br class=3D""></div><div class=3D""><pre class=3D"newpage" =
style=3D"font-size: 13.333333015441895px; margin-top: 0px; =
margin-bottom: 0px; break-before: page;">   The implementation of SNEs =
is not specified in this document, but one
   possible way is described here that can be used for either RCV.SNE,
   SND.SNE, or both.</pre></div><div class=3D""><div class=3D""><br =
class=3D""></div><div class=3D"">The corrections that Juhamatti posted =
don=E2=80=99t address the key issue, which is that there are several =
possible interpretations of any received SEQ. There is no simple way to =
fix this except to handle all the cases (which the code below does - I =
believe minimally):</div><div class=3D""><br class=3D""></div><div =
class=3D""><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>- it is in-order</div><div class=3D""><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">		</span>- =
moving the receive window forward</div><div class=3D""><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">		=
</span>-&nbsp;incrementing the SNE if it wraps the number =
space</div><div class=3D""><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>- it is out-of-order</div><div =
class=3D""><span class=3D"Apple-tab-span" style=3D"white-space:pre">		=
</span>- which might use either the current SNE or SNE-1, depending on =
whether it is pre-wrap of the number space</div><div class=3D""><br =
class=3D""></div><div class=3D"">Because the number space wraps, all =
received SEQ could be interpreted as in-order or out-of-order; the only =
way to know the difference is the distance between the two =E2=80=94 =
which cannot be larger than 0x7FFF FFFF (half the number space minus =
1).</div><div class=3D""><br class=3D""></div><div class=3D"">The =
following is code that covers the cases and passes all current tests. =
Given this is not as obvious as it might seem and might be more useful =
than for just RFC 5925 and the SNE alg in 5925 is =E2=80=9Cnot =
specified=E2=80=9D anyway, I propose to submit this as a stand-alone =
doc.</div><div class=3D""><br class=3D""></div><div class=3D"">So here =
are the ways forward:</div><div class=3D""><br class=3D""></div><div =
class=3D""><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>1) have that doc =E2=80=9CUPDATE=E2=80=9D RFC 5925</div><div =
class=3D""><span class=3D"Apple-tab-span" style=3D"white-space:pre">		=
</span>this is the cleanest because it would more definitely be caught =
by implementers</div><div class=3D""><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">		</span>but it requires the new =
alg be standards-track, which seems odd (though we could say what SNE is =
as a standard</div><div class=3D""><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">		</span>and still leave its =
implementation as informative rather than normative)</div><div =
class=3D""><br class=3D""></div><div class=3D""><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>2) =
publish that doc as informational</div><div class=3D""><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">		=
</span>then we could issue an errata on 5925 to point to the new =
RFC</div><div class=3D""><br class=3D""></div><div class=3D""><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>3) issue =
the new alg as an errata to 5925 only</div><div class=3D""><br =
class=3D""></div><div class=3D"">Any thoughts?</div><div class=3D""><br =
class=3D""></div><div class=3D"">Joe</div><div class=3D""><br =
class=3D""></div><div class=3D"">=E2=80=94=E2=80=94</div><div =
class=3D""><br class=3D""></div><div class=3D"">PS - here=E2=80=99s the =
alg:</div><div class=3D""><br class=3D""></div><div class=3D""><br =
class=3D""></div></div><div class=3D""><span style=3D"font-family: =
Menlo; background-color: rgb(255, 255, 255);" class=3D"">#DEFINE HALF =
(0x80000000)</span></div><div class=3D""><span style=3D"font-family: =
Menlo; background-color: rgb(255, 255, 255);" class=3D""><br =
class=3D""></span></div><div class=3D""><span style=3D"font-family: =
Menlo; background-color: rgb(255, 255, 255);" class=3D"">SNE =3D =
RCV_SNE; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&nbsp;</span><span =
style=3D"font-family: Menlo; background-color: rgb(255, 255, 255); =
color: rgb(93, 108, 121);" class=3D"">// use current SNE (as =
default)</span></div><div style=3D"margin: 0px; font-stretch: normal; =
line-height: normal; font-family: Menlo; background-color: rgb(255, 255, =
255);" class=3D""><span style=3D"color: #9b2393" class=3D"">if</span> =
(distance(SEG_SEQ,RCV_PREV_SEQ) &lt; <span style=3D"color: #1c00cf" =
class=3D"">HALF</span>) {&nbsp;<span style=3D"color: #5d6c79" =
class=3D"">// both in same SNE range?</span></div><div style=3D"margin: =
0px; font-stretch: normal; line-height: normal; font-family: Menlo; =
background-color: rgb(255, 255, 255);" class=3D"">&nbsp; if<span =
style=3D"" class=3D""> ((SEG_SEQ &gt;=3D </span><span style=3D"color: =
rgb(28, 0, 207);" class=3D"">HALF</span><span style=3D"" =
class=3D"">)</span><span style=3D"" class=3D"">&nbsp;</span></div><div =
style=3D"margin: 0px; font-stretch: normal; line-height: normal; =
font-family: Menlo; background-color: rgb(255, 255, 255);" =
class=3D""><span style=3D"" class=3D"">&nbsp; &nbsp; &nbsp; &amp;&amp; =
(RCV_PREV_SEQ &lt; </span><span style=3D"color: rgb(28, 0, 207);" =
class=3D"">HALF</span><span style=3D"" class=3D"">)) { &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp;</span><span style=3D"color: rgb(93, 108, =
121);" class=3D"">// jumps fwd over N/2?</span></div><div style=3D"margin:=
 0px; font-stretch: normal; line-height: normal; font-family: Menlo; =
background-color: rgb(255, 255, 255);" class=3D"">&nbsp; &nbsp; =
RCV_SNE_FLAG =3D <span style=3D"color: #1c00cf" class=3D"">0</span>; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;<span style=3D"color: #5d6c79" class=3D"">// reset wrap =
increment flag</span></div><div style=3D"margin: 0px; font-stretch: =
normal; line-height: normal; font-family: Menlo; background-color: =
rgb(255, 255, 255);" class=3D"">&nbsp; }</div><div style=3D"margin: 0px; =
font-stretch: normal; line-height: normal; font-family: Menlo; =
background-color: rgb(255, 255, 255);" class=3D"">&nbsp; RCV_PREV_SEQ =3D =
MAX(SEG_SEQ,RCV_PREV_SEQ); &nbsp;<span style=3D"color: #5d6c79" =
class=3D"">// move prev forward if needed</span></div><div =
style=3D"margin: 0px; font-stretch: normal; line-height: normal; =
font-family: Menlo; background-color: rgb(255, 255, 255);" class=3D"">} =
<span style=3D"color: #9b2393" class=3D"">else</span> {&nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<span style=3D"color: =
#5d6c79" class=3D"">// both in diff SNE ranges</span></div><div =
style=3D"margin: 0px; font-stretch: normal; line-height: normal; =
font-family: Menlo; background-color: rgb(255, 255, 255);" =
class=3D"">&nbsp; if<span style=3D"" class=3D""> (SEG_SEQ &lt; =
</span><span style=3D"color: rgb(28, 0, 207);" class=3D"">HALF</span><span=
 style=3D"" class=3D"">) { &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;</span><span style=3D"color: rgb(93, =
108, 121);" class=3D"">// jumps forward over zero?</span></div><div =
style=3D"margin: 0px; font-stretch: normal; line-height: normal; =
font-family: Menlo; background-color: rgb(255, 255, 255);" =
class=3D"">&nbsp; &nbsp; RCV_PREV_SEQ =3D SEG_SEQ; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<span style=3D"color: #5d6c79" =
class=3D"">// update prev</span></div><div style=3D"margin: 0px; =
font-stretch: normal; line-height: normal; font-family: Menlo; =
background-color: rgb(255, 255, 255);" class=3D"">&nbsp; &nbsp; if<span =
style=3D"" class=3D""> (RCV_SNE_FLAG =3D=3D </span><span style=3D"color: =
rgb(28, 0, 207);" class=3D"">0</span><span style=3D"" class=3D"">) {. =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;/</span><span =
style=3D"color: rgb(93, 108, 121);" class=3D"">/ first jump over zero? =
(wrap)</span></div><div style=3D"margin: 0px; font-stretch: normal; =
line-height: normal; font-family: Menlo; background-color: rgb(255, 255, =
255);" class=3D"">&nbsp; &nbsp; &nbsp; RCV_SNE_FLAG =3D <span =
style=3D"color: #1c00cf" class=3D"">1</span>; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<span =
style=3D"color: #5d6c79" class=3D"">// set flag so we increment =
once</span></div><div style=3D"margin: 0px; font-stretch: normal; =
line-height: normal; font-family: Menlo; background-color: rgb(255, 255, =
255);" class=3D"">&nbsp; &nbsp; &nbsp; RCV_SNE =3D RCV_SNE + <span =
style=3D"color: #1c00cf" class=3D"">1</span>; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&nbsp;<span style=3D"color: #5d6c79" =
class=3D"">// increment window</span></div><div style=3D"margin: 0px; =
font-stretch: normal; line-height: normal; font-family: Menlo; =
background-color: rgb(255, 255, 255);" class=3D"">&nbsp; &nbsp; &nbsp; =
SNE =3D RCV_SNE; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp;&nbsp;<span style=3D"color: #5d6c79" =
class=3D"">// use updated SNE value</span></div><div style=3D"margin: =
0px; font-stretch: normal; line-height: normal; font-family: Menlo; =
background-color: rgb(255, 255, 255);" class=3D"">&nbsp; &nbsp; =
}</div><div style=3D"margin: 0px; font-stretch: normal; line-height: =
normal; font-family: Menlo; background-color: rgb(255, 255, 255);" =
class=3D"">&nbsp; } <span style=3D"color: #9b2393" class=3D"">else</span> =
{ &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&nbsp;<span =
style=3D"color: #5d6c79" class=3D"">// jump backward over =
zero</span></div><div style=3D"margin: 0px; font-stretch: normal; =
line-height: normal; font-family: Menlo; background-color: rgb(255, 255, =
255);" class=3D"">&nbsp; &nbsp; SNE =3D RCV_SNE - <span style=3D"color: =
#1c00cf" class=3D"">1</span>; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&nbsp;<span style=3D"color: #5d6c79" =
class=3D"">// use pre-rollover SNE value</span></div><div style=3D"margin:=
 0px; font-stretch: normal; line-height: normal; font-family: Menlo; =
background-color: rgb(255, 255, 255);" class=3D"">&nbsp; }</div><div =
style=3D"margin: 0px; font-stretch: normal; line-height: normal; =
font-family: Menlo; background-color: rgb(255, 255, 255);" =
class=3D"">}</div></div>_______________________________________________<br=
 class=3D"">tcpm mailing list<br class=3D""><a =
href=3D"mailto:tcpm@ietf.org" class=3D"">tcpm@ietf.org</a><br =
class=3D"">https://www.ietf.org/mailman/listinfo/tcpm<br =
class=3D""></div></blockquote></div><br class=3D""></div></body></html>=

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Subject: Re: [tcpm] RFC 5925 SNE algorithm concern
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--Apple-Mail=_A7C2B710-EF89-40DF-9510-AD09176A1501
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	charset=utf-8

Here=E2=80=99s a full version, with everything including initialization =
and definitions.=20

// compute the distance from x to y assuming they are in the same SNE
#define DIST(x,y) (((x)<(y))?((y)-(x)):((x)-(y)))
#define HALF      (0x80000000)              // shorthand used in the =
code below

// variables, with inits where required
unsigned long RCV_SNE =3D 0;                  // high-watermark SNE, =
init per RFC5925
int           RCV_SNE_FLAG =3D 1;             // set during first half =
of rollover
unsigned long RCV_PREV_SEQ;                 // high-watermark SEQ, init =
by ISN
unsigned long SEG_SEQ;                      // input - received SEQ
unsigned long SNE;                          // output - SNE =
corresponding to received SEQ

if (SYN =3D=3D 1) {                             // SYN or SYN-ACK =
initialization
  RCV_PREV_SEQ =3D RECV_ISN;
}                                           // after SYN or SYN-ACK
SNE =3D RCV_SNE;                              // use current SNE (as =
default)
if (DIST(SEG_SEQ,RCV_PREV_SEQ) < HALF) {    // both in same SNE range?
  if ((SEG_SEQ >=3D HALF)=20
      && (RCV_PREV_SEQ < HALF)) {           // jumps fwd over N/2?
    RCV_SNE_FLAG =3D 0;                       // reset wrap increment =
flag
  }
  RCV_PREV_SEQ =3D MAX(SEG_SEQ,RCV_PREV_SEQ); // move prev forward if =
needed
} else {                                    // both in diff SNE ranges
  if (SEG_SEQ < HALF) {                     // jumps forward over zero?
    RCV_PREV_SEQ =3D SEG_SEQ;                 // update prev
    if (RCV_SNE_FLAG =3D=3D 0) {.               // first jump over zero? =
(wrap)
      RCV_SNE_FLAG =3D 1;                     // set flag so we =
increment once
      RCV_SNE =3D RCV_SNE + 1;                // increment window
      SNE =3D RCV_SNE;                        // use updated SNE value
    }
  } else {                                  // jump backward over zero
    SNE =3D RCV_SNE - 1;                      // use pre-rollover SNE =
value
  }
}


--Apple-Mail=_A7C2B710-EF89-40DF-9510-AD09176A1501
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; line-break: after-white-space;" =
class=3D"">Here=E2=80=99s a full version, with everything including =
initialization and definitions.&nbsp;<div class=3D""><br =
class=3D""></div><div class=3D""><font face=3D"Courier New" class=3D"">// =
compute the distance from x to y assuming they are in the same =
SNE</font></div><div class=3D""><font face=3D"Courier New" =
class=3D"">#define DIST(x,y)&nbsp;(((x)&lt;(y))?((y)-(x)):((x)-(y)))<br =
class=3D""></font></div><div class=3D""><span style=3D"font-family: =
&quot;Courier New&quot;;" class=3D"">#define HALF &nbsp; &nbsp; =
&nbsp;</span><span style=3D"font-family: &quot;Courier New&quot;;" =
class=3D"">(0x80000000) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;// shorthand used in the code below</span></div><div =
class=3D""><font face=3D"Courier New" class=3D""><br =
class=3D""></font></div><div class=3D""><font face=3D"Courier New" =
class=3D"">// variables, with inits where required</font></div><div =
class=3D""><font face=3D"Courier New" =
class=3D"">unsigned&nbsp;long&nbsp;RCV_SNE =3D&nbsp;0; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;// high-watermark SNE, =
init per RFC5925<br class=3D"">int&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&nbsp;RCV_SNE_FLAG =3D&nbsp;1;&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;&nbsp;// set during first half of&nbsp;rollover<br =
class=3D"">unsigned&nbsp;long&nbsp;RCV_PREV_SEQ;&nbsp;&nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&nbsp;// high-watermark SEQ, =
init by ISN<br class=3D"">unsigned&nbsp;long&nbsp;SEG_SEQ;&nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&nbsp;// =
input - received SEQ<br class=3D"">unsigned&nbsp;long&nbsp;SNE;&nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;&nbsp;// output - SNE corresponding&nbsp;to received =
SEQ</font></div><div class=3D""><font face=3D"Courier New" class=3D""><br =
class=3D""></font></div><div class=3D""><span style=3D"font-family: =
&quot;Courier New&quot;;" class=3D"">if (SYN =3D=3D 1) { &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; // SYN or SYN-ACK&nbsp;initialization</span></div><div =
class=3D""><font face=3D"Courier New" class=3D"">&nbsp; RCV_PREV_SEQ =3D =
RECV_ISN;</font></div><div class=3D""><span style=3D"font-family: =
&quot;Courier New&quot;;" class=3D"">} &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; // after SYN or =
SYN-ACK</span></div><div class=3D""><font face=3D"Courier New" =
class=3D"">SNE =3D RCV_SNE; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;// use =
current SNE (as default)<br class=3D"">if&nbsp;(DIST(SEG_SEQ,RCV_PREV_SEQ)=
 &lt;&nbsp;HALF) { &nbsp; &nbsp;// both in same SNE range?<br =
class=3D"">&nbsp; if&nbsp;((SEG_SEQ &gt;=3D&nbsp;HALF)&nbsp;<br =
class=3D"">&nbsp; &nbsp; &nbsp; &amp;&amp; (RCV_PREV_SEQ =
&lt;&nbsp;HALF)) { &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&nbsp;// jumps fwd =
over N/2?<br class=3D"">&nbsp; &nbsp; RCV_SNE_FLAG =3D&nbsp;0; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&nbsp;// reset wrap increment flag<br class=3D"">&nbsp; }<br =
class=3D"">&nbsp; RCV_PREV_SEQ =3D MAX(SEG_SEQ,RCV_PREV_SEQ); // move =
prev forward if needed<br class=3D"">}&nbsp;else&nbsp;{ &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;// both in diff SNE ranges<br =
class=3D"">&nbsp; if&nbsp;(SEG_SEQ &lt;&nbsp;HALF) { &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&nbsp;// jumps =
forward over zero?<br class=3D"">&nbsp; &nbsp; RCV_PREV_SEQ =3D SEG_SEQ; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&nbsp;// update =
prev<br class=3D"">&nbsp; &nbsp; if&nbsp;(RCV_SNE_FLAG =3D=3D&nbsp;0) {. =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; // first jump over =
zero? (wrap)<br class=3D"">&nbsp; &nbsp; &nbsp; RCV_SNE_FLAG =3D&nbsp;1; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&nbsp;// set flag so we increment once<br class=3D"">&nbsp; &nbsp; =
&nbsp; RCV_SNE =3D RCV_SNE +&nbsp;1; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp;// increment window<br class=3D"">&nbsp; &nbsp; =
&nbsp; SNE =3D RCV_SNE; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;// use updated SNE value<br =
class=3D"">&nbsp; &nbsp; }<br class=3D"">&nbsp; }&nbsp;else&nbsp;{ =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;// jump backward over =
zero<br class=3D"">&nbsp; &nbsp; SNE =3D RCV_SNE -&nbsp;1; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;// use =
pre-rollover SNE value<br class=3D"">&nbsp; }<br class=3D"">}</font><br =
class=3D""><div><br class=3D""></div></div></body></html>=

--Apple-Mail=_A7C2B710-EF89-40DF-9510-AD09176A1501--


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References: <CACS3ZpBLAO2=O6Vf2Vx766qAGqmY2yD_yiLag-K7C3sYFjH9hw@mail.gmail.com> <EAE578C4-4C60-429A-9826-821B3075DA08@strayalpha.com> <CACS3ZpCjWb2+-v8MCjRJvMCARZ1YDLAr5HyRUTSTteMefu76qQ@mail.gmail.com> <CACS3ZpC3z-D8C=g90vuz-1n-w4oX8y2vOzJWrhouNqnKeuZ42w@mail.gmail.com> <26C2355F-AAF8-405B-BA38-8E7F50013D27@strayalpha.com> <85AAF68F-DB45-475C-996C-8F99EF66D198@strayalpha.com> <4E2DD376-438F-416E-9D1C-0934F517E2C2@strayalpha.com>
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From: juhamatk@gmail.com
Date: Thu, 7 May 2020 09:00:34 +0300
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Subject: Re: [tcpm] RFC 5925 SNE algorithm concern
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Hello,

+1 for below algorithm.

The earlier improved algorithm I suggested did not handle middle-range
retransmissions correctly, this one does. It also passes other tests I
have (which found the original issues).

Thank you,
--
 Juhamatti

On Wed, 6 May 2020 at 22:36, Joseph Touch <touch@strayalpha.com> wrote:
>
> Here=E2=80=99s a full version, with everything including initialization a=
nd definitions.
>
> // compute the distance from x to y assuming they are in the same SNE
> #define DIST(x,y) (((x)<(y))?((y)-(x)):((x)-(y)))
> #define HALF      (0x80000000)              // shorthand used in the code=
 below
>
> // variables, with inits where required
> unsigned long RCV_SNE =3D 0;                  // high-watermark SNE, init=
 per RFC5925
> int           RCV_SNE_FLAG =3D 1;             // set during first half of=
 rollover
> unsigned long RCV_PREV_SEQ;                 // high-watermark SEQ, init b=
y ISN
> unsigned long SEG_SEQ;                      // input - received SEQ
> unsigned long SNE;                          // output - SNE corresponding=
 to received SEQ
>
> if (SYN =3D=3D 1) {                             // SYN or SYN-ACK initial=
ization
>   RCV_PREV_SEQ =3D RECV_ISN;
> }                                           // after SYN or SYN-ACK
> SNE =3D RCV_SNE;                              // use current SNE (as defa=
ult)
> if (DIST(SEG_SEQ,RCV_PREV_SEQ) < HALF) {    // both in same SNE range?
>   if ((SEG_SEQ >=3D HALF)
>       && (RCV_PREV_SEQ < HALF)) {           // jumps fwd over N/2?
>     RCV_SNE_FLAG =3D 0;                       // reset wrap increment fla=
g
>   }
>   RCV_PREV_SEQ =3D MAX(SEG_SEQ,RCV_PREV_SEQ); // move prev forward if nee=
ded
> } else {                                    // both in diff SNE ranges
>   if (SEG_SEQ < HALF) {                     // jumps forward over zero?
>     RCV_PREV_SEQ =3D SEG_SEQ;                 // update prev
>     if (RCV_SNE_FLAG =3D=3D 0) {.               // first jump over zero? =
(wrap)
>       RCV_SNE_FLAG =3D 1;                     // set flag so we increment=
 once
>       RCV_SNE =3D RCV_SNE + 1;                // increment window
>       SNE =3D RCV_SNE;                        // use updated SNE value
>     }
>   } else {                                  // jump backward over zero
>     SNE =3D RCV_SNE - 1;                      // use pre-rollover SNE val=
ue
>   }
> }
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Fri May  8 10:05:05 2020
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the TCP Maintenance and Minor Extensions WG of the IETF.

        Title           : Requirements for Time-Based Loss Detection
        Author          : Mark Allman
	Filename        : draft-ietf-tcpm-rto-consider-13.txt
	Pages           : 11
	Date            : 2020-05-08

Abstract:
    Many protocols must detect packet loss for various reasons (e.g., to
    ensure reliability using retransmissions or to understand the level
    of congestion along a network path).  While many mechanisms have
    been designed to detect loss, protocols ultimately can only count on
    the passage of time without delivery confirmation to declare a
    packet "lost".  Each implementation of a time-based loss detection
    mechanism represents a balance between correctness and timeliness
    and therefore no implementation suits all situations.  This document
    provides high-level requirements for time-based loss detectors
    appropriate for general use in the Internet.  Within the
    requirements, implementations have latitude to define particulars
    that best address each situation.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tcpm-rto-consider/

There are also htmlized versions available at:
https://tools.ietf.org/html/draft-ietf-tcpm-rto-consider-13
https://datatracker.ietf.org/doc/html/draft-ietf-tcpm-rto-consider-13

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-rto-consider-13


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/



From nobody Fri May  8 10:07:14 2020
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From: "Mark Allman" <mallman@icir.org>
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This version includes clarifications stemming from Martin's AD
review of the document.

allman




> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> This draft is a work item of the TCP Maintenance and Minor Extensions WG of the IETF.
>
>         Title           : Requirements for Time-Based Loss Detection
>         Author          : Mark Allman
> 	Filename        : draft-ietf-tcpm-rto-consider-13.txt
> 	Pages           : 11
> 	Date            : 2020-05-08
>
> Abstract:
>     Many protocols must detect packet loss for various reasons (e.g., to
>     ensure reliability using retransmissions or to understand the level
>     of congestion along a network path).  While many mechanisms have
>     been designed to detect loss, protocols ultimately can only count on
>     the passage of time without delivery confirmation to declare a
>     packet "lost".  Each implementation of a time-based loss detection
>     mechanism represents a balance between correctness and timeliness
>     and therefore no implementation suits all situations.  This document
>     provides high-level requirements for time-based loss detectors
>     appropriate for general use in the Internet.  Within the
>     requirements, implementations have latitude to define particulars
>     that best address each situation.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tcpm-rto-consider/
>
> There are also htmlized versions available at:
> https://tools.ietf.org/html/draft-ietf-tcpm-rto-consider-13
> https://datatracker.ietf.org/doc/html/draft-ietf-tcpm-rto-consider-13
>
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-rto-consider-13
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
>
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt

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From: Yuchung Cheng <ycheng@google.com>
Date: Fri, 8 May 2020 16:07:55 -0700
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Neal Cardwell <ncardwell@google.com>,  Nandita Dukkipati <nanditad@google.com>, Priyaranjan Jha <priyarjha@google.com>
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Subject: Re: [tcpm] Review of draft-ietf-tcpm-rack-07
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On Mon, Apr 6, 2020 at 2:01 PM Ilpo J=C3=A4rvinen
<ilpo.jarvinen@cs.helsinki.fi> wrote:
>
> Hi Yuchung,
>
> Certainly improved on many things, however, I'm surprised that the lost
> rexmit comment was not addressed at all. Aren't the rexmit losses
> supposed to be detected by RACK or have I misunderstood something?
>
> I don't understand how lost rexmits are detected by the algorithms in
> the draft. I think the algorithms are either flawed or lack something
> (mainly RACK_detect_loss is relevant I think). My original comment
> from below.
>
>        Is RACK_detect_loss() supposed to detect also lost rexmits? How?
>        The rexmit code itself is not in the draft so it's hard to evaluat=
e
>        how the Packet booleans are supposed to work in the first place.
>        I'm not saying the rexmit code should be included but currently it
>        surely looks like any rexmitted packet just remains .lost =3D TRUE
>        and .retransmitted =3D TRUE state until RTO?

Sorry to miss this point earlier -- RACK definitely detects
lost-retransmits, but the current algorithm does not properly define
some variables. Here is how I propose to address these by adding
these:


Per-packet variables


Theses variables indicate the status of the most recent transmission
of a data packet:
        =E2=80=9CPacket.lost=E2=80=9D is true if the most recent (re)transm=
ission of
the Packet has been considered lost and needs to be retransmitted.
False otherwise.
        =E2=80=9CPacket.retransmitted=E2=80=9D is tue if its sequence numbe=
rs were
retransmitted in the most recent transmission. False otherwise.
        =E2=80=9CPacket.xmit_ts=E2=80=9D is the time of the last transmissi=
on of a
data packet, including retransmissions, if any. The sender needs to
record the transmission time for each packet sent and not yet
acknowledged. The time MUST be stored at millisecond granularity or
finer.


Transmitting a data packet
Upon transmitting a new packet or retransmitting an old packet, record
the time in Packet.xmit_ts.

--- src

RACK_transmit_data(Packet):

Packet.time_ts =3D Now()

Packet.lost =3D FALSE


RACK_retransmit_data(Packet)

Packet.retransmitted =3D TRUE
      RACK_transmit_data(Packet)

---

When a packet is retransmitted again, its Packet.lost is reset and
transmission timestamp is renewed. Therefore, when its transmission
time falls out of the RACK reordering window in RACK_detect_loss(), it
will be marked as lost (again).




>
> Also my timeout "some" vs "all" internal inconsistency comment relates to
> this same algorithm:
>
>          "For timely loss detection, the sender MAY
>           install a "reordering settling" timer set to fire at the earlie=
st
>           moment at which it is safe to conclude that some packet is lost=
.  The
>           earliest moment is the time it takes to expire the reordering w=
indow
>           of the earliest unacked packet in flight."
>        vs
>           RACK_detect_loss():
>              ...
>                  timeout =3D max(remaining, timeout)
>        Which way it is, "some" or "all"? The use of max() in RACK_detect_=
loss()
>        seems to indicate that the timeout will be set according to the lo=
ngest
>        time, that is, the latest moment rather than earliest moments. Whi=
ch
>        is the intented behavior here? (I can there might be merits in bot=
h
>        approaches).

Comment 1.2: good catch. We changed to: 'For timely loss detection,
the sender is RECOMMENDED to install a "reordering settling" timer.
The timer expires at the earliest moment when it is safe to conclude
that all the unacknowledged packets within the reordering window are
lost.'


>
>
> Then there are some other comments I made which seem also unaddressed but
> I'll not duplicate them all here.
>
>
> --
>  i.
>
>
> On Mon, 9 Mar 2020, Yuchung Cheng wrote:
>
> > Hi Ilpo,
> > Thanks for a detailed review and the great suggestions. Here's all the =
edits
> > we made.
> > https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-rack-08
> >
> >  We defined the missing terms and addressed the nits, removed the less
> > critical parts like recommending PRR, and edited TLP algorithms (sectio=
ns
> > 7.4 to 7.6). I hope you find it's better now and would be happy to hear=
 any
> > improvements you'd like to make.
> >
> > Yuchung
> >
> > On Fri, Feb 14, 2020 at 4:22 AM Ilpo J=C3=A4rvinen <ilpo.jarvinen@cs.he=
lsinki.fi>
> > wrote:
> >       In general, I find the draft quite good shape already, it's
> >       quite easy
> >       to read and follow (that is, except the TLP sections that are
> >       specifically
> >       mentioned below).
> >
> >       Here's the list of issues I came across:
> >
> >       Sect 6.1
> >
> >       "RACK.pkts_sacked" lacks the empty line separator.
> >
> >       Sect 7.2
> >
> >         "This
> >          heuristic would fail to update RACK stats if the packet is
> >       spuriously
> >          retransmitted because of a recent minimum RTT decrease (e.g.
> >       path
> >          change)."
> >       This defies my logic as RTT decrease should not result in
> >       spurious
> >       retransmissions but help to avoid them.
> >
> >
> >         "Note that
> >          extensions that require additional TCP features (e.g.  DSACK)
> >       would
> >          work if the feature functions simply return false."
> >       Please rephrase this. I suspect you meant that if some functions
> >       relate
> >       to TCP features that are not available, the given code still
> >       works
> >       as long as the functions related to such features return false
> >       when
> >       the TCP feature is not available but that surely isn't what the
> >       sentence currently says.
> >
> >       Sect 7.2 Step 5
> >
> >         "For timely loss detection, the sender MAY
> >          install a "reordering settling" timer set to fire at the
> >       earliest
> >          moment at which it is safe to conclude that some packet is
> >       lost.  The
> >          earliest moment is the time it takes to expire the reordering
> >       window
> >          of the earliest unacked packet in flight."
> >       vs
> >          RACK_detect_loss():
> >             ...
> >                 timeout =3D max(remaining, timeout)
> >       Which way it is, "some" or "all"? The use of max() in
> >       RACK_detect_lost()
> >       seems to indicate that the timeout will be set according to the
> >       longest
> >       time, that is, the latest moment rather than earliest moments.
> >       Which
> >       is the intented behavior here? (I can there might be merits in
> >       both
> >       approaches).
> >
> >
> >       In general, I didn't like the presentation order here as this
> >       form looks
> >       the most obvious to me:
> >          now >=3D Packet.xmit_ts + RACK.reo_wnd + (RACK.ack_ts -
> >       RACK.xmit_ts)
> >       ...and I find the derivation of it just complicating things
> >       rather
> >       than helping but that's perhaps just my opinion.
> >
> >
> >          "It is also useful when the timestamp clock granularity is
> >       close to
> >           or longer than the actual round trip time."
> >       Given that some things based on timestamps will obviously stop
> >       working
> >       if the clock is slower than one tick per RTT I found this
> >       fragment a bit
> >       odd. RFC 7323 says "the clock SHOULD tick at least once per
> >       window's worth
> >       of data". I can understand though that a host might find clock
> >       tick
> >       granularity challenging due to extensive RTT range but on the
> >       same
> >       I'd not want such a "too slow clock" behavior to be endorsed as
> >       if
> >       it would be just normal. That is, I'm not against RACK handling
> >       also this case nicely but it would be nice to somehow downplay
> >       the
> >       normalness tone in it with something along the lines of "This
> >       check is
> >       also useful if the recommended one tick per RTT cannot be
> >       achieved due
> >       to timestamp clock granularity."
> >
> >       Is RACK_detect_lost() supposed to detect also lost rexmits? How?
> >       The rexmit code itself is not in the draft so it's hard to
> >       evaluate
> >       how the Packet booleans are supposed to work in the first place.
> >       I'm not saying the rexmit code should be included but currently
> >       it
> >       surely looks like any rexmitted packet just remains .lost =3D TRU=
E
> >       and
> >       .retransmitted =3D TRUE state until RTO?
> >
> >
> >       Sect 7.4
> >
> >       "PTO" just appears out of nowhere (and is not even written open
> >       in the first instance).
> >
> >          "2. ... in step (2) of the algorithm."
> >       "step (2)" is forward-refing, it would naturally refer back to
> >       "Step 2: Update RACK stats" so it makes very little sense as is.
> >
> >       Sect 7.5
> >
> >       Separator missing for PTO: and TLPRxtOut:
> >
> >       Sect 7.5.1
> >
> >         "3.  Upon receiving a SACK that selectively acknowledges data
> >       that was
> >              last sent before the segment with SEG.SEQ=3DSND.UNA was
> >       last
> >              (re)transmitted"
> >       This is hard to understand, perhaps rephrase it somehow?
> >
> >
> >         "But choosing PTO to be
> >          exactly an SRTT is likely to generate spurious probes given
> >       that
> >          network delay variance and even end-system timings can easily
> >       push an
> >          ACK to be above an SRTT.  We chose PTO to be the next
> >       integral
> >          multiple of SRTT.
> >
> >          Similarly, network delay variations and end-system processing
> >          latencies and timer granularities can easily delay ACKs
> >       beyond
> >          2*SRTT, so senders SHOULD add at least 2ms to a computed PTO
> >       value
> >          (and MAY add more if the sending host OS timer granularity is
> >       more
> >          coarse than 1ms)."
> >       Does this later paragraph depend on some assumptions (such as
> >       very
> >       low RTT?) as I don't otherwise understand why you say what you
> >       say
> >       in the given order. First you said x & y can push ACK over SRTT
> >       so
> >       we pick 2* but then you say the same things can push over 2*SRTT
> >       too?
> >
> >       WCDelAckT to terminology?
> >
> >       Sect 7.5.2
> >
> >       TLP_send_probe():
> >               - Perhaps set TLPRxtOut somewhere?
> >               - Add "Arm RTO" to the end?
> >
> >         "In the case where
> >          there is only one original outstanding segment of data (N=3D1)=
,
> >       the
> >          same logic (trivially) applies: an ACK for a single
> >       outstanding
> >          segment tells the sender the N-1=3D0 segments preceding that
> >       segment
> >          were lost.  Furthermore, whether there are N>1 or N=3D1
> >       outstanding
> >          segments,"
> >       If this distinction is useful to make at all, maybe just simply
> >       say:
> >       "Applies also in case N=3D1."
> >
> >       Sect 7.6
> >
> >         "SND.NXT at the time the episode started."
> >       Isn't this TLPHighRxt, why not use it here?
> >
> >       Sect 7.6.2
> >
> >         "Upon sending a TLP probe that is a retransmission, the sender
> >       sets
> >          TLPRxtOut to true and TLPHighRxt to SND.NXT."
> >       This should appear in TLP_send_probe().
> >
> >       "Detecting recoveries accomplished by loss probes"
> >       Did you want to start another section here?
> >
> >       "Step 1: Track ACKs ..." until what point? (It's told on later
> >       but
> >       this question comes into mind when reading the draft in order).
> >
> >
> >       Sect 7.4 - 7.6.2
> >
> >       In general, this particular part of the draft seems to require
> >       some
> >       further editing. After reading it all through, I feel most bits
> >       are
> >       present but the text is very hard to follow. I think the
> >       presentation
> >       order needs significant work. There are random bits here and
> >       there
> >       that are explained only later. Also, algorithm lacks some parts
> >       discussed only in the text but variable like items should
> >       definitely
> >       be included to the algorithm.
> >
> >
> >       Sect 8.4
> >
> >         "RACK is compatible with and does not interfere with the the
> >       standard
> >          RTO [RFC6298], RTO-restart [RFC7765], F-RTO [RFC5682] and
> >       Eifel
> >          algorithms [RFC3522]. ...
> >          It neither changes the RTO timer calculation nor detects
> >       spurious
> >          timeouts."
> >       For F-RTO, also send order is significant.
> >
> >       Sect 8.5
> >
> >       SHOULDs PRR (and also placed to normative references) that is
> >       only
> >       experimental and RACK aims to standards track. I feel that PRR
> >       is
> >       disjoint enough anyway, which I think was also your intention
> >       when
> >       maintaining the decoupling CC and RACK, that it's bit odd to
> >       take
> >       this strong stance on PRR (indepedent of the stds track vs
> >       experimental thing).
> >
> >         "due to congestion window validation."
> >       Does this refer to CWV as in RFC? Or some other mechanism in
> >       which
> >       case you shouldn't use the same term here.
> >
> >       I find the described scenario somewhat unconvincing becase
> >       the description ends with:
> >         "...
> >          The ending congestion window after recovery also impacts
> >          subsequent data transfer."
> >       ...First there wasn't enough data to fill cwnd of 20 packets and
> >       then suddenly transferring that non-existing data is impacted.
> >
> >       References
> >
> >       QUIC-LR is very old ref and even the filename has changed a few
> >       times from draft-tsvwg-...
> >
> >
> >       Nits
> >
> >       "runt" made the sentence incomprehendable for me, I had to look
> >       up
> >       that word (and only second source was descriptive enough that I
> >       think I could understand the meaning, and I'd guessed wrong
> >       because
> >       I thought there was some special intent here because of the
> >       quotes).
> >
> >       is dicussed -> is discussed
> >       RACT.RTT -> RACK.rtt
> >       First "TSO" -> TCP Segmentation Offload (TSO)
> >       grainularity -> granularity
> >       higher-sequenc -> higher-sequence
> >
> >
> >       --
> >        i.
> >
> >
> >


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On Fri, 8 May 2020, Yuchung Cheng wrote:

> On Mon, Apr 6, 2020 at 2:01 PM Ilpo J=C3=A4rvinen
> <ilpo.jarvinen@cs.helsinki.fi> wrote:
> >
> > Hi Yuchung,
> >
> > Certainly improved on many things, however, I'm surprised that the lo=
st
> > rexmit comment was not addressed at all. Aren't the rexmit losses
> > supposed to be detected by RACK or have I misunderstood something?
> >
> > I don't understand how lost rexmits are detected by the algorithms in
> > the draft. I think the algorithms are either flawed or lack something
> > (mainly RACK_detect_loss is relevant I think). My original comment
> > from below.
> >
> >        Is RACK_detect_loss() supposed to detect also lost rexmits? Ho=
w?
> >        The rexmit code itself is not in the draft so it's hard to eva=
luate
> >        how the Packet booleans are supposed to work in the first plac=
e.
> >        I'm not saying the rexmit code should be included but currentl=
y it
> >        surely looks like any rexmitted packet just remains .lost =3D =
TRUE
> >        and .retransmitted =3D TRUE state until RTO?
>=20
> Sorry to miss this point earlier -- RACK definitely detects
> lost-retransmits, but the current algorithm does not properly define
> some variables. Here is how I propose to address these by adding
> these:

Np and I was pleased to note already during the interim presentation=20
that you'd addressed this.

> Per-packet variables
>=20
>=20
> Theses variables indicate the status of the most recent transmission
> of a data packet:
>         =E2=80=9CPacket.lost=E2=80=9D is true if the most recent (re)tr=
ansmission of
> the Packet has been considered lost and needs to be retransmitted.
> False otherwise.
>         =E2=80=9CPacket.retransmitted=E2=80=9D is tue if its sequence n=
umbers were
> retransmitted in the most recent transmission. False otherwise.
>         =E2=80=9CPacket.xmit_ts=E2=80=9D is the time of the last transm=
ission of a
> data packet, including retransmissions, if any. The sender needs to
> record the transmission time for each packet sent and not yet
> acknowledged. The time MUST be stored at millisecond granularity or
> finer.
>=20
>=20
> Transmitting a data packet
> Upon transmitting a new packet or retransmitting an old packet, record
> the time in Packet.xmit_ts.
>=20
> --- src
>=20
> RACK_transmit_data(Packet):
>=20
> Packet.time_ts =3D Now()
>=20
> Packet.lost =3D FALSE
>=20
>=20
> RACK_retransmit_data(Packet)
>=20
> Packet.retransmitted =3D TRUE
>       RACK_transmit_data(Packet)
>=20
> ---
>=20
> When a packet is retransmitted again, its Packet.lost is reset and
> transmission timestamp is renewed. Therefore, when its transmission
> time falls out of the RACK reordering window in RACK_detect_loss(), it
> will be marked as lost (again).
>=20
>=20
>=20
>=20
> >
> > Also my timeout "some" vs "all" internal inconsistency comment relate=
s to
> > this same algorithm:
> >
> >          "For timely loss detection, the sender MAY
> >           install a "reordering settling" timer set to fire at the ea=
rliest
> >           moment at which it is safe to conclude that some packet is =
lost.  The
> >           earliest moment is the time it takes to expire the reorderi=
ng window
> >           of the earliest unacked packet in flight."
> >        vs
> >           RACK_detect_loss():
> >              ...
> >                  timeout =3D max(remaining, timeout)
> >        Which way it is, "some" or "all"? The use of max() in RACK_det=
ect_loss()
> >        seems to indicate that the timeout will be set according to th=
e longest
> >        time, that is, the latest moment rather than earliest moments. =
Which
> >        is the intented behavior here? (I can there might be merits in =
both
> >        approaches).
>=20
> Comment 1.2: good catch. We changed to: 'For timely loss detection,
> the sender is RECOMMENDED to install a "reordering settling" timer.
> The timer expires at the earliest moment when it is safe to conclude
> that all the unacknowledged packets within the reordering window are
> lost.'

I was under impression it's likely what you intented to propose right=20
from the start. In that case this change seems not enough to address my=20
concern, the algorithm seems to do exactly opposite on
  timeout =3D max(remaining,  timeout)
line, no? That is, if there is a moment further away in the future,=20
the line selects it rather than "the earliest moment". This inconsistency=20
between the text and algorithm is why I brought this up and I think the=20
algorithm requires some small tweak to remove the inconsistency here.


--=20
 i.

>=20
>=20
> >
> >
> > Then there are some other comments I made which seem also unaddressed =
but
> > I'll not duplicate them all here.
> >
> >
> > --
> >  i.
> >
> >
> > On Mon, 9 Mar 2020, Yuchung Cheng wrote:
> >
> > > Hi Ilpo,
> > > Thanks for a detailed review and the great suggestions. Here's all =
the edits
> > > we made.
> > > https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-rack-08
> > >
> > >  We defined the missing terms and addressed the nits, removed the l=
ess
> > > critical parts like recommending PRR, and edited TLP algorithms (se=
ctions
> > > 7.4 to 7.6). I hope you find it's better now and would be happy to =
hear any
> > > improvements you'd like to make.
> > >
> > > Yuchung
> > >
> > > On Fri, Feb 14, 2020 at 4:22 AM Ilpo J=C3=A4rvinen <ilpo.jarvinen@c=
s.helsinki.fi>
> > > wrote:
> > >       In general, I find the draft quite good shape already, it's
> > >       quite easy
> > >       to read and follow (that is, except the TLP sections that are
> > >       specifically
> > >       mentioned below).
> > >
> > >       Here's the list of issues I came across:
> > >
> > >       Sect 6.1
> > >
> > >       "RACK.pkts_sacked" lacks the empty line separator.
> > >
> > >       Sect 7.2
> > >
> > >         "This
> > >          heuristic would fail to update RACK stats if the packet is
> > >       spuriously
> > >          retransmitted because of a recent minimum RTT decrease (e.=
g.
> > >       path
> > >          change)."
> > >       This defies my logic as RTT decrease should not result in
> > >       spurious
> > >       retransmissions but help to avoid them.
> > >
> > >
> > >         "Note that
> > >          extensions that require additional TCP features (e.g.  DSA=
CK)
> > >       would
> > >          work if the feature functions simply return false."
> > >       Please rephrase this. I suspect you meant that if some functi=
ons
> > >       relate
> > >       to TCP features that are not available, the given code still
> > >       works
> > >       as long as the functions related to such features return fals=
e
> > >       when
> > >       the TCP feature is not available but that surely isn't what t=
he
> > >       sentence currently says.
> > >
> > >       Sect 7.2 Step 5
> > >
> > >         "For timely loss detection, the sender MAY
> > >          install a "reordering settling" timer set to fire at the
> > >       earliest
> > >          moment at which it is safe to conclude that some packet is
> > >       lost.  The
> > >          earliest moment is the time it takes to expire the reorder=
ing
> > >       window
> > >          of the earliest unacked packet in flight."
> > >       vs
> > >          RACK_detect_loss():
> > >             ...
> > >                 timeout =3D max(remaining, timeout)
> > >       Which way it is, "some" or "all"? The use of max() in
> > >       RACK_detect_lost()
> > >       seems to indicate that the timeout will be set according to t=
he
> > >       longest
> > >       time, that is, the latest moment rather than earliest moments=
.
> > >       Which
> > >       is the intented behavior here? (I can there might be merits i=
n
> > >       both
> > >       approaches).
> > >
> > >
> > >       In general, I didn't like the presentation order here as this
> > >       form looks
> > >       the most obvious to me:
> > >          now >=3D Packet.xmit_ts + RACK.reo_wnd + (RACK.ack_ts -
> > >       RACK.xmit_ts)
> > >       ...and I find the derivation of it just complicating things
> > >       rather
> > >       than helping but that's perhaps just my opinion.
> > >
> > >
> > >          "It is also useful when the timestamp clock granularity is
> > >       close to
> > >           or longer than the actual round trip time."
> > >       Given that some things based on timestamps will obviously sto=
p
> > >       working
> > >       if the clock is slower than one tick per RTT I found this
> > >       fragment a bit
> > >       odd. RFC 7323 says "the clock SHOULD tick at least once per
> > >       window's worth
> > >       of data". I can understand though that a host might find cloc=
k
> > >       tick
> > >       granularity challenging due to extensive RTT range but on the
> > >       same
> > >       I'd not want such a "too slow clock" behavior to be endorsed =
as
> > >       if
> > >       it would be just normal. That is, I'm not against RACK handli=
ng
> > >       also this case nicely but it would be nice to somehow downpla=
y
> > >       the
> > >       normalness tone in it with something along the lines of "This
> > >       check is
> > >       also useful if the recommended one tick per RTT cannot be
> > >       achieved due
> > >       to timestamp clock granularity."
> > >
> > >       Is RACK_detect_lost() supposed to detect also lost rexmits? H=
ow?
> > >       The rexmit code itself is not in the draft so it's hard to
> > >       evaluate
> > >       how the Packet booleans are supposed to work in the first pla=
ce.
> > >       I'm not saying the rexmit code should be included but current=
ly
> > >       it
> > >       surely looks like any rexmitted packet just remains .lost =3D =
TRUE
> > >       and
> > >       .retransmitted =3D TRUE state until RTO?
> > >
> > >
> > >       Sect 7.4
> > >
> > >       "PTO" just appears out of nowhere (and is not even written op=
en
> > >       in the first instance).
> > >
> > >          "2. ... in step (2) of the algorithm."
> > >       "step (2)" is forward-refing, it would naturally refer back t=
o
> > >       "Step 2: Update RACK stats" so it makes very little sense as =
is.
> > >
> > >       Sect 7.5
> > >
> > >       Separator missing for PTO: and TLPRxtOut:
> > >
> > >       Sect 7.5.1
> > >
> > >         "3.  Upon receiving a SACK that selectively acknowledges da=
ta
> > >       that was
> > >              last sent before the segment with SEG.SEQ=3DSND.UNA wa=
s
> > >       last
> > >              (re)transmitted"
> > >       This is hard to understand, perhaps rephrase it somehow?
> > >
> > >
> > >         "But choosing PTO to be
> > >          exactly an SRTT is likely to generate spurious probes give=
n
> > >       that
> > >          network delay variance and even end-system timings can eas=
ily
> > >       push an
> > >          ACK to be above an SRTT.  We chose PTO to be the next
> > >       integral
> > >          multiple of SRTT.
> > >
> > >          Similarly, network delay variations and end-system process=
ing
> > >          latencies and timer granularities can easily delay ACKs
> > >       beyond
> > >          2*SRTT, so senders SHOULD add at least 2ms to a computed P=
TO
> > >       value
> > >          (and MAY add more if the sending host OS timer granularity =
is
> > >       more
> > >          coarse than 1ms)."
> > >       Does this later paragraph depend on some assumptions (such as
> > >       very
> > >       low RTT?) as I don't otherwise understand why you say what yo=
u
> > >       say
> > >       in the given order. First you said x & y can push ACK over SR=
TT
> > >       so
> > >       we pick 2* but then you say the same things can push over 2*S=
RTT
> > >       too?
> > >
> > >       WCDelAckT to terminology?
> > >
> > >       Sect 7.5.2
> > >
> > >       TLP_send_probe():
> > >               - Perhaps set TLPRxtOut somewhere?
> > >               - Add "Arm RTO" to the end?
> > >
> > >         "In the case where
> > >          there is only one original outstanding segment of data (N=3D=
1),
> > >       the
> > >          same logic (trivially) applies: an ACK for a single
> > >       outstanding
> > >          segment tells the sender the N-1=3D0 segments preceding th=
at
> > >       segment
> > >          were lost.  Furthermore, whether there are N>1 or N=3D1
> > >       outstanding
> > >          segments,"
> > >       If this distinction is useful to make at all, maybe just simp=
ly
> > >       say:
> > >       "Applies also in case N=3D1."
> > >
> > >       Sect 7.6
> > >
> > >         "SND.NXT at the time the episode started."
> > >       Isn't this TLPHighRxt, why not use it here?
> > >
> > >       Sect 7.6.2
> > >
> > >         "Upon sending a TLP probe that is a retransmission, the sen=
der
> > >       sets
> > >          TLPRxtOut to true and TLPHighRxt to SND.NXT."
> > >       This should appear in TLP_send_probe().
> > >
> > >       "Detecting recoveries accomplished by loss probes"
> > >       Did you want to start another section here?
> > >
> > >       "Step 1: Track ACKs ..." until what point? (It's told on late=
r
> > >       but
> > >       this question comes into mind when reading the draft in order=
).
> > >
> > >
> > >       Sect 7.4 - 7.6.2
> > >
> > >       In general, this particular part of the draft seems to requir=
e
> > >       some
> > >       further editing. After reading it all through, I feel most bi=
ts
> > >       are
> > >       present but the text is very hard to follow. I think the
> > >       presentation
> > >       order needs significant work. There are random bits here and
> > >       there
> > >       that are explained only later. Also, algorithm lacks some par=
ts
> > >       discussed only in the text but variable like items should
> > >       definitely
> > >       be included to the algorithm.
> > >
> > >
> > >       Sect 8.4
> > >
> > >         "RACK is compatible with and does not interfere with the th=
e
> > >       standard
> > >          RTO [RFC6298], RTO-restart [RFC7765], F-RTO [RFC5682] and
> > >       Eifel
> > >          algorithms [RFC3522]. ...
> > >          It neither changes the RTO timer calculation nor detects
> > >       spurious
> > >          timeouts."
> > >       For F-RTO, also send order is significant.
> > >
> > >       Sect 8.5
> > >
> > >       SHOULDs PRR (and also placed to normative references) that is
> > >       only
> > >       experimental and RACK aims to standards track. I feel that PR=
R
> > >       is
> > >       disjoint enough anyway, which I think was also your intention
> > >       when
> > >       maintaining the decoupling CC and RACK, that it's bit odd to
> > >       take
> > >       this strong stance on PRR (indepedent of the stds track vs
> > >       experimental thing).
> > >
> > >         "due to congestion window validation."
> > >       Does this refer to CWV as in RFC? Or some other mechanism in
> > >       which
> > >       case you shouldn't use the same term here.
> > >
> > >       I find the described scenario somewhat unconvincing becase
> > >       the description ends with:
> > >         "...
> > >          The ending congestion window after recovery also impacts
> > >          subsequent data transfer."
> > >       ...First there wasn't enough data to fill cwnd of 20 packets =
and
> > >       then suddenly transferring that non-existing data is impacted=
.
> > >
> > >       References
> > >
> > >       QUIC-LR is very old ref and even the filename has changed a f=
ew
> > >       times from draft-tsvwg-...
> > >
> > >
> > >       Nits
> > >
> > >       "runt" made the sentence incomprehendable for me, I had to lo=
ok
> > >       up
> > >       that word (and only second source was descriptive enough that =
I
> > >       think I could understand the meaning, and I'd guessed wrong
> > >       because
> > >       I thought there was some special intent here because of the
> > >       quotes).
> > >
> > >       is dicussed -> is discussed
> > >       RACT.RTT -> RACK.rtt
> > >       First "TSO" -> TCP Segmentation Offload (TSO)
> > >       grainularity -> granularity
> > >       higher-sequenc -> higher-sequence
> > >
> > >
> > >       --
> > >        i.
> > >
> > >
> > >
>
--=_script-32421-1589013040-0001-2--


From nobody Mon May 11 09:40:37 2020
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From: Yuchung Cheng <ycheng@google.com>
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Neal Cardwell <ncardwell@google.com>,  Nandita Dukkipati <nanditad@google.com>, Priyaranjan Jha <priyarjha@google.com>
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Subject: Re: [tcpm] Review of draft-ietf-tcpm-rack-07
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On Sat, May 9, 2020 at 1:30 AM Ilpo J=C3=A4rvinen
<ilpo.jarvinen@cs.helsinki.fi> wrote:
>
> On Fri, 8 May 2020, Yuchung Cheng wrote:
>
> > On Mon, Apr 6, 2020 at 2:01 PM Ilpo J=C3=A4rvinen
> > <ilpo.jarvinen@cs.helsinki.fi> wrote:
> > >
> > > Hi Yuchung,
> > >
> > > Certainly improved on many things, however, I'm surprised that the lo=
st
> > > rexmit comment was not addressed at all. Aren't the rexmit losses
> > > supposed to be detected by RACK or have I misunderstood something?
> > >
> > > I don't understand how lost rexmits are detected by the algorithms in
> > > the draft. I think the algorithms are either flawed or lack something
> > > (mainly RACK_detect_loss is relevant I think). My original comment
> > > from below.
> > >
> > >        Is RACK_detect_loss() supposed to detect also lost rexmits? Ho=
w?
> > >        The rexmit code itself is not in the draft so it's hard to eva=
luate
> > >        how the Packet booleans are supposed to work in the first plac=
e.
> > >        I'm not saying the rexmit code should be included but currentl=
y it
> > >        surely looks like any rexmitted packet just remains .lost =3D =
TRUE
> > >        and .retransmitted =3D TRUE state until RTO?
> >
> > Sorry to miss this point earlier -- RACK definitely detects
> > lost-retransmits, but the current algorithm does not properly define
> > some variables. Here is how I propose to address these by adding
> > these:
>
> Np and I was pleased to note already during the interim presentation
> that you'd addressed this.
>
> > Per-packet variables
> >
> >
> > Theses variables indicate the status of the most recent transmission
> > of a data packet:
> >         =E2=80=9CPacket.lost=E2=80=9D is true if the most recent (re)tr=
ansmission of
> > the Packet has been considered lost and needs to be retransmitted.
> > False otherwise.
> >         =E2=80=9CPacket.retransmitted=E2=80=9D is tue if its sequence n=
umbers were
> > retransmitted in the most recent transmission. False otherwise.
> >         =E2=80=9CPacket.xmit_ts=E2=80=9D is the time of the last transm=
ission of a
> > data packet, including retransmissions, if any. The sender needs to
> > record the transmission time for each packet sent and not yet
> > acknowledged. The time MUST be stored at millisecond granularity or
> > finer.
> >
> >
> > Transmitting a data packet
> > Upon transmitting a new packet or retransmitting an old packet, record
> > the time in Packet.xmit_ts.
> >
> > --- src
> >
> > RACK_transmit_data(Packet):
> >
> > Packet.time_ts =3D Now()
> >
> > Packet.lost =3D FALSE
> >
> >
> > RACK_retransmit_data(Packet)
> >
> > Packet.retransmitted =3D TRUE
> >       RACK_transmit_data(Packet)
> >
> > ---
> >
> > When a packet is retransmitted again, its Packet.lost is reset and
> > transmission timestamp is renewed. Therefore, when its transmission
> > time falls out of the RACK reordering window in RACK_detect_loss(), it
> > will be marked as lost (again).
> >
> >
> >
> >
> > >
> > > Also my timeout "some" vs "all" internal inconsistency comment relate=
s to
> > > this same algorithm:
> > >
> > >          "For timely loss detection, the sender MAY
> > >           install a "reordering settling" timer set to fire at the ea=
rliest
> > >           moment at which it is safe to conclude that some packet is =
lost.  The
> > >           earliest moment is the time it takes to expire the reorderi=
ng window
> > >           of the earliest unacked packet in flight."
> > >        vs
> > >           RACK_detect_loss():
> > >              ...
> > >                  timeout =3D max(remaining, timeout)
> > >        Which way it is, "some" or "all"? The use of max() in RACK_det=
ect_loss()
> > >        seems to indicate that the timeout will be set according to th=
e longest
> > >        time, that is, the latest moment rather than earliest moments.=
 Which
> > >        is the intented behavior here? (I can there might be merits in=
 both
> > >        approaches).
> >
> > Comment 1.2: good catch. We changed to: 'For timely loss detection,
> > the sender is RECOMMENDED to install a "reordering settling" timer.
> > The timer expires at the earliest moment when it is safe to conclude
> > that all the unacknowledged packets within the reordering window are
> > lost.'
>
> I was under impression it's likely what you intented to propose right
> from the start. In that case this change seems not enough to address my
> concern, the algorithm seems to do exactly opposite on
>   timeout =3D max(remaining,  timeout)
> line, no? That is, if there is a moment further away in the future,
> the line selects it rather than "the earliest moment". This inconsistency
> between the text and algorithm is why I brought this up and I think the
> algorithm requires some small tweak to remove the inconsistency here.

The diff is not the "earliest moment" wording. it was "the earliest
moment of some packet" but it is now "the earliest moment of all
unacked packets"

Here is the old and proposed new text

...; but this
   risks a timeout (RTO) if no more ACKs come back (e.g, due to losses
   or application limit).  For timely loss detection, the sender MAY
   install a "reordering settling" timer set to fire at the earliest
   moment at which it is safe to conclude that some packet is lost.  The
   earliest moment is the time it takes to expire the reordering window
   of the earliest unacked packet in flight.

new text:

... but this risks a timeout (RTO) if no more ACKs come back (e.g, due
to losses or application-limited transmissions). For timely loss
detection, the sender is RECOMMENDED to install a reordering settling
timer. This timer expires at the earliest moment when it is safe to
conclude that all the unacknowledged packets within the reordering
window are lost.

so the keyword all would make the description match the pseudo code.

>
>
> --
>  i.
>
> >
> >
> > >
> > >
> > > Then there are some other comments I made which seem also unaddressed=
 but
> > > I'll not duplicate them all here.
> > >
> > >
> > > --
> > >  i.
> > >
> > >
> > > On Mon, 9 Mar 2020, Yuchung Cheng wrote:
> > >
> > > > Hi Ilpo,
> > > > Thanks for a detailed review and the great suggestions. Here's all =
the edits
> > > > we made.
> > > > https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-rack-08
> > > >
> > > >  We defined the missing terms and addressed the nits, removed the l=
ess
> > > > critical parts like recommending PRR, and edited TLP algorithms (se=
ctions
> > > > 7.4 to 7.6). I hope you find it's better now and would be happy to =
hear any
> > > > improvements you'd like to make.
> > > >
> > > > Yuchung
> > > >
> > > > On Fri, Feb 14, 2020 at 4:22 AM Ilpo J=C3=A4rvinen <ilpo.jarvinen@c=
s.helsinki.fi>
> > > > wrote:
> > > >       In general, I find the draft quite good shape already, it's
> > > >       quite easy
> > > >       to read and follow (that is, except the TLP sections that are
> > > >       specifically
> > > >       mentioned below).
> > > >
> > > >       Here's the list of issues I came across:
> > > >
> > > >       Sect 6.1
> > > >
> > > >       "RACK.pkts_sacked" lacks the empty line separator.
> > > >
> > > >       Sect 7.2
> > > >
> > > >         "This
> > > >          heuristic would fail to update RACK stats if the packet is
> > > >       spuriously
> > > >          retransmitted because of a recent minimum RTT decrease (e.=
g.
> > > >       path
> > > >          change)."
> > > >       This defies my logic as RTT decrease should not result in
> > > >       spurious
> > > >       retransmissions but help to avoid them.
> > > >
> > > >
> > > >         "Note that
> > > >          extensions that require additional TCP features (e.g.  DSA=
CK)
> > > >       would
> > > >          work if the feature functions simply return false."
> > > >       Please rephrase this. I suspect you meant that if some functi=
ons
> > > >       relate
> > > >       to TCP features that are not available, the given code still
> > > >       works
> > > >       as long as the functions related to such features return fals=
e
> > > >       when
> > > >       the TCP feature is not available but that surely isn't what t=
he
> > > >       sentence currently says.
> > > >
> > > >       Sect 7.2 Step 5
> > > >
> > > >         "For timely loss detection, the sender MAY
> > > >          install a "reordering settling" timer set to fire at the
> > > >       earliest
> > > >          moment at which it is safe to conclude that some packet is
> > > >       lost.  The
> > > >          earliest moment is the time it takes to expire the reorder=
ing
> > > >       window
> > > >          of the earliest unacked packet in flight."
> > > >       vs
> > > >          RACK_detect_loss():
> > > >             ...
> > > >                 timeout =3D max(remaining, timeout)
> > > >       Which way it is, "some" or "all"? The use of max() in
> > > >       RACK_detect_lost()
> > > >       seems to indicate that the timeout will be set according to t=
he
> > > >       longest
> > > >       time, that is, the latest moment rather than earliest moments=
.
> > > >       Which
> > > >       is the intented behavior here? (I can there might be merits i=
n
> > > >       both
> > > >       approaches).
> > > >
> > > >
> > > >       In general, I didn't like the presentation order here as this
> > > >       form looks
> > > >       the most obvious to me:
> > > >          now >=3D Packet.xmit_ts + RACK.reo_wnd + (RACK.ack_ts -
> > > >       RACK.xmit_ts)
> > > >       ...and I find the derivation of it just complicating things
> > > >       rather
> > > >       than helping but that's perhaps just my opinion.
> > > >
> > > >
> > > >          "It is also useful when the timestamp clock granularity is
> > > >       close to
> > > >           or longer than the actual round trip time."
> > > >       Given that some things based on timestamps will obviously sto=
p
> > > >       working
> > > >       if the clock is slower than one tick per RTT I found this
> > > >       fragment a bit
> > > >       odd. RFC 7323 says "the clock SHOULD tick at least once per
> > > >       window's worth
> > > >       of data". I can understand though that a host might find cloc=
k
> > > >       tick
> > > >       granularity challenging due to extensive RTT range but on the
> > > >       same
> > > >       I'd not want such a "too slow clock" behavior to be endorsed =
as
> > > >       if
> > > >       it would be just normal. That is, I'm not against RACK handli=
ng
> > > >       also this case nicely but it would be nice to somehow downpla=
y
> > > >       the
> > > >       normalness tone in it with something along the lines of "This
> > > >       check is
> > > >       also useful if the recommended one tick per RTT cannot be
> > > >       achieved due
> > > >       to timestamp clock granularity."
> > > >
> > > >       Is RACK_detect_lost() supposed to detect also lost rexmits? H=
ow?
> > > >       The rexmit code itself is not in the draft so it's hard to
> > > >       evaluate
> > > >       how the Packet booleans are supposed to work in the first pla=
ce.
> > > >       I'm not saying the rexmit code should be included but current=
ly
> > > >       it
> > > >       surely looks like any rexmitted packet just remains .lost =3D=
 TRUE
> > > >       and
> > > >       .retransmitted =3D TRUE state until RTO?
> > > >
> > > >
> > > >       Sect 7.4
> > > >
> > > >       "PTO" just appears out of nowhere (and is not even written op=
en
> > > >       in the first instance).
> > > >
> > > >          "2. ... in step (2) of the algorithm."
> > > >       "step (2)" is forward-refing, it would naturally refer back t=
o
> > > >       "Step 2: Update RACK stats" so it makes very little sense as =
is.
> > > >
> > > >       Sect 7.5
> > > >
> > > >       Separator missing for PTO: and TLPRxtOut:
> > > >
> > > >       Sect 7.5.1
> > > >
> > > >         "3.  Upon receiving a SACK that selectively acknowledges da=
ta
> > > >       that was
> > > >              last sent before the segment with SEG.SEQ=3DSND.UNA wa=
s
> > > >       last
> > > >              (re)transmitted"
> > > >       This is hard to understand, perhaps rephrase it somehow?
> > > >
> > > >
> > > >         "But choosing PTO to be
> > > >          exactly an SRTT is likely to generate spurious probes give=
n
> > > >       that
> > > >          network delay variance and even end-system timings can eas=
ily
> > > >       push an
> > > >          ACK to be above an SRTT.  We chose PTO to be the next
> > > >       integral
> > > >          multiple of SRTT.
> > > >
> > > >          Similarly, network delay variations and end-system process=
ing
> > > >          latencies and timer granularities can easily delay ACKs
> > > >       beyond
> > > >          2*SRTT, so senders SHOULD add at least 2ms to a computed P=
TO
> > > >       value
> > > >          (and MAY add more if the sending host OS timer granularity=
 is
> > > >       more
> > > >          coarse than 1ms)."
> > > >       Does this later paragraph depend on some assumptions (such as
> > > >       very
> > > >       low RTT?) as I don't otherwise understand why you say what yo=
u
> > > >       say
> > > >       in the given order. First you said x & y can push ACK over SR=
TT
> > > >       so
> > > >       we pick 2* but then you say the same things can push over 2*S=
RTT
> > > >       too?
> > > >
> > > >       WCDelAckT to terminology?
> > > >
> > > >       Sect 7.5.2
> > > >
> > > >       TLP_send_probe():
> > > >               - Perhaps set TLPRxtOut somewhere?
> > > >               - Add "Arm RTO" to the end?
> > > >
> > > >         "In the case where
> > > >          there is only one original outstanding segment of data (N=
=3D1),
> > > >       the
> > > >          same logic (trivially) applies: an ACK for a single
> > > >       outstanding
> > > >          segment tells the sender the N-1=3D0 segments preceding th=
at
> > > >       segment
> > > >          were lost.  Furthermore, whether there are N>1 or N=3D1
> > > >       outstanding
> > > >          segments,"
> > > >       If this distinction is useful to make at all, maybe just simp=
ly
> > > >       say:
> > > >       "Applies also in case N=3D1."
> > > >
> > > >       Sect 7.6
> > > >
> > > >         "SND.NXT at the time the episode started."
> > > >       Isn't this TLPHighRxt, why not use it here?
> > > >
> > > >       Sect 7.6.2
> > > >
> > > >         "Upon sending a TLP probe that is a retransmission, the sen=
der
> > > >       sets
> > > >          TLPRxtOut to true and TLPHighRxt to SND.NXT."
> > > >       This should appear in TLP_send_probe().
> > > >
> > > >       "Detecting recoveries accomplished by loss probes"
> > > >       Did you want to start another section here?
> > > >
> > > >       "Step 1: Track ACKs ..." until what point? (It's told on late=
r
> > > >       but
> > > >       this question comes into mind when reading the draft in order=
).
> > > >
> > > >
> > > >       Sect 7.4 - 7.6.2
> > > >
> > > >       In general, this particular part of the draft seems to requir=
e
> > > >       some
> > > >       further editing. After reading it all through, I feel most bi=
ts
> > > >       are
> > > >       present but the text is very hard to follow. I think the
> > > >       presentation
> > > >       order needs significant work. There are random bits here and
> > > >       there
> > > >       that are explained only later. Also, algorithm lacks some par=
ts
> > > >       discussed only in the text but variable like items should
> > > >       definitely
> > > >       be included to the algorithm.
> > > >
> > > >
> > > >       Sect 8.4
> > > >
> > > >         "RACK is compatible with and does not interfere with the th=
e
> > > >       standard
> > > >          RTO [RFC6298], RTO-restart [RFC7765], F-RTO [RFC5682] and
> > > >       Eifel
> > > >          algorithms [RFC3522]. ...
> > > >          It neither changes the RTO timer calculation nor detects
> > > >       spurious
> > > >          timeouts."
> > > >       For F-RTO, also send order is significant.
> > > >
> > > >       Sect 8.5
> > > >
> > > >       SHOULDs PRR (and also placed to normative references) that is
> > > >       only
> > > >       experimental and RACK aims to standards track. I feel that PR=
R
> > > >       is
> > > >       disjoint enough anyway, which I think was also your intention
> > > >       when
> > > >       maintaining the decoupling CC and RACK, that it's bit odd to
> > > >       take
> > > >       this strong stance on PRR (indepedent of the stds track vs
> > > >       experimental thing).
> > > >
> > > >         "due to congestion window validation."
> > > >       Does this refer to CWV as in RFC? Or some other mechanism in
> > > >       which
> > > >       case you shouldn't use the same term here.
> > > >
> > > >       I find the described scenario somewhat unconvincing becase
> > > >       the description ends with:
> > > >         "...
> > > >          The ending congestion window after recovery also impacts
> > > >          subsequent data transfer."
> > > >       ...First there wasn't enough data to fill cwnd of 20 packets =
and
> > > >       then suddenly transferring that non-existing data is impacted=
.
> > > >
> > > >       References
> > > >
> > > >       QUIC-LR is very old ref and even the filename has changed a f=
ew
> > > >       times from draft-tsvwg-...
> > > >
> > > >
> > > >       Nits
> > > >
> > > >       "runt" made the sentence incomprehendable for me, I had to lo=
ok
> > > >       up
> > > >       that word (and only second source was descriptive enough that=
 I
> > > >       think I could understand the meaning, and I'd guessed wrong
> > > >       because
> > > >       I thought there was some special intent here because of the
> > > >       quotes).
> > > >
> > > >       is dicussed -> is discussed
> > > >       RACT.RTT -> RACK.rtt
> > > >       First "TSO" -> TCP Segmentation Offload (TSO)
> > > >       grainularity -> granularity
> > > >       higher-sequenc -> higher-sequence
> > > >
> > > >
> > > >       --
> > > >        i.
> > > >
> > > >
> > > >
> >


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On Mon, 11 May 2020, Yuchung Cheng wrote:

> On Sat, May 9, 2020 at 1:30 AM Ilpo J=C3=A4rvinen
> <ilpo.jarvinen@cs.helsinki.fi> wrote:
> >
> > On Fri, 8 May 2020, Yuchung Cheng wrote:
> >
> > > On Mon, Apr 6, 2020 at 2:01 PM Ilpo J=C3=A4rvinen
> > > <ilpo.jarvinen@cs.helsinki.fi> wrote:
> > > >
> > > > Hi Yuchung,
> > > >
> > > > Certainly improved on many things, however, I'm surprised that th=
e lost
> > > > rexmit comment was not addressed at all. Aren't the rexmit losses
> > > > supposed to be detected by RACK or have I misunderstood something=
?
> > > >
> > > > I don't understand how lost rexmits are detected by the algorithm=
s in
> > > > the draft. I think the algorithms are either flawed or lack somet=
hing
> > > > (mainly RACK_detect_loss is relevant I think). My original commen=
t
> > > > from below.
> > > >
> > > >        Is RACK_detect_loss() supposed to detect also lost rexmits=
? How?
> > > >        The rexmit code itself is not in the draft so it's hard to =
evaluate
> > > >        how the Packet booleans are supposed to work in the first =
place.
> > > >        I'm not saying the rexmit code should be included but curr=
ently it
> > > >        surely looks like any rexmitted packet just remains .lost =
=3D TRUE
> > > >        and .retransmitted =3D TRUE state until RTO?
> > >
> > > Sorry to miss this point earlier -- RACK definitely detects
> > > lost-retransmits, but the current algorithm does not properly defin=
e
> > > some variables. Here is how I propose to address these by adding
> > > these:
> >
> > Np and I was pleased to note already during the interim presentation
> > that you'd addressed this.
> >
> > > Per-packet variables
> > >
> > >
> > > Theses variables indicate the status of the most recent transmissio=
n
> > > of a data packet:
> > >         =E2=80=9CPacket.lost=E2=80=9D is true if the most recent (r=
e)transmission of
> > > the Packet has been considered lost and needs to be retransmitted.
> > > False otherwise.
> > >         =E2=80=9CPacket.retransmitted=E2=80=9D is tue if its sequen=
ce numbers were
> > > retransmitted in the most recent transmission. False otherwise.
> > >         =E2=80=9CPacket.xmit_ts=E2=80=9D is the time of the last tr=
ansmission of a
> > > data packet, including retransmissions, if any. The sender needs to
> > > record the transmission time for each packet sent and not yet
> > > acknowledged. The time MUST be stored at millisecond granularity or
> > > finer.
> > >
> > >
> > > Transmitting a data packet
> > > Upon transmitting a new packet or retransmitting an old packet, rec=
ord
> > > the time in Packet.xmit_ts.
> > >
> > > --- src
> > >
> > > RACK_transmit_data(Packet):
> > >
> > > Packet.time_ts =3D Now()
> > >
> > > Packet.lost =3D FALSE
> > >
> > >
> > > RACK_retransmit_data(Packet)
> > >
> > > Packet.retransmitted =3D TRUE
> > >       RACK_transmit_data(Packet)
> > >
> > > ---
> > >
> > > When a packet is retransmitted again, its Packet.lost is reset and
> > > transmission timestamp is renewed. Therefore, when its transmission
> > > time falls out of the RACK reordering window in RACK_detect_loss(), =
it
> > > will be marked as lost (again).
> > >
> > >
> > >
> > >
> > > >
> > > > Also my timeout "some" vs "all" internal inconsistency comment re=
lates to
> > > > this same algorithm:
> > > >
> > > >          "For timely loss detection, the sender MAY
> > > >           install a "reordering settling" timer set to fire at th=
e earliest
> > > >           moment at which it is safe to conclude that some packet =
is lost.  The
> > > >           earliest moment is the time it takes to expire the reor=
dering window
> > > >           of the earliest unacked packet in flight."
> > > >        vs
> > > >           RACK_detect_loss():
> > > >              ...
> > > >                  timeout =3D max(remaining, timeout)
> > > >        Which way it is, "some" or "all"? The use of max() in RACK=
_detect_loss()
> > > >        seems to indicate that the timeout will be set according t=
o the longest
> > > >        time, that is, the latest moment rather than earliest mome=
nts. Which
> > > >        is the intented behavior here? (I can there might be merit=
s in both
> > > >        approaches).
> > >
> > > Comment 1.2: good catch. We changed to: 'For timely loss detection,
> > > the sender is RECOMMENDED to install a "reordering settling" timer.
> > > The timer expires at the earliest moment when it is safe to conclud=
e
> > > that all the unacknowledged packets within the reordering window ar=
e
> > > lost.'
> >
> > I was under impression it's likely what you intented to propose right
> > from the start. In that case this change seems not enough to address =
my
> > concern, the algorithm seems to do exactly opposite on
> >   timeout =3D max(remaining,  timeout)
> > line, no? That is, if there is a moment further away in the future,
> > the line selects it rather than "the earliest moment". This inconsist=
ency
> > between the text and algorithm is why I brought this up and I think t=
he
> > algorithm requires some small tweak to remove the inconsistency here.
>=20
> The diff is not the "earliest moment" wording. it was "the earliest
> moment of some packet" but it is now "the earliest moment of all
> unacked packets"
>=20
> Here is the old and proposed new text
>=20
> ...; but this
>    risks a timeout (RTO) if no more ACKs come back (e.g, due to losses
>    or application limit).  For timely loss detection, the sender MAY
>    install a "reordering settling" timer set to fire at the earliest
>    moment at which it is safe to conclude that some packet is lost.  Th=
e
>    earliest moment is the time it takes to expire the reordering window
>    of the earliest unacked packet in flight.
>=20
> new text:
>=20
> ... but this risks a timeout (RTO) if no more ACKs come back (e.g, due
> to losses or application-limited transmissions). For timely loss
> detection, the sender is RECOMMENDED to install a reordering settling
> timer. This timer expires at the earliest moment when it is safe to
> conclude that all the unacknowledged packets within the reordering
> window are lost.
>=20
> so the keyword all would make the description match the pseudo code.

Ah, you're of course right. It was sloppy reading (and too strong=20
preconceived idea of what the fix should look like) from my side.
So the correction is fine. My apologies for the noise.

--=20
 i.


>=20
> >
> >
> > --
> >  i.
> >
> > >
> > >
> > > >
> > > >
> > > > Then there are some other comments I made which seem also unaddre=
ssed but
> > > > I'll not duplicate them all here.
> > > >
> > > >
> > > > --
> > > >  i.
> > > >
> > > >
> > > > On Mon, 9 Mar 2020, Yuchung Cheng wrote:
> > > >
> > > > > Hi Ilpo,
> > > > > Thanks for a detailed review and the great suggestions. Here's =
all the edits
> > > > > we made.
> > > > > https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-rack-08
> > > > >
> > > > >  We defined the missing terms and addressed the nits, removed t=
he less
> > > > > critical parts like recommending PRR, and edited TLP algorithms =
(sections
> > > > > 7.4 to 7.6). I hope you find it's better now and would be happy =
to hear any
> > > > > improvements you'd like to make.
> > > > >
> > > > > Yuchung
> > > > >
> > > > > On Fri, Feb 14, 2020 at 4:22 AM Ilpo J=C3=A4rvinen <ilpo.jarvin=
en@cs.helsinki.fi>
> > > > > wrote:
> > > > >       In general, I find the draft quite good shape already, it=
's
> > > > >       quite easy
> > > > >       to read and follow (that is, except the TLP sections that =
are
> > > > >       specifically
> > > > >       mentioned below).
> > > > >
> > > > >       Here's the list of issues I came across:
> > > > >
> > > > >       Sect 6.1
> > > > >
> > > > >       "RACK.pkts_sacked" lacks the empty line separator.
> > > > >
> > > > >       Sect 7.2
> > > > >
> > > > >         "This
> > > > >          heuristic would fail to update RACK stats if the packe=
t is
> > > > >       spuriously
> > > > >          retransmitted because of a recent minimum RTT decrease =
(e.g.
> > > > >       path
> > > > >          change)."
> > > > >       This defies my logic as RTT decrease should not result in
> > > > >       spurious
> > > > >       retransmissions but help to avoid them.
> > > > >
> > > > >
> > > > >         "Note that
> > > > >          extensions that require additional TCP features (e.g.  =
DSACK)
> > > > >       would
> > > > >          work if the feature functions simply return false."
> > > > >       Please rephrase this. I suspect you meant that if some fu=
nctions
> > > > >       relate
> > > > >       to TCP features that are not available, the given code st=
ill
> > > > >       works
> > > > >       as long as the functions related to such features return =
false
> > > > >       when
> > > > >       the TCP feature is not available but that surely isn't wh=
at the
> > > > >       sentence currently says.
> > > > >
> > > > >       Sect 7.2 Step 5
> > > > >
> > > > >         "For timely loss detection, the sender MAY
> > > > >          install a "reordering settling" timer set to fire at t=
he
> > > > >       earliest
> > > > >          moment at which it is safe to conclude that some packe=
t is
> > > > >       lost.  The
> > > > >          earliest moment is the time it takes to expire the reo=
rdering
> > > > >       window
> > > > >          of the earliest unacked packet in flight."
> > > > >       vs
> > > > >          RACK_detect_loss():
> > > > >             ...
> > > > >                 timeout =3D max(remaining, timeout)
> > > > >       Which way it is, "some" or "all"? The use of max() in
> > > > >       RACK_detect_lost()
> > > > >       seems to indicate that the timeout will be set according =
to the
> > > > >       longest
> > > > >       time, that is, the latest moment rather than earliest mom=
ents.
> > > > >       Which
> > > > >       is the intented behavior here? (I can there might be meri=
ts in
> > > > >       both
> > > > >       approaches).
> > > > >
> > > > >
> > > > >       In general, I didn't like the presentation order here as =
this
> > > > >       form looks
> > > > >       the most obvious to me:
> > > > >          now >=3D Packet.xmit_ts + RACK.reo_wnd + (RACK.ack_ts =
-
> > > > >       RACK.xmit_ts)
> > > > >       ...and I find the derivation of it just complicating thin=
gs
> > > > >       rather
> > > > >       than helping but that's perhaps just my opinion.
> > > > >
> > > > >
> > > > >          "It is also useful when the timestamp clock granularit=
y is
> > > > >       close to
> > > > >           or longer than the actual round trip time."
> > > > >       Given that some things based on timestamps will obviously =
stop
> > > > >       working
> > > > >       if the clock is slower than one tick per RTT I found this
> > > > >       fragment a bit
> > > > >       odd. RFC 7323 says "the clock SHOULD tick at least once p=
er
> > > > >       window's worth
> > > > >       of data". I can understand though that a host might find =
clock
> > > > >       tick
> > > > >       granularity challenging due to extensive RTT range but on =
the
> > > > >       same
> > > > >       I'd not want such a "too slow clock" behavior to be endor=
sed as
> > > > >       if
> > > > >       it would be just normal. That is, I'm not against RACK ha=
ndling
> > > > >       also this case nicely but it would be nice to somehow dow=
nplay
> > > > >       the
> > > > >       normalness tone in it with something along the lines of "=
This
> > > > >       check is
> > > > >       also useful if the recommended one tick per RTT cannot be
> > > > >       achieved due
> > > > >       to timestamp clock granularity."
> > > > >
> > > > >       Is RACK_detect_lost() supposed to detect also lost rexmit=
s? How?
> > > > >       The rexmit code itself is not in the draft so it's hard t=
o
> > > > >       evaluate
> > > > >       how the Packet booleans are supposed to work in the first =
place.
> > > > >       I'm not saying the rexmit code should be included but cur=
rently
> > > > >       it
> > > > >       surely looks like any rexmitted packet just remains .lost =
=3D TRUE
> > > > >       and
> > > > >       .retransmitted =3D TRUE state until RTO?
> > > > >
> > > > >
> > > > >       Sect 7.4
> > > > >
> > > > >       "PTO" just appears out of nowhere (and is not even writte=
n open
> > > > >       in the first instance).
> > > > >
> > > > >          "2. ... in step (2) of the algorithm."
> > > > >       "step (2)" is forward-refing, it would naturally refer ba=
ck to
> > > > >       "Step 2: Update RACK stats" so it makes very little sense =
as is.
> > > > >
> > > > >       Sect 7.5
> > > > >
> > > > >       Separator missing for PTO: and TLPRxtOut:
> > > > >
> > > > >       Sect 7.5.1
> > > > >
> > > > >         "3.  Upon receiving a SACK that selectively acknowledge=
s data
> > > > >       that was
> > > > >              last sent before the segment with SEG.SEQ=3DSND.UN=
A was
> > > > >       last
> > > > >              (re)transmitted"
> > > > >       This is hard to understand, perhaps rephrase it somehow?
> > > > >
> > > > >
> > > > >         "But choosing PTO to be
> > > > >          exactly an SRTT is likely to generate spurious probes =
given
> > > > >       that
> > > > >          network delay variance and even end-system timings can =
easily
> > > > >       push an
> > > > >          ACK to be above an SRTT.  We chose PTO to be the next
> > > > >       integral
> > > > >          multiple of SRTT.
> > > > >
> > > > >          Similarly, network delay variations and end-system pro=
cessing
> > > > >          latencies and timer granularities can easily delay ACK=
s
> > > > >       beyond
> > > > >          2*SRTT, so senders SHOULD add at least 2ms to a comput=
ed PTO
> > > > >       value
> > > > >          (and MAY add more if the sending host OS timer granula=
rity is
> > > > >       more
> > > > >          coarse than 1ms)."
> > > > >       Does this later paragraph depend on some assumptions (suc=
h as
> > > > >       very
> > > > >       low RTT?) as I don't otherwise understand why you say wha=
t you
> > > > >       say
> > > > >       in the given order. First you said x & y can push ACK ove=
r SRTT
> > > > >       so
> > > > >       we pick 2* but then you say the same things can push over =
2*SRTT
> > > > >       too?
> > > > >
> > > > >       WCDelAckT to terminology?
> > > > >
> > > > >       Sect 7.5.2
> > > > >
> > > > >       TLP_send_probe():
> > > > >               - Perhaps set TLPRxtOut somewhere?
> > > > >               - Add "Arm RTO" to the end?
> > > > >
> > > > >         "In the case where
> > > > >          there is only one original outstanding segment of data =
(N=3D1),
> > > > >       the
> > > > >          same logic (trivially) applies: an ACK for a single
> > > > >       outstanding
> > > > >          segment tells the sender the N-1=3D0 segments precedin=
g that
> > > > >       segment
> > > > >          were lost.  Furthermore, whether there are N>1 or N=3D=
1
> > > > >       outstanding
> > > > >          segments,"
> > > > >       If this distinction is useful to make at all, maybe just =
simply
> > > > >       say:
> > > > >       "Applies also in case N=3D1."
> > > > >
> > > > >       Sect 7.6
> > > > >
> > > > >         "SND.NXT at the time the episode started."
> > > > >       Isn't this TLPHighRxt, why not use it here?
> > > > >
> > > > >       Sect 7.6.2
> > > > >
> > > > >         "Upon sending a TLP probe that is a retransmission, the =
sender
> > > > >       sets
> > > > >          TLPRxtOut to true and TLPHighRxt to SND.NXT."
> > > > >       This should appear in TLP_send_probe().
> > > > >
> > > > >       "Detecting recoveries accomplished by loss probes"
> > > > >       Did you want to start another section here?
> > > > >
> > > > >       "Step 1: Track ACKs ..." until what point? (It's told on =
later
> > > > >       but
> > > > >       this question comes into mind when reading the draft in o=
rder).
> > > > >
> > > > >
> > > > >       Sect 7.4 - 7.6.2
> > > > >
> > > > >       In general, this particular part of the draft seems to re=
quire
> > > > >       some
> > > > >       further editing. After reading it all through, I feel mos=
t bits
> > > > >       are
> > > > >       present but the text is very hard to follow. I think the
> > > > >       presentation
> > > > >       order needs significant work. There are random bits here =
and
> > > > >       there
> > > > >       that are explained only later. Also, algorithm lacks some =
parts
> > > > >       discussed only in the text but variable like items should
> > > > >       definitely
> > > > >       be included to the algorithm.
> > > > >
> > > > >
> > > > >       Sect 8.4
> > > > >
> > > > >         "RACK is compatible with and does not interfere with th=
e the
> > > > >       standard
> > > > >          RTO [RFC6298], RTO-restart [RFC7765], F-RTO [RFC5682] =
and
> > > > >       Eifel
> > > > >          algorithms [RFC3522]. ...
> > > > >          It neither changes the RTO timer calculation nor detec=
ts
> > > > >       spurious
> > > > >          timeouts."
> > > > >       For F-RTO, also send order is significant.
> > > > >
> > > > >       Sect 8.5
> > > > >
> > > > >       SHOULDs PRR (and also placed to normative references) tha=
t is
> > > > >       only
> > > > >       experimental and RACK aims to standards track. I feel tha=
t PRR
> > > > >       is
> > > > >       disjoint enough anyway, which I think was also your inten=
tion
> > > > >       when
> > > > >       maintaining the decoupling CC and RACK, that it's bit odd =
to
> > > > >       take
> > > > >       this strong stance on PRR (indepedent of the stds track v=
s
> > > > >       experimental thing).
> > > > >
> > > > >         "due to congestion window validation."
> > > > >       Does this refer to CWV as in RFC? Or some other mechanism =
in
> > > > >       which
> > > > >       case you shouldn't use the same term here.
> > > > >
> > > > >       I find the described scenario somewhat unconvincing becas=
e
> > > > >       the description ends with:
> > > > >         "...
> > > > >          The ending congestion window after recovery also impac=
ts
> > > > >          subsequent data transfer."
> > > > >       ...First there wasn't enough data to fill cwnd of 20 pack=
ets and
> > > > >       then suddenly transferring that non-existing data is impa=
cted.
> > > > >
> > > > >       References
> > > > >
> > > > >       QUIC-LR is very old ref and even the filename has changed =
a few
> > > > >       times from draft-tsvwg-...
> > > > >
> > > > >
> > > > >       Nits
> > > > >
> > > > >       "runt" made the sentence incomprehendable for me, I had t=
o look
> > > > >       up
> > > > >       that word (and only second source was descriptive enough =
that I
> > > > >       think I could understand the meaning, and I'd guessed wro=
ng
> > > > >       because
> > > > >       I thought there was some special intent here because of t=
he
> > > > >       quotes).
> > > > >
> > > > >       is dicussed -> is discussed
> > > > >       RACT.RTT -> RACK.rtt
> > > > >       First "TSO" -> TCP Segmentation Offload (TSO)
> > > > >       grainularity -> granularity
> > > > >       higher-sequenc -> higher-sequence
> > > > >
> > > > >
> > > > >       --
> > > > >        i.
> > > > >
> > > > >
> > > > >
> > >
>
--=_script-21394-1589216031-0001-2--


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References: <alpine.DEB.2.20.2002141414420.25645@whs-18.cs.helsinki.fi> <CAK6E8=dZN5PnCt7QFoyGJUnSoCcg0XUKxyZCbJAM_2tVN7A8wQ@mail.gmail.com> <alpine.DEB.2.20.2004062342530.14114@whs-18.cs.helsinki.fi> <CAK6E8=eVjMACZSkUWmY9Z+fiid6cdfG310y4O8G32cEeXFAhFQ@mail.gmail.com> <alpine.DEB.2.20.2005091114100.17711@whs-18.cs.helsinki.fi> <CAK6E8=dAvskPOg6-eNPUNmiUojcs81k+Fvu6cAfVgw8C_bOQbg@mail.gmail.com> <alpine.DEB.2.20.2005111949370.14773@whs-18.cs.helsinki.fi>
In-Reply-To: <alpine.DEB.2.20.2005111949370.14773@whs-18.cs.helsinki.fi>
From: Yuchung Cheng <ycheng@google.com>
Date: Mon, 11 May 2020 10:21:19 -0700
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To: =?UTF-8?Q?Ilpo_J=C3=A4rvinen?= <ilpo.jarvinen@cs.helsinki.fi>
Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Neal Cardwell <ncardwell@google.com>,  Nandita Dukkipati <nanditad@google.com>, Priyaranjan Jha <priyarjha@google.com>
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Subject: Re: [tcpm] Review of draft-ietf-tcpm-rack-07
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On Mon, May 11, 2020 at 9:53 AM Ilpo J=C3=A4rvinen
<ilpo.jarvinen@cs.helsinki.fi> wrote:
>
> On Mon, 11 May 2020, Yuchung Cheng wrote:
>
> > On Sat, May 9, 2020 at 1:30 AM Ilpo J=C3=A4rvinen
> > <ilpo.jarvinen@cs.helsinki.fi> wrote:
> > >
> > > On Fri, 8 May 2020, Yuchung Cheng wrote:
> > >
> > > > On Mon, Apr 6, 2020 at 2:01 PM Ilpo J=C3=A4rvinen
> > > > <ilpo.jarvinen@cs.helsinki.fi> wrote:
> > > > >
> > > > > Hi Yuchung,
> > > > >
> > > > > Certainly improved on many things, however, I'm surprised that th=
e lost
> > > > > rexmit comment was not addressed at all. Aren't the rexmit losses
> > > > > supposed to be detected by RACK or have I misunderstood something=
?
> > > > >
> > > > > I don't understand how lost rexmits are detected by the algorithm=
s in
> > > > > the draft. I think the algorithms are either flawed or lack somet=
hing
> > > > > (mainly RACK_detect_loss is relevant I think). My original commen=
t
> > > > > from below.
> > > > >
> > > > >        Is RACK_detect_loss() supposed to detect also lost rexmits=
? How?
> > > > >        The rexmit code itself is not in the draft so it's hard to=
 evaluate
> > > > >        how the Packet booleans are supposed to work in the first =
place.
> > > > >        I'm not saying the rexmit code should be included but curr=
ently it
> > > > >        surely looks like any rexmitted packet just remains .lost =
=3D TRUE
> > > > >        and .retransmitted =3D TRUE state until RTO?
> > > >
> > > > Sorry to miss this point earlier -- RACK definitely detects
> > > > lost-retransmits, but the current algorithm does not properly defin=
e
> > > > some variables. Here is how I propose to address these by adding
> > > > these:
> > >
> > > Np and I was pleased to note already during the interim presentation
> > > that you'd addressed this.
> > >
> > > > Per-packet variables
> > > >
> > > >
> > > > Theses variables indicate the status of the most recent transmissio=
n
> > > > of a data packet:
> > > >         =E2=80=9CPacket.lost=E2=80=9D is true if the most recent (r=
e)transmission of
> > > > the Packet has been considered lost and needs to be retransmitted.
> > > > False otherwise.
> > > >         =E2=80=9CPacket.retransmitted=E2=80=9D is tue if its sequen=
ce numbers were
> > > > retransmitted in the most recent transmission. False otherwise.
> > > >         =E2=80=9CPacket.xmit_ts=E2=80=9D is the time of the last tr=
ansmission of a
> > > > data packet, including retransmissions, if any. The sender needs to
> > > > record the transmission time for each packet sent and not yet
> > > > acknowledged. The time MUST be stored at millisecond granularity or
> > > > finer.
> > > >
> > > >
> > > > Transmitting a data packet
> > > > Upon transmitting a new packet or retransmitting an old packet, rec=
ord
> > > > the time in Packet.xmit_ts.
> > > >
> > > > --- src
> > > >
> > > > RACK_transmit_data(Packet):
> > > >
> > > > Packet.time_ts =3D Now()
> > > >
> > > > Packet.lost =3D FALSE
> > > >
> > > >
> > > > RACK_retransmit_data(Packet)
> > > >
> > > > Packet.retransmitted =3D TRUE
> > > >       RACK_transmit_data(Packet)
> > > >
> > > > ---
> > > >
> > > > When a packet is retransmitted again, its Packet.lost is reset and
> > > > transmission timestamp is renewed. Therefore, when its transmission
> > > > time falls out of the RACK reordering window in RACK_detect_loss(),=
 it
> > > > will be marked as lost (again).
> > > >
> > > >
> > > >
> > > >
> > > > >
> > > > > Also my timeout "some" vs "all" internal inconsistency comment re=
lates to
> > > > > this same algorithm:
> > > > >
> > > > >          "For timely loss detection, the sender MAY
> > > > >           install a "reordering settling" timer set to fire at th=
e earliest
> > > > >           moment at which it is safe to conclude that some packet=
 is lost.  The
> > > > >           earliest moment is the time it takes to expire the reor=
dering window
> > > > >           of the earliest unacked packet in flight."
> > > > >        vs
> > > > >           RACK_detect_loss():
> > > > >              ...
> > > > >                  timeout =3D max(remaining, timeout)
> > > > >        Which way it is, "some" or "all"? The use of max() in RACK=
_detect_loss()
> > > > >        seems to indicate that the timeout will be set according t=
o the longest
> > > > >        time, that is, the latest moment rather than earliest mome=
nts. Which
> > > > >        is the intented behavior here? (I can there might be merit=
s in both
> > > > >        approaches).
> > > >
> > > > Comment 1.2: good catch. We changed to: 'For timely loss detection,
> > > > the sender is RECOMMENDED to install a "reordering settling" timer.
> > > > The timer expires at the earliest moment when it is safe to conclud=
e
> > > > that all the unacknowledged packets within the reordering window ar=
e
> > > > lost.'
> > >
> > > I was under impression it's likely what you intented to propose right
> > > from the start. In that case this change seems not enough to address =
my
> > > concern, the algorithm seems to do exactly opposite on
> > >   timeout =3D max(remaining,  timeout)
> > > line, no? That is, if there is a moment further away in the future,
> > > the line selects it rather than "the earliest moment". This inconsist=
ency
> > > between the text and algorithm is why I brought this up and I think t=
he
> > > algorithm requires some small tweak to remove the inconsistency here.
> >
> > The diff is not the "earliest moment" wording. it was "the earliest
> > moment of some packet" but it is now "the earliest moment of all
> > unacked packets"
> >
> > Here is the old and proposed new text
> >
> > ...; but this
> >    risks a timeout (RTO) if no more ACKs come back (e.g, due to losses
> >    or application limit).  For timely loss detection, the sender MAY
> >    install a "reordering settling" timer set to fire at the earliest
> >    moment at which it is safe to conclude that some packet is lost.  Th=
e
> >    earliest moment is the time it takes to expire the reordering window
> >    of the earliest unacked packet in flight.
> >
> > new text:
> >
> > ... but this risks a timeout (RTO) if no more ACKs come back (e.g, due
> > to losses or application-limited transmissions). For timely loss
> > detection, the sender is RECOMMENDED to install a reordering settling
> > timer. This timer expires at the earliest moment when it is safe to
> > conclude that all the unacknowledged packets within the reordering
> > window are lost.
> >
> > so the keyword all would make the description match the pseudo code.
>
> Ah, you're of course right. It was sloppy reading (and too strong
> preconceived idea of what the fix should look like) from my side.
> So the correction is fine. My apologies for the noise.
No problem. We really appreciate your thorough review!

>
> --
>  i.
>
>
> >
> > >
> > >
> > > --
> > >  i.
> > >
> > > >
> > > >
> > > > >
> > > > >
> > > > > Then there are some other comments I made which seem also unaddre=
ssed but
> > > > > I'll not duplicate them all here.
> > > > >
> > > > >
> > > > > --
> > > > >  i.
> > > > >
> > > > >
> > > > > On Mon, 9 Mar 2020, Yuchung Cheng wrote:
> > > > >
> > > > > > Hi Ilpo,
> > > > > > Thanks for a detailed review and the great suggestions. Here's =
all the edits
> > > > > > we made.
> > > > > > https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-rack-08
> > > > > >
> > > > > >  We defined the missing terms and addressed the nits, removed t=
he less
> > > > > > critical parts like recommending PRR, and edited TLP algorithms=
 (sections
> > > > > > 7.4 to 7.6). I hope you find it's better now and would be happy=
 to hear any
> > > > > > improvements you'd like to make.
> > > > > >
> > > > > > Yuchung
> > > > > >
> > > > > > On Fri, Feb 14, 2020 at 4:22 AM Ilpo J=C3=A4rvinen <ilpo.jarvin=
en@cs.helsinki.fi>
> > > > > > wrote:
> > > > > >       In general, I find the draft quite good shape already, it=
's
> > > > > >       quite easy
> > > > > >       to read and follow (that is, except the TLP sections that=
 are
> > > > > >       specifically
> > > > > >       mentioned below).
> > > > > >
> > > > > >       Here's the list of issues I came across:
> > > > > >
> > > > > >       Sect 6.1
> > > > > >
> > > > > >       "RACK.pkts_sacked" lacks the empty line separator.
> > > > > >
> > > > > >       Sect 7.2
> > > > > >
> > > > > >         "This
> > > > > >          heuristic would fail to update RACK stats if the packe=
t is
> > > > > >       spuriously
> > > > > >          retransmitted because of a recent minimum RTT decrease=
 (e.g.
> > > > > >       path
> > > > > >          change)."
> > > > > >       This defies my logic as RTT decrease should not result in
> > > > > >       spurious
> > > > > >       retransmissions but help to avoid them.
> > > > > >
> > > > > >
> > > > > >         "Note that
> > > > > >          extensions that require additional TCP features (e.g. =
 DSACK)
> > > > > >       would
> > > > > >          work if the feature functions simply return false."
> > > > > >       Please rephrase this. I suspect you meant that if some fu=
nctions
> > > > > >       relate
> > > > > >       to TCP features that are not available, the given code st=
ill
> > > > > >       works
> > > > > >       as long as the functions related to such features return =
false
> > > > > >       when
> > > > > >       the TCP feature is not available but that surely isn't wh=
at the
> > > > > >       sentence currently says.
> > > > > >
> > > > > >       Sect 7.2 Step 5
> > > > > >
> > > > > >         "For timely loss detection, the sender MAY
> > > > > >          install a "reordering settling" timer set to fire at t=
he
> > > > > >       earliest
> > > > > >          moment at which it is safe to conclude that some packe=
t is
> > > > > >       lost.  The
> > > > > >          earliest moment is the time it takes to expire the reo=
rdering
> > > > > >       window
> > > > > >          of the earliest unacked packet in flight."
> > > > > >       vs
> > > > > >          RACK_detect_loss():
> > > > > >             ...
> > > > > >                 timeout =3D max(remaining, timeout)
> > > > > >       Which way it is, "some" or "all"? The use of max() in
> > > > > >       RACK_detect_lost()
> > > > > >       seems to indicate that the timeout will be set according =
to the
> > > > > >       longest
> > > > > >       time, that is, the latest moment rather than earliest mom=
ents.
> > > > > >       Which
> > > > > >       is the intented behavior here? (I can there might be meri=
ts in
> > > > > >       both
> > > > > >       approaches).
> > > > > >
> > > > > >
> > > > > >       In general, I didn't like the presentation order here as =
this
> > > > > >       form looks
> > > > > >       the most obvious to me:
> > > > > >          now >=3D Packet.xmit_ts + RACK.reo_wnd + (RACK.ack_ts =
-
> > > > > >       RACK.xmit_ts)
> > > > > >       ...and I find the derivation of it just complicating thin=
gs
> > > > > >       rather
> > > > > >       than helping but that's perhaps just my opinion.
> > > > > >
> > > > > >
> > > > > >          "It is also useful when the timestamp clock granularit=
y is
> > > > > >       close to
> > > > > >           or longer than the actual round trip time."
> > > > > >       Given that some things based on timestamps will obviously=
 stop
> > > > > >       working
> > > > > >       if the clock is slower than one tick per RTT I found this
> > > > > >       fragment a bit
> > > > > >       odd. RFC 7323 says "the clock SHOULD tick at least once p=
er
> > > > > >       window's worth
> > > > > >       of data". I can understand though that a host might find =
clock
> > > > > >       tick
> > > > > >       granularity challenging due to extensive RTT range but on=
 the
> > > > > >       same
> > > > > >       I'd not want such a "too slow clock" behavior to be endor=
sed as
> > > > > >       if
> > > > > >       it would be just normal. That is, I'm not against RACK ha=
ndling
> > > > > >       also this case nicely but it would be nice to somehow dow=
nplay
> > > > > >       the
> > > > > >       normalness tone in it with something along the lines of "=
This
> > > > > >       check is
> > > > > >       also useful if the recommended one tick per RTT cannot be
> > > > > >       achieved due
> > > > > >       to timestamp clock granularity."
> > > > > >
> > > > > >       Is RACK_detect_lost() supposed to detect also lost rexmit=
s? How?
> > > > > >       The rexmit code itself is not in the draft so it's hard t=
o
> > > > > >       evaluate
> > > > > >       how the Packet booleans are supposed to work in the first=
 place.
> > > > > >       I'm not saying the rexmit code should be included but cur=
rently
> > > > > >       it
> > > > > >       surely looks like any rexmitted packet just remains .lost=
 =3D TRUE
> > > > > >       and
> > > > > >       .retransmitted =3D TRUE state until RTO?
> > > > > >
> > > > > >
> > > > > >       Sect 7.4
> > > > > >
> > > > > >       "PTO" just appears out of nowhere (and is not even writte=
n open
> > > > > >       in the first instance).
> > > > > >
> > > > > >          "2. ... in step (2) of the algorithm."
> > > > > >       "step (2)" is forward-refing, it would naturally refer ba=
ck to
> > > > > >       "Step 2: Update RACK stats" so it makes very little sense=
 as is.
> > > > > >
> > > > > >       Sect 7.5
> > > > > >
> > > > > >       Separator missing for PTO: and TLPRxtOut:
> > > > > >
> > > > > >       Sect 7.5.1
> > > > > >
> > > > > >         "3.  Upon receiving a SACK that selectively acknowledge=
s data
> > > > > >       that was
> > > > > >              last sent before the segment with SEG.SEQ=3DSND.UN=
A was
> > > > > >       last
> > > > > >              (re)transmitted"
> > > > > >       This is hard to understand, perhaps rephrase it somehow?
> > > > > >
> > > > > >
> > > > > >         "But choosing PTO to be
> > > > > >          exactly an SRTT is likely to generate spurious probes =
given
> > > > > >       that
> > > > > >          network delay variance and even end-system timings can=
 easily
> > > > > >       push an
> > > > > >          ACK to be above an SRTT.  We chose PTO to be the next
> > > > > >       integral
> > > > > >          multiple of SRTT.
> > > > > >
> > > > > >          Similarly, network delay variations and end-system pro=
cessing
> > > > > >          latencies and timer granularities can easily delay ACK=
s
> > > > > >       beyond
> > > > > >          2*SRTT, so senders SHOULD add at least 2ms to a comput=
ed PTO
> > > > > >       value
> > > > > >          (and MAY add more if the sending host OS timer granula=
rity is
> > > > > >       more
> > > > > >          coarse than 1ms)."
> > > > > >       Does this later paragraph depend on some assumptions (suc=
h as
> > > > > >       very
> > > > > >       low RTT?) as I don't otherwise understand why you say wha=
t you
> > > > > >       say
> > > > > >       in the given order. First you said x & y can push ACK ove=
r SRTT
> > > > > >       so
> > > > > >       we pick 2* but then you say the same things can push over=
 2*SRTT
> > > > > >       too?
> > > > > >
> > > > > >       WCDelAckT to terminology?
> > > > > >
> > > > > >       Sect 7.5.2
> > > > > >
> > > > > >       TLP_send_probe():
> > > > > >               - Perhaps set TLPRxtOut somewhere?
> > > > > >               - Add "Arm RTO" to the end?
> > > > > >
> > > > > >         "In the case where
> > > > > >          there is only one original outstanding segment of data=
 (N=3D1),
> > > > > >       the
> > > > > >          same logic (trivially) applies: an ACK for a single
> > > > > >       outstanding
> > > > > >          segment tells the sender the N-1=3D0 segments precedin=
g that
> > > > > >       segment
> > > > > >          were lost.  Furthermore, whether there are N>1 or N=3D=
1
> > > > > >       outstanding
> > > > > >          segments,"
> > > > > >       If this distinction is useful to make at all, maybe just =
simply
> > > > > >       say:
> > > > > >       "Applies also in case N=3D1."
> > > > > >
> > > > > >       Sect 7.6
> > > > > >
> > > > > >         "SND.NXT at the time the episode started."
> > > > > >       Isn't this TLPHighRxt, why not use it here?
> > > > > >
> > > > > >       Sect 7.6.2
> > > > > >
> > > > > >         "Upon sending a TLP probe that is a retransmission, the=
 sender
> > > > > >       sets
> > > > > >          TLPRxtOut to true and TLPHighRxt to SND.NXT."
> > > > > >       This should appear in TLP_send_probe().
> > > > > >
> > > > > >       "Detecting recoveries accomplished by loss probes"
> > > > > >       Did you want to start another section here?
> > > > > >
> > > > > >       "Step 1: Track ACKs ..." until what point? (It's told on =
later
> > > > > >       but
> > > > > >       this question comes into mind when reading the draft in o=
rder).
> > > > > >
> > > > > >
> > > > > >       Sect 7.4 - 7.6.2
> > > > > >
> > > > > >       In general, this particular part of the draft seems to re=
quire
> > > > > >       some
> > > > > >       further editing. After reading it all through, I feel mos=
t bits
> > > > > >       are
> > > > > >       present but the text is very hard to follow. I think the
> > > > > >       presentation
> > > > > >       order needs significant work. There are random bits here =
and
> > > > > >       there
> > > > > >       that are explained only later. Also, algorithm lacks some=
 parts
> > > > > >       discussed only in the text but variable like items should
> > > > > >       definitely
> > > > > >       be included to the algorithm.
> > > > > >
> > > > > >
> > > > > >       Sect 8.4
> > > > > >
> > > > > >         "RACK is compatible with and does not interfere with th=
e the
> > > > > >       standard
> > > > > >          RTO [RFC6298], RTO-restart [RFC7765], F-RTO [RFC5682] =
and
> > > > > >       Eifel
> > > > > >          algorithms [RFC3522]. ...
> > > > > >          It neither changes the RTO timer calculation nor detec=
ts
> > > > > >       spurious
> > > > > >          timeouts."
> > > > > >       For F-RTO, also send order is significant.
> > > > > >
> > > > > >       Sect 8.5
> > > > > >
> > > > > >       SHOULDs PRR (and also placed to normative references) tha=
t is
> > > > > >       only
> > > > > >       experimental and RACK aims to standards track. I feel tha=
t PRR
> > > > > >       is
> > > > > >       disjoint enough anyway, which I think was also your inten=
tion
> > > > > >       when
> > > > > >       maintaining the decoupling CC and RACK, that it's bit odd=
 to
> > > > > >       take
> > > > > >       this strong stance on PRR (indepedent of the stds track v=
s
> > > > > >       experimental thing).
> > > > > >
> > > > > >         "due to congestion window validation."
> > > > > >       Does this refer to CWV as in RFC? Or some other mechanism=
 in
> > > > > >       which
> > > > > >       case you shouldn't use the same term here.
> > > > > >
> > > > > >       I find the described scenario somewhat unconvincing becas=
e
> > > > > >       the description ends with:
> > > > > >         "...
> > > > > >          The ending congestion window after recovery also impac=
ts
> > > > > >          subsequent data transfer."
> > > > > >       ...First there wasn't enough data to fill cwnd of 20 pack=
ets and
> > > > > >       then suddenly transferring that non-existing data is impa=
cted.
> > > > > >
> > > > > >       References
> > > > > >
> > > > > >       QUIC-LR is very old ref and even the filename has changed=
 a few
> > > > > >       times from draft-tsvwg-...
> > > > > >
> > > > > >
> > > > > >       Nits
> > > > > >
> > > > > >       "runt" made the sentence incomprehendable for me, I had t=
o look
> > > > > >       up
> > > > > >       that word (and only second source was descriptive enough =
that I
> > > > > >       think I could understand the meaning, and I'd guessed wro=
ng
> > > > > >       because
> > > > > >       I thought there was some special intent here because of t=
he
> > > > > >       quotes).
> > > > > >
> > > > > >       is dicussed -> is discussed
> > > > > >       RACT.RTT -> RACK.rtt
> > > > > >       First "TSO" -> TCP Segmentation Offload (TSO)
> > > > > >       grainularity -> granularity
> > > > > >       higher-sequenc -> higher-sequence
> > > > > >
> > > > > >
> > > > > >       --
> > > > > >        i.
> > > > > >
> > > > > >
> > > > > >
> > > >
> >


From nobody Mon May 11 15:57:33 2020
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Date: Mon, 11 May 2020 15:57:22 -0700
From: Christoph Paasch <cpaasch@apple.com>
To: gorry@erg.abdn.ac.uk, tcpm@ietf.org
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Subject: [tcpm] RFC7661 - assumes reno in Section 4.4.1 ?
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Hello,

I have a question regarding RFC7661. In Section 4.4.1, it specifies the
congestion-window when entering and exiting the recovery-period, namely:

	cwnd = (Max(pipeACK,LossFlightSize))/2

and

	cwnd = (Max(pipeACK,LossFlightSize) - R)/2


Is the division by 2 because it assumes Reno's multiplicative decrease and
if Cubic would be used that would be beta_cubic ?

Or is there another reason behind the division by 2?


Thanks,
Christoph


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From: Praveen Balasubramanian <pravb@microsoft.com>
To: Matt Mathis <mattmathis=40google.com@dmarc.ietf.org>, Neal Cardwell <ncardwell=40google.com@dmarc.ietf.org>
CC: Michael Tuexen <michael.tuexen@lurchi.franken.de>, "tcpm@ietf.org" <tcpm@ietf.org>, Jon Crowcroft <jon.crowcroft@cl.cam.ac.uk>
Thread-Topic: [EXTERNAL] Re: [tcpm] On Sender Control of Delayed Acks in TCP
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Subject: Re: [tcpm] [EXTERNAL] Re: On Sender Control of Delayed Acks in TCP
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From: "Mark Allman" <mallman@icir.org>
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Date: Tue, 12 May 2020 15:54:27 -0400
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Subject: [tcpm] hystart++ -03 review
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I read version -03 of the HyStart++ document and my view differs
from many, I guess.  IMHO, this document is not ready for adoption.

The draft is pretty anemic and I am not sure there is enough there
to actually judge whether it is or is not a good idea.  It seems
like perhaps the document could get there, but at this point I think
the authors have some work to do.  Review below.

allman




Section 1:

  - Nit: RFC 793 does not define slow start, remove ref.  (Could add
    a ref to original Jacobson paper.)

  - HyStart++ is "widely deployed", but is it widely used?  Is it
    enabled or just there for the enabling?  The document makes it
    sound like we have a lot of experience with this somehow, but
    none of that is shared (see below).

  - Nit: Spell out "LSS" on first use.

Section 3:

  - Nit: Totally fine to crib the definitions from RFC 5681, but
    it'd be good form to note that is where they came from.

  - Nit: Also, since you use rwnd in one of the definitions and
    you're copying things, you might as well copy that one, too.

Section 4:

  - There is so much unexplained magic in here.  When setting
    RttThresh, what is magic about 1/8-th of the lastRoundMinRtt?
    Why clamp this between 4msec and 16msec?  Would that really be
    reasonable for all networks?

    We need magic numbers sometimes and as long as they're
    resasonably chosen and motivated we have to live with having
    them around.  But, the draft really needs to explain these
    choices, how they were made and how broadly applicable they
    might be.  Further, the document suggests one MAY vary the
    constants, but it'd be great if it gave some sort of advice on
    how one might decide if different constants were OK or not OK.

  - The algorithm (series of equations) is here and clear.  But, the
    document really needs to give us a flavor of the why.  Talk us
    through what is happening and why in here, please.

General / meta:

  - The draft offers a little intuition on why this is a good idea
    (i.e., to prevent massive slow start over-shoot).  But, there is
    no sketch of evidence that shows it is effective.  I see this
    was previously mentioned on the mailing list and Praveen said:

      We presented lab measurement data at the last IETF. We are in
      the process of doing some A/B experiments at scale with real
      world workloads. We did fairness comparisons in our lab
      measurements between Hystart and non-Hystart and we didn't see
      any problems.

    The bar ought to be higher than "didn't see any problems".  It
    is great that there is some testing done and planned.  I'd like
    to see this all sketched in the document itself.  I am not even
    close to convinced that this is a good idea.

  - In some sense, this idea makes TCP more conservative (i.e., we
    stop exponential increase sooner than RFC5681 would suggest).
    So, the bar is likely not huge for saying this is "OK".  But, I
    think we should have some notion that this helps *something*
    before agreeing to suggest it as somehow a reasonable thing to
    do in an RFC.  Right now this document doesn't concretely show
    that HyStart++ is useful in any way.

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From nobody Wed May 13 05:00:52 2020
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Date: Wed, 13 May 2020 14:00:37 +0200
From: "Carles Gomez Montenegro" <carlesgo@entel.upc.edu>
To: "Praveen Balasubramanian" <pravb=40microsoft.com@dmarc.ietf.org>
Cc: "Matt Mathis" <mattmathis=40google.com@dmarc.ietf.org>, "Neal Cardwell" <ncardwell=40google.com@dmarc.ietf.org>, "Michael Tuexen" <michael.tuexen@lurchi.franken.de>, "tcpm@ietf.org" <tcpm@ietf.org>, "Jon Crowcroft" <jon.crowcroft@cl.cam.ac.uk>
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Subject: Re: [tcpm] [EXTERNAL] Re: On Sender Control of Delayed Acks in TCP
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Hi Praveen,

Thanks for your comments.

Please find below some inline responses and comments:

> I agree with the two use-case taxonomy. While it'd be quite nice to have a
> single mechanism like a TCP option that addresses both use cases, there
> are problems with a new option that's used in data packets:
>
> 1. middlebox traversal
>
> 2. hardware offloads like TSO and LRO might need updates?

Indeed, our analysis in draft-gomez-tcpm-delack-suppr-reqs-01 shows that
all potential solutions considered so far, including new TCP options,
appear to have pros and cons.

In order to evaluate solutions, I think the most critical part is
determining what could be an acceptable trade-off among different goals or
requirements.

In this thread, several WG participants expressed a positive opinion about
defining a new TCP option. (Although, as you mentioned, this approach also
has drawbacks.)

> If we go down the path of a new option for data packets that is broadly
> deployed, should we aim for a list of key-value pairs so that it is
> extensible?

Capturing what Neal and Michael outlined, we might consider a new TCP
option with the following format and intended functionality:


       Kind: TBD

       Length: 3 bytes

       Format:

              +---------+---------+---------+
              |Kind=TBD |Length=3 |    N    |
              +---------+---------+---------+
                   1         1         1

       N=0 --> "Please, ACK now (but don't change the ACK rate)"
       N>0 --> "Change the ACK rate to 1 every N received data segments"

Would the above (where the "N" field has a size of 1 byte) suffice, or
would you still envision the need for a list of key-value pairs (or did
you have a different approach in mind)?

> Also, I recall there was some discussion about usage of PSH bit for ACK
> pull. It’s already used for signaling message boundary, so an immediate
> ACK upon end of message might also help lower latency.

A problem that has been reported is that, when the PSH bit is set, it is
not always desirable to trigger an ACK immediately:
https://mailarchive.ietf.org/arch/msg/tcpm/seVEB4XAPjixXOPqfPea7zZAVmU/

Thanks,

Carles



> Thanks
>
> From: tcpm <tcpm-bounces@ietf.org> On Behalf Of Matt Mathis
> Sent: Saturday, May 2, 2020 7:55 AM
> To: Neal Cardwell <ncardwell=40google.com@dmarc.ietf.org>
> Cc: Michael Tuexen <michael.tuexen@lurchi.franken.de>; tcpm@ietf.org; Jon
> Crowcroft <jon.crowcroft@cl.cam.ac.uk>
> Subject: [EXTERNAL] Re: [tcpm] On Sender Control of Delayed Acks in TCP
>
> Another probably small opportunity would be to make the RTT estimators
> less noisy, by having the receiver always ACK the first (or last) TSO
> segment.   The RTT is currently fuzzed by both del ACK and TCP strides.
>
> Thanks,
> --MM--
> The best way to predict the future is to create it.  - Alan Kay
>
> We must not tolerate intolerance;
>        however our response must be carefully measured:
>             too strong would be hypocritical and risks spiraling out of
> control;
>             too weak risks being mistaken for tacit approval.
>
>
> On Sat, May 2, 2020 at 7:35 AM Neal Cardwell
> <ncardwell=40google.com@dmarc.ietf.org<mailto:40google.com@dmarc.ietf.org>>
> wrote:
> On Sat, May 2, 2020 at 10:26 AM Michael Tuexen
> <michael.tuexen@lurchi.franken.de<mailto:michael.tuexen@lurchi.franken.de>>
> wrote:
>>
>> > On 2. May 2020, at 16:10, Neal Cardwell
>> <ncardwell=40google.com@dmarc.ietf.org<mailto:40google.com@dmarc.ietf.org>>
>> wrote:
>> >
>> > On Sat, May 2, 2020 at 5:40 AM Carles Gomez Montenegro
> .....
>> >> [CG] It seems that there are two different areas where avoiding the
>> >> typical Delayed ACKs approach may be of particular interest: i)
>> scenarios
>> >> where very small window sizes are used (cwnd=1), and ii) scenarios
>> where
>> >> large window sizes, and high ACK ratios, can be used. We will
>> reflect
>> >> these considerations in the next draft update.
>> >
>> > Yes, I wholeheartedly agree with Bob's point that it's also extremely
>> > useful for TCP senders to be able to say things like: "I don't
>> > particularly mind which packets you ACK, but there's no need to do it
>> > more often than 1:16". QUIC has a mechanism like this, and it seems
>> to
>> > allow significant CPU savings, even allowing QUIC to run faster than
>> > TCP in some cases, AFAICT due to the reduced overhead for handling
>> > ACKs:
>> >  https://www.fastly.com/blog/measuring-quic-vs-tcp-computational-efficiency<https://nam06.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.fastly.com%2Fblog%2Fmeasuring-quic-vs-tcp-computational-efficiency&data=02%7C01%7Cpravb%40microsoft.com%7C38ff3d1d88b245d19a2b08d7eea8ebea%7C72f988bf86f141af91ab2d7cd011db47%7C1%7C0%7C637240282206886368&sdata=Uv%2B%2FgUOkvXss66hcGohqnt3tRHK4g3imDt0I0mu5xRw%3D&reserved=0>
>> >  https://tools.ietf.org/html/draft-iyengar-quic-delayed-ack-00<https://nam06.safelinks.protection.outlook.com/?url=https%3A%2F%2Ftools.ietf.org%2Fhtml%2Fdraft-iyengar-quic-delayed-ack-00&data=02%7C01%7Cpravb%40microsoft.com%7C38ff3d1d88b245d19a2b08d7eea8ebea%7C72f988bf86f141af91ab2d7cd011db47%7C1%7C0%7C637240282206886368&sdata=9vw3sjqTAPLPCNAQPCFza3RJSLY%2B25T8Odqq0PCec%2BE%3D&reserved=0>
>> >
>> > So I agree with the usefulness of the two-use-case taxonomy Carles has
>> outlined:
>> >
>> > 1: "please ACK now"; because sender's cwnd is low
>> >
>> > 2: "ACK at frequency K"; because sender's cwnd is high
>> >
>> > It seems those could both be addressed cleanly with one or two TCP
>> > option kinds. A single TCP option kind might suffice if the ACK
>> > frequency of 0 or 1 is taken to mean "please ACK now".
>> I might be useful if the sender can trigger an ACK for this particular
>> packet,
>> but leave the rate unchanged. And change the rate from now on.
>> If using an option, you could encode "send an ack for this packet, but
>> don't
>> change the ack rate" by using 0, and encode "change the ack rate to 1
>> for n"
>> by using n, n > 0.
>
> Yes, that sounds like a very nice idea.
>
> best,
> neal



From nobody Wed May 13 10:57:13 2020
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-rto-consider-14.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the TCP Maintenance and Minor Extensions WG of the IETF.

        Title           : Requirements for Time-Based Loss Detection
        Author          : Mark Allman
	Filename        : draft-ietf-tcpm-rto-consider-14.txt
	Pages           : 11
	Date            : 2020-05-13

Abstract:
    Many protocols must detect packet loss for various reasons (e.g., to
    ensure reliability using retransmissions or to understand the level
    of congestion along a network path).  While many mechanisms have
    been designed to detect loss, protocols ultimately can only count on
    the passage of time without delivery confirmation to declare a
    packet "lost".  Each implementation of a time-based loss detection
    mechanism represents a balance between correctness and timeliness
    and therefore no implementation suits all situations.  This document
    provides high-level requirements for time-based loss detectors
    appropriate for general use in the Internet.  Within the
    requirements, implementations have latitude to define particulars
    that best address each situation.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tcpm-rto-consider/

There are also htmlized versions available at:
https://tools.ietf.org/html/draft-ietf-tcpm-rto-consider-14
https://datatracker.ietf.org/doc/html/draft-ietf-tcpm-rto-consider-14

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-rto-consider-14


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/



From nobody Wed May 13 13:02:05 2020
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Yoshifumi Nishida has requested publication of draft-ietf-tcpm-rto-consider-14 as Best Current Practice on behalf of the TCPM working group.

Please verify the document's state at https://datatracker.ietf.org/doc/draft-ietf-tcpm-rto-consider/



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From: Michael Tuexen <tuexen@fh-muenster.de>
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Date: Wed, 13 May 2020 23:40:11 +0200
Cc: draft-balasubramanian-tcpm-hystartplusplus@ietf.org
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Subject: [tcpm] Request for feedback on WG adoption of draft-balasubramanian-tcpm-hystartplusplus-03
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Dear all,

this mail starts a WG adoption call for
=
https://tools.ietf.org/html/draft-balasubramanian-tcpm-hystartplusplus-03

The intended status would be PS.

So I would like to solicit feedback regarding support for or objections =
against the
adoption of the document as a WG document in TCPM, if it hasn't been =
already sent to
the list.

Please provide feedback before May 21st.

Best regards
Michael

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From nobody Wed May 13 19:20:11 2020
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Subject: [tcpm] Milestones changed for tcpm WG
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Changed milestone "Submit document on Retransmission Timeout Considerations
as Best Current Practice RFC", resolved as "Done", added
draft-ietf-tcpm-rto-consider to milestone.

URL: https://datatracker.ietf.org/wg/tcpm/about/


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From: "Mark Allman" <mallman@icir.org>
To: "Michael Tuexen" <tuexen@fh-muenster.de>
Cc: "tcpm IETF list" <tcpm@ietf.org>, draft-balasubramanian-tcpm-hystartplusplus@ietf.org
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Subject: Re: [tcpm] Request for feedback on WG adoption of draft-balasubramanian-tcpm-hystartplusplus-03
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> this mail starts a WG adoption call for
> https://tools.ietf.org/html/draft-balasubramanian-tcpm-hystartplusplus-03
>
> The intended status would be PS.

Per note from the other day, I think this could get there, but the
current document doesn't offer enough (any, really) motivation that
suggests HyStart++ is useful or effective.  Since I can't figure
that out, I can't support the document for WG adoption at the moment
(and especially so at PS).

allman

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From: Neal Cardwell <ncardwell@google.com>
Date: Thu, 14 May 2020 11:13:47 -0400
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To: Mark Allman <mallman@icir.org>
Cc: Michael Tuexen <tuexen@fh-muenster.de>, tcpm IETF list <tcpm@ietf.org>,  draft-balasubramanian-tcpm-hystartplusplus@ietf.org
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Subject: Re: [tcpm] Request for feedback on WG adoption of draft-balasubramanian-tcpm-hystartplusplus-03
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"On Thu, May 14, 2020 at 8:06 AM Mark Allman <mallman@icir.org> wrote:
>
>
> > this mail starts a WG adoption call for
> > https://tools.ietf.org/html/draft-balasubramanian-tcpm-hystartplusplus-03
> >
> > The intended status would be PS.
>
> Per note from the other day, I think this could get there, but the
> current document doesn't offer enough (any, really) motivation that
> suggests HyStart++ is useful or effective.  Since I can't figure
> that out, I can't support the document for WG adoption at the moment
> (and especially so at PS).

Fair enough. However, I would urge that these concerns be addressed in
future editorial revisions, but that we not allow them to derail
progress toward WG adoption working toward PS for
draft-balasubramanian-tcpm-hystartplusplus-03.

I would make several arguments for WG adoption, working toward the PS
level, for draft-balasubramanian-tcpm-hystartplusplus-03:

(1) From previous production experience with large-scale Internet
services, our team has seen that Hystart is very important in reducing
packet loss rates from new flows entering the public Internet. So
Hystart is important and very useful on a technical level.

(2) The original Hystart has already been ubiquitously deployed on the
Internet for a decade, since it has been default-enabled in Linux TCP
for a decade (as part of the default-enabled CUBIC congestion control
module for Linux). Just as it was important, IMHO, to document the
core CUBIC algorithm in RFC8312, I think it's important to document
some version of Hystart as an RFC.

(3) At an algorithmic level, I see several important improvements that
Hystart++ makes over the original Hystart; at least:

  (a) using only the delay-based variant of Hystart, and not the
ack-train-based variant of Hystart (the ack-train-based variant our
team has also found is liable to false positives, especially with
paced flows)

 (b) "mitigating poor performance which can result from false
positives when HyStart is used alone", by ensuring that after
triggering Hystart++, a connection increases its cwnd at least as fast
as Limited Slow Start allows.

  (c) decoupling the delay-based algorithm from the global min_rtt of
the connection, and instead using lastRoundMinRTT

Presuming that multiple implementations or deployments can validate
that these changes are indeed resulting in good performance, I think
it would be good to document this improved version, Hystart++, rather
than the 10-year-old original Hystart.

(4) Yes, there are magic numbers in Hystart++, but (aside from the new
LSS_DIVISOR)  those other four magic numbers are all identical to the
magic numbers in the Hystart code that has been doing a good job of
improving performance for much of the traffic on the Internet for the
last decade (see (2)). So the fact that the magic numbers are not
discussed or justified is a point that can be cleared up in the
editing process, and IMHO need not delay progress through the RFC
process. I would argue that, like the constants used for SRTT and RTO
calculations, in the future we could try to do more science to try
other values and quantify the performance of the alternative values,
but long-term experience has shown that these magic values work well
enough in practice, and thus we do not need to bikeshed them now, and
so we should not let the magicness of these numbers hold back their
publication as a PS RFC. We should just document the rationale for
each magic number in the RFC, and explain in the document that they
have stood the test of time and wide deployment/use. Slides at the
meetings can quantify the performance impact of alternative values.

best regards,
neal


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From: "Mark Allman" <mallman@icir.org>
To: "Neal Cardwell" <ncardwell@google.com>
Cc: "Michael Tuexen" <tuexen@fh-muenster.de>, "tcpm IETF list" <tcpm@ietf.org>, draft-balasubramanian-tcpm-hystartplusplus@ietf.org
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Subject: Re: [tcpm] Request for feedback on WG adoption of draft-balasubramanian-tcpm-hystartplusplus-03
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Neal-

Based on your note it seems that it should be straightforward to add
solid motivation in the document.  (And, hopefully your high-level
notions of finding that it works well can readily be turned into
objective statements.)  So, why don't we just do that?

> However, I would urge that these concerns be addressed in future
> editorial revisions,

My issue is that this is **not** editorial.  There is simply nothing
in the document that suggests it is a good idea.  That fact is
completely independent of whether it is or is not a good idea.  It
just means there is no basis to judge given in the document.  We
should expect some explanation of why this is reasonable before the
WG is asked to adopt and work on something.  IMHO.

allman

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Subject: Re: [tcpm] Request for feedback on WG adoption of draft-balasubramanian-tcpm-hystartplusplus-03
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I would like to see this work adopted.

Am 13.05.2020 um 23:40 schrieb Michael Tuexen:
> Dear all,
>
> this mail starts a WG adoption call for
> https://tools.ietf.org/html/draft-balasubramanian-tcpm-hystartplusplus-0=
3
>
> The intended status would be PS.
>
> So I would like to solicit feedback regarding support for or objections =
against the
> adoption of the document as a WG document in TCPM, if it hasn't been alr=
eady sent to
> the list.
>
> Please provide feedback before May 21st.
>
> Best regards
> Michael
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>


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From: "Rodney W. Grimes" <ietf@gndrsh.dnsmgr.net>
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To: Michael Tuexen <tuexen@fh-muenster.de>
Date: Thu, 14 May 2020 19:21:18 -0700 (PDT)
CC: tcpm IETF list <tcpm@ietf.org>, draft-balasubramanian-tcpm-hystartplusplus@ietf.org
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Subject: Re: [tcpm] Request for feedback on WG adoption of draft-balasubramanian-tcpm-hystartplusplus-03
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Dear tcpm'ers,
	I have read the 03 version of this draft, and support this draft
being adopted by the WG.

	I however believe this work should be consider only a starting
point for formalizing hystartplusplus, for one the description seems to
ignore ECN (RFC3168) and refers to drops or loss, though I believe the
implementation actually does react to a CE.

	I am also concerned as a "Proposed Standard" it has the text
indicated one can "experiment" with magic constants, that would be fine
for an experimental track, but I do not feel comfortable with that text
in a PS.  If these constants are not well defined perhaps this should be
an experiment.

Regards,
Rod Grimes

> Dear all,
> 
> this mail starts a WG adoption call for
> https://tools.ietf.org/html/draft-balasubramanian-tcpm-hystartplusplus-03
> 
> The intended status would be PS.
> 
> So I would like to solicit feedback regarding support for or objections against the
> adoption of the document as a WG document in TCPM, if it hasn't been already sent to
> the list.
> 
> Please provide feedback before May 21st.
> 
> Best regards
> Michael

> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm

-- 
Rod Grimes                                                 rgrimes@freebsd.org


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From: Mirja Kuehlewind <mirja.kuehlewind@ericsson.com>
To: Mark Allman <mallman@icir.org>, Neal Cardwell <ncardwell@google.com>
CC: tcpm IETF list <tcpm@ietf.org>, Michael Tuexen <tuexen@fh-muenster.de>, "draft-balasubramanian-tcpm-hystartplusplus@ietf.org" <draft-balasubramanian-tcpm-hystartplusplus@ietf.org>
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From: "Scharf, Michael" <Michael.Scharf@hs-esslingen.de>
To: Mirja Kuehlewind <mirja.kuehlewind=40ericsson.com@dmarc.ietf.org>, "Mark Allman" <mallman@icir.org>, Neal Cardwell <ncardwell@google.com>
CC: Michael Tuexen <tuexen@fh-muenster.de>, tcpm IETF list <tcpm@ietf.org>, "draft-balasubramanian-tcpm-hystartplusplus@ietf.org" <draft-balasubramanian-tcpm-hystartplusplus@ietf.org>
Thread-Topic: [tcpm] Request for feedback on WG adoption of draft-balasubramanian-tcpm-hystartplusplus-03
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Subject: Re: [tcpm] Request for feedback on WG adoption of draft-balasubramanian-tcpm-hystartplusplus-03
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From nobody Fri May 15 06:26:32 2020
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From: "Mark Allman" <mallman@icir.org>
To: "Mirja Kuehlewind" <mirja.kuehlewind@ericsson.com>
Cc: "Neal Cardwell" <ncardwell@google.com>, "tcpm IETF list" <tcpm@ietf.org>,  "Michael Tuexen" <tuexen@fh-muenster.de>, draft-balasubramanian-tcpm-hystartplusplus@ietf.org
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Subject: Re: [tcpm] Request for feedback on WG adoption of draft-balasubramanian-tcpm-hystartplusplus-03
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> this work has been presented multiple time in tcpm which a bunch
> of measurement numbers of real deployments. I really don't think
> this material should end up in the draft. The draft provides the
> specification of the mechanism (preferably as comprehensive and
> short as possible). The IETF proceedings provide the record for
> results and background information.

I have never actually heard quite that framing before.

I am not looking for a research paper here.  But, some sketch of why
this has been found to be reasonable, that the draft is just
documenting existing practice, etc. would seem required to me.

And, incorporating a comprehensive set of references for more and
deeper information is fine.  The substantive reference in this draft
is from over a decade ago (yes, "the" because there is only one).  I
am not the sharpest knife in the drawer, but I would never think to
go dig through the IETF proceedings to try to find such.  Not that
it might not be there, just that that would be a long slog without a
reference.

I also bristle somewhat at this notion that our job is just to give
the "what" and not the "why".  I know we are much better at the
former.  And, I know I have been guilty of too little of the latter
at times.  I know this because I have seen new folks read these RFCs
and not understand why something was done.  Or, that something is
important for some unstated, but crucial reason.  Or, decide that we
do X because of Y and so if Y is different then X can be, too---even
though we did X because of Z and just didn't say it well.

I am clearly not consensus, but to me the adoption question is
completely premature when the document doesn't offer motivation that
it is reasonable (beyond an intuitive hand wave that we're
preventing slow start overshoot).  Neal's note does sketch out why
this might be an OK thing to do.  And, probably IETF talks have, as
well.  But, some of that motivation has to make it into the
document, it seems to me.

allman

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Subject: [tcpm] TCP Connection ID
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Dear WG,

As you all know L3 LB is a common element of today's cloud infrastructure.
Its original design was based on the connection state that maps backend IPs
(real IPs) to connection. Recently there were several papers that were
suggesting the move to the stateless L3 LB, where the state (connection id)
becomes the property of connection and it kept at the client, so the L3 LB
can work without keeping state for every connection. As far as I
understand, for QUIC it becomes native with destination connection ID in
its header.

I've seen suggestions for getting this functionality in TCP by overloading
the Timestamps option or even TCP SEQ - can't say that such design looks
good for me. I wonder was there an attempt to add connection id in the TCP
options, and if it was - why it wasn't successful?

-- 
Best regards,
Alexander Azimov

--00000000000093fb6405a5b12d77
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<div dir=3D"ltr">Dear WG,<div><br></div><div>As you all know L3 LB is a com=
mon element of today&#39;s cloud infrastructure. Its original design was ba=
sed on the connection state that maps backend IPs (real IPs) to connection.=
 Recently=C2=A0there were several papers that were suggesting the move to t=
he stateless L3 LB, where the state (connection id) becomes the property of=
 connection and it kept at the client, so the L3 LB can work without keepin=
g state for every connection. As far as I understand, for QUIC it becomes n=
ative with destination connection ID in its header.<br><br>I&#39;ve seen su=
ggestions for getting this functionality=C2=A0in TCP by overloading the Tim=
estamps option or even TCP SEQ - can&#39;t say that such design looks good =
for me. I wonder was there an attempt to add connection id in the TCP optio=
ns, and if it was - why it wasn&#39;t successful?<br></div><div><div></div>=
<div><br></div>-- <br><div dir=3D"ltr" class=3D"gmail_signature" data-smart=
mail=3D"gmail_signature"><div dir=3D"ltr">Best regards,<div>Alexander Azimo=
v</div></div></div></div></div>

--00000000000093fb6405a5b12d77--


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From: Mirja Kuehlewind <mirja.kuehlewind@ericsson.com>
To: Mark Allman <mallman@icir.org>, "Scharf, Michael" <Michael.Scharf@hs-esslingen.de>
CC: Neal Cardwell <ncardwell@google.com>, tcpm IETF list <tcpm@ietf.org>, Michael Tuexen <tuexen@fh-muenster.de>, "draft-balasubramanian-tcpm-hystartplusplus@ietf.org" <draft-balasubramanian-tcpm-hystartplusplus@ietf.org>
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Subject: Re: [tcpm] Request for feedback on WG adoption of draft-balasubramanian-tcpm-hystartplusplus-03
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Cg==


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To: tcpm@ietf.org
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From: marcelo bagnulo braun <marcelo@it.uc3m.es>
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Date: Fri, 15 May 2020 18:39:07 +0200
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Subject: Re: [tcpm] Request for feedback on WG adoption of draft-balasubramanian-tcpm-hystartplusplus-03
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I support its adoption.

El 13/5/20 a las 23:40, Michael Tuexen escribió:
> Dear all,
>
> this mail starts a WG adoption call for
> https://tools.ietf.org/html/draft-balasubramanian-tcpm-hystartplusplus-03
>
> The intended status would be PS.
>
> So I would like to solicit feedback regarding support for or objections against the
> adoption of the document as a WG document in TCPM, if it hasn't been already sent to
> the list.
>
> Please provide feedback before May 21st.
>
> Best regards
> Michael
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm



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Subject: Re: [tcpm] TCP Connection ID
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RFC7974


> On May 15, 2020, at 8:07 AM, Alexander Azimov <a.e.azimov@gmail.com> =
wrote:
>=20
> Dear WG,
>=20
> As you all know L3 LB is a common element of today's cloud =
infrastructure. Its original design was based on the connection state =
that maps backend IPs (real IPs) to connection. Recently there were =
several papers that were suggesting the move to the stateless L3 LB, =
where the state (connection id) becomes the property of connection and =
it kept at the client, so the L3 LB can work without keeping state for =
every connection. As far as I understand, for QUIC it becomes native =
with destination connection ID in its header.
>=20
> I've seen suggestions for getting this functionality in TCP by =
overloading the Timestamps option or even TCP SEQ - can't say that such =
design looks good for me. I wonder was there an attempt to add =
connection id in the TCP options, and if it was - why it wasn't =
successful?
>=20
> --=20
> Best regards,
> Alexander Azimov
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


--Apple-Mail=_04A42C1A-ED56-4660-9209-40D05F714729
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dus-ascii"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; line-break: after-white-space;" class=3D""><pre =
style=3D"font-size: 13.333333015441895px; margin-top: 0px; =
margin-bottom: 0px; caret-color: rgb(0, 0, 0); color: rgb(0, 0, 0);" =
class=3D"">RFC7974</pre><div class=3D""><br class=3D""></div><div =
style=3D""><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D"">On May 15, 2020, at 8:07 AM, Alexander Azimov &lt;<a =
href=3D"mailto:a.e.azimov@gmail.com" =
class=3D"">a.e.azimov@gmail.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><div dir=3D"ltr" =
class=3D"">Dear WG,<div class=3D""><br class=3D""></div><div class=3D"">As=
 you all know L3 LB is a common element of today's cloud infrastructure. =
Its original design was based on the connection state that maps backend =
IPs (real IPs) to connection. Recently&nbsp;there were several papers =
that were suggesting the move to the stateless L3 LB, where the state =
(connection id) becomes the property of connection and it kept at the =
client, so the L3 LB can work without keeping state for every =
connection. As far as I understand, for QUIC it becomes native with =
destination connection ID in its header.<br class=3D""><br class=3D"">I've=
 seen suggestions for getting this functionality&nbsp;in TCP by =
overloading the Timestamps option or even TCP SEQ - can't say that such =
design looks good for me. I wonder was there an attempt to add =
connection id in the TCP options, and if it was - why it wasn't =
successful?<br class=3D""></div><div class=3D""><div class=3D""></div><div=
 class=3D""><br class=3D""></div>-- <br class=3D""><div dir=3D"ltr" =
class=3D"gmail_signature" data-smartmail=3D"gmail_signature"><div =
dir=3D"ltr" class=3D"">Best regards,<div class=3D"">Alexander =
Azimov</div></div></div></div></div>
_______________________________________________<br class=3D"">tcpm =
mailing list<br class=3D""><a href=3D"mailto:tcpm@ietf.org" =
class=3D"">tcpm@ietf.org</a><br =
class=3D"">https://www.ietf.org/mailman/listinfo/tcpm<br =
class=3D""></div></blockquote></div><br class=3D""></body></html>=

--Apple-Mail=_04A42C1A-ED56-4660-9209-40D05F714729--


From nobody Mon May 18 07:15:58 2020
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Subject: [tcpm] Last Call: <draft-ietf-tcpm-rto-consider-14.txt> (Requirements for Time-Based Loss Detection) to Best Current Practice
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The IESG has received a request from the TCP Maintenance and Minor Extensions
WG (tcpm) to consider the following document: - 'Requirements for Time-Based
Loss Detection'
  <draft-ietf-tcpm-rto-consider-14.txt> as Best Current Practice

The IESG plans to make a decision in the next few weeks, and solicits final
comments on this action. Please send substantive comments to the
last-call@ietf.org mailing lists by 2020-06-01. Exceptionally, comments may
be sent to iesg@ietf.org instead. In either case, please retain the beginning
of the Subject line to allow automated sorting.

Abstract


    Many protocols must detect packet loss for various reasons (e.g., to
    ensure reliability using retransmissions or to understand the level
    of congestion along a network path).  While many mechanisms have
    been designed to detect loss, protocols ultimately can only count on
    the passage of time without delivery confirmation to declare a
    packet "lost".  Each implementation of a time-based loss detection
    mechanism represents a balance between correctness and timeliness
    and therefore no implementation suits all situations.  This document
    provides high-level requirements for time-based loss detectors
    appropriate for general use in the Internet.  Within the
    requirements, implementations have latitude to define particulars
    that best address each situation.




The file can be obtained via
https://datatracker.ietf.org/doc/draft-ietf-tcpm-rto-consider/



No IPR declarations have been submitted directly on this I-D.






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To: Carles Gomez Montenegro <carlesgo@entel.upc.edu>
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Subject: Re: [tcpm] [EXTERNAL] Re: On Sender Control of Delayed Acks in TCP
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> On May 13, 2020, at 5:00 AM, Carles Gomez Montenegro =
<carlesgo@entel.upc.edu> wrote:
>=20
> In this thread, several WG participants expressed a positive opinion =
about
> defining a new TCP option. (Although, as you mentioned, this approach =
also
> has drawbacks.)

Why not start as defining an Experimental Option (with a corresponding =
code point)?

Joe=

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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dus-ascii"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; line-break: after-white-space;" class=3D""><br =
class=3D""><div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D"">On May 13, 2020, at 5:00 AM, Carles Gomez Montenegro &lt;<a =
href=3D"mailto:carlesgo@entel.upc.edu" =
class=3D"">carlesgo@entel.upc.edu</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
12px; font-style: normal; font-variant-caps: normal; font-weight: =
normal; letter-spacing: normal; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; text-decoration: none; float: none; =
display: inline !important;" class=3D"">In this thread, several WG =
participants expressed a positive opinion about</span><br =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
12px; font-style: normal; font-variant-caps: normal; font-weight: =
normal; letter-spacing: normal; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; text-decoration: none;" class=3D""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
12px; font-style: normal; font-variant-caps: normal; font-weight: =
normal; letter-spacing: normal; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; text-decoration: none; float: none; =
display: inline !important;" class=3D"">defining a new TCP option. =
(Although, as you mentioned, this approach also</span><br =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
12px; font-style: normal; font-variant-caps: normal; font-weight: =
normal; letter-spacing: normal; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; text-decoration: none;" class=3D""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
12px; font-style: normal; font-variant-caps: normal; font-weight: =
normal; letter-spacing: normal; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; text-decoration: none; float: none; =
display: inline !important;" class=3D"">has =
drawbacks.)</span></div></blockquote></div><br class=3D""><div =
class=3D"">Why not start as defining an Experimental Option (with a =
corresponding code point)?</div><div class=3D""><br class=3D""></div><div =
class=3D"">Joe</div></body></html>=

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From: Alexander Azimov <a.e.azimov@gmail.com>
Date: Wed, 20 May 2020 11:29:03 +0300
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Subject: Re: [tcpm] TCP Connection ID
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Dear colleagues,

I haven't got any reply which increases level my confusion.

If there was no such proposal I will be glad to volunteer to write such a
document. But if you believe that such functionality is not needed in TCP -
I will be glad to learn your opinion in advance.

=D0=BF=D1=82, 15 =D0=BC=D0=B0=D1=8F 2020 =D0=B3. =D0=B2 18:07, Alexander Az=
imov <a.e.azimov@gmail.com>:

> Dear WG,
>
> As you all know L3 LB is a common element of today's cloud infrastructure=
.
> Its original design was based on the connection state that maps backend I=
Ps
> (real IPs) to connection. Recently there were several papers that were
> suggesting the move to the stateless L3 LB, where the state (connection i=
d)
> becomes the property of connection and it kept at the client, so the L3 L=
B
> can work without keeping state for every connection. As far as I
> understand, for QUIC it becomes native with destination connection ID in
> its header.
>
> I've seen suggestions for getting this functionality in TCP by overloadin=
g
> the Timestamps option or even TCP SEQ - can't say that such design looks
> good for me. I wonder was there an attempt to add connection id in the TC=
P
> options, and if it was - why it wasn't successful?
>
> --
> Best regards,
> Alexander Azimov
>


--=20
Best regards,
Alexander Azimov

--000000000000fc259105a610303f
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<div dir=3D"ltr">Dear colleagues,<div><br></div><div>I haven&#39;t got any =
reply which increases level=C2=A0my confusion.=C2=A0</div><div><br></div><d=
iv>If there was no such proposal I will be glad to volunteer=C2=A0to write =
such a document. But if you believe that such functionality is not needed i=
n TCP - I will be glad to learn your opinion in=C2=A0advance.=C2=A0<br></di=
v><br><div class=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">=D0=
=BF=D1=82, 15 =D0=BC=D0=B0=D1=8F 2020 =D0=B3. =D0=B2 18:07, Alexander Azimo=
v &lt;<a href=3D"mailto:a.e.azimov@gmail.com" target=3D"_blank">a.e.azimov@=
gmail.com</a>&gt;:<br></div><blockquote class=3D"gmail_quote" style=3D"marg=
in:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1e=
x"><div dir=3D"ltr">Dear WG,<div><br></div><div>As you all know L3 LB is a =
common element of today&#39;s cloud infrastructure. Its original design was=
 based on the connection state that maps backend IPs (real IPs) to connecti=
on. Recently=C2=A0there were several papers that were suggesting the move t=
o the stateless L3 LB, where the state (connection id) becomes the property=
 of connection and it kept at the client, so the L3 LB can work without kee=
ping state for every connection. As far as I understand, for QUIC it become=
s native with destination connection ID in its header.<br><br>I&#39;ve seen=
 suggestions for getting this functionality=C2=A0in TCP by overloading the =
Timestamps option or even TCP SEQ - can&#39;t say that such design looks go=
od for me. I wonder was there an attempt to add connection id in the TCP op=
tions, and if it was - why it wasn&#39;t successful?<br></div><div><div></d=
iv><div><br></div>-- <br><div dir=3D"ltr"><div dir=3D"ltr">Best regards,<di=
v>Alexander Azimov</div></div></div></div></div>
</blockquote></div></div><br clear=3D"all"><div><br></div>-- <br><div dir=
=3D"ltr"><div dir=3D"ltr">Best regards,<div>Alexander Azimov</div></div></d=
iv>

--000000000000fc259105a610303f--


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From: Jonathan Morton <chromatix99@gmail.com>
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Date: Wed, 20 May 2020 11:34:07 +0300
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Subject: Re: [tcpm] TCP Connection ID
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> On 20 May, 2020, at 11:29 am, Alexander Azimov <a.e.azimov@gmail.com> =
wrote:
>=20
> If there was no such proposal I will be glad to volunteer to write =
such a document. But if you believe that such functionality is not =
needed in TCP - I will be glad to learn your opinion in advance.=20

Joe Touch replied with a simple mention of RFC-7974.  Did you not see =
that?

I would also point to the Flow ID field in the IPv6 header, which is =
intended to carry this sort of information.

 - Jonathan Morton=


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From: Alexander Azimov <a.e.azimov@gmail.com>
Date: Wed, 20 May 2020 16:08:11 +0300
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Subject: Re: [tcpm] TCP Connection ID
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--000000000000318db005a61417dc
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Hi all,

The email from Joe got into the spam box, so I was able to find it only
after your notice. Thank you.

The Host Id presented in RFC7974 is a very different approach: it is
client-side id generated at the moment of the SYN which should help to
distinguish different hosts behind NAT.

What I'm looking for is a connection id, generated at the server-side and
sent back with a SYN/ACK, which is next mirrored by the client. It is not
meant for host identification, but it can be used for connection
identification by the middlebox under the same administrative control. And
there is no need to have a 1:1 map id to a server, instead, each server
should use randomized ids with encryption.

And speaking about IPv6 Flow Label it's not working for IPv4 + it current
implementation in Linux already has great use cases, so even in the world
of IPv6 only I would prefer to keep it without overloading its functions.

--000000000000318db005a61417dc
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<div dir=3D"ltr">Hi all,<div><br></div><div>The email from=C2=A0Joe got int=
o the spam box, so I was able to find it only after your notice. Thank you.=
</div><div><br></div><div>The Host Id presented in RFC7974 is a very differ=
ent approach: it is client-side id generated at the moment of the SYN which=
 should help to distinguish different hosts behind NAT.<br></div><div><br><=
/div><div>What I&#39;m looking for is a connection id, generated at the ser=
ver-side and sent back with a SYN/ACK, which is next mirrored by the client=
. It is not meant for host identification, but it can be used for connectio=
n identification=C2=A0by the middlebox under the same administrative contro=
l. And there is no need to have a 1:1 map id to a server, instead, each ser=
ver should use randomized ids with encryption.</div><div><br></div><div>And=
 speaking about IPv6 Flow Label it&#39;s not working for IPv4=C2=A0+ it cur=
rent implementation in Linux already has great use cases, so even in the wo=
rld of IPv6 only I would prefer to keep it without=C2=A0overloading its fun=
ctions.</div></div>

--000000000000318db005a61417dc--


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To: Alexander Azimov <a.e.azimov@gmail.com>, Jonathan Morton <chromatix99@gmail.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/tcpm/F2fi3yfKfOL37U1kg161t9sExGY>
Subject: Re: [tcpm] TCP Connection ID
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Read the caveats in the RFC that Joe cited. How would a server chosen direct index fair any better than client selected?
 

 

 

 

 
 
 
 
 
>  
> On May 20, 2020 at 8:08 AM, Alexander Azimov  <a.e.azimov@gmail.com>  wrote:
>  
>    
> Hi all, 
>
>  
> The email from   Joe got into the spam box, so I was able to find it only after your notice. Thank you.
>  
>
>  
> The Host Id presented in RFC7974 is a very different approach: it is client-side id generated at the moment of the SYN which should help to distinguish different hosts behind NAT.
>  
>
>  
> What I'm looking for is a connection id, generated at the server-side and sent back with a SYN/ACK, which is next mirrored by the client.. It is not meant for host identification, but it can be used for connection identification   by the middlebox under the same administrative control. And there is no need to have a 1:1 map id to a server, instead, each server should use randomized ids with encryption.
>  
>
>  
> And speaking about IPv6 Flow Label it's not working for IPv4   + it current implementation in Linux already has great use cases, so even in the world of IPv6 only I would prefer to keep it without   overloading its functions.
>  
>  
>  
>  _______________________________________________ tcpm mailing list tcpm@ietf.org https://www.ietf.org/mailman/listinfo/tcpm 
>  
     
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<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dutf-8"><d=
iv id=3D"edo-message"><div><div>Read the caveats in the RFC that Joe cited.=
 How would a server chosen direct index fair any better than client selecte=
d?</div><div><br></div><div><br></div><div><br></div><div id=3D"edo-signatu=
re"><pre></pre></div></div></div><div id=3D"edo-original"><div><blockquote =
type=3D"cite" style=3D"margin: 1ex 0 0 0 !important; border-left: 1px #ccc =
solid !important; padding-left: 0.4ex !important;"><div id=3D"edo-meta">On =
May 20, 2020 at 8:08 AM, Alexander Azimov &lt;<a href=3D"mailto:a.e.azimov@=
gmail.com">a.e.azimov@gmail.com</a>&gt; wrote: <br><br></div><div dir=3D"lt=
r">Hi all,<div><br></div><div>The email from&nbsp;Joe got into the spam box=
, so I was able to find it only after your notice. Thank you.</div><div><br=
></div><div>The Host Id presented in RFC7974 is a very different approach: =
it is client-side id generated at the moment of the SYN which should help t=
o distinguish different hosts behind NAT.<br></div><div><br></div><div>What=
 I'm looking for is a connection id, generated at the server-side and sent =
back with a SYN/ACK, which is next mirrored by the client.. It is not meant=
 for host identification, but it can be used for connection identification&=
nbsp;by the middlebox under the same administrative control. And there is n=
o need to have a 1:1 map id to a server, instead, each server should use ra=
ndomized ids with encryption.</div><div><br></div><div>And speaking about I=
Pv6 Flow Label it's not working for IPv4&nbsp;&#43; it current implementati=
on in Linux already has great use cases, so even in the world of IPv6 only =
I would prefer to keep it without&nbsp;overloading its functions.</div></di=
v>
<br><pre>_______________________________________________
tcpm mailing list
tcpm@ietf.org
https://www.ietf.org/mailman/listinfo/tcpm
</pre> =20
 <br> </blockquote></div></div>=

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From: Joseph Touch <touch@strayalpha.com>
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Date: Wed, 20 May 2020 10:52:48 -0700
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Subject: Re: [tcpm] TCP Connection ID
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Right - I think that if RFC7974 doesn=E2=80=99t work for you, you=E2=80=99=
ll need to provide a much more precise description of what you=E2=80=99re =
seeking.

=46rom your description below, it=E2=80=99s not a connection ID you =
want; you really seem to be re-inventing cookies (something that lets =
the server re-establish state with a client BETWEEN connections, not =
just within one), and you want that cookie to govern how the middle box =
works.

Good luck with that. I think any decent security mechanism would (or =
should) prevent that.

Joe

> On May 20, 2020, at 10:32 AM, cait <cait@asomi.com> wrote:
>=20
> Read the caveats in the RFC that Joe cited. How would a server chosen =
direct index fair any better than client selected?
>=20
>=20
>=20
>> On May 20, 2020 at 8:08 AM, Alexander Azimov <a.e.azimov@gmail.com =
<mailto:a.e.azimov@gmail.com>> wrote:=20
>>=20
>> Hi all,
>>=20
>> The email from Joe got into the spam box, so I was able to find it =
only after your notice. Thank you.
>>=20
>> The Host Id presented in RFC7974 is a very different approach: it is =
client-side id generated at the moment of the SYN which should help to =
distinguish different hosts behind NAT.
>>=20
>> What I'm looking for is a connection id, generated at the server-side =
and sent back with a SYN/ACK, which is next mirrored by the client.. It =
is not meant for host identification, but it can be used for connection =
identification by the middlebox under the same administrative control. =
And there is no need to have a 1:1 map id to a server, instead, each =
server should use randomized ids with encryption.
>>=20
>> And speaking about IPv6 Flow Label it's not working for IPv4 + it =
current implementation in Linux already has great use cases, so even in =
the world of IPv6 only I would prefer to keep it without overloading its =
functions.
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; line-break: after-white-space;" class=3D"">Right=
 - I think that if RFC7974 doesn=E2=80=99t work for you, you=E2=80=99ll =
need to provide a much more precise description of what you=E2=80=99re =
seeking.<div class=3D""><br class=3D""></div><div class=3D"">=46rom your =
description below, it=E2=80=99s not a connection ID you want; you really =
seem to be re-inventing cookies (something that lets the server =
re-establish state with a client BETWEEN connections, not just within =
one), and you want that cookie to govern how the middle box =
works.</div><div class=3D""><br class=3D""></div><div class=3D"">Good =
luck with that. I think any decent security mechanism would (or should) =
prevent that.</div><div class=3D""><br class=3D""></div><div =
class=3D"">Joe</div><div class=3D""><div><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D"">On May 20, 2020, at 10:32 AM, =
cait &lt;<a href=3D"mailto:cait@asomi.com" =
class=3D"">cait@asomi.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><meta =
http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dutf-8" =
class=3D""><div id=3D"edo-message" class=3D""><div class=3D""><div =
class=3D"">Read the caveats in the RFC that Joe cited. How would a =
server chosen direct index fair any better than client =
selected?</div><div class=3D""><br class=3D""></div><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div><div =
id=3D"edo-signature" class=3D""><pre =
class=3D""></pre></div></div></div><div id=3D"edo-original" =
class=3D""><div class=3D""><blockquote type=3D"cite" style=3D"margin: =
1ex 0 0 0 !important; border-left: 1px #ccc solid !important; =
padding-left: 0.4ex !important;" class=3D""><div id=3D"edo-meta" =
class=3D"">On May 20, 2020 at 8:08 AM, Alexander Azimov &lt;<a =
href=3D"mailto:a.e.azimov@gmail.com" =
class=3D"">a.e.azimov@gmail.com</a>&gt; wrote: <br class=3D""><br =
class=3D""></div><div dir=3D"ltr" class=3D"">Hi all,<div class=3D""><br =
class=3D""></div><div class=3D"">The email from&nbsp;Joe got into the =
spam box, so I was able to find it only after your notice. Thank =
you.</div><div class=3D""><br class=3D""></div><div class=3D"">The Host =
Id presented in RFC7974 is a very different approach: it is client-side =
id generated at the moment of the SYN which should help to distinguish =
different hosts behind NAT.<br class=3D""></div><div class=3D""><br =
class=3D""></div><div class=3D"">What I'm looking for is a connection =
id, generated at the server-side and sent back with a SYN/ACK, which is =
next mirrored by the client.. It is not meant for host identification, =
but it can be used for connection identification&nbsp;by the middlebox =
under the same administrative control. And there is no need to have a =
1:1 map id to a server, instead, each server should use randomized ids =
with encryption.</div><div class=3D""><br class=3D""></div><div =
class=3D"">And speaking about IPv6 Flow Label it's not working for =
IPv4&nbsp;+ it current implementation in Linux already has great use =
cases, so even in the world of IPv6 only I would prefer to keep it =
without&nbsp;overloading its functions.</div></div>
<br class=3D""><pre =
class=3D"">_______________________________________________
tcpm mailing list
<a href=3D"mailto:tcpm@ietf.org" class=3D"">tcpm@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" =
class=3D"">https://www.ietf.org/mailman/listinfo/tcpm</a>
</pre> =20
 <br class=3D""> =
</blockquote></div></div>_______________________________________________<b=
r class=3D"">tcpm mailing list<br class=3D""><a =
href=3D"mailto:tcpm@ietf.org" class=3D"">tcpm@ietf.org</a><br =
class=3D"">https://www.ietf.org/mailman/listinfo/tcpm<br =
class=3D""></div></blockquote></div><br class=3D""></div></body></html>=

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From: Alexander Azimov <a.e.azimov@gmail.com>
Date: Wed, 20 May 2020 21:44:39 +0300
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To: Joseph Touch <touch@strayalpha.com>
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Subject: Re: [tcpm] TCP Connection ID
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Thank you all for the comments. I will combine my answers all-in-one reply.

Michael:
The mechanics of TFO has significant intersection, but with a certain
difference. The TFO is applied on the client src IP, while the connection
ID should depend on the server IP, which is invisible for the client. And
while TFO is just cookie (you can't retrieve the value since it was
truncated) connection ID should be reversible. And of course, the
Connection Id must be present in all packets except the first SYN.

Cait (I don't see your full name, sorry if I got it wrong):
Can you please clarify your comment? Are you speaking about IESG note? If
yes, I don't see how privacy concerns of host id are applicable in this
case.

Joseph:
If you read my suggestions as something cached between connection you got
it wrong. What I'm trying to discuss is an option similar to DSID in QUIC,
which is put in the unencrypted part of the header for packets sent from
the client to the server. The straightforward way is to apply AES on 128bit
block where the first 64 bits go from high bits server IP (real server IP,
not VIP) + random 64 bits/timestamp. The middlebox (L3 LB), knowing the
secret, will be able to retrieve real IP and make balancing without keeping
the state of the connection. Lack of space in TCP options might be a
problem for such an approach.



=D1=81=D1=80, 20 =D0=BC=D0=B0=D1=8F 2020 =D0=B3. =D0=B2 20:52, Joseph Touch=
 <touch@strayalpha.com>:

> Right - I think that if RFC7974 doesn=E2=80=99t work for you, you=E2=80=
=99ll need to
> provide a much more precise description of what you=E2=80=99re seeking.
>
> From your description below, it=E2=80=99s not a connection ID you want; y=
ou really
> seem to be re-inventing cookies (something that lets the server
> re-establish state with a client BETWEEN connections, not just within one=
),
> and you want that cookie to govern how the middle box works.
>
> Good luck with that. I think any decent security mechanism would (or
> should) prevent that.
>
> Joe
>
> On May 20, 2020, at 10:32 AM, cait <cait@asomi.com> wrote:
>
> Read the caveats in the RFC that Joe cited. How would a server chosen
> direct index fair any better than client selected?
>
>
>
> On May 20, 2020 at 8:08 AM, Alexander Azimov <a.e.azimov@gmail.com>
> wrote:
>
> Hi all,
>
> The email from Joe got into the spam box, so I was able to find it only
> after your notice. Thank you.
>
> The Host Id presented in RFC7974 is a very different approach: it is
> client-side id generated at the moment of the SYN which should help to
> distinguish different hosts behind NAT.
>
> What I'm looking for is a connection id, generated at the server-side and
> sent back with a SYN/ACK, which is next mirrored by the client.. It is no=
t
> meant for host identification, but it can be used for connection
> identification by the middlebox under the same administrative control. An=
d
> there is no need to have a 1:1 map id to a server, instead, each server
> should use randomized ids with encryption.
>
> And speaking about IPv6 Flow Label it's not working for IPv4 + it current
> implementation in Linux already has great use cases, so even in the world
> of IPv6 only I would prefer to keep it without overloading its functions.
>
> _______________________________________________
> tcpm mailing listtcpm@ietf.orghttps://www.ietf.org/mailman/listinfo/tcpm
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>
>
>

--=20
Best regards,
Alexander Azimov

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<div dir=3D"ltr"><div>Thank you all for the comments. I will combine my ans=
wers all-in-one reply.</div><div><br></div><div>Michael:</div><div>The mech=
anics of TFO has significant intersection, but with a certain difference. T=
he TFO is applied on the client src IP, while the connection ID should depe=
nd on the server IP, which is invisible for the client. And while TFO is ju=
st cookie=C2=A0(you can&#39;t retrieve the value since it was truncated) co=
nnection ID should be reversible. And of course, the Connection Id must be =
present in all packets except the first SYN.</div><div><br>Cait (I don&#39;=
t see your full name, sorry if I got it wrong):<br></div><div>Can you pleas=
e clarify your comment? Are you speaking about IESG note? If yes, I don&#39=
;t see how privacy concerns=C2=A0of host=C2=A0id=C2=A0are applicable in=C2=
=A0this case.</div><div><br></div>Joseph:=C2=A0<div>If you read my suggesti=
ons as something cached between connection you got it wrong. What I&#39;m t=
rying to discuss is an option similar to DSID in QUIC, which is put in the =
unencrypted=C2=A0part of the header for packets sent from the client=C2=A0t=
o the server. The straightforward=C2=A0way is to apply AES on 128bit block =
where the first 64 bits go from high bits server IP (real server IP, not VI=
P) + random 64 bits/timestamp. The middlebox (L3 LB), knowing the secret, w=
ill be able to retrieve real IP and make balancing without keeping the stat=
e of the connection. Lack of space in TCP options might be a problem for su=
ch an approach.<br></div><div><br></div><div><br></div></div><br><div class=
=3D"gmail_quote"><div dir=3D"ltr" class=3D"gmail_attr">=D1=81=D1=80, 20 =D0=
=BC=D0=B0=D1=8F 2020 =D0=B3. =D0=B2 20:52, Joseph Touch &lt;<a href=3D"mail=
to:touch@strayalpha.com">touch@strayalpha.com</a>&gt;:<br></div><blockquote=
 class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px so=
lid rgb(204,204,204);padding-left:1ex"><div style=3D"overflow-wrap: break-w=
ord;">Right - I think that if RFC7974 doesn=E2=80=99t work for you, you=E2=
=80=99ll need to provide a much more precise description of what you=E2=80=
=99re seeking.<div><br></div><div>From your description below, it=E2=80=99s=
 not a connection ID you want; you really seem to be re-inventing cookies (=
something that lets the server re-establish state with a client BETWEEN con=
nections, not just within one), and you want that cookie to govern how the =
middle box works.</div><div><br></div><div>Good luck with that. I think any=
 decent security mechanism would (or should) prevent that.</div><div><br></=
div><div>Joe</div><div><div><br><blockquote type=3D"cite"><div>On May 20, 2=
020, at 10:32 AM, cait &lt;<a href=3D"mailto:cait@asomi.com" target=3D"_bla=
nk">cait@asomi.com</a>&gt; wrote:</div><br><div><div id=3D"gmail-m_55022787=
62794127888edo-message"><div><div>Read the caveats in the RFC that Joe cite=
d. How would a server chosen direct index fair any better than client selec=
ted?</div><div><br></div><div><br></div><div><br></div><div id=3D"gmail-m_5=
502278762794127888edo-signature"><pre></pre></div></div></div><div id=3D"gm=
ail-m_5502278762794127888edo-original"><div><blockquote type=3D"cite" style=
=3D"margin:1ex 0px 0px;border-left:1px solid rgb(204,204,204);padding-left:=
0.4ex"><div id=3D"gmail-m_5502278762794127888edo-meta">On May 20, 2020 at 8=
:08 AM, Alexander Azimov &lt;<a href=3D"mailto:a.e.azimov@gmail.com" target=
=3D"_blank">a.e.azimov@gmail.com</a>&gt; wrote: <br><br></div><div dir=3D"l=
tr">Hi all,<div><br></div><div>The email from=C2=A0Joe got into the spam bo=
x, so I was able to find it only after your notice. Thank you.</div><div><b=
r></div><div>The Host Id presented in RFC7974 is a very different approach:=
 it is client-side id generated at the moment of the SYN which should help =
to distinguish different hosts behind NAT.<br></div><div><br></div><div>Wha=
t I&#39;m looking for is a connection id, generated at the server-side and =
sent back with a SYN/ACK, which is next mirrored by the client.. It is not =
meant for host identification, but it can be used for connection identifica=
tion=C2=A0by the middlebox under the same administrative control. And there=
 is no need to have a 1:1 map id to a server, instead, each server should u=
se randomized ids with encryption.</div><div><br></div><div>And speaking ab=
out IPv6 Flow Label it&#39;s not working for IPv4=C2=A0+ it current impleme=
ntation in Linux already has great use cases, so even in the world of IPv6 =
only I would prefer to keep it without=C2=A0overloading its functions.</div=
></div>
<br><pre>_______________________________________________
tcpm mailing list
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tcpm</a>
</pre> =20
 <br> </blockquote></div></div>____________________________________________=
___<br>tcpm mailing list<br><a href=3D"mailto:tcpm@ietf.org" target=3D"_bla=
nk">tcpm@ietf.org</a><br><a href=3D"https://www.ietf.org/mailman/listinfo/t=
cpm" target=3D"_blank">https://www.ietf.org/mailman/listinfo/tcpm</a><br></=
div></blockquote></div><br></div></div></blockquote></div><br clear=3D"all"=
><div><br></div>-- <br><div dir=3D"ltr" class=3D"gmail_signature"><div dir=
=3D"ltr">Best regards,<div>Alexander Azimov</div></div></div>

--0000000000008f2c7205a618ca9e--


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Subject: Re: [tcpm] TCP Connection ID
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> On May 20, 2020, at 11:44 AM, Alexander Azimov <a.e.azimov@gmail.com> =
wrote:
>=20
> Joseph:=20
> If you read my suggestions as something cached between connection you =
got it wrong. What I'm trying to discuss is an option similar to DSID in =
QUIC, which is put in the unencrypted part of the header for packets =
sent from the client to the server. The straightforward way is to apply =
AES on 128bit block where the first 64 bits go from high bits server IP =
(real server IP, not VIP) + random 64 bits/timestamp. The middlebox (L3 =
LB), knowing the secret, will be able to retrieve real IP and make =
balancing without keeping the state of the connection. Lack of space in =
TCP options might be a problem for such an approach.

You can do this by combining the existing TCP Host ID option (RFC 7974) =
with IP LSR.

You only really need:
	- a TCP option in the SYN and SYN-ACK (8-bytes using the RFC7874 =
variant) but not in later segments
	- IP LSR at the client
	- a middlebox that rewrites IP headers of SYN packets with LSR

Here=E2=80=99s how:

Servers are assigned two addresses each - one unique (P, Q, R, =E2=80=A6),=
 one common (let=E2=80=99s say =E2=80=9CX=E2=80=9D)
All servers are configured to forward

Clients send SYN to LSR X,X with TCP HID option with an empty value
	You need IP to have LSR because the middlebox should NEVER make =
a packet longer; it could mess up MTU discovery
	You need the TCP HID option by the rules of that option
LB middlebox picks a server (P) and rewrites the IP LSR X, X with the =
LSR P,X
	So load balancing happens only on SYNs
The SYN proceeds to server P, which performs LSR; in the process of =
forwarding, it accepts the packet at X
Server P needs to see the IP LSR P,X and extract P
Server P puts P in the SYN-ACK TCP HID option and sends it back to the =
client
	now that you receive the TCP HID option back, the client no =
longer needs to include it in future packets

The client sees P in the SYN-ACK option and configures that TCP =
connection to send all future packets with IP LSR P,X

At that point the middlebox doesn=E2=80=99t need to do anything anymore; =
everything is handled by the client, who has to keep state for the =
connection anyway.

This can work even with end-to-end TCP encryption or authentication. The =
only thing it needs is an outermost IP header whose LSR can be =
rewritten.

Joe=

--Apple-Mail=_88D41AE7-96F4-4DE0-83D9-19F0F4AD7250
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=utf-8

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; line-break: after-white-space;" class=3D""><br =
class=3D""><div><blockquote type=3D"cite" class=3D""><div class=3D"">On =
May 20, 2020, at 11:44 AM, Alexander Azimov &lt;<a =
href=3D"mailto:a.e.azimov@gmail.com" =
class=3D"">a.e.azimov@gmail.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
12px; font-style: normal; font-variant-caps: normal; font-weight: =
normal; letter-spacing: normal; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; text-decoration: none; float: none; =
display: inline !important;" class=3D"">Joseph:&nbsp;</span><div =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
12px; font-style: normal; font-variant-caps: normal; font-weight: =
normal; letter-spacing: normal; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; text-decoration: none;" class=3D"">If =
you read my suggestions as something cached between connection you got =
it wrong. What I'm trying to discuss is an option similar to DSID in =
QUIC, which is put in the unencrypted&nbsp;part of the header for =
packets sent from the client&nbsp;to the server. The =
straightforward&nbsp;way is to apply AES on 128bit block where the first =
64 bits go from high bits server IP (real server IP, not VIP) + random =
64 bits/timestamp. The middlebox (L3 LB), knowing the secret, will be =
able to retrieve real IP and make balancing without keeping the state of =
the connection. Lack of space in TCP options might be a problem for such =
an approach.</div></div></blockquote><br class=3D""></div><div>You can =
do this by combining the existing TCP Host ID option (RFC&nbsp;<span =
style=3D"font-size: 13.333333015441895px;" =
class=3D"">7974)</span>&nbsp;with IP LSR.</div><br class=3D""><div =
class=3D"">You only really need:</div><div class=3D""><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>-&nbsp;a =
TCP option in the SYN and SYN-ACK (8-bytes using the RFC7874 variant) =
but not in later segments</div><div class=3D""><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>-&nbsp;IP =
LSR at the client</div><div class=3D""><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>- a middlebox that rewrites IP =
headers of SYN packets with LSR</div><div class=3D""><br =
class=3D""></div><div class=3D"">Here=E2=80=99s how:</div><div =
class=3D""><br class=3D""></div><div class=3D"">Servers are assigned two =
addresses each - one unique (P, Q, R, =E2=80=A6), one common (let=E2=80=99=
s say =E2=80=9CX=E2=80=9D)</div><div class=3D"">All servers are =
configured to forward</div><div class=3D""><br class=3D""></div><div =
class=3D"">Clients send SYN to LSR X,X with TCP HID option with an empty =
value</div><div class=3D""><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>You need IP to have LSR because =
the middlebox should NEVER make a packet longer; it could mess up MTU =
discovery</div><div class=3D""><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>You need the TCP HID option by =
the rules of that option</div><div class=3D"">LB middlebox picks a =
server (P) and rewrites the IP LSR X, X with the LSR P,X</div><div =
class=3D""><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>So load balancing happens only on SYNs</div><div class=3D"">The =
SYN proceeds to server P, which performs LSR; in the process of =
forwarding, it accepts the packet at X</div><div class=3D"">Server P =
needs to see the IP LSR P,X and extract P</div><div class=3D"">Server P =
puts P in the SYN-ACK TCP HID option and sends it back to the =
client</div><div class=3D""><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>now that you receive the TCP HID =
option back, the client no longer needs to include it in future =
packets</div><div class=3D""><br class=3D""></div><div class=3D"">The =
client sees P in the SYN-ACK option and configures that TCP connection =
to send all future packets with IP LSR P,X</div><div class=3D""><br =
class=3D""></div><div class=3D"">At that point the middlebox doesn=E2=80=99=
t need to do anything anymore; everything is handled by the client, who =
has to keep state for the connection anyway.</div><div class=3D""><br =
class=3D""></div><div class=3D"">This can work even with end-to-end TCP =
encryption or authentication. The only thing it needs is an outermost IP =
header whose LSR can be rewritten.</div><div class=3D""><br =
class=3D""></div><div class=3D"">Joe</div></body></html>=

--Apple-Mail=_88D41AE7-96F4-4DE0-83D9-19F0F4AD7250--


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From: Olivier Bonaventure <olivier.bonaventure@tessares.net>
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To: Alexander Azimov <a.e.azimov@gmail.com>
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Subject: Re: [tcpm] TCP Connection ID
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Dear Alexander,

>=20
> As you all know L3 LB is a common element of today's cloud infrastructure=
. Its original design was based on the connection state that maps backend I=
Ps (real IPs) to connection. Recently there were several papers that were s=
uggesting the move to the stateless L3 LB, where the state (connection id) =
becomes the property of connection and it kept at the client, so the L3 LB =
can work without keeping state for every connection. As far as I understand=
, for QUIC it becomes native with destination connection ID in its header.
>=20
> I've seen suggestions for getting this functionality in TCP by overloadin=
g the Timestamps option or even TCP SEQ - can't say that such design looks =
good for me. I wonder was there an attempt to add connection id in the TCP =
options, and if it was - why it wasn't successful?

Overloading TCP sequence numbers or TCP timestamps to support this feature =
could lead to problems that are difficult to troubleshoot as there are midd=
leboxes that modify these fields.

The latest version of Multipath TCP (RFC8684) includes a feature to support=
 load-balancers. The connection is established via the load balancer that f=
orwards the SYN to a server and this server advertises its own address to t=
he client. This enables the client to stick the new subflows of the Multipa=
th TCP connection to this server and thus completely bypass the load balanc=
er. This is described in https://inl.info.ucl.ac.be/publications/making-mul=
tipath-tcp-friendlier-load-balancers-and-anycast.html <https://inl.info.ucl=
.ac.be/publications/making-multipath-tcp-friendlier-load-balancers-and-anyc=
ast.html>

Olivier
--=20


Disclaimer: https://www.tessares.net/mail-disclaimer/=20
<https://www.tessares.net/mail-disclaimer/>



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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html; charset=
=3Dus-ascii"></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode=
: space; line-break: after-white-space;" class=3D""><div>Dear Alexander,</d=
iv><div><br class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""=
><div dir=3D"ltr" class=3D""><div class=3D"gmail_quote"><blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rg=
b(204,204,204);padding-left:1ex"><div dir=3D"ltr" class=3D""><div class=3D"=
"><br class=3D""></div><div class=3D"">As you all know L3 LB is a common el=
ement of today's cloud infrastructure. Its original design was based on the=
 connection state that maps backend IPs (real IPs) to connection. Recently&=
nbsp;there were several papers that were suggesting the move to the statele=
ss L3 LB, where the state (connection id) becomes the property of connectio=
n and it kept at the client, so the L3 LB can work without keeping state fo=
r every connection. As far as I understand, for QUIC it becomes native with=
 destination connection ID in its header.<br class=3D""><br class=3D"">I've=
 seen suggestions for getting this functionality&nbsp;in TCP by overloading=
 the Timestamps option or even TCP SEQ - can't say that such design looks g=
ood for me. I wonder was there an attempt to add connection id in the TCP o=
ptions, and if it was - why it wasn't successful?<br class=3D""></div><div =
class=3D""><div class=3D""></div></div></div></blockquote></div></div></div=
></blockquote></div><br class=3D""><div class=3D"">Overloading TCP sequence=
 numbers or TCP timestamps to support this feature could lead to problems t=
hat are difficult to troubleshoot as there are middleboxes that modify thes=
e fields.</div><div class=3D""><br class=3D""></div><div class=3D"">The lat=
est version of Multipath TCP (RFC8684) includes a feature to support load-b=
alancers. The connection is established via the load balancer that forwards=
 the SYN to a server and this server advertises its own address to the clie=
nt. This enables the client to stick the new subflows of the Multipath TCP =
connection to this server and thus completely bypass the load balancer. Thi=
s is described in&nbsp;<a href=3D"https://inl.info.ucl.ac.be/publications/m=
aking-multipath-tcp-friendlier-load-balancers-and-anycast.html" class=3D"">=
https://inl.info.ucl.ac.be/publications/making-multipath-tcp-friendlier-loa=
d-balancers-and-anycast.html</a></div><div class=3D""><br class=3D""></div>=
<div class=3D"">Olivier</div></body></html>
<br>
<div><hr><br></div><div>Disclaimer: <a href=3D"https://www.tessares.net/mai=
l-disclaimer/" target=3D"_blank">https://www.tessares.net/mail-<wbr>disclai=
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From: Alexander Azimov <a.e.azimov@gmail.com>
Date: Thu, 21 May 2020 18:47:13 +0300
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To: Olivier Bonaventure <olivier.bonaventure@tessares.net>
Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
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Subject: Re: [tcpm] TCP Connection ID
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Hi everyone, here is another all-in-one reply.

Joseph:
As far as I understand you are suggesting to use IP LSR/RH which
SHOULD be turned
off by default (RFC7196). Besides, P and X, in your terms may belong to
different address families. It seems that you still have confusion on the
purpose of my proposal:

   - There is a browser, that sends a WEB-request. The request is sent to
   the virtual IP (VIP). The browser/OS is out of our control.
   - There is a load balancer inside the datacenter, which receives an SYN
   packet and needs to send it to the server with real IP. Normally it is d=
one
   via tunnel: IPIP or GRE. Both load balancer and server are under the sam=
e
   administrative control.
   - The idea is to introduce a connection id, that is generated by the
   server at the time of SYN/ACK and kept at the client for the time of TCP
   session, thus removing the need to keep the state at the load balancer.
   This brings advantages in the case of DDoS attacks, load drain from
   servers, and traffic rerouting between LB.


Oliver:
Thank you for the link! I was not aware of this paper. The option to
advertise VIP looks interesting, but seem to have limitations:

   - What if real server IP is IPv6 address while the client is working
   from IPv4 only network?
   - The attacker may learn the full list of backend servers, which can be
   used for DDoS attacks. The dynamic IP allocation for connections may wor=
k
   as a partial workaround, have you implemented it in the lab?
   - The load balancer still needs to keep states of the connections. But
   it wasn't the problem that was addressed in the paper.

Anyway, the connection id, as I see previously described, is a function on
top of real server IP, and with proper encoding/decoding it will be working
for Anycast when connection jumps from one to another LB. It should
be working for MPTCP case too.

PS: previously in this thread, I used the term L3LB, while in Oliver's
paper it is named L4 load balancer. I hope it hasn't brought additional
confusion.

=D1=87=D1=82, 21 =D0=BC=D0=B0=D1=8F 2020 =D0=B3. =D0=B2 16:38, Olivier Bona=
venture <
olivier.bonaventure@tessares.net>:

> Dear Alexander,
>
>
>> As you all know L3 LB is a common element of today's cloud
>> infrastructure. Its original design was based on the connection state th=
at
>> maps backend IPs (real IPs) to connection. Recently there were several
>> papers that were suggesting the move to the stateless L3 LB, where the
>> state (connection id) becomes the property of connection and it kept at =
the
>> client, so the L3 LB can work without keeping state for every connection=
.
>> As far as I understand, for QUIC it becomes native with destination
>> connection ID in its header.
>>
>> I've seen suggestions for getting this functionality in TCP by
>> overloading the Timestamps option or even TCP SEQ - can't say that such
>> design looks good for me. I wonder was there an attempt to add connectio=
n
>> id in the TCP options, and if it was - why it wasn't successful?
>>
>
> Overloading TCP sequence numbers or TCP timestamps to support this featur=
e
> could lead to problems that are difficult to troubleshoot as there are
> middleboxes that modify these fields.
>
> The latest version of Multipath TCP (RFC8684) includes a feature to
> support load-balancers. The connection is established via the load balanc=
er
> that forwards the SYN to a server and this server advertises its own
> address to the client. This enables the client to stick the new subflows =
of
> the Multipath TCP connection to this server and thus completely bypass th=
e
> load balancer. This is described in
> https://inl.info.ucl.ac.be/publications/making-multipath-tcp-friendlier-l=
oad-balancers-and-anycast.html
>
> Olivier
>
> ------------------------------
>
> Disclaimer: https://www.tessares.net/mail-disclaimer/
>
>
>

--=20
Best regards,
Alexander Azimov

--000000000000cbb45c05a62a6d54
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<div dir=3D"ltr">Hi everyone, here is another all-in-one reply.<div><br></d=
iv><div>Joseph:</div><div>As far as I understand you are suggesting to use =
IP=20

 LSR/RH which=C2=A0<span style=3D"color:rgb(0,0,0);font-size:13.3333px">SHO=
ULD be</span>=C2=A0turned off by default (RFC7196). Besides, P and X, in yo=
ur terms may belong to different address families. It seems that you still =
have=C2=A0confusion=C2=A0on the purpose of my proposal:</div><div><ul><li>T=
here is a browser, that sends a WEB-request. The request=C2=A0is sent to th=
e virtual IP (VIP). The browser/OS is out of our control.</li><li>There is =
a load balancer inside the datacenter, which receives an SYN packet and nee=
ds to send it to the server with real IP. Normally it is done via tunnel: I=
PIP or GRE. Both load balancer and server are under the same administrative=
 control.</li><li>The idea is to introduce a connection id, that is generat=
ed by the server at the time of SYN/ACK and kept at the client for the time=
 of TCP session, thus removing the need to keep the state at the load balan=
cer. This brings advantages in the case of DDoS attacks, load drain from se=
rvers, and traffic rerouting between LB.<br></li></ul></div><div><br></div>=
<div>Oliver:</div><div>Thank you for the link! I was not aware of this pape=
r. The option to advertise=C2=A0VIP looks interesting, but seem to have lim=
itations:</div><div><ul><li>What if real server=C2=A0IP is IPv6 address whi=
le the client is working from IPv4 only network?</li><li>The attacker may l=
earn the full list of backend servers, which can be used for DDoS attacks. =
The dynamic IP allocation for connections may work as a partial workaround,=
 have you implemented it in the lab?</li><li>The load balancer still needs =
to keep states of the connections. But it wasn&#39;t the problem that was a=
ddressed in the paper.</li></ul><div>Anyway, the connection id, as I see pr=
eviously described, is a function on top of real server IP, and with proper=
 encoding/decoding it will be working for Anycast when connection jumps fro=
m one to another LB. It should be=C2=A0working for MPTCP case too.</div></d=
iv><div><br></div><div>PS: previously in this thread, I used the term L3LB,=
 while in Oliver&#39;s paper it is named L4 load balancer. I hope it hasn&#=
39;t brought=C2=A0additional confusion.</div></div><br><div class=3D"gmail_=
quote"><div dir=3D"ltr" class=3D"gmail_attr">=D1=87=D1=82, 21 =D0=BC=D0=B0=
=D1=8F 2020 =D0=B3. =D0=B2 16:38, Olivier Bonaventure &lt;<a href=3D"mailto=
:olivier.bonaventure@tessares.net">olivier.bonaventure@tessares.net</a>&gt;=
:<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.=
8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div style=3D"=
overflow-wrap: break-word;"><div>Dear Alexander,</div><div><br><blockquote =
type=3D"cite"><div><div dir=3D"ltr"><div class=3D"gmail_quote"><blockquote =
class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px sol=
id rgb(204,204,204);padding-left:1ex"><div dir=3D"ltr"><div><br></div><div>=
As you all know L3 LB is a common element of today&#39;s cloud infrastructu=
re. Its original design was based on the connection state that maps backend=
 IPs (real IPs) to connection. Recently=C2=A0there were several papers that=
 were suggesting the move to the stateless L3 LB, where the state (connecti=
on id) becomes the property of connection and it kept at the client, so the=
 L3 LB can work without keeping state for every connection. As far as I und=
erstand, for QUIC it becomes native with destination connection ID in its h=
eader.<br><br>I&#39;ve seen suggestions for getting this functionality=C2=
=A0in TCP by overloading the Timestamps option or even TCP SEQ - can&#39;t =
say that such design looks good for me. I wonder was there an attempt to ad=
d connection id in the TCP options, and if it was - why it wasn&#39;t succe=
ssful?<br></div><div><div></div></div></div></blockquote></div></div></div>=
</blockquote></div><br><div>Overloading TCP sequence numbers or TCP timesta=
mps to support this feature could lead to problems that are difficult to tr=
oubleshoot as there are middleboxes that modify these fields.</div><div><br=
></div><div>The latest version of Multipath TCP (RFC8684) includes a featur=
e to support load-balancers. The connection is established via the load bal=
ancer that forwards the SYN to a server and this server advertises its own =
address to the client. This enables the client to stick the new subflows of=
 the Multipath TCP connection to this server and thus completely bypass the=
 load balancer. This is described in=C2=A0<a href=3D"https://inl.info.ucl.a=
c.be/publications/making-multipath-tcp-friendlier-load-balancers-and-anycas=
t.html" target=3D"_blank">https://inl.info.ucl.ac.be/publications/making-mu=
ltipath-tcp-friendlier-load-balancers-and-anycast.html</a></div><div><br></=
div><div>Olivier</div></div>
<br>
<div><hr><br></div><div>Disclaimer: <a href=3D"https://www.tessares.net/mai=
l-disclaimer/" target=3D"_blank">https://www.tessares.net/mail-disclaimer/<=
/a></div><br><br></blockquote></div><br clear=3D"all"><div><br></div>-- <br=
><div dir=3D"ltr" class=3D"gmail_signature"><div dir=3D"ltr">Best regards,<=
div>Alexander Azimov</div></div></div>

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Subject: Re: [tcpm] TCP Connection ID
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> On May 21, 2020, at 8:47 AM, Alexander Azimov <a.e.azimov@gmail.com> =
wrote:
>=20
> Hi everyone, here is another all-in-one reply.
>=20
> Joseph:
> As far as I understand you are suggesting to use IP LSR/RH which =
SHOULD be turned off by default (RFC7196).

There=E2=80=99s no mention of LSR in RFC7196. Perhaps you mean RFC7126 - =
in which case, yes - another blow to the flexibility of the Internet by =
BCPs that treat anything unexpected as an attack ;-(

> Besides, P and X, in your terms may belong to different address =
families.

You=E2=80=99re adding new constraints, but OK.

> It seems that you still have confusion on the purpose of my proposal:
> There is a browser, that sends a WEB-request. The request is sent to =
the virtual IP (VIP). The browser/OS is out of our control.
Then you wouldn=E2=80=99t be talking about TCP option anymore (options =
cannot be added en-route). You also wouldn=E2=80=99t be talking about =
MPTCP either - in general.
> There is a load balancer inside the datacenter, which receives an SYN =
packet and needs to send it to the server with real IP. Normally it is =
done via tunnel: IPIP or GRE. Both load balancer and server are under =
the same administrative control.
> The idea is to introduce a connection id, that is generated by the =
server at the time of SYN/ACK and kept at the client for the time of TCP =
session,
The server TCP sequence number and timestamps are the only things that =
most TCP clients already keep track of that the server has control ove).=20=


Both of those have other meanings and change during a connection, which =
means that the middlebox would have to keep state to track them. But you =
also said the middlebox doesn=E2=80=99t keep state.
> thus removing the need to keep the state at the load balancer. This =
brings advantages in the case of DDoS attacks, load drain from servers, =
and traffic rerouting between LB.

I see more clearly what you want, but no way to get there if the client =
browser/OS is unmodified.

Joe


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; line-break: after-white-space;" class=3D""><br =
class=3D""><div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D"">On May 21, 2020, at 8:47 AM, Alexander Azimov &lt;<a =
href=3D"mailto:a.e.azimov@gmail.com" =
class=3D"">a.e.azimov@gmail.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><div dir=3D"ltr" =
class=3D"">Hi everyone, here is another all-in-one reply.<div =
class=3D""><br class=3D""></div><div class=3D"">Joseph:</div><div =
class=3D"">As far as I understand you are suggesting to use IP=20

 LSR/RH which&nbsp;<span style=3D"font-size: 13.3333px;" class=3D"">SHOULD=
 be</span>&nbsp;turned off by default =
(RFC7196).</div></div></div></blockquote><div><br =
class=3D""></div><div>There=E2=80=99s no mention of LSR in RFC7196. =
Perhaps you mean RFC7126 - in which case, yes - another blow to the =
flexibility of the Internet by BCPs that treat anything unexpected as an =
attack ;-(</div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><div dir=3D"ltr" class=3D""><div class=3D""> Besides, P and =
X, in your terms may belong to different address families. =
</div></div></div></blockquote><div><br class=3D""></div><div>You=E2=80=99=
re adding new constraints, but OK.</div><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><div dir=3D"ltr" class=3D""><div =
class=3D"">It seems that you still have&nbsp;confusion&nbsp;on the =
purpose of my proposal:</div><div class=3D""><ul class=3D""><li =
class=3D"">There is a browser, that sends a WEB-request. The =
request&nbsp;is sent to the virtual IP (VIP). The browser/OS is out of =
our control.</li></ul></div></div></div></blockquote><div>Then you =
wouldn=E2=80=99t be talking about TCP option anymore (options cannot be =
added en-route). You also wouldn=E2=80=99t be talking about MPTCP either =
- in general.</div><blockquote type=3D"cite" class=3D""><div =
class=3D""><div dir=3D"ltr" class=3D""><div class=3D""><ul class=3D""><li =
class=3D"">There is a load balancer inside the datacenter, which =
receives an SYN packet and needs to send it to the server with real IP. =
Normally it is done via tunnel: IPIP or GRE. Both load balancer and =
server are under the same administrative control.</li><li class=3D"">The =
idea is to introduce a connection id, that is generated by the server at =
the time of SYN/ACK and kept at the client for the time of TCP session, =
</li></ul></div></div></div></blockquote><div>The server TCP sequence =
number and timestamps are the only things that most TCP clients already =
keep track of that the server has control ove).&nbsp;</div><div><br =
class=3D""></div><div>Both of those have other meanings and change =
during a connection, which means that the middlebox would have to keep =
state to track them. But you also said the middlebox doesn=E2=80=99t =
keep state.</div><blockquote type=3D"cite" class=3D""><div class=3D""><div=
 dir=3D"ltr" class=3D""><div class=3D""><ul class=3D""><li class=3D"">thus=
 removing the need to keep the state at the load balancer. This brings =
advantages in the case of DDoS attacks, load drain from servers, and =
traffic rerouting between LB.<br =
class=3D""></li></ul></div></div></div></blockquote></div><div>I see =
more clearly what you want, but no way to get there if the client =
browser/OS is unmodified.</div><div><br class=3D""></div><div>Joe</div><br=
 class=3D""></body></html>=

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From: Alexander Azimov <a.e.azimov@gmail.com>
Date: Thu, 21 May 2020 22:48:39 +0300
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To: Joseph Touch <touch@strayalpha.com>
Cc: Olivier Bonaventure <olivier.bonaventure@tessares.net>,  "tcpm@ietf.org Extensions" <tcpm@ietf.org>
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Subject: Re: [tcpm] TCP Connection ID
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--00000000000045e2bc05a62dcdcb
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Joseph,

Then you wouldn=E2=80=99t be talking about TCP option anymore (options cann=
ot be
> added en-route). You also wouldn=E2=80=99t be talking about MPTCP either =
- in
> general.
>
I believe that regular TCP options (not experimental) in general have
higher chances to be supported globally.

The server TCP sequence number and timestamps are the only things that most
> TCP clients already keep track of that the server has control ove).
>
> Both of those have other meanings and change during a connection, which
> means that the middlebox would have to keep state to track them. But you
> also said the middlebox doesn=E2=80=99t keep state.
>
That's the purpose of my proposal. To have a reverse function for
connection id that returns the real IP, so the load balancer doesn't need
to perform the connection tracking. Though you need to know the secret to
restore value(s), so it will not have negative security effects.


> I see more clearly what you want, but no way to get there if the client
> browser/OS is unmodified.
>
I've meant that OS/browser is out of administrative control. So you can't
rely on a proprietary software/agent on the client-side. But the open
standard may be successful especially taking into account that main OS
vendors are content providers at the same time, thus already trying to
solve the problem of LB in different ways.

The incremental deployment is crucial, but IMO it looks doable.

--=20
Best regards,
Alexander Azimov

--00000000000045e2bc05a62dcdcb
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<div dir=3D"ltr"><div>Joseph,</div><div><br></div><div class=3D"gmail_quote=
"><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;borde=
r-left:1px solid rgb(204,204,204);padding-left:1ex"><div style=3D"overflow-=
wrap: break-word;"><div><div>Then you wouldn=E2=80=99t be talking about TCP=
 option anymore (options cannot be added en-route). You also wouldn=E2=80=
=99t be talking about MPTCP either - in general.</div></div></div></blockqu=
ote><div>I believe that regular TCP options (not e<span style=3D"color:rgb(=
0,0,0);font-size:13.3333px">xperimental)</span>=C2=A0in general have higher=
 chances to be supported globally.<br></div><div><br></div><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid r=
gb(204,204,204);padding-left:1ex"><div style=3D"overflow-wrap: break-word;"=
><div><div>The server TCP sequence number and timestamps are the only thing=
s that most TCP clients already keep track of that the server has control o=
ve).=C2=A0</div><div><br></div><div>Both of those have other meanings and c=
hange during a connection, which means that the middlebox would have to kee=
p state to track them. But you also said the middlebox doesn=E2=80=99t keep=
 state.</div></div><div></div></div></blockquote><div>That&#39;s the purpos=
e of my proposal. To have a reverse function for connection id that returns=
 the real IP, so the load balancer doesn&#39;t need to perform the connecti=
on tracking. Though you need to know the secret to restore value(s), so it =
will not have negative security effects.=C2=A0</div><div>=C2=A0<br></div><b=
lockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-le=
ft:1px solid rgb(204,204,204);padding-left:1ex"><div style=3D"overflow-wrap=
: break-word;"><div>I see more clearly what you want, but no way to get the=
re if the client browser/OS is unmodified.</div></div></blockquote><div>I&#=
39;ve meant that OS/browser is out of administrative control. So you can&#3=
9;t rely on a proprietary=C2=A0software/agent on the client-side. But the o=
pen standard may be successful especially taking into account that main OS =
vendors are content providers at the same time,=C2=A0thus already trying to=
 solve the=C2=A0problem of LB in different ways.<br></div><div><br></div><d=
iv>The incremental deployment is crucial, but IMO it looks doable.</div></d=
iv><div><br></div>-- <br><div dir=3D"ltr" class=3D"gmail_signature"><div di=
r=3D"ltr">Best regards,<div>Alexander Azimov</div></div></div></div>

--00000000000045e2bc05a62dcdcb--


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> On May 21, 2020, at 12:48 PM, Alexander Azimov <a.e.azimov@gmail.com> =
wrote:
>=20
> I see more clearly what you want, but no way to get there if the =
client browser/OS is unmodified.
> I've meant that OS/browser is out of administrative control.

When (or if) you can formulate a clear, complete problem statement, let =
us know.

Joe=

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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dus-ascii"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; line-break: after-white-space;" class=3D""><br =
class=3D""><div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D"">On May 21, 2020, at 12:48 PM, Alexander Azimov &lt;<a =
href=3D"mailto:a.e.azimov@gmail.com" =
class=3D"">a.e.azimov@gmail.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><blockquote =
class=3D"gmail_quote" style=3D"caret-color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant-caps: =
normal; font-weight: normal; letter-spacing: normal; text-align: start; =
text-indent: 0px; text-transform: none; white-space: normal; =
word-spacing: 0px; -webkit-text-stroke-width: 0px; text-decoration: =
none; margin: 0px 0px 0px 0.8ex; border-left-width: 1px; =
border-left-style: solid; border-left-color: rgb(204, 204, 204); =
padding-left: 1ex;"><div style=3D"overflow-wrap: break-word;" =
class=3D""><div class=3D"">I see more clearly what you want, but no way =
to get there if the client browser/OS is =
unmodified.</div></div></blockquote><div style=3D"caret-color: rgb(0, 0, =
0); font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: normal; =
text-align: start; text-indent: 0px; text-transform: none; white-space: =
normal; word-spacing: 0px; -webkit-text-stroke-width: 0px; =
text-decoration: none;" class=3D"">I've meant that OS/browser is out of =
administrative control.</div></div></blockquote></div><br class=3D""><div =
class=3D"">When (or if) you can formulate a clear, complete problem =
statement, let us know.</div><div class=3D""><br class=3D""></div><div =
class=3D"">Joe</div></body></html>=

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A new meeting session request has just been submitted by Michael Scharf, a Chair of the tcpm working group.


---------------------------------------------------------
Working Group Name: TCP Maintenance and Minor Extensions
Area Name: Transport Area
Session Requester: Michael Scharf


Number of Sessions: 2
Length of Session(s):  100 Minutes, 50 Minutes
Number of Attendees: 60
Conflicts to Avoid: 

 Technology Overlap: tsvwg quic taps tsvarea iccrg panrg webtrans masque






People who must be present:
  Yoshifumi Nishida
  Michael Tuexen
  Martin Duke
  Michael Scharf

Resources Requested:

Special Requests:
  
---------------------------------------------------------



From nobody Mon May 25 17:56:08 2020
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Reviewer: Liang Xia
Review result: Ready

This document is good written, and no new security issues as it's a network
mechanism requirements BCP document.



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From: Olivier Bonaventure <olivier.bonaventure@tessares.net>
Date: Tue, 26 May 2020 08:51:40 +0200
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Subject: Re: [tcpm] TCP Connection ID
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Alexander,

> As far as I understand you are suggesting to use IP LSR/RH which SHOULD b=
e turned off by default (RFC7196). Besides, P and X, in your terms may belo=
ng to different address families. It seems that you still have confusion on=
 the purpose of my proposal:
>
> There is a browser, that sends a WEB-request. The request is sent to the =
virtual IP (VIP). The browser/OS is out of our control.
> There is a load balancer inside the datacenter, which receives an SYN pac=
ket and needs to send it to the server with real IP. Normally it is done vi=
a tunnel: IPIP or GRE. Both load balancer and server are under the same adm=
inistrative control.
> The idea is to introduce a connection id, that is generated by the server=
 at the time of SYN/ACK and kept at the client for the time of TCP session,=
 thus removing the need to keep the state at the load balancer. This brings=
 advantages in the case of DDoS attacks, load drain from servers, and traff=
ic rerouting between LB.

Another possibility would be to place the connection id in the network
layer, see doi:10.1109/TNET.2018.2799242

> Oliver:
> Thank you for the link! I was not aware of this paper. The option to adve=
rtise VIP looks interesting, but seem to have limitations:
>
> What if real server IP is IPv6 address while the client is working from I=
Pv4 only network?

If your VIP is IPv4 and the server behind the LB only support IPv6,
then you need the packets to pass through the LB anyway

> The attacker may learn the full list of backend servers, which can be use=
d for DDoS attacks. The dynamic IP allocation for connections may work as a=
 partial workaround, have you implemented it in the lab?

We did some tests but did not deploy that on busy servers

> The load balancer still needs to keep states of the connections. But it w=
asn't the problem that was addressed in the paper.
>

Indeed. The main objective of the paper was to show how to efficiently
support MPTCP on load balancers


Olivier

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--Apple-Mail=_F14C23DA-A79F-4E46-A63B-B086002E9A34
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> On May 25, 2020, at 11:51 PM, Olivier Bonaventure =
<olivier.bonaventure@tessares.net> wrote:
>=20
> Another possibility would be to place the connection id in the network
> layer, see doi:10.1109/TNET.2018.2799242

Not unless the network layer is establishing the connection, IMO.

In IP, connections are transport layer and above; they shouldn=E2=80=99t =
be buried elsewhere. (And no, a flow ID is not a connection ID, because =
one flow may represent multiple connections or one connection may use =
multiple flows).

Joe=

--Apple-Mail=_F14C23DA-A79F-4E46-A63B-B086002E9A34
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	charset=utf-8

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; line-break: after-white-space;" class=3D""><br =
class=3D""><div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D"">On May 25, 2020, at 11:51 PM, Olivier Bonaventure &lt;<a =
href=3D"mailto:olivier.bonaventure@tessares.net" =
class=3D"">olivier.bonaventure@tessares.net</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><span =
style=3D"caret-color: rgb(0, 0, 0); font-family: Helvetica; font-size: =
12px; font-style: normal; font-variant-caps: normal; font-weight: =
normal; letter-spacing: normal; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; text-decoration: none; float: none; =
display: inline !important;" class=3D"">Another possibility would be to =
place the connection id in the network</span><br style=3D"caret-color: =
rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: =
normal; font-variant-caps: normal; font-weight: normal; letter-spacing: =
normal; text-align: start; text-indent: 0px; text-transform: none; =
white-space: normal; word-spacing: 0px; -webkit-text-stroke-width: 0px; =
text-decoration: none;" class=3D""><span style=3D"caret-color: rgb(0, 0, =
0); font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: normal; =
text-align: start; text-indent: 0px; text-transform: none; white-space: =
normal; word-spacing: 0px; -webkit-text-stroke-width: 0px; =
text-decoration: none; float: none; display: inline !important;" =
class=3D"">layer, see =
doi:10.1109/TNET.2018.2799242</span></div></blockquote></div><br =
class=3D""><div class=3D"">Not unless the network layer is establishing =
the connection, IMO.</div><div class=3D""><br class=3D""></div><div =
class=3D"">In IP, connections are transport layer and above; they =
shouldn=E2=80=99t be buried elsewhere. (And no, a flow ID is not a =
connection ID, because one flow may represent multiple connections or =
one connection may use multiple flows).</div><div class=3D""><br =
class=3D""></div><div class=3D"">Joe</div></body></html>=

--Apple-Mail=_F14C23DA-A79F-4E46-A63B-B086002E9A34--


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From: tom petch <ietfa@btconnect.com>
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CC: "tcpm@ietf.org" <tcpm@ietf.org>, "draft-ietf-tcpm-rto-consider@ietf.org" <draft-ietf-tcpm-rto-consider@ietf.org>, "tcpm-chairs@ietf.org" <tcpm-chairs@ietf.org>
Thread-Topic: [tcpm] Last Call: <draft-ietf-tcpm-rto-consider-14.txt> (Requirements for Time-Based Loss Detection) to Best Current Practice
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Archived-At: <https://mailarchive.ietf.org/arch/msg/tcpm/VXxfC9sTEGJuMvf3Si5SzpnhCHk>
Subject: Re: [tcpm] Last Call: <draft-ietf-tcpm-rto-consider-14.txt> (Requirements for Time-Based Loss Detection) to Best Current Practice
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Looking at this takes me back.=0A=
=0A=
The format of the references is not one I have seen in a long time, likewis=
e the running footing, the front matter and the placing of the Abstract whi=
le Terminology is also from a bygone age.  Ah, I see that the I-D is a deca=
de old - all is explained but I do think that an update is called for.=0A=
=0A=
Tom Petch=0A=
=0A=
________________________________________=0A=
From: tcpm <tcpm-bounces@ietf.org> on behalf of The IESG <iesg-secretary@ie=
tf.org>=0A=
Sent: 18 May 2020 15:15=0A=
To: IETF-Announce=0A=
Cc: tcpm@ietf.org; draft-ietf-tcpm-rto-consider@ietf.org; tcpm-chairs@ietf.=
org=0A=
Subject: [tcpm] Last Call: <draft-ietf-tcpm-rto-consider-14.txt> (Requireme=
nts for Time-Based Loss Detection) to Best Current Practice=0A=
=0A=
=0A=
The IESG has received a request from the TCP Maintenance and Minor Extensio=
ns=0A=
WG (tcpm) to consider the following document: - 'Requirements for Time-Base=
d=0A=
Loss Detection'=0A=
  <draft-ietf-tcpm-rto-consider-14.txt> as Best Current Practice=0A=
=0A=
The IESG plans to make a decision in the next few weeks, and solicits final=
=0A=
comments on this action. Please send substantive comments to the=0A=
last-call@ietf.org mailing lists by 2020-06-01. Exceptionally, comments may=
=0A=
be sent to iesg@ietf.org instead. In either case, please retain the beginni=
ng=0A=
of the Subject line to allow automated sorting.=0A=
=0A=
Abstract=0A=
=0A=
=0A=
    Many protocols must detect packet loss for various reasons (e.g., to=0A=
    ensure reliability using retransmissions or to understand the level=0A=
    of congestion along a network path).  While many mechanisms have=0A=
    been designed to detect loss, protocols ultimately can only count on=0A=
    the passage of time without delivery confirmation to declare a=0A=
    packet "lost".  Each implementation of a time-based loss detection=0A=
    mechanism represents a balance between correctness and timeliness=0A=
    and therefore no implementation suits all situations.  This document=0A=
    provides high-level requirements for time-based loss detectors=0A=
    appropriate for general use in the Internet.  Within the=0A=
    requirements, implementations have latitude to define particulars=0A=
    that best address each situation.=0A=
=0A=
=0A=
=0A=
=0A=
The file can be obtained via=0A=
https://datatracker.ietf.org/doc/draft-ietf-tcpm-rto-consider/=0A=
=0A=
=0A=
=0A=
No IPR declarations have been submitted directly on this I-D.=0A=
=0A=
=0A=
=0A=
=0A=
=0A=
_______________________________________________=0A=
tcpm mailing list=0A=
tcpm@ietf.org=0A=
https://www.ietf.org/mailman/listinfo/tcpm=0A=


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Subject: Re: [tcpm] [Last-Call] Last Call: <draft-ietf-tcpm-rto-consider-14.txt> (Requirements for Time-Based Loss Detection) to Best Current Practice
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The content of this I-D puzzles me, a bit as if an academic paper had
been removed from its context and allowed to float away.

It talks of  'Within the standards process ...' which the rest of the
I-D implies is the IETF Standards process but it then recommends a
minimum RTO of one second which would be a catastrophe for some IETF
protocols which must detect failure in 10 milli-second so that repair is
effected within 50.

There are many references to safety so I looked at the Security
Considerations to see what the dangers are; none.  I am unclear what
these references to safety mean. What is the danger, what is the threat?

" When this time period passes without delivery confirmation the sender
concludes the data was lost in transit...." or the response was lost or
the response was witheld.  I hope that you never get an e-mail delivery
confirmation from me because while the protocol includes it, I will
always configure my MUA to suppress it; breach of security.

"Some protocols use keepalives, heartbeats or other messages to exchange
control information."  I struggle to think of an example of this.  On
the other hand, it came as a surprise to one protocol designer that when
he removed the plug on his connection, literally, nothing happened.
Layer 2 did not tell layer 3, layer 3 did not tell layer 4 and the
application was blissfully unaware, typical of the IETF protocol stack.
Ok failure detection is out of scope but I see heartbeat as primarily
for that purpose and have recommended it on a number of occasions.  (In
a similar vein, some IETF protocols are simplex so the best that can be
done is to add a heartbeat so that the urgent alarm, when it comes, has
a better chance of getting through). I find it hard to draw a hard 
distinction between failure detection and
loss detection.

I am ignorant of the technology encountered in server farms nowadays but
wonder if there is a degree of spoofing there, of ACK being generated at a
front end giving an illusion of a shorter FT, like a recursive resolver
but more so.

Overall the flavour I have is a document about TCP over the
ever-unreliable IP and lacking more general application.  TCP already 
has a rich literature of documents about timeout.  Not wrong, but
likely to be useful to the designer(s) of a new layer four protocol.

Tom Petch

>> ________________________________________
>> From: tcpm <tcpm-bounces@ietf.org> on behalf of The IESG
> <iesg-secretary@ietf.org>
>> Sent: 18 May 2020 15:15
>> To: IETF-Announce
>> Cc: tcpm@ietf.org; draft-ietf-tcpm-rto-consider@ietf.org;
> tcpm-chairs@ietf.org
>> Subject: [tcpm] Last Call: <draft-ietf-tcpm-rto-consider-14.txt>
> (Requirements for Time-Based Loss Detection) to Best Current Practice
>>
>>
>> The IESG has received a request from the TCP Maintenance and Minor
> Extensions
>> WG (tcpm) to consider the following document: - 'Requirements for
> Time-Based
>> Loss Detection'
>>    <draft-ietf-tcpm-rto-consider-14.txt> as Best Current Practice
>>
>> The IESG plans to make a decision in the next few weeks, and solicits
> final
>> comments on this action. Please send substantive comments to the
>> last-call@ietf.org mailing lists by 2020-06-01. Exceptionally,
> comments may
>> be sent to iesg@ietf.org instead. In either case, please retain the
> beginning
>> of the Subject line to allow automated sorting.
>>
>> Abstract
>>
>>
>>      Many protocols must detect packet loss for various reasons (e.g.,
> to
>>      ensure reliability using retransmissions or to understand the
> level
>>      of congestion along a network path).  While many mechanisms have
>>      been designed to detect loss, protocols ultimately can only count
> on
>>      the passage of time without delivery confirmation to declare a
>>      packet "lost".  Each implementation of a time-based loss detection
>>      mechanism represents a balance between correctness and timeliness
>>      and therefore no implementation suits all situations.  This
> document
>>      provides high-level requirements for time-based loss detectors
>>      appropriate for general use in the Internet.  Within the
>>      requirements, implementations have latitude to define particulars
>>      that best address each situation.
>>
>>
>>
>>
>> The file can be obtained via
>> https://datatracker.ietf.org/doc/draft-ietf-tcpm-rto-consider/
>>
>>
>>
>> No IPR declarations have been submitted directly on this I-D.
>>
>>
>>


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From: "Mark Allman" <mallman@icir.org>
To: "tom petch" <daedulus@btconnect.com>
Cc: last-call@ietf.org, tcpm@ietf.org, draft-ietf-tcpm-rto-consider@ietf.org,  tcpm-chairs@ietf.org
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Subject: Re: [tcpm] [Last-Call] Last Call: <draft-ietf-tcpm-rto-consider-14.txt> (Requirements for Time-Based Loss Detection) to Best Current Practice
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I just want to quickly correct one thing here ...

[the draft] then recommends a minimum RTO of one second

(a) It DOES NOT say this at all.  The document does not specify ANY
    minimum RTO.  The document does says this:

        In the absence of any knowledge about the latency of a path,
        the initial RTO MUST be conservatively set to no less than 1
        second.

    But, as soon as there is some notion (e.g., a RTT measurement)
    then there is knowledge and this no longer applies.  I.e., this
    is for the startup case where we are beginning transmission into
    an unknown network path.

(b) If this is too hefty for some application, that's fine.  Do what
    we do now and get consensus to use something different.  Again,
    the document says:

        The correct way to view this document is as the default
        case.

        [...]

        The requirements in this document may not be appropriate in
        all cases and, therefore, inconsistent deviations may be
        necessary (hence the "SHOULD" in the last bullet).  However,
        inconsistencies MUST be (a) explained and (b) gather
        consensus.

    In other words, the worst case is the current case.

I am not entirely sure I understand the remaining points in the
review as it's pretty rambling to me.  Certainly we use heartbeats
(in things like SCTP) and control packets (think TCP zero window
probes or keep-alives).  The document is very simply saying that if
these are used in some fashion we can also use them to measure FT
information.  That seems pretty reasonable to me and I can't figure
out what your complaint about that is.  Perhaps you could try again
so my pea brain can understand the complaint?

Thanks!

allman
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Cc: Last Call <last-call@ietf.org>, tcpm <tcpm@ietf.org>, draft-ietf-tcpm-rto-consider@ietf.org, tcpm-chairs@ietf.org
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Subject: Re: [tcpm] [Last-Call] Last Call: <draft-ietf-tcpm-rto-consider-14.txt> (Requirements for Time-Based Loss Detection) to Best Current Practice
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Mark

Yes, my comments ramble:-)  The underlying thread is that for me the
document lacks context, lacks assumptions, or at least ones explicit
enough for me to pick up.

When I got to
"Within the standards process"
I thought IEEE standards process? mmm no, but the absence of IETF is the
sort of detail that I find missing albeit trivial in this case.

I see an underlying assumption that the network is unreliable and that
the problem is packets discarded as a result of congestion.  A common
assumption but untrue in some cases.  If that is the basis of this I-D,
then I think that it needs spelling out up front.

Likewise, I see an assumption that the recipient will always generate an
acknowledgement when asked.  Again, untrue for some protocols so again an
assumption I would like to see stated.

Safety is mentioned many times; what is the danger?  I do not know and
would like to be told (congestive collapse?).

"protocols ultimately can only count on the passage of time without
delivery confirmation "
true for some networks, not true for others.  I do see a steady if small
increase in IETF protocols to detect and react to problems rather than
waiting for someone else - like TCP - to notice and do something so
again I am seeing an assumption.

Apologies for misquoting you on the figure of one second; yes you offer
a more nuanced guideline.  Yet again it makes me think this is about
e.g. web access with HTTP while other IETF protocols are dealing with
timescales an order or two of magnitude shorter.

So I am conscious that the I-D comes out of the TCPM WG  but it seems as
if it tries to be all encompassing; and I struggle to think of an
application of the I-D which is not TCP until some standards body 
develops a new
transport protocol; but even then it would only be applicable if the
underlying network had the assumed characteristics of IPv4.

Tom Petch

> ----- Original Message -----
> From: "Mark Allman" <mallman@icsi.berkeley.edu>
> Sent: Thursday, May 28, 2020 6:09 PM
>>
>> I just want to quickly correct one thing here ...
>>
>>> [the draft] then recommends a minimum RTO of one second
>>
>> (a) It DOES NOT say this at all.  The document does not specify ANY
>>      minimum RTO.  The document does says this:
>>
>>          In the absence of any knowledge about the latency of a path,
>>          the initial RTO MUST be conservatively set to no less than 1
>>          second.
>>
>>      But, as soon as there is some notion (e.g., a RTT measurement)
>>      then there is knowledge and this no longer applies.  I.e., this
>>      is for the startup case where we are beginning transmission into
>>      an unknown network path.
>>
>> (b) If this is too hefty for some application, that's fine.  Do what
>>      we do now and get consensus to use something different.  Again,
>>      the document says:
>>
>>          The correct way to view this document is as the default
>>          case.
>>
>>          [...]
>>
>>          The requirements in this document may not be appropriate in
>>          all cases and, therefore, inconsistent deviations may be
>>          necessary (hence the "SHOULD" in the last bullet).  However,
>>          inconsistencies MUST be (a) explained and (b) gather
>>          consensus.
>>
>>      In other words, the worst case is the current case.
>>
>> I am not entirely sure I understand the remaining points in the
>> review as it's pretty rambling to me.  Certainly we use heartbeats
>> (in things like SCTP) and control packets (think TCP zero window
>> probes or keep-alives).  The document is very simply saying that if
>> these are used in some fashion we can also use them to measure FT
>> information.  That seems pretty reasonable to me and I can't figure
>> out what your complaint about that is.  Perhaps you could try again
>> so my pea brain can understand the complaint?
>>
>> Thanks!
>>
>> allman
>>
>>


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From: "Mark Allman" <mallman@icir.org>
To: "tom petch" <daedulus@btconnect.com>
Cc: "Last Call" <last-call@ietf.org>, tcpm <tcpm@ietf.org>, draft-ietf-tcpm-rto-consider@ietf.org, tcpm-chairs@ietf.org
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Subject: Re: [tcpm] [Last-Call] Last Call: <draft-ietf-tcpm-rto-consider-14.txt> (Requirements for Time-Based Loss Detection) to Best Current Practice
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Hi Tom!

I am going to re-arrange slightly and deal with the last one first.

> So I am conscious that the I-D comes out of the TCPM WG but it
> seems as if it tries to be all encompassing;

Yes; we have found the guidelines are broadly applicable.

> and I struggle to think of an application of the I-D which is not
> TCP until some standards body develops a new transport protocol;

SCTP, QUIC, DNS, ... all manner of UDP applications.  Several
things:

  - In fact, RFC8085 (on UDP usage) includes many of the same things
    (cribbed from this I-D in some cases, actually!).  So, I think
    your framing this as applying TCP guidance too widely is
    incorrect.

  - The WG chairs were conscious of this and explicitly got input
    from folks working in these other (non-TCP) areas, as well.

  - The WGLC was run in both tcpm and tsvwg.

So I do think this is quite a bit more widely applicable than TCP.
It did start its life as a TCP document, but grew in breadth as
folks have said "hey, that applies [here]" (e.g., most recently when
Jana noted this was quite applicable within QUIC).  It has been
developed and reviewed as pretty general for a while now.  I don't
think we're somehow trying to sneak something through that is
broader than it either should be or has been reviewed to be.

I'll say something quick about the rest of the things, but basically
I am happy to add bits of clarification and fix sloppy writing.

> When I got to "Within the standards process"

Will add "IETF".

> I see an underlying assumption that the network is unreliable and
> that the problem is packets discarded as a result of congestion.
> A common assumption but untrue in some cases.

If your network is reliable then I presume you won't be reading
documents about loss detection.  But, I dunno.

We can add a sentence to the intro to explicitly say we are
discussing unreliable networks if that helps.

> Likewise, I see an assumption that the recipient will always
> generate an acknowledgement when asked.  Again, untrue for some
> protocols so again an assumption I would like to see stated.

I don't think we *assume* this is *always* the case.  This is from
the document:

    Various mechanisms are used to detect losses in a packet stream.
    Often we use continuous or periodic acknowledgments from the
    recipient to inform the sender's notion of which pieces of data are
    missing.

Without details, there is an acknowledgment there that there are
different ways to detect loss other than continuous ACKs.  And, we
state "often we use" ACKs, which I think runs counter to your notion
that we "assume" "always".  So, I am not sure what you want here.

> Safety is mentioned many times; what is the danger?  I do not know and
> would like to be told (congestive collapse?).

Safety in a congestion, not security, sense.  I can add a quick
clarification on this.

> "protocols ultimately can only count on the passage of time
> without delivery confirmation " true for some networks, not true
> for others.

I struggle to think of a case where it doesn't apply.  There are for
sure cases where we code the stream so that we get probabilistically
damn close to ensuring delivery without a continuous stream of ACKs
for specific packets.  But, without some feedback ("delivery
confirmation") to the sender how could we ever be sure the data
arrived?

> I do see a steady if small increase in IETF protocols to detect
> and react to problems rather than waiting for someone else - like
> TCP - to notice and do something so again I am seeing an
> assumption.

I can't follow this.  I don't mean to disagree with your
characterization of IETF protocols, but I don't understand how
you're applying that statement to the document.

allman

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From: "Mark Allman" <mallman@icsi.berkeley.edu>
To: "tom petch" <daedulus@btconnect.com>
Cc: last-call@ietf.org, tcpm@ietf.org, draft-ietf-tcpm-rto-consider@ietf.org,  tcpm-chairs@ietf.org
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Subject: Re: [tcpm] [Last-Call] Last Call: <draft-ietf-tcpm-rto-consider-14.txt> (Requirements for Time-Based Loss Detection) to Best Current Practice
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I just want to quickly correct one thing here ...

> [the draft] then recommends a minimum RTO of one second

(a) It DOES NOT say this at all.  The document does not specify ANY
    minimum RTO.  The document does says this:

        In the absence of any knowledge about the latency of a path,
        the initial RTO MUST be conservatively set to no less than 1
        second.

    But, as soon as there is some notion (e.g., a RTT measurement)
    then there is knowledge and this no longer applies.  I.e., this
    is for the startup case where we are beginning transmission into
    an unknown network path.

(b) If this is too hefty for some application, that's fine.  Do what
    we do now and get consensus to use something different.  Again,
    the document says:

        The correct way to view this document is as the default
        case.

        [...]

        The requirements in this document may not be appropriate in
        all cases and, therefore, inconsistent deviations may be
        necessary (hence the "SHOULD" in the last bullet).  However,
        inconsistencies MUST be (a) explained and (b) gather
        consensus.

    In other words, the worst case is the current case.

I am not entirely sure I understand the remaining points in the
review as it's pretty rambling to me.  Certainly we use heartbeats
(in things like SCTP) and control packets (think TCP zero window
probes or keep-alives).  The document is very simply saying that if
these are used in some fashion we can also use them to measure FT
information.  That seems pretty reasonable to me and I can't figure
out what your complaint about that is.  Perhaps you could try again
so my pea brain can understand the complaint?

Thanks!

allman

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From nobody Sat May 30 04:27:03 2020
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Reviewer: Stewart Bryant
Review result: Not Ready

I am the assigned Gen-ART reviewer for this draft. The General Area
Review Team (Gen-ART) reviews all IETF documents being processed
by the IESG for the IETF Chair.  Please treat these comments just
like any other last call comments.

For more information, please see the FAQ at

<https://trac.ietf.org/trac/gen/wiki/GenArtfaq>.

Document: draft-ietf-tcpm-rto-consider-14
Reviewer: Stewart Bryant
Review Date: 2020-05-30
IETF LC End Date: 2020-06-01
IESG Telechat date: Not scheduled for a telechat

Summary:

I have concerns that there is inadequate scoping information provided in this
proposed BCP.

The authors are clearly focused on some L4 cases and some applications. I am
concerned that this document, if published as is, will result in excessive work
and delay if reviewers insist that it applies to many network infrastructure
cases. I am also concerned that past
 experience may not be a good guide to some new applications, particularly those
from the deterministic stable.

Major issues:

As far as I can see this text only applies to exchanges between applications
and network support applications such as DNS. I.e. this is targeted at layer 4
and above. Given the religious nature of BCPs in the eyes of some reviewers,
and to prevent endless explanations by those that design routing protocols, OAM
and other lower layer sub-system I think there needs to a scoping text in block
capitals at the at the very start of the documnet.

=========

      - The requirements in this document may not be appropriate in all
        cases and, therefore, inconsistent deviations may be necessary
        (hence the "SHOULD" in the last bullet).  However,
        inconsistencies MUST be (a) explained and (b) gather consensus.

SB> That can be quite an onerous obligation  and provide scope for endless
argument when reviewers are not domain experts in the protocol being
designed.

=======

          While there are a bevy of uses for timers in protocols---from
          rate-based pacing to connection failure detection and
          beyond---these are outside the scope of this document.

SB> I am not sure what that means for the applicability of this document.

=========

    (1) As we note above, loss detection happens when a sender does not
        receive delivery confirmation within an some expected period of
        time.  In the absence of any knowledge about the latency of a
        path, the initial RTO MUST be conservatively set to no less than
        1 second.

SB> This issue may be addressed by the scoping text, but 1s is no use
when you are trying to detect sub 50ms of packet loss in the infrastructure.

=============

    (3) Each time the RTO is used to detect a loss, the value of the RTO
        MUST be exponentially backed off such that the next firing
        requires a longer interval.  The backoff SHOULD be removed after
        either (a) the subsequent successful transmission of
        non-retransmitted data, or (b) an RTO passes without detecting
        additional losses.  The former will generally be quicker.  The
        latter covers cases where loss is detected, but not repaired.

        A maximum value MAY be placed on the RTO.  The maximum RTO MUST
        NOT be less than 60 seconds (as specified in [RFC6298]).

        This ensures network safety.

SB> This does not work in OAM applications.

Minor issues:

 "By waiting long enough that we are unambiguously
  certain a packet has been lost we cannot repair losses in a timely
  manner and we risk prolonging network congestion."

I have a concern here that the emphasis is on classical operation. We are
beginning to see application to run over the network where the timely delivery
of a packet is critical for correct operation of even SoL. As a BCP the text
needs to recognise that the scope and purpose of IP is changing and that
classical learning and rules derived from them may not apply.

Also if not ruled out of scope earlier we need to be clear at this point that
things like BFD have different considerations.

==========

      "- This document does not update or obsolete any existing RFC.
        These previous specifications---while generally consistent with
        the requirements in this document---reflect community consensus
        and this document does not change that consensus."

I think it needs to be clear that adherence to this RFC is not required for
minor updates and extensions to existing RFCs. Having seen minor routing
extension held up by security concerns related to underlying protocols rather
than the extension itself there is a lot of sensitivity on this point in some
quarters of the IETF.

========

It might be useful to make it clear that there are some applications that would
prefer no data to late data.

Nits/editorial comments:

The terminology section confuses ID-nits - I think it should be a section in
its own right later in the document.

The following nits issues need looking at

  == Missing Reference: 'RFC5681' is mentioned on line 377, but not defined

  == Unused Reference: 'RFC3940' is defined on line 515, but no explicit
     reference was found in the text

  == Unused Reference: 'RFC4340' is defined on line 519, but no explicit
     reference was found in the text

  == Unused Reference: 'RFC6582' is defined on line 540, but no explicit
     reference was found in the text



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Cc: Last Call <last-call@ietf.org>, tcpm <tcpm@ietf.org>, draft-ietf-tcpm-rto-consider@ietf.org, tcpm-chairs@ietf.org
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Subject: Re: [tcpm] [Last-Call] Last Call: <draft-ietf-tcpm-rto-consider-14.txt> (Requirements for Time-Based Loss Detection) to Best Current Practice
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> ----- Original Message -----
> From: "Mark Allman" <mallman@icir.org>
> Sent: Friday, May 29, 2020 1:45 PM
>
>> Hi Tom!
>>
>> I am going to re-arrange slightly and deal with the last one first.
>>
>>> So I am conscious that the I-D comes out of the TCPM WG but it
>>> seems as if it tries to be all encompassing;
>>
>> Yes; we have found the guidelines are broadly applicable.
>>
>>> and I struggle to think of an application of the I-D which is not
>>> TCP until some standards body develops a new transport protocol;
>>
>> SCTP, QUIC, DNS, ... all manner of UDP applications.  Several
>> things:
>>
>>    - In fact, RFC8085 (on UDP usage) includes many of the same things
>>      (cribbed from this I-D in some cases, actually!).  So, I think
>>      your framing this as applying TCP guidance too widely is
>>      incorrect.
>>
>>    - The WG chairs were conscious of this and explicitly got input
>>      from folks working in these other (non-TCP) areas, as well.
>>
>>    - The WGLC was run in both tcpm and tsvwg.
>>
>> So I do think this is quite a bit more widely applicable than TCP.
>> It did start its life as a TCP document, but grew in breadth as
>> folks have said "hey, that applies [here]" (e.g., most recently when
>> Jana noted this was quite applicable within QUIC).  It has been
>> developed and reviewed as pretty general for a while now.  I don't
>> think we're somehow trying to sneak something through that is
>> broader than it either should be or has been reviewed to be.

  Mark
  This is good to hear but I wish that more detail of this was in the I-D.
  I have followed a number of Transport AD review of xxxx-over-UDP and it
  was always about the need for flow control but never with the sort of
  techniques that TCP uses, usually just a requirement for a statement on
  rate limiting by the sender so if there are xxxx-over-UDP which measure
  RTO and lost packets and act accordingly, then that would be excellent
  to hear about.  I know of none (and yes I am familiar with RFC8085) so
  including non-TCP examples would counter the impression I have got that
  this is TCP!   SCTP I have always seen as son-of-TCP and so not much
  different in this context.  QUIC I have yet to encounter.  As it stands,
  the example of DNS caching server stands out to me as an example of
  broader application, of the I-D in general; I would value more such
  concrete examples. And below ...


>> I'll say something quick about the rest of the things, but basically
>> I am happy to add bits of clarification and fix sloppy writing.
>>
>>> When I got to "Within the standards process"
>>
>> Will add "IETF".
>>
>>> I see an underlying assumption that the network is unreliable and
>>> that the problem is packets discarded as a result of congestion.
>>> A common assumption but untrue in some cases.
>>
>> If your network is reliable then I presume you won't be reading
>> documents about loss detection.  But, I dunno.

  No, I am being unclear.  It is the 'packets are lost due to congestion
  and that is THE problem' that I see as the unstated assumption for this
  and many IETF documents.  Is it correct? Often, perhaps, but I struggle
  to recall anyone testing the hypothesis and there is growing use of IETF
  protocols over different kinds of networks where the problem is e.g.
  interference and packet corruption so the recommended techniques will
  give the user a worse experience - the corrupt packet needs
  re-transmitting as soon as the interference is over and backing off the
  flow rate is bad news for the user.  And below...

>> We can add a sentence to the intro to explicitly say we are
>> discussing unreliable networks if that helps.
>>
>>> Likewise, I see an assumption that the recipient will always
>>> generate an acknowledgement when asked.  Again, untrue for some
>>> protocols so again an assumption I would like to see stated.
>>
>> I don't think we *assume* this is *always* the case.  This is from
>> the document:
>>
>>      Various mechanisms are used to detect losses in a packet stream.
>>      Often we use continuous or periodic acknowledgments from the
>>      recipient to inform the sender's notion of which pieces of data
> are
>>      missing.
>>
>> Without details, there is an acknowledgment there that there are
>> different ways to detect loss other than continuous ACKs.  And, we
>> state "often we use" ACKs, which I think runs counter to your notion
>> that we "assume" "always".  So, I am not sure what you want here.
>>
>>> Safety is mentioned many times; what is the danger?  I do not know
> and
>>> would like to be told (congestive collapse?).
>>
>> Safety in a congestion, not security, sense.  I can add a quick
>> clarification on this.

  Yes please, this is an example of what I generically refer to as a
  lack of context, that for me safety is usually something quite
  different in an I-D.  And below ..

>> "protocols ultimately can only count on the passage of time
>>> without delivery confirmation " true for some networks, not true
>>> for others.
>>
>> I struggle to think of a case where it doesn't apply.  There are for
>> sure cases where we code the stream so that we get probabilistically
>> damn close to ensuring delivery without a continuous stream of ACKs
>> for specific packets.  But, without some feedback ("delivery
>> confirmation") to the sender how could we ever be sure the data
>> arrived?
>>
>>> I do see a steady if small increase in IETF protocols to detect
>>> and react to problems rather than waiting for someone else - like
>>> TCP - to notice and do something so again I am seeing an
>>> assumption.
>>
>> I can't follow this.  I don't mean to disagree with your
>> characterization of IETF protocols, but I don't understand how
>> you're applying that statement to the document.

It comes back to the assumption of a unreliable network, unreliable for
any reason, not just congestion.  An alternative approach is to make the
network reliable and this I see as a strong interest of WG such as TEAS,
CCAMP, MPLS and such like.  A statement up front about the assumption of
unreliability would address this.

Tom Petch
>
>> allman
>>
>>


From nobody Sat May 30 11:06:31 2020
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From: "Scharf, Michael" <Michael.Scharf@hs-esslingen.de>
To: "tcpm@ietf.org" <tcpm@ietf.org>
CC: tcpm-chairs <tcpm-chairs@ietf.org>
Thread-Topic: WGLC for draft-ietf-tcpm-2140bis
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Subject: [tcpm] WGLC for draft-ietf-tcpm-2140bis
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Hi all,

The document "TCP Control Block Interdependence" (draft-ietf-tcpm-2140bis) =
is out there for a quite some time already. Given that there has been a bit=
 less activity on the list after the interim, let's use this opportunity to=
 finish one of our milestones...
=20
This e-mail starts a WGLG for the document draft-ietf-tcpm-2140bis. The WGL=
C will run until ***June 14***.

The intended status is an informational RFC. The current version of the doc=
ument can be found at:

https://tools.ietf.org/html/draft-ietf-tcpm-2140bis-05

Please review the latest version and please let us know any comments or sug=
gestions. TCPM needs reviews to ensure that the content is ready to move fo=
rward. Feedback supporting publication ("+1") is also very welcome.

Thanks

Michael, on behalf of the TCPM chairs


From nobody Sun May 31 11:54:19 2020
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Date: Sun, 31 May 2020 11:54:01 -0700
From: Benjamin Kaduk <kaduk@mit.edu>
To: Mark Allman <mallman@icsi.berkeley.edu>
Cc: tom petch <daedulus@btconnect.com>, last-call@ietf.org, tcpm@ietf.org, draft-ietf-tcpm-rto-consider@ietf.org, tcpm-chairs@ietf.org
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Subject: Re: [tcpm] [Last-Call] Last Call: <draft-ietf-tcpm-rto-consider-14.txt> (Requirements for Time-Based Loss Detection) to Best Current Practice
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On Thu, May 28, 2020 at 01:09:41PM -0400, Mark Allman wrote:
> 
> I am not entirely sure I understand the remaining points in the
> review as it's pretty rambling to me.  Certainly we use heartbeats
> (in things like SCTP) and control packets (think TCP zero window
> probes or keep-alives).  The document is very simply saying that if
> these are used in some fashion we can also use them to measure FT
> information.  That seems pretty reasonable to me and I can't figure

There was a quote from the document: "Some protocols use keepalives,
heartbeats or other messages to exchange control information."
A naive reading of this text is that it is discussing ways to exchange
control information, and giving specific examples of keepalives and
heartbeats as messages used to exchange control information (while admitting
the existence of other such messages).  This is as surprising to me as it
is to Tom -- keepalives and heartbeats do not, in my experience, contain
control information!

Your follow-up discussion suggests that the intent is to say that "some
protocols use regular/periodically/frequently exchanged messages, such as
for keepalives, heartbeats, or control-information exchange" and to
leverage that traffic for FT purposes.  I suggest that this be clarified in
the text of the document.

Thanks,

Ben

