[tcpm] WGLC review of draft-ietf-lwig-tcp-constrained-node-networks

Markku Kojo <kojo@cs.helsinki.fi> Thu, 02 May 2019 09:16 UTC

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Date: Thu, 2 May 2019 12:15:59 +0300 (EEST)
From: Markku Kojo <kojo@cs.helsinki.fi>
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Subject: [tcpm] WGLC review of draft-ietf-lwig-tcp-constrained-node-networks
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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,


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?

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?

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?

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!

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 ..."

  "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].

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.

Sec 4.2.

  "This section discusses TCP stacks that focus on transferring a single

Maybe better:

  "This section discusses TCP stacks that allow transferring only a single
   MSS at a time."

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.

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.

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

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.

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.

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.

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.

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.

Sec. 4.2.4.

RTO is not estimated but calculated using estimated RTT and deviation
from it. That is, modify:

RTO estimation -> RTO calculation

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.

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?

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.

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.

  "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


  "... 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).

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.

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.


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


  "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"