Re: [tcpm] [Lwip] WGLC review ofdraft-ietf-lwig-tcp-constrained-node-networks

"Carles Gomez Montenegro" <> Sun, 17 November 2019 09:49 UTC

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Date: Sun, 17 Nov 2019 10:49:33 +0100
From: Carles Gomez Montenegro <>
To: Markku Kojo <>
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Subject: Re: [tcpm] [Lwip] WGLC review ofdraft-ietf-lwig-tcp-constrained-node-networks
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Dear Markku,

First of all, our apologies for the delay in this response!

We would like to thank you once again for your thorough review and update

We recently published -09, with the aim of addressing the remaining points.

Please find below our inline responses:

> Dear Carles, all,
> it took longer to find time to pass through the draft than I thought a
> week or so ago. My apologies and thanks for your patience.
> It looks fine to me with a few exceptions that still seem to need some
> work, I think.
> Please see inline.
> Cheers,
> /Markku
> On Wed, 5 Jun 2019, Carles Gomez Montenegro wrote:
>> Dear Markku,
>> Thank you very much for your comprehensive and detailed review of the
>> draft. Your constructive comments have been very useful to address
>> issues
>> and improve the quality of the document. Our updates can be found in
>> revision -08.
>> Please find below our inline responses to your comments.
>>> Hi all,
>>> I have reviewed the -07 version of this draft for the WGLC.
>>> The draft is very useful for many developpers using or considering of
>>> using TCP in CNN scenarious. It has improved a lot from the previous
>>> versions but there are still a number of issues worth addressing.
>>> See comments below.
>>> Best regards,
>>> /Markku
>>> Sec 4.1.
>>> Title: Path properties
>>> - Would it be better to use "Addressing path properties" or something
>>> similar as TCP cannot (much) affect the properties?
>> Sounds good!
>>> Sec. 4.1.1.
>>> If I understand it correctly, the discussion in this section is
>>> intended
>>> to be all about avoiding Path MTU discovery when running TCP oevr IPv6?
>>> Therefore, "Avoiding Path MTU Discovery in IPv6" would probably be
>>> a better title?
>>> From our point of view, the section is actually about setting the MSS
>>> to a
>> suitable value, which then may help avoiding the need to support Path
>> MTU
>> Discovery, but also the need to perform IP-layer fragmentation at the
>> source. We have explicitly added the latter in -08.
> This seems fine now, except I noticed another problem that I didn't spot
> last time. The text says a few of times something about
> limiting/setting MTU, where you actually want to advice limiting IP
> datagram size (by setting the MSS), just like you say above. But the text
> reads, for example:
>   ... it may be desirable to limit the MTU to 1280 bytes ...
> which I believe should read somethig like:
>   ... it may be desirable to limit the IP datagram size to 1280 bytes ...
> MTU is the property of the network link at hand, not controllable same way
> as the IP datagram size which can be limited by setting the TCP MSS. And,
> if MTU was settable, avoiding framentation would call for setting it to as
> large value as possible, not limiting it.


>>> In addition, to my understanding TCP implementations typically address
>>> the presence of TCP options such that they eat the necessary space for
>>> TCP options from the payload, not by increasing the IP datagram size
>>> if TCP options are present. For example, if SMSS is set to, let's say
>>> 1460 octets, and a TCP sender adds a TCP timestamp option (12 bytes) it
>>> will send only 1448 bytes of payload in a TCP segment?
>>> What the draft now says in this respect is on the safe side, but it
>>> might be overcautious. I don't remember any RFC saying how SMSS and
>>> adding options to a TCP segment are related. Maybe someone of the TCP
>>> implementors may shed more light to this how?
>> We have tried to address your two comments above. On this matter, we had
>> received feedback that this measure (advertising an MSS smaller than
>> 1220
>> bytes) would be safe, but we have tried to reflect that this might not
>> be
>> necessary, and even overcautious.
> Seems fine, thanks.
>>> The second but last para discussing IPv4 in this context is very
>>> confusing. In particular, the 2nd sentence
>>>   "In IPv4, the MTU is 576 bytes."
>>> is simply incorrect. In IPv4 the requirement is that any host must be
>>> able to accept datagrams of up to 576 octets, but there is no upper
>>> limit of 576 for IPv4 MTU!
>> Indeed. The 2nd sentence missed the word ?minimum? before ?MTU?, and we
>> have made several updates to the paragraph.
> Maybe I was unclear in my comment. There is no upper nor lower limit of
> 576 for IPv4 MTU. Definitely MTU can be less than 576. Moreover,
> IPv4 requires that every node must be able to forward an IP packet of 68
> bytes without fragmentation, but even this is not exactly a minimum MTU
> requiremt. It is because IP is not necessarily able to fragment packets
> smaller than 68 bytes. In other words, there is no similar requirement
> for link layers to support a certain minimum MTU with IPv4 as there is
> for IPv6 that requires link layers to support an MTU of at least 1280
> bytes.
> Therefore, I think it is hard to give similar advise for IPv4 as the
> draft gives for IPv6.

Agreed. We removed the intended guidance text for IPv4.

>>> Sec. 4.1.2.
>>>   "In such traffic patterns, it is more difficult to
>>>    detect packet loss without retransmission timeouts ..."
>>> ->
>>>   "In such traffic patterns, it is more difficult and often
>>>    impossible to detect packet loss without retransmission timeouts
>>>    unless ECN is enabled ..."
>> Done.
>>>   "When the congestion window of a TCP sender has a
>>>   size of one segment, the TCP sender resets the retransmit timer, and
>>>   the sender will only be able to send a new packet when the retransmit
>>>   timer expires [RFC3168]. Effectively, the TCP sender reduces at that
>>>   moment its sending rate from 1 segment per Round Trip Time (RTT) to 1
>>>   segment per RTO, which can result in a very low throughput.  In
>>>   addition to better throughput, ECN can also help reducing latency and
>>>   ECN can also help reducing latency and	jitter."
>>> This text is somewhat inaccurate in terms of how ECN works if only a
>>> single segment is in flight (cwnd = 1 MSS) and confusing when it says
>>> "which can result in a very low throughput". The latter is kind a true,
>>> but also necessary to avoid congestion and, after all, it may result in
>>> higher throughput compared to the case where ECN is not used and
>>> retransmissions are needed.
>>> I'd rephrase the above to something along the lines:
>>>   "When the congestion window of a TCP sender has a
>>>   size of one segment and a TCP ACK with an ECN signal (ECE flag)
>>> arrives
>>>   at the TCP sender, the TCP sender resets the retransmit timer, and
>>>   the sender will only be able to send a new packet when the retransmit
>>>   timer expires. Effectively, the TCP sender reduces at that
>>>   moment its sending rate from 1 segment per Round Trip Time (RTT) to 1
>>>   segment per RTO and reduces the sending rate further on each ECN
>>> signal
>>>   received in subsequent TCP ACKs. Otherwise, if an ECN signal is not
>>>   present in a subsequent TCP ACK the TCP sender resumes the normal
>>>   ack-clocked transmission of segments [RFC 3168].
>> Thank you very much for the proposed text. The draft has been updated
>> accordingly.
>>> It might be also good to rearrange the text in the second and third
>>> paragraphs to disscuss the effect of retrasmission and timeouts more
>>> coherently. I may suggest text for this.
>> We will welcome any suggestion you may have in this regard.
> I'm fine with the text as is. It seems to me after all that rearranging
> the text is not crucial. A reader should get the intended message,
> although rewrite might help the reader.
>>> Sec 4.2.
>>>   "This section discusses TCP stacks that focus on transferring a
>>> single
>>>    MSS."
>>> Maybe better:
>>>   "This section discusses TCP stacks that allow transferring only a
>>> single
>>>    MSS at a time."
>> Done.
> Sorry, but now I am a bit confused here with the new naming. Earlier this
> was titled "single MSS-windows and buffers" and "single-MSS stacks" and
> the text indicated that such TCP stack reduces its maximum advertized
> window to one MSS and may only hold one MSS of data in its send and
> receive buffers. Further, the text indicated and still indicates that
> such a stack may, however, have several TCP segments in flight as long as
> they carry at most one MSS worth of data in total. So, my comments were
> based on that assumption.
> Now the new way of naming them as "single-segment stacks" hints that these
> are even more restricted such that the stack has e.g., limited capability
> for bookkeeping and may only hold a single TCP segment in its one MSS send
> or receive buffer at a time regardless of the payload size of the
> segment and thereby possibly have only(?) a single data segment in
> flight in each direction at any point of time. That is, once such a TCP
> sender has received some data from the application, it does not accept
> more data from the application until a cumulative Ack has arrived and
> released the send buffer for the next piece of app data. And such a TCP
> receiver advertises zero window once it receives a data segment carrying
> any amount of data (e.g., only a single byte).
> Or, there obviously can be different variants of these two behaviors
> above, for example uIP that releases the send buffer after transmitting
> the segment and requires the app to provide the same data for possible
> retransmission. Or, a receiving TCP always delivers the segment
> immediately to the application (and the receive buffer is possibly
> provided by the application).
> So, what is exactly meant by a "single-segment stack"? Both variants
> above are possible, or various flavors of them, and possibly
> both types of stacks exist at least for the TCP send buffer
> implementation? Single-segment reveive buffer implementations that keep
> the segment in the TCP reveive buffer until an application receives it
> would have hard time to work decently with a regular TCP stack in certain
> fully legitimate scenarios, though. Therefore, they possibly are
> non-existing?
> The definition affects the text in many parts of the draft as well my
> comments. For now, I continue with my original interpretation and assume
> the definition of "single-MSS buffers" hold. Otherwise, the split hack
> would not be possible at all if the TCP sender is not able to compose
> more than one segment at a time and  would result in really bad
> performance with a receiver stack that may accept only a single segment
> at time, right?

We realized that "single-MSS" is a better term, as in fact it reflects
what is supported by the most constrained major TCP implementations for
IoT scenarios.

>>> Sec 4.2.1.
>>> Last sentence:
>>>   "For this use of CoAP, a maximum TCP window
>>>    of one MSS will be sufficient."
>>> This is not necessarily true. If both TCP stacks involved allow a TCP
>>> window larger than 1 MSS and a CoAP request or response larger than one
>>> MSS is in use, it can be delivered more efficiently than in case where
>>> max TCP window is one MSS.
> The above part of the comment is addressed in the new text, thank you.
>>> Furthermore, this may also not be the case if a CoAP over TCP
>>> application uses short-lived TCP connections. Why? Because then the
>>> mandatory CSM message that each CoAP endpoint sends after the 3WHS may
>>> introduce an additional RTT as it cannot necessarily be sent during the
>>> same RTT with the first CoAP request/response. Of course, if the CSM
>>> message and the first CoAP request/response message fit into a single
>>> MSS and the TCP Nagle algorithm is disabled, such a single MSS window
>>> does not result in an additional RTT.
> .. but this not?
> It is true that a CoAP endpoint is not allowed to send a new
> application message until a response to the previous one has been
> received. However, sending a CMS message is a mandatory artefact
> of CoAP over TCP and it is sent as the first msg over the TCP
> connection and in both directions. This CMS message eats a part of the one
> MSS window. The CoAP over TCP spec allows sending the first application
> message immediately after the CMS has been transmitted. However, a TCP
> window of one MSS may prevent TCP from transmitting the app msg until
> the TCP Ack for the TCP segment carrying CSM has arrived (depends on the
> size of these msgs and the MSS). Therefore, the performance may degrade
> notably for apps running on a single-segment or single-MSS stack at either
> end and using short-lived TCP connections as they may need to first wait
> for the TCP Ack for the CSM and only then can the first application
> message be transmitted. The need for waiting for the TCP Ack of CSM
> arises always if Nagle is enabled, and if Nagle is disabled it may arise
> due to the lack of buffer space to hold both CSM and the first data
> segment carrying the first application message.
> Not quite sure how to address this nicely, since it is a small
> CoAP over TCP detail that mostly affects short-lived TCP connections but
> depends on the size of the first msgs and MSS (as well as Nagle). That
> is, it has an effect under certain conditions, not necessarily always. And
> it may have a significant effect or only a negligible effect.

We added text intended to point out the above.

>> We have updated the sentence accordingly.
>>> Sec. 4.2.2.
>>>   "A TCP implementation for a constrained device that uses a single-MSS
>>>   TCP receive or transmit window size may not benefit from supporting
>>>   the following TCP options: Window scale [RFC7323], TCP Timestamps
>>>   [RFC7323], Selective Acknowledgments (SACK) and SACK-Permitted
>>>   [RFC2018]."
>>> It may be useful to mention that a TCP sender can benefit from
>>> Timestamps in detecting spurious RTOs that are quite likely to occur
>>> in CNN scenarios.
>> Done.
>>> Sec. 4.2.3.
>>> 2nd para:
>>>   "A device that advertises a single-MSS receive window should avoid
>>> use
>>>   of Delayed ACKs in order to avoid contributing unnecessary delay (of
>>>   up to 500 ms) to the RTT [RFC5681], which limits the throughput and
>>>   can increase the data delivery time."
>>> This should not appear as a generic recommendation as it is not
>>> correct for some typical usage scenarios such as request-response
>>> traffic where the node with a single-MSS receive window is the server
>>> sending the responses. With delayed ACKs it can biggyback the TCP ACK
>>> with the response if the response is sent before the delayed ACK timer
>>> expires, thus avoiding unnecessary pure TCP ACKs.
>>> So, here, like in Sec 4.3.2., it is important to indicate that it
>>> depends
>>> on the communication pattern whether delayed ACKs are useful or
>>> harmful.
>> We have modified this text accordingly.
>>> 3rd para:
>>>   "A device that can send at most one MSS of data is significantly
>>>   affected if the receiver uses Delayed ACKs, e.g., if a TCP server or
>>>   receiver is outside the CNN."
>>> Again, this does not hold in all cases. E.g., if the server is outside
>>> of the CNN and request-response communication is used.
>> We have modified and reorganized the text accordingly.
>>> The "split hack" is not advisable workaround. First for the reason
>>> stated in the end of the para, but more importantly because it simply
>>> does not necessarily even work; a TCP receiver is requited to
>>> acknowledge every second full-sized segment, but not two consecutive
>>> small segments.
>> Agreed. This comment has been incorporated into the text. We discuss the
>> ?split hack?, but we do not recommend it.
> Fine. May I suggest a minor additional tweak:
> "A standard compliant TCP receiver will acknowledge the second MSS of
> data, ..."
>   ->
> "A standard compliant TCP receiver may immediately acknowledge the second
> MSS of data, ..."

Done, thanks!

>>> 4th para:
>>>   "Similar issues happen when a sender uses the Nagle algorithm.
>>>   Disabling the algorithm will not have impact if the sender can only
>>>   handle stop-and-wait operation."
>>> Actually it does have an impact in some specific usage scenarios, e.g.,
>>> with CoAP over TCP disabling the Nagle algorithm allows sending the
>>> mandatory CSM message and the first CoAP  msg (request) and possibly
>>> also CSM and the first response without unnecessarily waiting for a TCP
>>> ACK of the CSM msg. This is of particular impact if short-lived TCP
>>> connections are in use with CoAP over TCP.
>> The text above refers to stop-and-wait operation. In your example above,
>> were you considering stop-and-wait operation?
> Yes, I was considering stop-and-wait operation at the application level in
> the context of CoAP over TCP which happens to be a weird thing. Even
> though a single CoAP request msg may be issued at a time, the
> transmission of the CSM message in the beginning of the TCP connection
> breaks the stop-and-wait operation.
> But, I'm not anymore sure how to interpret the text. Where does "similar
> issues" now refer to?
> It may be useful to consider moving the discussion on Nagle to Sec 4.2.1
> in the context of CoAP over TCP where it has an impact and not discuss it
> here at all, because it is not that necessary to say here that "Nagle is
> no-op"?

We tried to clarify the text in -09.

>>> Sec. 4.2.4.
>>> RTO is not estimated but calculated using estimated RTT and deviation
>>> from it. That is, modify:
>>> RTO estimation -> RTO calculation
>> Done.
>>> 2nd para:
>>>   "[RFC6298] describes the standard TCP RTO algorithm."
>>> You may delete this sentence and cite RFC 6298 in the first sentence
>>> of the Sec 4.2.4 where the algorithm is first mentioned.
>> Done.
>>> 3rd para:
>>>   "As an example, an adaptive RTO algorithm for CoAP over UDP has been
>>>   defined [I-D.ietf-core-cocoa] that has been found to perform  well in
>>>   CNN scenarios [Commag]."
>>> Maybe not a good idea to cite the current version of CoCoA RTO
>>> algorithm (v3) that have been found also to have detrimental behavior?
>> Done.
> Maybe I should not say this but possibly could cite here more than one
> alternative that have been experimentally shown to perform well ;)

Done ;)

>>> Sec. 4.3.1.
>>>   "Assuming that Delayed ACKs are used by the receiver, the
>>>   mentioned algorithms work efficiently for window sizes of at least 5
>>>   MSS: If in a given TCP transmission of segments 1, 2, 3, 4, 5, and 6
>>>   the segment 2 gets lost, the sender should get an ACK for segment 1
>>>   when 3 arrives and duplicate acknowledgements when 4, 5, and 6
>>>   arrive. It will retransmit segment 2 when the third duplicate ACK
>>>   arrives. In order to have segment 2, 3, 4, 5, and 6 sent, the window
>>>   has to be at least 5 MSS. With an MSS of 1220 byte, a buffer of the
>>>   size of 5 MSS would require 6100 bytes."
>>> The requirement for the window size to be of at least 5 segments does
>>> not hold if Limited Transmit is in use.
>> We have updated the text (including that now we mention ?up to 5 MSS?),
>> along with updates in the Limited Transmit paragraph.
>>> Also, the requirement of at least 5 segments is valid only if the ACK
>>> for the segment 1 was held by the DelAck timer, i.e., the requirement
>>> holds approx. with 50% probability. That is, if the segment 1 got
>>> acknowledged (because there was also a segment before segment 1 and
>>> that was held by DelAck timer), only a window size of 4 MSS is needed.
>> We have modified the text accordingly.
> I have similar comment as Ilpo here. It's worth clarifying the example
> by removing the potential interpretation that segments 1-6 can be in
> flight as a starting point with a cwnd of 5 MSS (assuming they all are
> full-sized segments).

We believe that we corrected the flawed text.

> Note also that in some traffic scenarios where Nagle is disabled and a TCP
> sender does not send MSS-sized segments but smaller segments, it is quite
> possible to levarage FR/FR even with window sizes smaller than 5 MSS
> (actually even with a window size of one MSS 3 dupacks and FR/FR may be
> possible).


>>>   "For bulk data transfers further TCP improvements may also be useful,
>>>   such as limited transmit [RFC3042].
>>> Limited Transmit is not useful only for bulk data transfers but for any
>>> transfer that has more than one segment in flight. Small transfers tend
>>> to benefit more, because they are more likely to not receive enough
>>> dupacks.
>> We have modified the paragraph accordingly.
> Fine, but it might be useful to start the 2nd para that discusses
> Limited Transmit by noting that the example in the 1st para assumed
> that limited Transmit is not in use.
> In addition, actually a cwnd allowing 2 segments in flight would be
> enough to trigger sending segments 1-5. Only difference to cwnd of 3
> segments is that one needs to wait for the DelAck timer to expire for
> segment 1.

Agreed, and done!

>>> Sec.
>>>   "... a sender (having previously sent the SACK-Permitted
>>>   option) can avoid performing unnecessary retransmissions, saving
>>>   energy and bandwidth, as well as reducing latency."
>>> It might be worth mentioning also that SACK often allows for faster
>>> loss recovery when there is more than one lost segment in a window of
>>> data (i.e., recovery with less RTTs).
>> Done.
> Fine. I'd suggest editing the sentence slightly because it is not
> guaranteed that with SACK recovery takes less RTTs:
>   "...since with SACK recovery requires less RTTs."
>   -->
>   "...since with SACK recovery may complete with less RTTs.


>>> Sec. 4.3.2.
>>> Disabling delayed ACKs on a client for infrequent request-response
>>> traffic with small messages might be advisable, too. It would allow
>>> an immediate ACK for the data segment carrying the response.
>>> This comment holds for Sec. 4.2.3 as well.
>> Added, thanks.
> The text here would benefit from more accurate expression of at which end
> the delayed Acks should be turned on and where turned off (e.g.,  "sender"
> does not specify the end point as there is both app. level (data) sender
> as well as TCP sender at each end for request-response type of traffic.
> Maybe something like:
>   For request-response traffic, enabling Delayed ACKs is recommended,
>   in order to allow combining a response with the ACK into
>   a single segment, thus increasing efficiency.  In this case,
>   disabling Delayed ACKs at the sender allows an immediate
>   ACK for the data segment carrying the response.
>   -->
>   For request-response traffic, enabling Delayed ACKs is recommended at
>   the server end, in order to allow combining a response with the ACK into
>   a single  segment, thus increasing efficiency.  In addition, if
>   a client issues requests infrequently, disabling Delayed
>   ACKs at the client allows an immediate ACK for the data segment
>   carrying the response.


Once again, thanks a lot for all your comments and proposed updates!

This is very much appreciated.



>>> Sec. 5.3.
>>>   "A mean TCP NAT binding timeout of 386
>>>    minutes has been reported, while in some cases, inactivity timeouts
>>>    are in the order of a few minutes [HomeGateway].
>>> Reporting just the mean TCP NAT binding timeout from [HomeGateway] does
>>> not give a correct view of the results in this study, because the
>>> meaasured timeouts were highly variable and some devices had a very
>>> long
>>> timeout (or no timeout at all), yielding a very high mean timeout
>>> value.
>>> Therefore, we reported median and it would be more descriptive to
>>> report
>>> it here as well. The median of the measured TCP NAT binding timeouts in
>>> this study was around 60 mins, the shortest being around 2 mins. That
>>> is,
>>> clearly more than 50% of the devices had timeout shorter than RFC
>>> 5382 recommended minimum of 124 mins.
>>> In the light of these results, it may be hard to find a proper timeout
>>> value for the application-layer heartbeat messages, and it might be
>>> worth mentioning, I think.
>> We have updated the section based on these comments.
>>> Nits:
>> All done, except for the ?Bit-Error Rate? suggestion. The Collins
>> English
>> dictionary uses the non-hyphenated form.
>>> Sec 1:
>>> 1st para: Add references and cite  6LoWPAN, RPL, and CoAP.
>>> 3rd para: "At the application layer, CoAP was developed over UDP
>>> [RFC7252]."
>>> - this seems to cite UDP incorrectly while the intent is to cite CoAP.
>>>    If you cite CoAP in the first para, you do not need to cite here at
>>> all.
>>>   "This the main reason..." -> "This is the main reason..."
>>> 5th para:
>>>   "Given the limited resources on constrained devices, careful "tuning"
>>>    of the TCP implementation can make an implementation more
>>> lightweight."
>>> Instead of saying "tuning" of the TCP implementation, I'd say that
>>> careful
>>> selection of optional TCP features can make an implementation more
>>> lightweight (and improve operation in CNNs).
>>> 6th para:
>>>   "This document provides guidance on how to implement and use TCP in
>>>    CNNs.
>>> ->
>>>   "This document provides guidance on how to implement and configure
>>> TCP
>>>    as well as how TCP is advisable to be used by applications in CNNs.
>>> Sec 3.1., last para:
>>>   high bit error rate ->  high bit-error rate
>>> Sec 5., first sentence:
>>>   "how a TCP stack can be used" ->  "how TCP can be used"
>> Once again, thank you very much for your comprehensive review and
>> constructive suggestions!
>> Cheers,
>> Carles (on behalf of all authors)