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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: Neal Cardwell <ncardwell@google.com>
Thread-Topic: [tcpm] [e2e] TCP HyStart patch deployment
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Date: Fri, 2 Sep 2016 09:11:39 +0000
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Cc: Kristofer Sandlund <kristofer.sandlund@ericsson.com>, "tcpm@ietf.org" <tcpm@ietf.org>, Piers O'Hanlon <p.ohanlon@gmail.com>, Sangtae Ha <sangtae.ha@gmail.com>, Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, Eric Dumazet <edumazet@google.com>, "end2end-interest@postel.org" <end2end-interest@postel.org>
Subject: Re: [tcpm] [e2e] TCP HyStart patch deployment
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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
Date: Sun, 4 Sep 2016 03:53:54 -0700
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Subject: Re: [tcpm] (ref. RFC 7323) Increase of shift count from 14 to 15.
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Hi Sanjeev,

I think there might be some useful cases if we can change shift count
in the middle of the session.
However, one tricky thing for this is you'll need to send new shift
count reliably. Because when an end node sends a new shift count, it
will be problematic if the peer cannot receive it.
But, TCP doesn't have a good mechanism to send TCP options reliably
during data transfer. So, in order to build this feature, we'll need
to think about how to solve it. That's the reason I think it'll be a
rather broad topic.

Thanks,
--
Yoshi



On Tue, Aug 30, 2016 at 9:02 PM, Sanjeev Ranot
<sanjeev.ranot.81@gmail.com> wrote:
> Hi Yoshi,
>
> Yes, using shf.cnt of 15 in connection establishment phase is sufficient
> enough, but  while thinking from migration and interoperability this
> suggestion of changing shf.cnt during Data Transfer Phase came to mind.
>
> Present behavior is that if window limit is reached then we cannot
> transfer no matter how big cwnd is. This limits the rate of growth of
> congestion window if not growth completely. Now with high speed interfaces
> it will not always be the case that we increase size of MTU (as for jumbo
> frames) to get the bandwidth with support of bundling (etherChannels).
>
> Generally when shf.cnt is chosen it defines the size of SND.WND
> within range from 0 to max SND.WND. SND.WND proves to be a
> constraint in situation discussed below.
>
> Note: With PAWS we need to make sure that it is sufficiently large
> enough to provide ample monotonically non-decreasing numbers so that
> they can be differentiated or make its use in such a way so that
> its always sufficient to distinguish at-least between 2 successive
> windows. RFC 7323 defines such a way by putting same TSval for all
> segments belonging to the same window.
>
> Before citing situation for the need of dynamic exchange of shf.cnt we
> need to discuss why sequence number of greater than 32 bit is not needed
> because of PAWS and PAWS timestamp Interface transfer speed and window
> size are related
>
> With PAWS, each wrap of timestamp must be >= 2*MSL. This MSL is based
> on transfer rate. Assuming 1 timestamp tick is 1 ms, hence if we
> transfer 2^30 bytes of data in each window i.e. with shf.cnt = 14 then
> maximum transfer rate which we need to get this big a window is of
> 1 Tera bytes (TbE) with PAWS in place
>
> 1000 ms (1s) = 1000 Gb (1TbE)
> 1ms  = 1 Gb ~ 2^30
>
> Therefore with PAWS timestamp tick of 1ms allows 1 Gb of window size that
> we can transmit safely without data corruption due to duplicate data or
> ACK segments. This minimum timestamp click needed for PAWS safety is
> provided with a base consideration of sequence number of 32 bits in size.
> It takes 49.6 days for a 32 bit sequence number to wrap again when
> timestamp tick is 1ms.  If in future we were to increase sequence numbers
> to 64 bits, then this timestamp tick can be reduced or we MAY never need
> PAWS again thereafter if 64 bit sequence number is used and we can never
> increase bandwidth speed no matter how much we bundle cables.
>
> The situation discussed assume that we are never going to increase the
> sequence number size and we are good with PAWS to provide us safety from
> duplicate data and ACK segment or of old packets from previous incarnations
> of the same TCP connection.
>
> Another example with shf.cnt = 15 :
> With shf.cnt = 15
> 1000ms  = 2000 Gb (2TbE)
> 1ms        = 2000000000 bits = 2 Gb ~ 2^31
> Therefore with a speed of 2Tbps and 1 ms timestamp tick for PAWS we can
> get a window size of 2 Gb.
>
> Migration Notes:
> This means that when we introduce a shf.cnt of 15, then we MAY either let
> PAWS timestamp tick to remain same as used for 1 Gb of window size with
> shf.cnt of 14 if its sufficient enough or we may think to readjust if we
> want to gain more speed by lowering the previous PAWS timestamp.
>
> Similary with a speed of 4Tbps and 1 ms timestamp tick for PAWS we can
> get a window size of 4 Gb.
>
> Situation:
> TCP connection is going on for days using same shf.cnt which was
> exchanged on day 1 when this TCP connection was established. Suppose
> due to incredible high transfer rate of the inter-continent link
> cwnd == SND.WND was already reached at end of day 1 and now the
> transfer rate is limited by SND.WND (decided by shf.cnt) when only
> very small fraction of link speed is utilized. Now suppose a very
> huge surplus of memory is made available by hot-swappable memory
> card inserted on both ends while TCP connection is running its
> Data Transfer Phase. Shf.cnt was already decided in beginning of
> this TCP connection and it was decided on the basis of amount on
> total memory available at that time to be allocated to its RCV.BUF.
> It required restarting this ongoing connection to have a new shf.cnt
> that will now consider this new additional memory surplus (i.e.
> increase RCV.BUF) and start all over again from start or further
> from the last index received is area of improvement.
>
> My proposal is that (this is separate from the on going work on shf.cnt =
> 15) :
> If TCP stack have support of exchanging shf.cnt in data transfer phase
> when large amount of memory is available dynamically then we can achieve
> significant high transfer rates without ever having to restart the
> connection.
>
> This feature comes in heading "Dynamically adjustable TCP window" and
> shf.cnt_interoperability.pdf mentions one of the possible mechanisms
> for implementing this introducing 7 new state variables and 1 TCP flag.
>
> Also, just to reiterate regarding migration purposes please look into
> PAWS and discuss with stake holders if they are ok with existing
> timestamp for new shf.cnt of 15.
>
>
> Thanks & Regards,
> Sanjeev Ranot
>
>
>
> On 29 August 2016 at 13:02, Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
> wrote:
>>
>> Hi Sanjeev,
>>
>> On Sun, Aug 28, 2016 at 1:55 AM, Sanjeev Ranot
>> <sanjeev.ranot.81@gmail.com> wrote:
>> > Hi Yoshi,
>> >
>> > thanks for replying. I have done some work in these couple of days on
>> > how exchange and interoperability of shift count 15 or greater will
>> > work.
>> >
>> > Basically my idea is for the new TCP stack (which will support shf.cnt
>> > >=
>> > 15)
>> > to exchange shf.cnt even in Data Transfer Phase and not only in
>> > connection
>> > establishment. I have presented this exchange work in the document
>> > enclosed.
>> >
>> > Please have a look at the document and let me know for any info or
>> > clarification
>>
>> Thanks for the doc.
>> I personally think you might want to clarify why we want to change
>> shift count in data transfer phase a bit more.
>> This is because even if we use shift count=15, I think we can specify
>> rather small window size by using a small number for the timestamp
>> value. So, if the stack supports shift count=15, I think it can just
>> use it from the beginning of the data transfer in most cases.
>>
>> If your focus is to have a feature to update shift count during data
>> transfer phase in general, I think it can be a separate proposal from
>> increasing max shift count. Because it looks a more broad topic to me.
>>
>> Thanks,
>> --
>> Yoshi
>
>


From nobody Sun Sep  4 03:55:13 2016
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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
Date: Sun, 4 Sep 2016 03:55:05 -0700
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Subject: Re: [tcpm] question about if a recent Linux patch is compliant with RFC7323 on window scaling
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Hi Cheng,

On Tue, Aug 30, 2016 at 7:16 AM, Cui, Cheng <Cheng.Cui@netapp.com> wrote:
> Six days have passed and no response yet. Can anyone in this board make a comment?
>
> Thanks,
> --Cheng Cui
> NetApp Scale Out Networking
>
>
> On 8/24/16, 5:06 PM, "Cui, Cheng" <Cheng.Cui@netapp.com> wrote:
>
>     Hello David,
>
>     I hope this email could reach you well, because I found related
>     discussions about this topic on window scaling and the case of window
>     shrinking (or retraction or loss of precision). And I try to make this
>     question simple.
>
>     There is a recent Linux patch at receiver side to round-up advertised
>     window due to precision loss of window scaling. It reaches my attention
>     because the same problem could also happen between a pair of Linux and
>     FreeBSD nodes, and I am not aware of any similar patch in FreeBSD yet.
>
>     The Linux patch is this:
>     http://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/commit/?id=6
>     07bfbf2d55dd1cfe5368b41c2a81a8c9ccf4723
>
>     And I quote some description of the Linux patch below:
>     > If the sender uses up the entire window before it is shrunk, this can
>     >have
>     > chaotic effects on the connection. When sending ACKs,
>     >tcp_acceptable_seq()
>     > will notice that the window has been shrunk since tcp_wnd_end() is before
>     > tp->snd_nxt, which makes it choose tcp_wnd_end() as sequence number.
>     > This will fail the receivers checks in tcp_sequence() however since it
>     > is before it's tp->rcv_wup, making it respond with a dupack.
>
>
>     If the Linux sender?s behavior is right ("ACK-only packets should be sent
>     with the largest in-window sequence number that has ever been sent." ref:
>     https://www.ietf.org/mail-archive/web/tcpm/current/msg10512.html), it
>     actually chooses "tp->snd_una+tp->snd_wnd" (tcp_wnd_end()) instead of
>     tp->snd_nxt, as it thought tp->snd_nxt is out of window, in case of
>     precision loss which made the receiver?s advertise-window smaller.
>     However, the RFC7323 explicitly specifies the receiver should right shift
>     the exact scale factor, without mentioning any further round-up:
>
>     https://tools.ietf.org/html/rfc7323#page-9
>     >    o  The window field (SEG.WND) of every outgoing segment, with the
>     >       exception of <SYN> segments, MUST be right-shifted by
>     >       Rcv.Wind.Shift bits:
>     >
>     >                     SEG.WND = RCV.WND >> Rcv.Wind.Shift
>
>     And also if I am understanding correctly, RFC7323 page 10 (2.4. Addressing
>     Window Retraction) specifies it is the sender?s responsibility to handle
>     the sequence number out of window:
>     https://tools.ietf.org/html/rfc7323#page-10
>     >    4)  On first retransmission, or if the sequence number is out of
>     >        window by less than 2^Rcv.Wind.Shift, then do normal
>     >        retransmission(s) without regard to the receiver window as long
>     >        as the original segment was in window when it was sent.
>     >
>     >    5)  Subsequent retransmissions MAY only be sent if they are within
>     >        the window announced by the most recent <ACK>.
>
>     In your discussion of the link below, if I am understanding correctly, you
>     were also in favor of "Yes. It is ok to have more receive buffer space
>     available than you advertise in the window, but not ok to advertise a
>     larger window than you are willing to accept." So I think you were not in
>     favor of rounding-up the advertise-window.
>
>     https://www.ietf.org/mail-archive/web/tsvwg/current/msg06475.html
>
>
>     So my question is: Is the Linux patch (on receiver side to round-up the
>     advertise-window) RFC compliant? Or is it just the right thing (vendor
>     specific) and other systems like FreeBSD have to follow up, so that their
>     talks can have no problem?

I think the Linux patch at least follows the following principle in RFC793.

  The mechanisms provided allow a TCP to advertise a large window and to
  subsequently advertise a much smaller window without having accepted
  that much data.  This, so called "shrinking the window," is strongly
  discouraged.  The robustness principle dictates that TCPs will not
  shrink the window themselves, but will be prepared for such behavior
  on the part of other TCPs.

I personally think a real question would be how FreeBSD prepares for
shrinking window.
Linux chose one way for it, but I guess all implementations don't need
to follow the same way.

--
Yoshi


From nobody Mon Sep  5 08:01:46 2016
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From: Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
Date: Mon, 5 Sep 2016 09:48:11 +0530
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Subject: Re: [tcpm] (ref. RFC 7323) Increase of shift count from 14 to 15.
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Hi Yoshi,

thanks for the reply. I really want you and team to take forward this idea
of
dynamically changing the shift count whenever in future you start work on
this.
I will be glad if it gets implemented.

At present its out of my capacity to take it forward because I'm looking to
relocate permanently abroad by getting some job and so is trying to meet
the ends meet.

As per your concerns, I too had similar concerns of reliably communicating
this information to the peer therefore in the document which contains
implementation of the idea i.e.
*"shf.cnt_interoperability.pdf"*
has a note:




*"The condition is that these shf.cnt update will be transmitted only
reliablyon data segment or on non-empty ack segment (NOT using empty
acksegment or pure ack)"*

Thanks again for your and team's valuable time and input.

Thanks & Regards,
Sanjeev Ranot



On 4 September 2016 at 16:23, Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
wrote:

> Hi Sanjeev,
>
> I think there might be some useful cases if we can change shift count
> in the middle of the session.
> However, one tricky thing for this is you'll need to send new shift
> count reliably. Because when an end node sends a new shift count, it
> will be problematic if the peer cannot receive it.
> But, TCP doesn't have a good mechanism to send TCP options reliably
> during data transfer. So, in order to build this feature, we'll need
> to think about how to solve it. That's the reason I think it'll be a
> rather broad topic.
>
> Thanks,
> --
> Yoshi
>
>
>
> On Tue, Aug 30, 2016 at 9:02 PM, Sanjeev Ranot
> <sanjeev.ranot.81@gmail.com> wrote:
> > Hi Yoshi,
> >
> > Yes, using shf.cnt of 15 in connection establishment phase is sufficient
> > enough, but  while thinking from migration and interoperability this
> > suggestion of changing shf.cnt during Data Transfer Phase came to mind.
> >
> > Present behavior is that if window limit is reached then we cannot
> > transfer no matter how big cwnd is. This limits the rate of growth of
> > congestion window if not growth completely. Now with high speed
> interfaces
> > it will not always be the case that we increase size of MTU (as for jumbo
> > frames) to get the bandwidth with support of bundling (etherChannels).
> >
> > Generally when shf.cnt is chosen it defines the size of SND.WND
> > within range from 0 to max SND.WND. SND.WND proves to be a
> > constraint in situation discussed below.
> >
> > Note: With PAWS we need to make sure that it is sufficiently large
> > enough to provide ample monotonically non-decreasing numbers so that
> > they can be differentiated or make its use in such a way so that
> > its always sufficient to distinguish at-least between 2 successive
> > windows. RFC 7323 defines such a way by putting same TSval for all
> > segments belonging to the same window.
> >
> > Before citing situation for the need of dynamic exchange of shf.cnt we
> > need to discuss why sequence number of greater than 32 bit is not needed
> > because of PAWS and PAWS timestamp Interface transfer speed and window
> > size are related
> >
> > With PAWS, each wrap of timestamp must be >= 2*MSL. This MSL is based
> > on transfer rate. Assuming 1 timestamp tick is 1 ms, hence if we
> > transfer 2^30 bytes of data in each window i.e. with shf.cnt = 14 then
> > maximum transfer rate which we need to get this big a window is of
> > 1 Tera bytes (TbE) with PAWS in place
> >
> > 1000 ms (1s) = 1000 Gb (1TbE)
> > 1ms  = 1 Gb ~ 2^30
> >
> > Therefore with PAWS timestamp tick of 1ms allows 1 Gb of window size that
> > we can transmit safely without data corruption due to duplicate data or
> > ACK segments. This minimum timestamp click needed for PAWS safety is
> > provided with a base consideration of sequence number of 32 bits in size.
> > It takes 49.6 days for a 32 bit sequence number to wrap again when
> > timestamp tick is 1ms.  If in future we were to increase sequence numbers
> > to 64 bits, then this timestamp tick can be reduced or we MAY never need
> > PAWS again thereafter if 64 bit sequence number is used and we can never
> > increase bandwidth speed no matter how much we bundle cables.
> >
> > The situation discussed assume that we are never going to increase the
> > sequence number size and we are good with PAWS to provide us safety from
> > duplicate data and ACK segment or of old packets from previous
> incarnations
> > of the same TCP connection.
> >
> > Another example with shf.cnt = 15 :
> > With shf.cnt = 15
> > 1000ms  = 2000 Gb (2TbE)
> > 1ms        = 2000000000 bits = 2 Gb ~ 2^31
> > Therefore with a speed of 2Tbps and 1 ms timestamp tick for PAWS we can
> > get a window size of 2 Gb.
> >
> > Migration Notes:
> > This means that when we introduce a shf.cnt of 15, then we MAY either let
> > PAWS timestamp tick to remain same as used for 1 Gb of window size with
> > shf.cnt of 14 if its sufficient enough or we may think to readjust if we
> > want to gain more speed by lowering the previous PAWS timestamp.
> >
> > Similary with a speed of 4Tbps and 1 ms timestamp tick for PAWS we can
> > get a window size of 4 Gb.
> >
> > Situation:
> > TCP connection is going on for days using same shf.cnt which was
> > exchanged on day 1 when this TCP connection was established. Suppose
> > due to incredible high transfer rate of the inter-continent link
> > cwnd == SND.WND was already reached at end of day 1 and now the
> > transfer rate is limited by SND.WND (decided by shf.cnt) when only
> > very small fraction of link speed is utilized. Now suppose a very
> > huge surplus of memory is made available by hot-swappable memory
> > card inserted on both ends while TCP connection is running its
> > Data Transfer Phase. Shf.cnt was already decided in beginning of
> > this TCP connection and it was decided on the basis of amount on
> > total memory available at that time to be allocated to its RCV.BUF.
> > It required restarting this ongoing connection to have a new shf.cnt
> > that will now consider this new additional memory surplus (i.e.
> > increase RCV.BUF) and start all over again from start or further
> > from the last index received is area of improvement.
> >
> > My proposal is that (this is separate from the on going work on shf.cnt =
> > 15) :
> > If TCP stack have support of exchanging shf.cnt in data transfer phase
> > when large amount of memory is available dynamically then we can achieve
> > significant high transfer rates without ever having to restart the
> > connection.
> >
> > This feature comes in heading "Dynamically adjustable TCP window" and
> > shf.cnt_interoperability.pdf mentions one of the possible mechanisms
> > for implementing this introducing 7 new state variables and 1 TCP flag.
> >
> > Also, just to reiterate regarding migration purposes please look into
> > PAWS and discuss with stake holders if they are ok with existing
> > timestamp for new shf.cnt of 15.
> >
> >
> > Thanks & Regards,
> > Sanjeev Ranot
> >
> >
> >
> > On 29 August 2016 at 13:02, Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
> > wrote:
> >>
> >> Hi Sanjeev,
> >>
> >> On Sun, Aug 28, 2016 at 1:55 AM, Sanjeev Ranot
> >> <sanjeev.ranot.81@gmail.com> wrote:
> >> > Hi Yoshi,
> >> >
> >> > thanks for replying. I have done some work in these couple of days on
> >> > how exchange and interoperability of shift count 15 or greater will
> >> > work.
> >> >
> >> > Basically my idea is for the new TCP stack (which will support shf.cnt
> >> > >=
> >> > 15)
> >> > to exchange shf.cnt even in Data Transfer Phase and not only in
> >> > connection
> >> > establishment. I have presented this exchange work in the document
> >> > enclosed.
> >> >
> >> > Please have a look at the document and let me know for any info or
> >> > clarification
> >>
> >> Thanks for the doc.
> >> I personally think you might want to clarify why we want to change
> >> shift count in data transfer phase a bit more.
> >> This is because even if we use shift count=15, I think we can specify
> >> rather small window size by using a small number for the timestamp
> >> value. So, if the stack supports shift count=15, I think it can just
> >> use it from the beginning of the data transfer in most cases.
> >>
> >> If your focus is to have a feature to update shift count during data
> >> transfer phase in general, I think it can be a separate proposal from
> >> increasing max shift count. Because it looks a more broad topic to me.
> >>
> >> Thanks,
> >> --
> >> Yoshi
> >
> >
>

--001a11439ef2d8b65b053bbaf71f
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div><div><div><div><div><div>Hi Yoshi,<br><br></div>thank=
s for the reply. I really want you and team to take forward this idea of <b=
r>dynamically changing the shift count whenever in future you start work on=
 this. <br>I will be glad if it gets implemented. <br></div></div><div><br>=
</div>At present its out of my capacity to take it forward because I&#39;m =
looking to <br>relocate permanently abroad by getting some job and so is tr=
ying to meet <br></div>the ends meet.=C2=A0 <br><br></div></div><div>As per=
 your concerns, I too had similar concerns of reliably communicating <br></=
div><div>this information to the peer therefore in the document which conta=
ins <br></div><div>implementation of the idea i.e. <b>&quot;shf.cnt_interop=
erability.pdf&quot;<br></b></div><div>has a note: <br><i><br>&quot;The cond=
ition is that these shf.cnt update will be transmitted only reliably<br>on =
data segment or on non-empty ack segment (NOT using empty ack<br>segment or=
 pure ack)&quot;<br></i></div><div><div><div><div><br></div><div>Thanks aga=
in for your and team&#39;s valuable time and input.<br><br></div><div>Thank=
s &amp; Regards,<br></div><div>Sanjeev Ranot<br></div><div><br><div><br></d=
iv></div></div></div></div></div><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">On 4 September 2016 at 16:23, Yoshifumi Nishida <span dir=
=3D"ltr">&lt;<a href=3D"mailto:nishida@sfc.wide.ad.jp" target=3D"_blank">ni=
shida@sfc.wide.ad.jp</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex=
">Hi Sanjeev,<br>
<br>
I think there might be some useful cases if we can change shift count<br>
in the middle of the session.<br>
However, one tricky thing for this is you&#39;ll need to send new shift<br>
count reliably. Because when an end node sends a new shift count, it<br>
will be problematic if the peer cannot receive it.<br>
But, TCP doesn&#39;t have a good mechanism to send TCP options reliably<br>
during data transfer. So, in order to build this feature, we&#39;ll need<br=
>
to think about how to solve it. That&#39;s the reason I think it&#39;ll be =
a<br>
rather broad topic.<br>
<br>
Thanks,<br>
--<br>
Yoshi<br>
<br>
<br>
<br>
On Tue, Aug 30, 2016 at 9:02 PM, Sanjeev Ranot<br>
<div class=3D"HOEnZb"><div class=3D"h5">&lt;<a href=3D"mailto:sanjeev.ranot=
.81@gmail.com">sanjeev.ranot.81@gmail.com</a>&gt; wrote:<br>
&gt; Hi Yoshi,<br>
&gt;<br>
&gt; Yes, using shf.cnt of 15 in connection establishment phase is sufficie=
nt<br>
&gt; enough, but=C2=A0 while thinking from migration and interoperability t=
his<br>
&gt; suggestion of changing shf.cnt during Data Transfer Phase came to mind=
.<br>
&gt;<br>
&gt; Present behavior is that if window limit is reached then we cannot<br>
&gt; transfer no matter how big cwnd is. This limits the rate of growth of<=
br>
&gt; congestion window if not growth completely. Now with high speed interf=
aces<br>
&gt; it will not always be the case that we increase size of MTU (as for ju=
mbo<br>
&gt; frames) to get the bandwidth with support of bundling (etherChannels).=
<br>
&gt;<br>
&gt; Generally when shf.cnt is chosen it defines the size of SND.WND<br>
&gt; within range from 0 to max SND.WND. SND.WND proves to be a<br>
&gt; constraint in situation discussed below.<br>
&gt;<br>
&gt; Note: With PAWS we need to make sure that it is sufficiently large<br>
&gt; enough to provide ample monotonically non-decreasing numbers so that<b=
r>
&gt; they can be differentiated or make its use in such a way so that<br>
&gt; its always sufficient to distinguish at-least between 2 successive<br>
&gt; windows. RFC 7323 defines such a way by putting same TSval for all<br>
&gt; segments belonging to the same window.<br>
&gt;<br>
&gt; Before citing situation for the need of dynamic exchange of shf.cnt we=
<br>
&gt; need to discuss why sequence number of greater than 32 bit is not need=
ed<br>
&gt; because of PAWS and PAWS timestamp Interface transfer speed and window=
<br>
&gt; size are related<br>
&gt;<br>
&gt; With PAWS, each wrap of timestamp must be &gt;=3D 2*MSL. This MSL is b=
ased<br>
&gt; on transfer rate. Assuming 1 timestamp tick is 1 ms, hence if we<br>
&gt; transfer 2^30 bytes of data in each window i.e. with shf.cnt =3D 14 th=
en<br>
&gt; maximum transfer rate which we need to get this big a window is of<br>
&gt; 1 Tera bytes (TbE) with PAWS in place<br>
&gt;<br>
&gt; 1000 ms (1s) =3D 1000 Gb (1TbE)<br>
&gt; 1ms=C2=A0 =3D 1 Gb ~ 2^30<br>
&gt;<br>
&gt; Therefore with PAWS timestamp tick of 1ms allows 1 Gb of window size t=
hat<br>
&gt; we can transmit safely without data corruption due to duplicate data o=
r<br>
&gt; ACK segments. This minimum timestamp click needed for PAWS safety is<b=
r>
&gt; provided with a base consideration of sequence number of 32 bits in si=
ze.<br>
&gt; It takes 49.6 days for a 32 bit sequence number to wrap again when<br>
&gt; timestamp tick is 1ms.=C2=A0 If in future we were to increase sequence=
 numbers<br>
&gt; to 64 bits, then this timestamp tick can be reduced or we MAY never ne=
ed<br>
&gt; PAWS again thereafter if 64 bit sequence number is used and we can nev=
er<br>
&gt; increase bandwidth speed no matter how much we bundle cables.<br>
&gt;<br>
&gt; The situation discussed assume that we are never going to increase the=
<br>
&gt; sequence number size and we are good with PAWS to provide us safety fr=
om<br>
&gt; duplicate data and ACK segment or of old packets from previous incarna=
tions<br>
&gt; of the same TCP connection.<br>
&gt;<br>
&gt; Another example with shf.cnt =3D 15 :<br>
&gt; With shf.cnt =3D 15<br>
&gt; 1000ms=C2=A0 =3D 2000 Gb (2TbE)<br>
&gt; 1ms=C2=A0 =C2=A0 =C2=A0 =C2=A0 =3D 2000000000 bits =3D 2 Gb ~ 2^31<br>
&gt; Therefore with a speed of 2Tbps and 1 ms timestamp tick for PAWS we ca=
n<br>
&gt; get a window size of 2 Gb.<br>
&gt;<br>
&gt; Migration Notes:<br>
&gt; This means that when we introduce a shf.cnt of 15, then we MAY either =
let<br>
&gt; PAWS timestamp tick to remain same as used for 1 Gb of window size wit=
h<br>
&gt; shf.cnt of 14 if its sufficient enough or we may think to readjust if =
we<br>
&gt; want to gain more speed by lowering the previous PAWS timestamp.<br>
&gt;<br>
&gt; Similary with a speed of 4Tbps and 1 ms timestamp tick for PAWS we can=
<br>
&gt; get a window size of 4 Gb.<br>
&gt;<br>
&gt; Situation:<br>
&gt; TCP connection is going on for days using same shf.cnt which was<br>
&gt; exchanged on day 1 when this TCP connection was established. Suppose<b=
r>
&gt; due to incredible high transfer rate of the inter-continent link<br>
&gt; cwnd =3D=3D SND.WND was already reached at end of day 1 and now the<br=
>
&gt; transfer rate is limited by SND.WND (decided by shf.cnt) when only<br>
&gt; very small fraction of link speed is utilized. Now suppose a very<br>
&gt; huge surplus of memory is made available by hot-swappable memory<br>
&gt; card inserted on both ends while TCP connection is running its<br>
&gt; Data Transfer Phase. Shf.cnt was already decided in beginning of<br>
&gt; this TCP connection and it was decided on the basis of amount on<br>
&gt; total memory available at that time to be allocated to its RCV.BUF.<br=
>
&gt; It required restarting this ongoing connection to have a new shf.cnt<b=
r>
&gt; that will now consider this new additional memory surplus (i.e.<br>
&gt; increase RCV.BUF) and start all over again from start or further<br>
&gt; from the last index received is area of improvement.<br>
&gt;<br>
&gt; My proposal is that (this is separate from the on going work on shf.cn=
t =3D<br>
&gt; 15) :<br>
&gt; If TCP stack have support of exchanging shf.cnt in data transfer phase=
<br>
&gt; when large amount of memory is available dynamically then we can achie=
ve<br>
&gt; significant high transfer rates without ever having to restart the<br>
&gt; connection.<br>
&gt;<br>
&gt; This feature comes in heading &quot;Dynamically adjustable TCP window&=
quot; and<br>
&gt; shf.cnt_interoperability.pdf mentions one of the possible mechanisms<b=
r>
&gt; for implementing this introducing 7 new state variables and 1 TCP flag=
.<br>
&gt;<br>
&gt; Also, just to reiterate regarding migration purposes please look into<=
br>
&gt; PAWS and discuss with stake holders if they are ok with existing<br>
&gt; timestamp for new shf.cnt of 15.<br>
&gt;<br>
&gt;<br>
&gt; Thanks &amp; Regards,<br>
&gt; Sanjeev Ranot<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On 29 August 2016 at 13:02, Yoshifumi Nishida &lt;<a href=3D"mailto:ni=
shida@sfc.wide.ad.jp">nishida@sfc.wide.ad.jp</a>&gt;<br>
&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; Hi Sanjeev,<br>
&gt;&gt;<br>
&gt;&gt; On Sun, Aug 28, 2016 at 1:55 AM, Sanjeev Ranot<br>
&gt;&gt; &lt;<a href=3D"mailto:sanjeev.ranot.81@gmail.com">sanjeev.ranot.81=
@gmail.com</a>&gt; wrote:<br>
&gt;&gt; &gt; Hi Yoshi,<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; thanks for replying. I have done some work in these couple of=
 days on<br>
&gt;&gt; &gt; how exchange and interoperability of shift count 15 or greate=
r will<br>
&gt;&gt; &gt; work.<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; Basically my idea is for the new TCP stack (which will suppor=
t shf.cnt<br>
&gt;&gt; &gt; &gt;=3D<br>
&gt;&gt; &gt; 15)<br>
&gt;&gt; &gt; to exchange shf.cnt even in Data Transfer Phase and not only =
in<br>
&gt;&gt; &gt; connection<br>
&gt;&gt; &gt; establishment. I have presented this exchange work in the doc=
ument<br>
&gt;&gt; &gt; enclosed.<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; Please have a look at the document and let me know for any in=
fo or<br>
&gt;&gt; &gt; clarification<br>
&gt;&gt;<br>
&gt;&gt; Thanks for the doc.<br>
&gt;&gt; I personally think you might want to clarify why we want to change=
<br>
&gt;&gt; shift count in data transfer phase a bit more.<br>
&gt;&gt; This is because even if we use shift count=3D15, I think we can sp=
ecify<br>
&gt;&gt; rather small window size by using a small number for the timestamp=
<br>
&gt;&gt; value. So, if the stack supports shift count=3D15, I think it can =
just<br>
&gt;&gt; use it from the beginning of the data transfer in most cases.<br>
&gt;&gt;<br>
&gt;&gt; If your focus is to have a feature to update shift count during da=
ta<br>
&gt;&gt; transfer phase in general, I think it can be a separate proposal f=
rom<br>
&gt;&gt; increasing max shift count. Because it looks a more broad topic to=
 me.<br>
&gt;&gt;<br>
&gt;&gt; Thanks,<br>
&gt;&gt; --<br>
&gt;&gt; Yoshi<br>
&gt;<br>
&gt;<br>
</div></div></blockquote></div><br></div>

--001a11439ef2d8b65b053bbaf71f--


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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
Subject: Re: [tcpm] (ref. RFC 7323) Increase of shift count from 14 to 15.
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The biggest problem with changing the shift count midstream is that TCP =
has no reliable mechanism for doing option negotiation midstream.  The =
only packets that are reliably delivered are packets that contain data =
or have the SYN or FIN bit set.  If the receiver changes its shift count =
mid-stream, the sender side needs to know where in the packet stream the =
new value applies.  You have to allow for lost or reordered packets, and =
the receiver, in a one-way data transfer, is only sending ACK packets, =
none of which are reliably transferred and have no ordering, the =
receivers sequence number remains constant, making it impossible to =
reliably identify a point in time where the new value takes effect.

One way to reliably change the shift count is to put it into every =
packet, for interoperability older implementations could be identified =
because they would only have it in the SYN packets.  This was considered =
years ago but was ruled out: the limitations of only negotiating the =
shift count during connection establishment was considered acceptable =
compared to putting it into every packet and using up valuable TCP =
option space.

To use a shift count of 15, you have to be able to determine whether or =
not the other side can accept a value of 15.  If you send a 15, but are =
talking to an older implementation that clamps it at 14 and you don=E2=80=99=
t know that, then your window values, sent assuming a shift count of 15, =
will be halved by the receiver who clamped it at 14; that is probably =
not desirable as the whole point of have a shift count of 15 is to allow =
larger windows, not smaller windows.

As to why the value is 14, that is explained in RFC 7323, section 2.4.  =
The logic is based on the assumption that if you allow a scale of 15, =
that at some point someone will try to use all of it, and limiting it to =
14 ensured that we wouldn=E2=80=99t run into any sequence space =
wrap-around problems.  Whether or not that logic is correct is a =
separate discussion, but that=E2=80=99s why 14 was chosen.

			-David Borman

> On Sep 4, 2016, at 5:53 AM, Yoshifumi Nishida <nishida@sfc.wide.ad.jp> =
wrote:
>=20
> Hi Sanjeev,
>=20
> I think there might be some useful cases if we can change shift count
> in the middle of the session.
> However, one tricky thing for this is you'll need to send new shift
> count reliably. Because when an end node sends a new shift count, it
> will be problematic if the peer cannot receive it.
> But, TCP doesn't have a good mechanism to send TCP options reliably
> during data transfer. So, in order to build this feature, we'll need
> to think about how to solve it. That's the reason I think it'll be a
> rather broad topic.
>=20
> Thanks,
> --
> Yoshi
>=20
>=20
>=20
> On Tue, Aug 30, 2016 at 9:02 PM, Sanjeev Ranot
> <sanjeev.ranot.81@gmail.com> wrote:
>> Hi Yoshi,
>>=20
>> Yes, using shf.cnt of 15 in connection establishment phase is =
sufficient
>> enough, but  while thinking from migration and interoperability this
>> suggestion of changing shf.cnt during Data Transfer Phase came to =
mind.
>>=20
>> Present behavior is that if window limit is reached then we cannot
>> transfer no matter how big cwnd is. This limits the rate of growth of
>> congestion window if not growth completely. Now with high speed =
interfaces
>> it will not always be the case that we increase size of MTU (as for =
jumbo
>> frames) to get the bandwidth with support of bundling =
(etherChannels).
>>=20
>> Generally when shf.cnt is chosen it defines the size of SND.WND
>> within range from 0 to max SND.WND. SND.WND proves to be a
>> constraint in situation discussed below.
>>=20
>> Note: With PAWS we need to make sure that it is sufficiently large
>> enough to provide ample monotonically non-decreasing numbers so that
>> they can be differentiated or make its use in such a way so that
>> its always sufficient to distinguish at-least between 2 successive
>> windows. RFC 7323 defines such a way by putting same TSval for all
>> segments belonging to the same window.
>>=20
>> Before citing situation for the need of dynamic exchange of shf.cnt =
we
>> need to discuss why sequence number of greater than 32 bit is not =
needed
>> because of PAWS and PAWS timestamp Interface transfer speed and =
window
>> size are related
>>=20
>> With PAWS, each wrap of timestamp must be >=3D 2*MSL. This MSL is =
based
>> on transfer rate. Assuming 1 timestamp tick is 1 ms, hence if we
>> transfer 2^30 bytes of data in each window i.e. with shf.cnt =3D 14 =
then
>> maximum transfer rate which we need to get this big a window is of
>> 1 Tera bytes (TbE) with PAWS in place
>>=20
>> 1000 ms (1s) =3D 1000 Gb (1TbE)
>> 1ms  =3D 1 Gb ~ 2^30
>>=20
>> Therefore with PAWS timestamp tick of 1ms allows 1 Gb of window size =
that
>> we can transmit safely without data corruption due to duplicate data =
or
>> ACK segments. This minimum timestamp click needed for PAWS safety is
>> provided with a base consideration of sequence number of 32 bits in =
size.
>> It takes 49.6 days for a 32 bit sequence number to wrap again when
>> timestamp tick is 1ms.  If in future we were to increase sequence =
numbers
>> to 64 bits, then this timestamp tick can be reduced or we MAY never =
need
>> PAWS again thereafter if 64 bit sequence number is used and we can =
never
>> increase bandwidth speed no matter how much we bundle cables.
>>=20
>> The situation discussed assume that we are never going to increase =
the
>> sequence number size and we are good with PAWS to provide us safety =
from
>> duplicate data and ACK segment or of old packets from previous =
incarnations
>> of the same TCP connection.
>>=20
>> Another example with shf.cnt =3D 15 :
>> With shf.cnt =3D 15
>> 1000ms  =3D 2000 Gb (2TbE)
>> 1ms        =3D 2000000000 bits =3D 2 Gb ~ 2^31
>> Therefore with a speed of 2Tbps and 1 ms timestamp tick for PAWS we =
can
>> get a window size of 2 Gb.
>>=20
>> Migration Notes:
>> This means that when we introduce a shf.cnt of 15, then we MAY either =
let
>> PAWS timestamp tick to remain same as used for 1 Gb of window size =
with
>> shf.cnt of 14 if its sufficient enough or we may think to readjust if =
we
>> want to gain more speed by lowering the previous PAWS timestamp.
>>=20
>> Similary with a speed of 4Tbps and 1 ms timestamp tick for PAWS we =
can
>> get a window size of 4 Gb.
>>=20
>> Situation:
>> TCP connection is going on for days using same shf.cnt which was
>> exchanged on day 1 when this TCP connection was established. Suppose
>> due to incredible high transfer rate of the inter-continent link
>> cwnd =3D=3D SND.WND was already reached at end of day 1 and now the
>> transfer rate is limited by SND.WND (decided by shf.cnt) when only
>> very small fraction of link speed is utilized. Now suppose a very
>> huge surplus of memory is made available by hot-swappable memory
>> card inserted on both ends while TCP connection is running its
>> Data Transfer Phase. Shf.cnt was already decided in beginning of
>> this TCP connection and it was decided on the basis of amount on
>> total memory available at that time to be allocated to its RCV.BUF.
>> It required restarting this ongoing connection to have a new shf.cnt
>> that will now consider this new additional memory surplus (i.e.
>> increase RCV.BUF) and start all over again from start or further
>> from the last index received is area of improvement.
>>=20
>> My proposal is that (this is separate from the on going work on =
shf.cnt =3D
>> 15) :
>> If TCP stack have support of exchanging shf.cnt in data transfer =
phase
>> when large amount of memory is available dynamically then we can =
achieve
>> significant high transfer rates without ever having to restart the
>> connection.
>>=20
>> This feature comes in heading "Dynamically adjustable TCP window" and
>> shf.cnt_interoperability.pdf mentions one of the possible mechanisms
>> for implementing this introducing 7 new state variables and 1 TCP =
flag.
>>=20
>> Also, just to reiterate regarding migration purposes please look into
>> PAWS and discuss with stake holders if they are ok with existing
>> timestamp for new shf.cnt of 15.
>>=20
>>=20
>> Thanks & Regards,
>> Sanjeev Ranot
>>=20
>>=20
>>=20
>> On 29 August 2016 at 13:02, Yoshifumi Nishida =
<nishida@sfc.wide.ad.jp>
>> wrote:
>>>=20
>>> Hi Sanjeev,
>>>=20
>>> On Sun, Aug 28, 2016 at 1:55 AM, Sanjeev Ranot
>>> <sanjeev.ranot.81@gmail.com> wrote:
>>>> Hi Yoshi,
>>>>=20
>>>> thanks for replying. I have done some work in these couple of days =
on
>>>> how exchange and interoperability of shift count 15 or greater will
>>>> work.
>>>>=20
>>>> Basically my idea is for the new TCP stack (which will support =
shf.cnt
>>>>> =3D
>>>> 15)
>>>> to exchange shf.cnt even in Data Transfer Phase and not only in
>>>> connection
>>>> establishment. I have presented this exchange work in the document
>>>> enclosed.
>>>>=20
>>>> Please have a look at the document and let me know for any info or
>>>> clarification
>>>=20
>>> Thanks for the doc.
>>> I personally think you might want to clarify why we want to change
>>> shift count in data transfer phase a bit more.
>>> This is because even if we use shift count=3D15, I think we can =
specify
>>> rather small window size by using a small number for the timestamp
>>> value. So, if the stack supports shift count=3D15, I think it can =
just
>>> use it from the beginning of the data transfer in most cases.
>>>=20
>>> If your focus is to have a feature to update shift count during data
>>> transfer phase in general, I think it can be a separate proposal =
from
>>> increasing max shift count. Because it looks a more broad topic to =
me.
>>>=20
>>> Thanks,
>>> --
>>> Yoshi
>>=20
>>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Tue Sep  6 11:14:54 2016
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To: David Borman <dab@weston.borman.com>, Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
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From: Joe Touch <touch@isi.edu>
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
Subject: Re: [tcpm] (ref. RFC 7323) Increase of shift count from 14 to 15.
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Hi, all,

On 9/6/2016 7:24 AM, David Borman wrote:
> The biggest problem with changing the shift count midstream is that TCP has no reliable mechanism for doing option negotiation midstream.
Agreed.

>   The only packets that are reliably delivered are packets that contain data or have the SYN or FIN bit set.

AFAICT, this isn't strictly true.

The data and connection state are reliably "delivered" (or
synchronized), but not the packets per se. Options in those packets
aren't reliably delivered any more than any other packet, esp. because
TCP has no strict requirement that the options of retransmitted segments
don't vary. TCP gets around this by confirming the options exchanged
during the SYN, but no other time during the connection.

You *can* build your own handshake on top of TCP, but it's complicated.
We did this for TCP-AO to support key rollover - the sender offers a new
key, which cannot be used until the receiver confirms it. The same
happens in the opposite direction. As Dave pointed out, this takes up
option space (one extra byte in AO, in addition to the confirmation
being overloaded with the current keyID) and requires that the option
"protocol" consider how to handle handshake failure, packet reordering, etc.

Joe


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From: "Cui, Cheng" <Cheng.Cui@netapp.com>
To: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
Thread-Topic: [tcpm] question about if a recent Linux patch is compliant with RFC7323 on window scaling
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Subject: Re: [tcpm] question about if a recent Linux patch is compliant with RFC7323 on window scaling
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On 9/4/16, 6:55 AM, "Yoshifumi Nishida" <nishida@sfc.wide.ad.jp> wrote:
> I think the Linux patch at least follows the following principle in RFC79=
3.
>=20
>   The mechanisms provided allow a TCP to advertise a large window and to
>   subsequently advertise a much smaller window without having accepted
>   that much data.  This, so called =B3shrinking the window,=B2 is strongl=
y
>   discouraged.  The robustness principle dictates that TCPs will not
>   shrink the window themselves, but will be prepared for such behavior
>   on the part of other TCPs.
>=20
> I personally think a real question would be how FreeBSD prepares for
> shrinking window.
> Linux chose one way for it, but I guess all implementations don=B9t need
> to follow the same way.
>=20
> --
> Yoshi

Hi Yoshi,

Thanks a lot for the reply.

I agree that's a choice for Linux. But I think FreeBSD has to follow the sa=
me
way; otherwise, Linux has to do something at its sender side to send a corr=
ect
th_seq number.

I think this forum may not be a best place to talk about FreeBSD
implementation. But talking about FreeBSD, I did not find anything in FreeB=
SD
to round-up the shrinking window due to window scale precision loss, neithe=
r
did I find anything about how FreeBSD can tolerate the incoming left-shifte=
d
th_seq ("tp->snd_una+tp->snd_wnd") in an ACK packet sent by Linux in this
particular case, as Linux had sent all the data in a previous larger window=
. I
guess the FreeBSD will just drop the Linux ACK packet as a DUP (see "todrop=
 =3D
tp->rcv_nxt - th->th_seq;").

I also sent a question to the FreeBSD mailing lists. But so far no reply ye=
t:
https://lists.freebsd.org/pipermail/freebsd-transport/2016-August/000141.ht=
ml
https://lists.freebsd.org/pipermail/freebsd-net/2016-September/046048.html

Therefore, I think my best plan is to send separate patches to Linux and
FreeBSD. For Linux, the patch will be correct the th_seq number at sender s=
ide
if window scale precision loss is detected. For FreebSD, I think the patch =
has
to round-up the receive-advertise-window at receiver side.

-- Cheng



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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
Date: Wed, 7 Sep 2016 11:55:38 -0700
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Subject: Re: [tcpm] (ref. RFC 7323) Increase of shift count from 14 to 15.
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Hi David,

> On 9/6/2016 7:24 AM, David Borman wrote:
> To use a shift count of 15, you have to be able to determine whether or n=
ot the other side can accept a value of 15.  If you send a 15, but are
> talking to an older implementation that clamps it at 14 and you don=E2=80=
=99t know that, then your window values, sent assuming a shift count of 15,=
 will
> be halved by the receiver who clamped it at 14; that is probably not desi=
rable as the whole point of have a shift count of 15 is to allow larger
> windows, not smaller windows.

Yes, exactly.
But, I think we might be able to presume people who want to use shift
count 15 will allocate more than 65535 * 2**15 + 65535 * 2**14 bytes
memory for the connection.
This won't be so difficult as I believe they should be able to
allocate at least 65535 * 2 ** 14 bytes memory. Otherwise they don't
need to use shift count 15.
If this is possible, it can always offer 65535 window value when the
peer uses shift count 14.
This allows the peer to use the maximum window size it can handle.

Thanks,
--
Yoshi


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Subject: [tcpm] [Errata Held for Document Update] RFC5961 (4772)
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The following errata report has been held for document update 
for RFC5961, "Improving TCP's Robustness to Blind In-Window Attacks". 

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=5961&eid=4772

--------------------------------------
Status: Held for Document Update
Type: Technical

Reported by: StÃ©phane Bortzmeyer <bortzmeyer+ietf@nic.fr>
Date Reported: 2016-08-10
Held by: Mirja KÃ¼hlewind (IESG)

Section: 7

Original Text
-------------
[The entire section]

Corrected Text
--------------
No suggested text because it requires a much more serious analysis. 
May be adding that the rate-limit counter SHOULD be per-connection, 
in the spirit of RFC 6528?

Notes
-----
It appears the section does not specify that the counter for ACK throttling SHOULD be per-connection. In Linux, it is apparently global, which allowed its use as a side channel enabling nasty attacks (CVE-2016-5696 and the paper "Off-Path TCP Exploits: Global Rate Limit Considered Dangerous" <http://www.cs.ucr.edu/~zhiyunq/pub/sec16_TCP_pure_offpath.pdf>).
Also see discussion on tcpm list about this reported errata!

--------------------------------------
RFC5961 (draft-ietf-tcpm-tcpsecure-13)
--------------------------------------
Title               : Improving TCP's Robustness to Blind In-Window Attacks
Publication Date    : August 2010
Author(s)           : A. Ramaiah, R. Stewart, M. Dalal
Category            : PROPOSED STANDARD
Source              : TCP Maintenance and Minor Extensions
Area                : Transport
Stream              : IETF
Verifying Party     : IESG


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Subject: [tcpm] Fwd: [Technical Errata Reported] RFC0793 (4753)
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FYI.

> Anfang der weitergeleiteten Nachricht:
>=20
> Von: RFC Errata System <rfc-editor@rfc-editor.org>
> Betreff: [Technical Errata Reported] RFC0793 (4753)
> Datum: 30. Juli 2016 um 09:26:29 MESZ
> An: iesg@ietf.org
> Kopie: sanjeev.ranot.81@gmail.com, rfc-editor@rfc-editor.org
>=20
> The following errata report has been submitted for RFC0793,
> "Transmission Control Protocol".
>=20
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=3D793&eid=3D4753
>=20
> --------------------------------------
> Type: Technical
> Reported by: Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
>=20
> Section: 1.5
>=20
> Original Text
> -------------
> A pair of sockets uniquely identifies each connection.
> That is, a socket may be simultaneously used in multiple
> connections
>=20
> Corrected Text
> --------------
> A pair of sockets uniquely identifies each connection. Sockets can be
> classified into client and server sockets. Typically a server socket=20=

> may be simultaneously used in multiple connections.
>=20
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> Notes
> -----
> TCP is connection oriented therefore when we say "sockets used in =
multiple connections" it implies that the context is TCP.  Considering =
their use in TCP,  though a single client socket can be implemented in a =
way to multiplex it for connection with multiple server sockets and =
exchange different SYN segments but then its  same what a server process =
listening for connections on server port does typically.=20
>=20
> I feel classification of sockets here is vital to facilitate =
understand implicitly that in what use-case can
> a socket be typically multiplexed while still keeping generality of =
the statement.
>=20
> Instructions:
> -------------
> This erratum is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary.=20
>=20
> --------------------------------------
> RFC0793 (no draft string recorded)
> --------------------------------------
> Title               : Transmission Control Protocol
> Publication Date    : September 1981
> Author(s)           : J. Postel
> Category            : INTERNET STANDARD
> Source              : Legacy
> Area                : Legacy
> Stream              : IETF
> Verifying Party     : IESG
>=20


From nobody Mon Sep 12 06:42:36 2016
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To: draft-ietf-tcpm-cubic@ietf.org, tcpm IETF list <tcpm@ietf.org>
From: marcelo bagnulo braun <marcelo@it.uc3m.es>
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Date: Mon, 12 Sep 2016 15:42:36 +0200
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Subject: [tcpm] Questions and comments about draft-ietf-tcpm-cubic-02
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Hi,


I have been reading draft-ietf-tcpm-cubic-02 and I have a few questions 
and comments about it that I attach below.

In general, I found the draft easy to follow and overall in good shape.

Dont want to open a can of worms here. If this has been debated before, 
point me to the proper thread and i will check it. I am not sure I 
understand why the intended status of this document is Informational if 
the status of RFC 5681 is Standards Track. Shouldnt this be standards 
track, or experimental if the WG feels that more experience is needed?


In the Introduction, it reads:

    CUBIC maintains the same window growth rate independent of RTTs
    outside of the TCP-friendly region, and flows with different RTTs
    have the similar window sizes under steady state when they operate
    outside the TCP-friendly region.  This ensures CUBIC flows with
    different RTTs to have their bandwidth shares linearly proportional
    to the inverse of their RTT ratio (the longer RTT, the smaller the
    share).

I find this part a bit confusing. I mean, the first two sentences seem to say that in cubic, the rate is almost independent of the RTT and then the last sentence, seem to conclude from the two previous sentences that the longer the RTT, the smaller the rate.

I understand what is meant is that cubic is less biased than classic TCP in terms of RTT but it is still somehow biased, but in the same sense than TCP. If this is the case, the last sentence should be in oposition to the two previous one maybe?

In section 3.1, the formula for K is presented. I understnad this results from including in Eq.1 that W(t=0) is equal to W_max*beta_cubic. Maybe it would be worthwhile stating that explicitly.

Later in section 3.1 it talks about W_cubic(t+RTT). How do you calculate RTT? Is it the RTT of the last ACK? do you do some kind of weighted average?

Section 3.2.
I would like to understand when is the sender int he TCP-friendly region.
If we call t=0 the moment of the last congestion notification (i.e. last loss) and we call t=K the moment when the cwnd is Wmax, then at which point in time is the sender in the TCP-friendly region? I would assume that in an interval centered in t=K, is that the case?
I mean, the way it is described i.e. first the TCP-friendly region and then the concave region and then the convex region, made me thought initially that they were sequential (i.e. at t=0 it starts in the TCP-friendly region and then it moves to the concave region and then to the convex) but I understand this is not the case?
Maybe stating explicitly when the different regions occur may facilitate the reading?

Section 3.3 and 3.4.
I understand that the function that governs the growth in both regions is the same and that the difference is conceptual rather than in the algorithm, correct? Maybe it would be clearer to state this explicitly and maybe merge these two sections.

Also, I think that having one section with a summary of the algorithm, where all the congestion control algorithm is described toghether would be beneficial. Maybe an appendix?

Section 4.3 reads:

    CUBIC is designed to remedy the poor performance of TCP in fast long-
    distance networks.  It is not designed for wireless networks.

I find this very wierd. does this means that cubic doesnt work well if my first link is a wifi or a cellular link?
This is probably/hopefully not what you mean. Maybe you are referring to wireless mesh/ad-hoc networks? If so, I suggest making it more explicit.

Anyway, good work, thanks for doing it.

Regards, marcelo


From nobody Mon Sep 12 07:25:12 2016
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From: David Borman <dab@weston.borman.com>
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To: "Mirja Kuehlewind (IETF)" <ietf@kuehlewind.net>
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Subject: Re: [tcpm] [Technical Errata Reported] RFC0793 (4753)
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I don=E2=80=99t agree with this change.  Sockets are generic; while many =
applications that use sockets put a client/server model on top of =
sockets, that is the semantics of the application, not the underlying =
socket connection.  At the TCP layer the only distinction is that you =
have passive (LISTEN/ACCEPT) and active (CONNECT) sockets.  For example, =
FTP can set up the data socket pair in either direction.

Even with the client/server model, a client can use bind() to use the =
exact same socket (address/port) to connect to multiple machines, so =
there=E2=80=99s no reason to imply that only the server side can use the =
same socket for multiple connections.

And then there is the simultaneous open case, where both sockets use =
CONNECT and there is no distinction between them.

			-David Borman

> On Sep 12, 2016, at 7:33 AM, Mirja Kuehlewind (IETF) =
<ietf@kuehlewind.net> wrote:
>=20
> FYI.
>=20
>> Anfang der weitergeleiteten Nachricht:
>>=20
>> Von: RFC Errata System <rfc-editor@rfc-editor.org>
>> Betreff: [Technical Errata Reported] RFC0793 (4753)
>> Datum: 30. Juli 2016 um 09:26:29 MESZ
>> An: iesg@ietf.org
>> Kopie: sanjeev.ranot.81@gmail.com, rfc-editor@rfc-editor.org
>>=20
>> The following errata report has been submitted for RFC0793,
>> "Transmission Control Protocol".
>>=20
>> --------------------------------------
>> You may review the report below and at:
>> http://www.rfc-editor.org/errata_search.php?rfc=3D793&eid=3D4753
>>=20
>> --------------------------------------
>> Type: Technical
>> Reported by: Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
>>=20
>> Section: 1.5
>>=20
>> Original Text
>> -------------
>> A pair of sockets uniquely identifies each connection.
>> That is, a socket may be simultaneously used in multiple
>> connections
>>=20
>> Corrected Text
>> --------------
>> A pair of sockets uniquely identifies each connection. Sockets can be
>> classified into client and server sockets. Typically a server socket=20=

>> may be simultaneously used in multiple connections.
>>=20
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>> Notes
>> -----
>> TCP is connection oriented therefore when we say "sockets used in =
multiple connections" it implies that the context is TCP.  Considering =
their use in TCP,  though a single client socket can be implemented in a =
way to multiplex it for connection with multiple server sockets and =
exchange different SYN segments but then its  same what a server process =
listening for connections on server port does typically.=20
>>=20
>> I feel classification of sockets here is vital to facilitate =
understand implicitly that in what use-case can
>> a socket be typically multiplexed while still keeping generality of =
the statement.
>>=20
>> Instructions:
>> -------------
>> This erratum is currently posted as "Reported". If necessary, please
>> use "Reply All" to discuss whether it should be verified or
>> rejected. When a decision is reached, the verifying party (IESG)
>> can log in to change the status and edit the report, if necessary.=20
>>=20
>> --------------------------------------
>> RFC0793 (no draft string recorded)
>> --------------------------------------
>> Title               : Transmission Control Protocol
>> Publication Date    : September 1981
>> Author(s)           : J. Postel
>> Category            : INTERNET STANDARD
>> Source              : Legacy
>> Area                : Legacy
>> Stream              : IETF
>> Verifying Party     : IESG
>>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Mon Sep 12 11:19:33 2016
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References: <20160730072629.D628FB80333@rfc-editor.org> <430A1E51-A608-4149-AF68-508D7E15488D@kuehlewind.net>
From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] Fwd: [Technical Errata Reported] RFC0793 (4753)
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Hi, all,

IMO this proposed errata confuses the concept of an OS socket (a
mechanism used to attach network communications to a process) with a TCP
socket (which is an <IP address:port> pair.

A TCP connection is defined by two TCP sockets, one on each end (a
socket pair).

A single TCP socket (the one addr:port pair) can have multiple of either
incoming or outgoing connections or both.

A Unix socket doesn't correspond to a TCP socket pair until after
accept() or connect() returns at best, and may have other semantic
differences.

As a result, this errata should be rejected.

Joe


On 9/12/2016 5:33 AM, Mirja Kuehlewind (IETF) wrote:
> FYI.
>
>> Anfang der weitergeleiteten Nachricht:
>>
>> Von: RFC Errata System <rfc-editor@rfc-editor.org>
>> Betreff: [Technical Errata Reported] RFC0793 (4753)
>> Datum: 30. Juli 2016 um 09:26:29 MESZ
>> An: iesg@ietf.org
>> Kopie: sanjeev.ranot.81@gmail.com, rfc-editor@rfc-editor.org
>>
>> The following errata report has been submitted for RFC0793,
>> "Transmission Control Protocol".
>>
>> --------------------------------------
>> You may review the report below and at:
>> http://www.rfc-editor.org/errata_search.php?rfc=793&eid=4753
>>
>> --------------------------------------
>> Type: Technical
>> Reported by: Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
>>
>> Section: 1.5
>>
>> Original Text
>> -------------
>> A pair of sockets uniquely identifies each connection.
>> That is, a socket may be simultaneously used in multiple
>> connections
>>
>> Corrected Text
>> --------------
>> A pair of sockets uniquely identifies each connection. Sockets can be
>> classified into client and server sockets. Typically a server socket 
>> may be simultaneously used in multiple connections.
>>
>>
>>
>>
>>
>>
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>> Notes
>> -----
>> TCP is connection oriented therefore when we say "sockets used in multiple connections" it implies that the context is TCP.  Considering their use in TCP,  though a single client socket can be implemented in a way to multiplex it for connection with multiple server sockets and exchange different SYN segments but then its  same what a server process listening for connections on server port does typically. 
>>
>> I feel classification of sockets here is vital to facilitate understand implicitly that in what use-case can
>> a socket be typically multiplexed while still keeping generality of the statement.
>>
>> Instructions:
>> -------------
>> This erratum is currently posted as "Reported". If necessary, please
>> use "Reply All" to discuss whether it should be verified or
>> rejected. When a decision is reached, the verifying party (IESG)
>> can log in to change the status and edit the report, if necessary. 
>>
>> --------------------------------------
>> RFC0793 (no draft string recorded)
>> --------------------------------------
>> Title               : Transmission Control Protocol
>> Publication Date    : September 1981
>> Author(s)           : J. Postel
>> Category            : INTERNET STANDARD
>> Source              : Legacy
>> Area                : Legacy
>> Stream              : IETF
>> Verifying Party     : IESG
>>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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Subject: Re: [tcpm] Fwd: [Technical Errata Reported] RFC0793 (4753)
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--001a11370a8a11773a053c53982b
Content-Type: text/plain; charset=UTF-8

i agree w/ Joe and David that this errata is not necessary.

On Mon, Sep 12, 2016 at 11:17 AM, Joe Touch <touch@isi.edu> wrote:

> Hi, all,
>
> IMO this proposed errata confuses the concept of an OS socket (a
> mechanism used to attach network communications to a process) with a TCP
> socket (which is an <IP address:port> pair.
>
> A TCP connection is defined by two TCP sockets, one on each end (a
> socket pair).
>
> A single TCP socket (the one addr:port pair) can have multiple of either
> incoming or outgoing connections or both.
>
> A Unix socket doesn't correspond to a TCP socket pair until after
> accept() or connect() returns at best, and may have other semantic
> differences.
>
> As a result, this errata should be rejected.
>
> Joe
>
>
> On 9/12/2016 5:33 AM, Mirja Kuehlewind (IETF) wrote:
> > FYI.
> >
> >> Anfang der weitergeleiteten Nachricht:
> >>
> >> Von: RFC Errata System <rfc-editor@rfc-editor.org>
> >> Betreff: [Technical Errata Reported] RFC0793 (4753)
> >> Datum: 30. Juli 2016 um 09:26:29 MESZ
> >> An: iesg@ietf.org
> >> Kopie: sanjeev.ranot.81@gmail.com, rfc-editor@rfc-editor.org
> >>
> >> The following errata report has been submitted for RFC0793,
> >> "Transmission Control Protocol".
> >>
> >> --------------------------------------
> >> You may review the report below and at:
> >> http://www.rfc-editor.org/errata_search.php?rfc=793&eid=4753
> >>
> >> --------------------------------------
> >> Type: Technical
> >> Reported by: Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
> >>
> >> Section: 1.5
> >>
> >> Original Text
> >> -------------
> >> A pair of sockets uniquely identifies each connection.
> >> That is, a socket may be simultaneously used in multiple
> >> connections
> >>
> >> Corrected Text
> >> --------------
> >> A pair of sockets uniquely identifies each connection. Sockets can be
> >> classified into client and server sockets. Typically a server socket
> >> may be simultaneously used in multiple connections.
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
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> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >> Notes
> >> -----
> >> TCP is connection oriented therefore when we say "sockets used in
> multiple connections" it implies that the context is TCP.  Considering
> their use in TCP,  though a single client socket can be implemented in a
> way to multiplex it for connection with multiple server sockets and
> exchange different SYN segments but then its  same what a server process
> listening for connections on server port does typically.
> >>
> >> I feel classification of sockets here is vital to facilitate understand
> implicitly that in what use-case can
> >> a socket be typically multiplexed while still keeping generality of the
> statement.
> >>
> >> Instructions:
> >> -------------
> >> This erratum is currently posted as "Reported". If necessary, please
> >> use "Reply All" to discuss whether it should be verified or
> >> rejected. When a decision is reached, the verifying party (IESG)
> >> can log in to change the status and edit the report, if necessary.
> >>
> >> --------------------------------------
> >> RFC0793 (no draft string recorded)
> >> --------------------------------------
> >> Title               : Transmission Control Protocol
> >> Publication Date    : September 1981
> >> Author(s)           : J. Postel
> >> Category            : INTERNET STANDARD
> >> Source              : Legacy
> >> Area                : Legacy
> >> Stream              : IETF
> >> Verifying Party     : IESG
> >>
> > _______________________________________________
> > 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
>

--001a11370a8a11773a053c53982b
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">i agree w/ Joe and David that this errata is not necessary=
.<div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Mon, Sep 12, =
2016 at 11:17 AM, Joe Touch <span dir=3D"ltr">&lt;<a href=3D"mailto:touch@i=
si.edu" target=3D"_blank">touch@isi.edu</a>&gt;</span> wrote:<br><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex">Hi, all,<br>
<br>
IMO this proposed errata confuses the concept of an OS socket (a<br>
mechanism used to attach network communications to a process) with a TCP<br=
>
socket (which is an &lt;IP address:port&gt; pair.<br>
<br>
A TCP connection is defined by two TCP sockets, one on each end (a<br>
socket pair).<br>
<br>
A single TCP socket (the one addr:port pair) can have multiple of either<br=
>
incoming or outgoing connections or both.<br>
<br>
A Unix socket doesn&#39;t correspond to a TCP socket pair until after<br>
accept() or connect() returns at best, and may have other semantic<br>
differences.<br>
<br>
As a result, this errata should be rejected.<br>
<span class=3D"m_-792831573261369468HOEnZb"><font color=3D"#888888"><br>
Joe<br>
</font></span><div class=3D"m_-792831573261369468HOEnZb"><div class=3D"m_-7=
92831573261369468h5"><br>
<br>
On 9/12/2016 5:33 AM, Mirja Kuehlewind (IETF) wrote:<br>
&gt; FYI.<br>
&gt;<br>
&gt;&gt; Anfang der weitergeleiteten Nachricht:<br>
&gt;&gt;<br>
&gt;&gt; Von: RFC Errata System &lt;<a href=3D"mailto:rfc-editor@rfc-editor=
.org" target=3D"_blank">rfc-editor@rfc-editor.org</a>&gt;<br>
&gt;&gt; Betreff: [Technical Errata Reported] RFC0793 (4753)<br>
&gt;&gt; Datum: 30. Juli 2016 um 09:26:29 MESZ<br>
&gt;&gt; An: <a href=3D"mailto:iesg@ietf.org" target=3D"_blank">iesg@ietf.o=
rg</a><br>
&gt;&gt; Kopie: <a href=3D"mailto:sanjeev.ranot.81@gmail.com" target=3D"_bl=
ank">sanjeev.ranot.81@gmail.com</a>, <a href=3D"mailto:rfc-editor@rfc-edito=
r.org" target=3D"_blank">rfc-editor@rfc-editor.org</a><br>
&gt;&gt;<br>
&gt;&gt; The following errata report has been submitted for RFC0793,<br>
&gt;&gt; &quot;Transmission Control Protocol&quot;.<br>
&gt;&gt;<br>
&gt;&gt; ------------------------------<wbr>--------<br>
&gt;&gt; You may review the report below and at:<br>
&gt;&gt; <a href=3D"http://www.rfc-editor.org/errata_search.php?rfc=3D793&a=
mp;eid=3D4753" rel=3D"noreferrer" target=3D"_blank">http://www.rfc-editor.o=
rg/erra<wbr>ta_search.php?rfc=3D793&amp;eid=3D4753</a><br>
&gt;&gt;<br>
&gt;&gt; ------------------------------<wbr>--------<br>
&gt;&gt; Type: Technical<br>
&gt;&gt; Reported by: Sanjeev Ranot &lt;<a href=3D"mailto:sanjeev.ranot.81@=
gmail.com" target=3D"_blank">sanjeev.ranot.81@gmail.com</a>&gt;<br>
&gt;&gt;<br>
&gt;&gt; Section: 1.5<br>
&gt;&gt;<br>
&gt;&gt; Original Text<br>
&gt;&gt; -------------<br>
&gt;&gt; A pair of sockets uniquely identifies each connection.<br>
&gt;&gt; That is, a socket may be simultaneously used in multiple<br>
&gt;&gt; connections<br>
&gt;&gt;<br>
&gt;&gt; Corrected Text<br>
&gt;&gt; --------------<br>
&gt;&gt; A pair of sockets uniquely identifies each connection. Sockets can=
 be<br>
&gt;&gt; classified into client and server sockets. Typically a server sock=
et<br>
&gt;&gt; may be simultaneously used in multiple connections.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; Notes<br>
&gt;&gt; -----<br>
&gt;&gt; TCP is connection oriented therefore when we say &quot;sockets use=
d in multiple connections&quot; it implies that the context is TCP.=C2=A0 C=
onsidering their use in TCP,=C2=A0 though a single client socket can be imp=
lemented in a way to multiplex it for connection with multiple server socke=
ts and exchange different SYN segments but then its=C2=A0 same what a serve=
r process listening for connections on server port does typically.<br>
&gt;&gt;<br>
&gt;&gt; I feel classification of sockets here is vital to facilitate under=
stand implicitly that in what use-case can<br>
&gt;&gt; a socket be typically multiplexed while still keeping generality o=
f the statement.<br>
&gt;&gt;<br>
&gt;&gt; Instructions:<br>
&gt;&gt; -------------<br>
&gt;&gt; This erratum is currently posted as &quot;Reported&quot;. If neces=
sary, please<br>
&gt;&gt; use &quot;Reply All&quot; to discuss whether it should be verified=
 or<br>
&gt;&gt; rejected. When a decision is reached, the verifying party (IESG)<b=
r>
&gt;&gt; can log in to change the status and edit the report, if necessary.=
<br>
&gt;&gt;<br>
&gt;&gt; ------------------------------<wbr>--------<br>
&gt;&gt; RFC0793 (no draft string recorded)<br>
&gt;&gt; ------------------------------<wbr>--------<br>
&gt;&gt; Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: Tran=
smission Control Protocol<br>
&gt;&gt; Publication Date=C2=A0 =C2=A0 : September 1981<br>
&gt;&gt; Author(s)=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: J. Postel<br>
&gt;&gt; Category=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : INTERNET STAND=
ARD<br>
&gt;&gt; Source=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Legacy<br=
>
&gt;&gt; Area=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Lega=
cy<br>
&gt;&gt; Stream=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : IETF<br>
&gt;&gt; Verifying Party=C2=A0 =C2=A0 =C2=A0: IESG<br>
&gt;&gt;<br>
&gt; ______________________________<wbr>_________________<br>
&gt; tcpm mailing list<br>
&gt; <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><b=
r>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferr=
er" target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</a><b=
r>
<br>
______________________________<wbr>_________________<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/l<wbr>istinfo/tcpm</a><br>
</div></div></blockquote></div><br></div></div>

--001a11370a8a11773a053c53982b--


From nobody Mon Sep 12 15:14:06 2016
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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
Date: Mon, 12 Sep 2016 15:13:58 -0700
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To: marcelo bagnulo braun <marcelo@it.uc3m.es>
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Cc: tcpm IETF list <tcpm@ietf.org>, draft-ietf-tcpm-cubic@ietf.org
Subject: Re: [tcpm] Questions and comments about draft-ietf-tcpm-cubic-02
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Hi Marcelo,
Thanks for the review.

On Mon, Sep 12, 2016 at 6:42 AM, marcelo bagnulo braun
<marcelo@it.uc3m.es> wrote:
> Hi,
>
>
> I have been reading draft-ietf-tcpm-cubic-02 and I have a few questions and
> comments about it that I attach below.
>
> In general, I found the draft easy to follow and overall in good shape.
>
> Dont want to open a can of worms here. If this has been debated before,
> point me to the proper thread and i will check it. I am not sure I
> understand why the intended status of this document is Informational if the
> status of RFC 5681 is Standards Track. Shouldnt this be standards track, or
> experimental if the WG feels that more experience is needed?

Yes, we had discussed it before and I believe it is our consensus.
Please check the following threads.

https://www.ietf.org/mail-archive/web/tcpm/current/msg09600.html
https://www.ietf.org/mail-archive/web/tcpm/current/msg09639.html

--
Yoshi


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From: "Mirja Kuehlewind (IETF)" <ietf@kuehlewind.net>
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Subject: Re: [tcpm] [Technical Errata Reported] RFC0793 (4753)
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Hi all,

thanks for your feedback. I also thought that the clarification is at =
least not necessary. I=E2=80=99ll wait a few more days, just to give =
others a chance to comment, and then reject if nobody shouts in the mean =
time.

Mirja


> Am 12.09.2016 um 20:22 schrieb Yuchung Cheng <ycheng@google.com>:
>=20
> i agree w/ Joe and David that this errata is not necessary.
>=20
> On Mon, Sep 12, 2016 at 11:17 AM, Joe Touch <touch@isi.edu> wrote:
> Hi, all,
>=20
> IMO this proposed errata confuses the concept of an OS socket (a
> mechanism used to attach network communications to a process) with a =
TCP
> socket (which is an <IP address:port> pair.
>=20
> A TCP connection is defined by two TCP sockets, one on each end (a
> socket pair).
>=20
> A single TCP socket (the one addr:port pair) can have multiple of =
either
> incoming or outgoing connections or both.
>=20
> A Unix socket doesn't correspond to a TCP socket pair until after
> accept() or connect() returns at best, and may have other semantic
> differences.
>=20
> As a result, this errata should be rejected.
>=20
> Joe
>=20
>=20
> On 9/12/2016 5:33 AM, Mirja Kuehlewind (IETF) wrote:
> > FYI.
> >
> >> Anfang der weitergeleiteten Nachricht:
> >>
> >> Von: RFC Errata System <rfc-editor@rfc-editor.org>
> >> Betreff: [Technical Errata Reported] RFC0793 (4753)
> >> Datum: 30. Juli 2016 um 09:26:29 MESZ
> >> An: iesg@ietf.org
> >> Kopie: sanjeev.ranot.81@gmail.com, rfc-editor@rfc-editor.org
> >>
> >> The following errata report has been submitted for RFC0793,
> >> "Transmission Control Protocol".
> >>
> >> --------------------------------------
> >> You may review the report below and at:
> >> http://www.rfc-editor.org/errata_search.php?rfc=3D793&eid=3D4753
> >>
> >> --------------------------------------
> >> Type: Technical
> >> Reported by: Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
> >>
> >> Section: 1.5
> >>
> >> Original Text
> >> -------------
> >> A pair of sockets uniquely identifies each connection.
> >> That is, a socket may be simultaneously used in multiple
> >> connections
> >>
> >> Corrected Text
> >> --------------
> >> A pair of sockets uniquely identifies each connection. Sockets can =
be
> >> classified into client and server sockets. Typically a server =
socket
> >> may be simultaneously used in multiple connections.
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >> Notes
> >> -----
> >> TCP is connection oriented therefore when we say "sockets used in =
multiple connections" it implies that the context is TCP.  Considering =
their use in TCP,  though a single client socket can be implemented in a =
way to multiplex it for connection with multiple server sockets and =
exchange different SYN segments but then its  same what a server process =
listening for connections on server port does typically.
> >>
> >> I feel classification of sockets here is vital to facilitate =
understand implicitly that in what use-case can
> >> a socket be typically multiplexed while still keeping generality of =
the statement.
> >>
> >> Instructions:
> >> -------------
> >> This erratum is currently posted as "Reported". If necessary, =
please
> >> use "Reply All" to discuss whether it should be verified or
> >> rejected. When a decision is reached, the verifying party (IESG)
> >> can log in to change the status and edit the report, if necessary.
> >>
> >> --------------------------------------
> >> RFC0793 (no draft string recorded)
> >> --------------------------------------
> >> Title               : Transmission Control Protocol
> >> Publication Date    : September 1981
> >> Author(s)           : J. Postel
> >> Category            : INTERNET STANDARD
> >> Source              : Legacy
> >> Area                : Legacy
> >> Stream              : IETF
> >> Verifying Party     : IESG
> >>
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Cc: tcpm IETF list <tcpm@ietf.org>, draft-ietf-tcpm-cubic@ietf.org
Subject: Re: [tcpm] Questions and comments about draft-ietf-tcpm-cubic-02
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Dear Marcel,

Thanks for your comments! I will revise the draft accordingly. Below are 
some short and quick answers.

Lisong


On 9/12/2016 8:42 AM, marcelo bagnulo braun wrote:
> Hi,
>
>
> I have been reading draft-ietf-tcpm-cubic-02 and I have a few 
> questions and comments about it that I attach below.
>
> In general, I found the draft easy to follow and overall in good shape.
>
> Dont want to open a can of worms here. If this has been debated 
> before, point me to the proper thread and i will check it. I am not 
> sure I understand why the intended status of this document is 
> Informational if the status of RFC 5681 is Standards Track. Shouldnt 
> this be standards track, or experimental if the WG feels that more 
> experience is needed?
>
>
> In the Introduction, it reads:
>
>    CUBIC maintains the same window growth rate independent of RTTs
>    outside of the TCP-friendly region, and flows with different RTTs
>    have the similar window sizes under steady state when they operate
>    outside the TCP-friendly region.  This ensures CUBIC flows with
>    different RTTs to have their bandwidth shares linearly proportional
>    to the inverse of their RTT ratio (the longer RTT, the smaller the
>    share).
>
> I find this part a bit confusing. I mean, the first two sentences seem 
> to say that in cubic, the rate is almost independent of the RTT and 
> then the last sentence, seem to conclude from the two previous 
> sentences that the longer the RTT, the smaller the rate.
>

I will revise it to make it more clearly. The first two sentences are 
about the congestion window size, and the last sentence is about the 
throughput=window/RTT.

> I understand what is meant is that cubic is less biased than classic 
> TCP in terms of RTT but it is still somehow biased, but in the same 
> sense than TCP. If this is the case, the last sentence should be in 
> oposition to the two previous one maybe?
>
> In section 3.1, the formula for K is presented. I understnad this 
> results from including in Eq.1 that W(t=0) is equal to 
> W_max*beta_cubic. Maybe it would be worthwhile stating that explicitly.

Thanks
>
> Later in section 3.1 it talks about W_cubic(t+RTT). How do you 
> calculate RTT? Is it the RTT of the last ACK? do you do some kind of 
> weighted average?

We just use the average RTT estimated by TCP.

>
> Section 3.2.
> I would like to understand when is the sender int he TCP-friendly region.
> If we call t=0 the moment of the last congestion notification (i.e. 
> last loss) and we call t=K the moment when the cwnd is Wmax, then at 
> which point in time is the sender in the TCP-friendly region? I would 
> assume that in an interval centered in t=K, is that the case?
> I mean, the way it is described i.e. first the TCP-friendly region and 
> then the concave region and then the convex region, made me thought 
> initially that they were sequential (i.e. at t=0 it starts in the 
> TCP-friendly region and then it moves to the concave region and then 
> to the convex) but I understand this is not the case?
> Maybe stating explicitly when the different regions occur may 
> facilitate the reading?
>
Thanks, I will revise the draft to clearly explain it. Basically, 
TCP-friendly region is determined based on whether W_tcp(t)>W_cubic(t).

> Section 3.3 and 3.4.
> I understand that the function that governs the growth in both regions 
> is the same and that the difference is conceptual rather than in the 
> algorithm, correct? Maybe it would be clearer to state this explicitly 
> and maybe merge these two sections.
>
> Also, I think that having one section with a summary of the algorithm, 
> where all the congestion control algorithm is described toghether 
> would be beneficial. Maybe an appendix?

Yes, Thanks, I will revise draft to clearly explain it.

>
> Section 4.3 reads:
>
>    CUBIC is designed to remedy the poor performance of TCP in fast long-
>    distance networks.  It is not designed for wireless networks.
>
> I find this very wierd. does this means that cubic doesnt work well if 
> my first link is a wifi or a cellular link?
> This is probably/hopefully not what you mean. Maybe you are referring 
> to wireless mesh/ad-hoc networks? If so, I suggest making it more 
> explicit.
>

The traditional TCP does not work well in fast long-distance networks, 
and does not work well in wireless networks. CUBIC only addresses the 
first issue.

Thanks again for your comments.
Lisong

> Anyway, good work, thanks for doing it.
>
> Regards, marcelo
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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To: Lisong Xu <xu@unl.edu>
References: <fa024ff0-3fc9-5c60-6a04-66717e871158@it.uc3m.es> <cf2d24f5-69e6-e80d-5c56-114f69ff5e55@unl.edu>
From: marcelo bagnulo braun <marcelo@it.uc3m.es>
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Date: Tue, 13 Sep 2016 07:50:20 +0200
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Cc: tcpm IETF list <tcpm@ietf.org>, draft-ietf-tcpm-cubic@ietf.org
Subject: Re: [tcpm] Questions and comments about draft-ietf-tcpm-cubic-02
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Thanks for the quick answer.
some replies below...

El 13/09/16 a las 03:22, Lisong Xu escribió:
> Section 3.2.
>> I would like to understand when is the sender int he TCP-friendly 
>> region.
>> If we call t=0 the moment of the last congestion notification (i.e. 
>> last loss) and we call t=K the moment when the cwnd is Wmax, then at 
>> which point in time is the sender in the TCP-friendly region? I would 
>> assume that in an interval centered in t=K, is that the case?
>> I mean, the way it is described i.e. first the TCP-friendly region 
>> and then the concave region and then the convex region, made me 
>> thought initially that they were sequential (i.e. at t=0 it starts in 
>> the TCP-friendly region and then it moves to the concave region and 
>> then to the convex) but I understand this is not the case?
>> Maybe stating explicitly when the different regions occur may 
>> facilitate the reading?
>>
> Thanks, I will revise the draft to clearly explain it. Basically, 
> TCP-friendly region is determined based on whether W_tcp(t)>W_cubic(t).
>

I understood that much :-)
But I would like to have a more intuitive perspective when does that happen.
Maybe an example of a window evolution over time that goes through all 
the 3 phases would be enough?

>
>>
>> Section 4.3 reads:
>>
>>    CUBIC is designed to remedy the poor performance of TCP in fast long-
>>    distance networks.  It is not designed for wireless networks.
>>
>> I find this very wierd. does this means that cubic doesnt work well 
>> if my first link is a wifi or a cellular link?
>> This is probably/hopefully not what you mean. Maybe you are referring 
>> to wireless mesh/ad-hoc networks? If so, I suggest making it more 
>> explicit.
>>
>
> The traditional TCP does not work well in fast long-distance networks, 
> and does not work well in wireless networks.

This last one seems like a too broad statement to me. Maybe including a 
few references about what problems are you referring to and maybe point 
to some solutions that have been developped and deployed would make it 
clearer what you mean?

Thanks again.
Regards, marcelo


> CUBIC only addresses the first issue.
>
> Thanks again for your comments.
> Lisong
>
>> Anyway, good work, thanks for doing it.
>>
>> Regards, marcelo
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>
>


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From: Loganaden Velvindron <loganaden@gmail.com>
Date: Tue, 13 Sep 2016 10:45:39 +0400
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] [Technical Errata Reported] RFC5961 (4772)
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On Fri, Aug 19, 2016 at 5:43 PM, Mirja K=C3=BChlewind
<mirja.kuehlewind@tik.ee.ethz.ch> wrote:
>> sounds fine with me but do documents really get updated?
> That depends on the working group. However the point here is, if the doc =
gets updated, then this is still listed.
>
> Mirja
>
>

I received feedback on the draft I posted. I can update it, based on
the feedback I received so far, assuming there is interest to make the
specs more clear.


From nobody Tue Sep 13 00:13:32 2016
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From: Fernando Gont <fgont@si6networks.com>
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Subject: Re: [tcpm] [Technical Errata Reported] RFC0793 (4753)
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On 09/12/2016 08:32 PM, Mirja Kuehlewind (IETF) wrote:
> Hi all,
> 
> thanks for your feedback. I also thought that the clarification is at least not necessary. Iâ€™ll wait a few more days, just to give others a chance to comment, and then reject if nobody shouts in the mean time.

+1 to reject this errata.

Thanks,
-- 
Fernando Gont
SI6 Networks
e-mail: fgont@si6networks.com
PGP Fingerprint: 6666 31C6 D484 63B2 8FB1 E3C4 AE25 0D55 1D4E 7492





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In-Reply-To: <628F252F-2173-4812-BEC0-4EC4E97E789E@kuehlewind.net>
References: <20160730072629.D628FB80333@rfc-editor.org> <430A1E51-A608-4149-AF68-508D7E15488D@kuehlewind.net> <24a111cf-1e20-bc4e-2a9b-9eb0f3ee5821@isi.edu> <CAK6E8=eem=L+zW_M5+R7+bVTRMq4OBmCx4EsAC+gpEJpfySXsA@mail.gmail.com> <628F252F-2173-4812-BEC0-4EC4E97E789E@kuehlewind.net>
From: Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
Date: Tue, 13 Sep 2016 09:42:02 +0530
Message-ID: <CA+EURmq02MtSghXn7kLBkFtJp59jGn-VYpBgNPDnzOEOwvwUHw@mail.gmail.com>
To: "Mirja Kuehlewind (IETF)" <ietf@kuehlewind.net>
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Cc: dab@weston.borman.com, "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] [Technical Errata Reported] RFC0793 (4753)
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Hi Mirja,

wishing you on occasion of Eid al-Adha.

I accept the team's decision and you are green flag from my side.

The main thought when putting forward this errata was to mention
typical practical use of TCP sockets so that the reader/ implementer
may gain technical clarity. The multiplexing feature of a socket is
realized most in client-server relationship of sockets in a TCP
connection when almost always a server socket gets multiplexed.

With significant duration past by since the RFC was introduced I think the
readers/ implementers with legacy of various sources of information
on sockets and their typical use-case may already have this
clarification and hence render this explanation optional.

Thanks for the team's valuable time and inputs.

Thanks & Regards,
Sanjeev Ranot





On 13 September 2016 at 05:02, Mirja Kuehlewind (IETF) <ietf@kuehlewind.net=
>
wrote:

> Hi all,
>
> thanks for your feedback. I also thought that the clarification is at
> least not necessary. I=E2=80=99ll wait a few more days, just to give othe=
rs a
> chance to comment, and then reject if nobody shouts in the mean time.
>
> Mirja
>
>
> > Am 12.09.2016 um 20:22 schrieb Yuchung Cheng <ycheng@google.com>:
> >
> > i agree w/ Joe and David that this errata is not necessary.
> >
> > On Mon, Sep 12, 2016 at 11:17 AM, Joe Touch <touch@isi.edu> wrote:
> > Hi, all,
> >
> > IMO this proposed errata confuses the concept of an OS socket (a
> > mechanism used to attach network communications to a process) with a TC=
P
> > socket (which is an <IP address:port> pair.
> >
> > A TCP connection is defined by two TCP sockets, one on each end (a
> > socket pair).
> >
> > A single TCP socket (the one addr:port pair) can have multiple of eithe=
r
> > incoming or outgoing connections or both.
> >
> > A Unix socket doesn't correspond to a TCP socket pair until after
> > accept() or connect() returns at best, and may have other semantic
> > differences.
> >
> > As a result, this errata should be rejected.
> >
> > Joe
> >
> >
> > On 9/12/2016 5:33 AM, Mirja Kuehlewind (IETF) wrote:
> > > FYI.
> > >
> > >> Anfang der weitergeleiteten Nachricht:
> > >>
> > >> Von: RFC Errata System <rfc-editor@rfc-editor.org>
> > >> Betreff: [Technical Errata Reported] RFC0793 (4753)
> > >> Datum: 30. Juli 2016 um 09:26:29 MESZ
> > >> An: iesg@ietf.org
> > >> Kopie: sanjeev.ranot.81@gmail.com, rfc-editor@rfc-editor.org
> > >>
> > >> The following errata report has been submitted for RFC0793,
> > >> "Transmission Control Protocol".
> > >>
> > >> --------------------------------------
> > >> You may review the report below and at:
> > >> http://www.rfc-editor.org/errata_search.php?rfc=3D793&eid=3D4753
> > >>
> > >> --------------------------------------
> > >> Type: Technical
> > >> Reported by: Sanjeev Ranot <sanjeev.ranot.81@gmail.com>
> > >>
> > >> Section: 1.5
> > >>
> > >> Original Text
> > >> -------------
> > >> A pair of sockets uniquely identifies each connection.
> > >> That is, a socket may be simultaneously used in multiple
> > >> connections
> > >>
> > >> Corrected Text
> > >> --------------
> > >> A pair of sockets uniquely identifies each connection. Sockets can b=
e
> > >> classified into client and server sockets. Typically a server socket
> > >> may be simultaneously used in multiple connections.
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >> Notes
> > >> -----
> > >> TCP is connection oriented therefore when we say "sockets used in
> multiple connections" it implies that the context is TCP.  Considering
> their use in TCP,  though a single client socket can be implemented in a
> way to multiplex it for connection with multiple server sockets and
> exchange different SYN segments but then its  same what a server process
> listening for connections on server port does typically.
> > >>
> > >> I feel classification of sockets here is vital to facilitate
> understand implicitly that in what use-case can
> > >> a socket be typically multiplexed while still keeping generality of
> the statement.
> > >>
> > >> Instructions:
> > >> -------------
> > >> This erratum is currently posted as "Reported". If necessary, please
> > >> use "Reply All" to discuss whether it should be verified or
> > >> rejected. When a decision is reached, the verifying party (IESG)
> > >> can log in to change the status and edit the report, if necessary.
> > >>
> > >> --------------------------------------
> > >> RFC0793 (no draft string recorded)
> > >> --------------------------------------
> > >> Title               : Transmission Control Protocol
> > >> Publication Date    : September 1981
> > >> Author(s)           : J. Postel
> > >> Category            : INTERNET STANDARD
> > >> Source              : Legacy
> > >> Area                : Legacy
> > >> Stream              : IETF
> > >> Verifying Party     : IESG
> > >>
> > > _______________________________________________
> > > 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
> >
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www.ietf.org/mailman/listinfo/tcpm
>
>

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<div dir=3D"ltr"><div>Hi Mirja,<br><br>wishing you on occasion of Eid al-Ad=
ha.<br><br></div><div>I accept the team&#39;s decision and you are green fl=
ag from my side.<br><br> The main thought when putting forward this errata =
was to mention <br>typical practical use of TCP sockets so that the reader/=
 implementer <br>may gain technical clarity. The multiplexing feature of a =
socket is <br>realized most in client-server relationship of sockets in a T=
CP <br>connection when almost always a server socket gets multiplexed. <br>=
<br>With significant duration past by since the RFC was introduced I think =
the <br>readers/ implementers with legacy of various sources of information=
 <br>on sockets and their typical use-case may already have this <br>clarif=
ication and hence render this explanation optional. <br></div><div></div><d=
iv><br></div><div>Thanks for the team&#39;s valuable time and inputs.<br><b=
r></div><div>Thanks &amp; Regards,<br></div><div>Sanjeev Ranot<br><br></div=
><div><b><br><br><br></b><div class=3D"gmail_extra"><br><div class=3D"gmail=
_quote">On 13 September 2016 at 05:02, Mirja Kuehlewind (IETF) <span dir=3D=
"ltr">&lt;<a target=3D"_blank" href=3D"mailto:ietf@kuehlewind.net">ietf@kue=
hlewind.net</a>&gt;</span> wrote:<br><blockquote style=3D"margin:0px 0px 0p=
x 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class=3D"g=
mail_quote">Hi all,<br>
<br>
thanks for your feedback. I also thought that the clarification is at least=
 not necessary. I=E2=80=99ll wait a few more days, just to give others a ch=
ance to comment, and then reject if nobody shouts in the mean time.<br>
<br>
Mirja<br>
<br>
<br>
&gt; Am 12.09.2016 um 20:22 schrieb Yuchung Cheng &lt;<a href=3D"mailto:ych=
eng@google.com">ycheng@google.com</a>&gt;:<br>
&gt;<br>
&gt; i agree w/ Joe and David that this errata is not necessary.<br>
&gt;<br>
&gt; On Mon, Sep 12, 2016 at 11:17 AM, Joe Touch &lt;<a href=3D"mailto:touc=
h@isi.edu">touch@isi.edu</a>&gt; wrote:<br>
&gt; Hi, all,<br>
&gt;<br>
&gt; IMO this proposed errata confuses the concept of an OS socket (a<br>
&gt; mechanism used to attach network communications to a process) with a T=
CP<br>
&gt; socket (which is an &lt;IP address:port&gt; pair.<br>
&gt;<br>
&gt; A TCP connection is defined by two TCP sockets, one on each end (a<br>
&gt; socket pair).<br>
&gt;<br>
&gt; A single TCP socket (the one addr:port pair) can have multiple of eith=
er<br>
&gt; incoming or outgoing connections or both.<br>
&gt;<br>
&gt; A Unix socket doesn&#39;t correspond to a TCP socket pair until after<=
br>
&gt; accept() or connect() returns at best, and may have other semantic<br>
&gt; differences.<br>
&gt;<br>
&gt; As a result, this errata should be rejected.<br>
&gt;<br>
&gt; Joe<br>
&gt;<br>
&gt;<br>
&gt; On 9/12/2016 5:33 AM, Mirja Kuehlewind (IETF) wrote:<br>
&gt; &gt; FYI.<br>
&gt; &gt;<br>
&gt; &gt;&gt; Anfang der weitergeleiteten Nachricht:<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Von: RFC Errata System &lt;<a href=3D"mailto:rfc-editor@rfc-e=
ditor.org">rfc-editor@rfc-editor.org</a>&gt;<br>
&gt; &gt;&gt; Betreff: [Technical Errata Reported] RFC0793 (4753)<br>
&gt; &gt;&gt; Datum: 30. Juli 2016 um 09:26:29 MESZ<br>
&gt; &gt;&gt; An: <a href=3D"mailto:iesg@ietf.org">iesg@ietf.org</a><br>
&gt; &gt;&gt; Kopie: <a href=3D"mailto:sanjeev.ranot.81@gmail.com">sanjeev.=
ranot.81@gmail.com</a>, <a href=3D"mailto:rfc-editor@rfc-editor.org">rfc-ed=
itor@rfc-editor.org</a><br>
<div><div class=3D"gmail-h5">&gt; &gt;&gt;<br>
&gt; &gt;&gt; The following errata report has been submitted for RFC0793,<b=
r>
&gt; &gt;&gt; &quot;Transmission Control Protocol&quot;.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; ------------------------------<wbr>--------<br>
&gt; &gt;&gt; You may review the report below and at:<br>
&gt; &gt;&gt; <a target=3D"_blank" rel=3D"noreferrer" href=3D"http://www.rf=
c-editor.org/errata_search.php?rfc=3D793&amp;eid=3D4753">http://www.rfc-edi=
tor.org/<wbr>errata_search.php?rfc=3D793&amp;eid=3D<wbr>4753</a><br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; ------------------------------<wbr>--------<br>
&gt; &gt;&gt; Type: Technical<br>
&gt; &gt;&gt; Reported by: Sanjeev Ranot &lt;<a href=3D"mailto:sanjeev.rano=
t.81@gmail.com">sanjeev.ranot.81@gmail.com</a>&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Section: 1.5<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Original Text<br>
&gt; &gt;&gt; -------------<br>
&gt; &gt;&gt; A pair of sockets uniquely identifies each connection.<br>
&gt; &gt;&gt; That is, a socket may be simultaneously used in multiple<br>
&gt; &gt;&gt; connections<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Corrected Text<br>
&gt; &gt;&gt; --------------<br>
&gt; &gt;&gt; A pair of sockets uniquely identifies each connection. Socket=
s can be<br>
&gt; &gt;&gt; classified into client and server sockets. Typically a server=
 socket<br>
&gt; &gt;&gt; may be simultaneously used in multiple connections.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
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&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Notes<br>
&gt; &gt;&gt; -----<br>
&gt; &gt;&gt; TCP is connection oriented therefore when we say &quot;socket=
s used in multiple connections&quot; it implies that the context is TCP.=C2=
=A0 Considering their use in TCP,=C2=A0 though a single client socket can b=
e implemented in a way to multiplex it for connection with multiple server =
sockets and exchange different SYN segments but then its=C2=A0 same what a =
server process listening for connections on server port does typically.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; I feel classification of sockets here is vital to facilitate =
understand implicitly that in what use-case can<br>
&gt; &gt;&gt; a socket be typically multiplexed while still keeping general=
ity of the statement.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Instructions:<br>
&gt; &gt;&gt; -------------<br>
&gt; &gt;&gt; This erratum is currently posted as &quot;Reported&quot;. If =
necessary, please<br>
&gt; &gt;&gt; use &quot;Reply All&quot; to discuss whether it should be ver=
ified or<br>
&gt; &gt;&gt; rejected. When a decision is reached, the verifying party (IE=
SG)<br>
&gt; &gt;&gt; can log in to change the status and edit the report, if neces=
sary.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; ------------------------------<wbr>--------<br>
&gt; &gt;&gt; RFC0793 (no draft string recorded)<br>
&gt; &gt;&gt; ------------------------------<wbr>--------<br>
&gt; &gt;&gt; Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 Transmission Control Protocol<br>
&gt; &gt;&gt; Publication Date=C2=A0 =C2=A0 : September 1981<br>
&gt; &gt;&gt; Author(s)=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: J. Postel=
<br>
&gt; &gt;&gt; Category=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : INTERNET =
STANDARD<br>
&gt; &gt;&gt; Source=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Lega=
cy<br>
&gt; &gt;&gt; Area=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 Legacy<br>
&gt; &gt;&gt; Stream=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : IETF=
<br>
&gt; &gt;&gt; Verifying Party=C2=A0 =C2=A0 =C2=A0: IESG<br>
&gt; &gt;&gt;<br>
</div></div>&gt; &gt; ______________________________<wbr>_________________<=
br>
&gt; &gt; tcpm mailing list<br>
&gt; &gt; <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br>
&gt; &gt; <a target=3D"_blank" rel=3D"noreferrer" href=3D"https://www.ietf.=
org/mailman/listinfo/tcpm">https://www.ietf.org/mailman/<wbr>listinfo/tcpm<=
/a><br>
&gt;<br>
&gt; ______________________________<wbr>_________________<br>
&gt; tcpm mailing list<br>
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From: Greg White <g.white@CableLabs.com>
To: tcpm IETF list <tcpm@ietf.org>, "draft-ietf-tcpm-cubic@ietf.org" <draft-ietf-tcpm-cubic@ietf.org>
Thread-Topic: some editorial comments on draft-ietf-tcpm-cubic-02
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Date: Tue, 13 Sep 2016 14:39:18 -0700
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Subject: Re: [tcpm] some editorial comments on draft-ietf-tcpm-cubic-02
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On Tue, Sep 13, 2016 at 12:02 PM, Greg White <g.white@cablelabs.com> wrote:
>
> Seeing Marcelo=E2=80=99s comments prompted me to read the draft as well. =
 I have just a few comments, and also I found a couple of typos.
>
>
>
>
>
> Comment 1:  Section 4.8 contains a lower case =E2=80=9Cshould not=E2=80=
=9D.  This seems like it should be a =E2=80=9CSHOULD NOT=E2=80=9D.
>
>
>
>    In cases of idle periods, t in Eq. 1 SHOULD NOT include the
>
>    idle time; otherwise, W_cubic(t) might be very high after restarting
>
>    from a long idle time.
why SHOULD vs MUST? I can't think of any corner case that we'd want to
include the idle time.

>
>
>
>
>
>
>
> Comment 2:  Section 4.3 reads:
>
>
>
> 4.3.  Difficult Environments
>
>
>
>    CUBIC is designed to remedy the poor performance of TCP in fast long-
>
>    distance networks.  It is not designed for wireless networks.
>
>
>
> Since a large proportion of TCP flows today cross a wireless link (WiFi o=
r LTE), this statement and the heading seem kind of anachronistic.   Perhap=
s it would be better to describe what some of the concerns might be with op=
eration across today=E2=80=99s networks that are likely to contain wireless=
 links (if there are any), or state what properties of a wireless network w=
ould cause problems for CUBIC.
>
>
>
>
>
>
>
> Comment 3: Section 4.5 states:
>
>
>
>    In case that there is congestion collapse, CUBIC behaves likely
>
>    standard TCP ...
>
>
>
> I believe that the whole point of congestion avoidance algorithms such as=
 CUBIC is to prevent congestion collapse.  (Also note the typo =E2=80=9Clik=
ely=E2=80=9D in that sentence.)  Maybe this would be better written as:
>
>
>
>    With regard to the potential of causing congestion collapse,
>
>    CUBIC behaves like standard TCP ...
>
>
>
>
>
>
>
> Typo:
>
>
>
> Section 3.2, first sentence reads:
>
>
>
>   When receiving an ACK in congestion avoidance, we first check whether
>
>    the protocol is in the TCP region or not.
>
>
>
> I believe it should be:
>
>
>
>    When receiving an ACK in congestion avoidance, we first check whether
>
>    the protocol is in the TCP **friendly** region or not.
>
>
>
>
>
>
>
> Best Regards,
>
> Greg
>
>
>
>
>
>
> _______________________________________________
> tcpm mailing list
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To: Greg White <g.white@CableLabs.com>, tcpm IETF list <tcpm@ietf.org>, "draft-ietf-tcpm-cubic@ietf.org" <draft-ietf-tcpm-cubic@ietf.org>
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From: Lisong Xu <xu@unl.edu>
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Subject: Re: [tcpm] some editorial comments on draft-ietf-tcpm-cubic-02
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Thanks, Greg, for your comments!

Lisong


On 9/13/2016 2:02 PM, Greg White wrote:
>
> Seeing Marcelo’s comments prompted me to read the draft as well.  I 
> have just a few comments, and also I found a couple of typos.
>
> Comment 1: Section 4.8 contains a lower case “should not”.  This seems 
> like it should be a “SHOULD NOT”.
>
>    In cases of idle periods, t in Eq. 1 SHOULD NOT include the
>
>    idle time; otherwise, W_cubic(t) might be very high after restarting
>
>    from a long idle time.
>
> Comment 2: Section 4.3 reads:
>
> 4.3.  Difficult Environments
>
>    CUBIC is designed to remedy the poor performance of TCP in fast long-
>
>    distance networks.  It is not designed for wireless networks.
>
> Since a large proportion of TCP flows today cross a wireless link 
> (WiFi or LTE), this statement and the heading seem kind of 
> anachronistic.   Perhaps it would be better to describe what some of 
> the concerns might be with operation across today’s networks that are 
> likely to contain wireless links (if there are any), or state what 
> properties of a wireless network would cause problems for CUBIC.
>
> Comment 3: Section 4.5 states:
>
>    In case that there is congestion collapse, CUBIC behaves likely
>
>    standard TCP ...
>
> I believe that the whole point of congestion avoidance algorithms such 
> as CUBIC is to prevent congestion collapse.  (Also note the typo 
> “likely” in that sentence.)  Maybe this would be better written as:
>
> With regard to the potential of causing congestion collapse,
>
>    CUBIC behaves like standard TCP ...
>
> Typo:
>
> Section 3.2, first sentence reads:
>
>   When receiving an ACK in congestion avoidance, we first check whether
>
> the protocol is in the TCP region or not.
>
> I believe it should be:
>
> When receiving an ACK in congestion avoidance, we first check whether
>
> the protocol is in the TCP **friendly** region or not.
>
> Best Regards,
>
> Greg
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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    <p>Thanks, Greg, for your comments!</p>
    <p>Lisong<br>
    </p>
    <br>
    <div class="moz-cite-prefix">On 9/13/2016 2:02 PM, Greg White wrote:<br>
    </div>
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      <div class="WordSection1">
        <p class="MsoNormal"><span style="font-size:11.0pt">Seeing
            Marcelo’s comments prompted me to read the draft as well.  I
            have just a few comments, and also I found a couple of
            typos.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt">Comment 1: 
            Section 4.8 contains a lower case “should not”.  This seems
            like it should be a “SHOULD NOT”.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal" style="text-autospace:none"><span
            style="font-size:10.0pt;font-family:&quot;PT
            Mono&quot;;color:#1A1A1A">   In cases of idle periods, t in
            Eq. 1 SHOULD NOT include the<o:p></o:p></span></p>
        <p class="MsoNormal" style="text-autospace:none"><span
            style="font-size:10.0pt;font-family:&quot;PT
            Mono&quot;;color:#1A1A1A">   idle time; otherwise,
            W_cubic(t) might be very high after restarting<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;PT
            Mono&quot;;color:#1A1A1A">   from a long idle time.</span><span
            style="font-size:10.0pt"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt">Comment 2: 
            Section 4.3 reads:
            <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal" style="text-autospace:none"><span
            style="font-size:10.0pt;font-family:&quot;PT
            Mono&quot;;color:#1A1A1A">4.3.  Difficult Environments<o:p></o:p></span></p>
        <p class="MsoNormal" style="text-autospace:none"><span
            style="font-size:10.0pt;font-family:&quot;PT
            Mono&quot;;color:#1A1A1A"><o:p> </o:p></span></p>
        <p class="MsoNormal" style="text-autospace:none"><span
            style="font-size:10.0pt;font-family:&quot;PT
            Mono&quot;;color:#1A1A1A">   CUBIC is designed to remedy the
            poor performance of TCP in fast long-<o:p></o:p></span></p>
        <p class="MsoNormal" style="text-autospace:none"><span
            style="font-size:10.0pt;font-family:&quot;PT
            Mono&quot;;color:#1A1A1A">   distance networks.  It is not
            designed for wireless networks.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:10.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt">Since a
            large proportion of TCP flows today cross a wireless link
            (WiFi or LTE), this statement and the heading seem kind of
            anachronistic.   Perhaps it would be better to describe what
            some of the concerns might be with operation across today’s
            networks that are likely to contain wireless links (if there
            are any), or state what properties of a wireless network
            would cause problems for CUBIC.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt">Comment 3:
            Section 4.5 states:<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal" style="text-autospace:none"><span
            style="font-size:10.0pt;font-family:&quot;PT
            Mono&quot;;color:#1A1A1A">   In case that there is
            congestion collapse, CUBIC behaves likely<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;PT
            Mono&quot;;color:#1A1A1A">   standard TCP ...</span><span
            style="font-size:10.0pt"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt">I believe
            that the whole point of congestion avoidance algorithms such
            as CUBIC is to prevent congestion collapse.  (Also note the
            typo “likely” in that sentence.)  Maybe this would be better
            written as: 
            <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;PT Mono&quot;">  
            With regard to the potential of causing congestion collapse,
            <o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;PT Mono&quot;">   CUBIC
            behaves like standard TCP ...<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoPlainText">Typo:<o:p></o:p></p>
        <p class="MsoPlainText"><o:p> </o:p></p>
        <p class="MsoPlainText">Section 3.2, first sentence reads:  <o:p></o:p></p>
        <p class="MsoPlainText"><o:p> </o:p></p>
        <p class="MsoPlainText"><span
            style="font-size:10.0pt;font-family:&quot;PT Mono&quot;">  When
            receiving an ACK in congestion avoidance, we first check
            whether<o:p></o:p></span></p>
        <p class="MsoPlainText"><span
            style="font-size:10.0pt;font-family:&quot;PT Mono&quot;">  
            the protocol is in the TCP region or not.<o:p></o:p></span></p>
        <p class="MsoPlainText"><o:p> </o:p></p>
        <p class="MsoPlainText">I believe it should be:<o:p></o:p></p>
        <p class="MsoPlainText"><o:p> </o:p></p>
        <p class="MsoPlainText"><span
            style="font-size:10.0pt;font-family:&quot;PT Mono&quot;">  
            When receiving an ACK in congestion avoidance, we first
            check whether<o:p></o:p></span></p>
        <p class="MsoPlainText"><span
            style="font-size:10.0pt;font-family:&quot;PT Mono&quot;">  
            the protocol is in the TCP **<span
              style="background:yellow;mso-highlight:yellow">friendly</span>**
            region or not.<o:p></o:p></span></p>
        <p class="MsoPlainText"><o:p> </o:p></p>
        <p class="MsoPlainText"><o:p> </o:p></p>
        <p class="MsoPlainText"><o:p> </o:p></p>
        <p class="MsoPlainText">Best Regards,<o:p></o:p></p>
        <p class="MsoPlainText">Greg<o:p></o:p></p>
        <p class="MsoPlainText"><o:p> </o:p></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
tcpm mailing list
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</pre>
    </blockquote>
    <br>
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Cc: tcpm IETF list <tcpm@ietf.org>, "draft-ietf-tcpm-cubic@ietf.org" <draft-ietf-tcpm-cubic@ietf.org>
Subject: Re: [tcpm] some editorial comments on draft-ietf-tcpm-cubic-02
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Thanks, Yuchung!  If the idle time is very short, it might be ok to 
include the idle time?
Lisong


On 9/13/2016 4:39 PM, Yuchung Cheng wrote:
> On Tue, Sep 13, 2016 at 12:02 PM, Greg White <g.white@cablelabs.com> wrote:
>> Seeing Marceloâ€™s comments prompted me to read the draft as well.  I have just a few comments, and also I found a couple of typos.
>>
>>
>>
>>
>>
>> Comment 1:  Section 4.8 contains a lower case â€œshould notâ€�.  This seems like it should be a â€œSHOULD NOTâ€�.
>>
>>
>>
>>     In cases of idle periods, t in Eq. 1 SHOULD NOT include the
>>
>>     idle time; otherwise, W_cubic(t) might be very high after restarting
>>
>>     from a long idle time.
> why SHOULD vs MUST? I can't think of any corner case that we'd want to
> include the idle time.
>
>>
>>
>>
>>
>>
>>
>> Comment 2:  Section 4.3 reads:
>>
>>
>>
>> 4.3.  Difficult Environments
>>
>>
>>
>>     CUBIC is designed to remedy the poor performance of TCP in fast long-
>>
>>     distance networks.  It is not designed for wireless networks.
>>
>>
>>
>> Since a large proportion of TCP flows today cross a wireless link (WiFi or LTE), this statement and the heading seem kind of anachronistic.   Perhaps it would be better to describe what some of the concerns might be with operation across todayâ€™s networks that are likely to contain wireless links (if there are any), or state what properties of a wireless network would cause problems for CUBIC.
>>
>>
>>
>>
>>
>>
>>
>> Comment 3: Section 4.5 states:
>>
>>
>>
>>     In case that there is congestion collapse, CUBIC behaves likely
>>
>>     standard TCP ...
>>
>>
>>
>> I believe that the whole point of congestion avoidance algorithms such as CUBIC is to prevent congestion collapse.  (Also note the typo â€œlikelyâ€� in that sentence.)  Maybe this would be better written as:
>>
>>
>>
>>     With regard to the potential of causing congestion collapse,
>>
>>     CUBIC behaves like standard TCP ...
>>
>>
>>
>>
>>
>>
>>
>> Typo:
>>
>>
>>
>> Section 3.2, first sentence reads:
>>
>>
>>
>>    When receiving an ACK in congestion avoidance, we first check whether
>>
>>     the protocol is in the TCP region or not.
>>
>>
>>
>> I believe it should be:
>>
>>
>>
>>     When receiving an ACK in congestion avoidance, we first check whether
>>
>>     the protocol is in the TCP **friendly** region or not.
>>
>>
>>
>>
>>
>>
>>
>> Best Regards,
>>
>> Greg
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> 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


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From: Yuchung Cheng <ycheng@google.com>
Date: Tue, 13 Sep 2016 21:19:00 -0700
Message-ID: <CAK6E8=ckOM+70QThq2Y+h1ER15nSnshh69nXKUBQoJDoLpnY0g@mail.gmail.com>
To: Lisong Xu <xu@unl.edu>
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Cc: tcpm IETF list <tcpm@ietf.org>, "draft-ietf-tcpm-cubic@ietf.org" <draft-ietf-tcpm-cubic@ietf.org>
Subject: Re: [tcpm] some editorial comments on draft-ietf-tcpm-cubic-02
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On Tue, Sep 13, 2016 at 7:26 PM, Lisong Xu <xu@unl.edu> wrote:

> Thanks, Yuchung!  If the idle time is very short, it might be ok to
> include the idle time?
> Lisong

Sure but to me a SHOULD means there is some legit cases where we would
include idle time. Since any system has finite timer granularity, the case
you mentioned does not violate MUST imo.


>
>
> On 9/13/2016 4:39 PM, Yuchung Cheng wrote:
>
>> On Tue, Sep 13, 2016 at 12:02 PM, Greg White <g.white@cablelabs.com>
>> wrote:
>>
>>> Seeing Marcelo=E2=80=99s comments prompted me to read the draft as well=
.  I have
>>> just a few comments, and also I found a couple of typos.
>>>
>>>
>>>
>>>
>>>
>>> Comment 1:  Section 4.8 contains a lower case =E2=80=9Cshould not=E2=80=
=9D.  This seems
>>> like it should be a =E2=80=9CSHOULD NOT=E2=80=9D.
>>>
>>>
>>>
>>>     In cases of idle periods, t in Eq. 1 SHOULD NOT include the
>>>
>>>     idle time; otherwise, W_cubic(t) might be very high after restartin=
g
>>>
>>>     from a long idle time.
>>>
>> why SHOULD vs MUST? I can't think of any corner case that we'd want to
>> include the idle time.
>>
>>
>>>
>>>
>>>
>>>
>>>
>>> Comment 2:  Section 4.3 reads:
>>>
>>>
>>>
>>> 4.3.  Difficult Environments
>>>
>>>
>>>
>>>     CUBIC is designed to remedy the poor performance of TCP in fast lon=
g-
>>>
>>>     distance networks.  It is not designed for wireless networks.
>>>
>>>
>>>
>>> Since a large proportion of TCP flows today cross a wireless link (WiFi
>>> or LTE), this statement and the heading seem kind of anachronistic.
>>>  Perhaps it would be better to describe what some of the concerns might=
 be
>>> with operation across today=E2=80=99s networks that are likely to conta=
in wireless
>>> links (if there are any), or state what properties of a wireless networ=
k
>>> would cause problems for CUBIC.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> Comment 3: Section 4.5 states:
>>>
>>>
>>>
>>>     In case that there is congestion collapse, CUBIC behaves likely
>>>
>>>     standard TCP ...
>>>
>>>
>>>
>>> I believe that the whole point of congestion avoidance algorithms such
>>> as CUBIC is to prevent congestion collapse.  (Also note the typo =E2=80=
=9Clikely=E2=80=9D
>>> in that sentence.)  Maybe this would be better written as:
>>>
>>>
>>>
>>>     With regard to the potential of causing congestion collapse,
>>>
>>>     CUBIC behaves like standard TCP ...
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> Typo:
>>>
>>>
>>>
>>> Section 3.2, first sentence reads:
>>>
>>>
>>>
>>>    When receiving an ACK in congestion avoidance, we first check whethe=
r
>>>
>>>     the protocol is in the TCP region or not.
>>>
>>>
>>>
>>> I believe it should be:
>>>
>>>
>>>
>>>     When receiving an ACK in congestion avoidance, we first check wheth=
er
>>>
>>>     the protocol is in the TCP **friendly** region or not.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> Best Regards,
>>>
>>> Greg
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>>
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Tue, Sep 13, 2016 at 7:26 PM, Lisong Xu <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:xu@unl.edu" target=3D"_blank">xu@unl.edu</a>&gt;</span> wrote:<=
br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex">Thanks, Yuchung!=C2=A0 If the idle time i=
s very short, it might be ok to include the idle time?<span class=3D"HOEnZb=
"><font color=3D"#888888"><br>
Lisong</font></span></blockquote><div>Sure but to me a SHOULD means there i=
s some legit cases where we would include idle time. Since any system has f=
inite timer granularity, the case you mentioned does not violate MUST imo.=
=C2=A0</div><div><br></div><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class=3D"HOE=
nZb"><div class=3D"h5"><br>
<br>
<br>
On 9/13/2016 4:39 PM, Yuchung Cheng wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On Tue, Sep 13, 2016 at 12:02 PM, Greg White &lt;<a href=3D"mailto:g.white@=
cablelabs.com" target=3D"_blank">g.white@cablelabs.com</a>&gt; wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Seeing Marcelo=E2=80=99s comments prompted me to read the draft as well.=C2=
=A0 I have just a few comments, and also I found a couple of typos.<br>
<br>
<br>
<br>
<br>
<br>
Comment 1:=C2=A0 Section 4.8 contains a lower case =E2=80=9Cshould not=E2=
=80=9D.=C2=A0 This seems like it should be a =E2=80=9CSHOULD NOT=E2=80=9D.<=
br>
<br>
<br>
<br>
=C2=A0 =C2=A0 In cases of idle periods, t in Eq. 1 SHOULD NOT include the<b=
r>
<br>
=C2=A0 =C2=A0 idle time; otherwise, W_cubic(t) might be very high after res=
tarting<br>
<br>
=C2=A0 =C2=A0 from a long idle time.<br>
</blockquote>
why SHOULD vs MUST? I can&#39;t think of any corner case that we&#39;d want=
 to<br>
include the idle time.<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
<br>
<br>
<br>
<br>
Comment 2:=C2=A0 Section 4.3 reads:<br>
<br>
<br>
<br>
4.3.=C2=A0 Difficult Environments<br>
<br>
<br>
<br>
=C2=A0 =C2=A0 CUBIC is designed to remedy the poor performance of TCP in fa=
st long-<br>
<br>
=C2=A0 =C2=A0 distance networks.=C2=A0 It is not designed for wireless netw=
orks.<br>
<br>
<br>
<br>
Since a large proportion of TCP flows today cross a wireless link (WiFi or =
LTE), this statement and the heading seem kind of anachronistic.=C2=A0 =C2=
=A0Perhaps it would be better to describe what some of the concerns might b=
e with operation across today=E2=80=99s networks that are likely to contain=
 wireless links (if there are any), or state what properties of a wireless =
network would cause problems for CUBIC.<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
Comment 3: Section 4.5 states:<br>
<br>
<br>
<br>
=C2=A0 =C2=A0 In case that there is congestion collapse, CUBIC behaves like=
ly<br>
<br>
=C2=A0 =C2=A0 standard TCP ...<br>
<br>
<br>
<br>
I believe that the whole point of congestion avoidance algorithms such as C=
UBIC is to prevent congestion collapse.=C2=A0 (Also note the typo =E2=80=9C=
likely=E2=80=9D in that sentence.)=C2=A0 Maybe this would be better written=
 as:<br>
<br>
<br>
<br>
=C2=A0 =C2=A0 With regard to the potential of causing congestion collapse,<=
br>
<br>
=C2=A0 =C2=A0 CUBIC behaves like standard TCP ...<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
Typo:<br>
<br>
<br>
<br>
Section 3.2, first sentence reads:<br>
<br>
<br>
<br>
=C2=A0 =C2=A0When receiving an ACK in congestion avoidance, we first check =
whether<br>
<br>
=C2=A0 =C2=A0 the protocol is in the TCP region or not.<br>
<br>
<br>
<br>
I believe it should be:<br>
<br>
<br>
<br>
=C2=A0 =C2=A0 When receiving an ACK in congestion avoidance, we first check=
 whether<br>
<br>
=C2=A0 =C2=A0 the protocol is in the TCP **friendly** region or not.<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
Best Regards,<br>
<br>
Greg<br>
<br>
<br>
<br>
<br>
<br>
<br>
______________________________<wbr>_________________<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/l<wbr>istinfo/tcpm</a><br>
<br>
</blockquote>
______________________________<wbr>_________________<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/l<wbr>istinfo/tcpm</a><br>
</blockquote>
<br>
</div></div></blockquote></div><br></div></div>

--94eb2c055ea4bbed3c053c700955--


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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>, "g.white@CableLabs.com" <g.white@CableLabs.com>
Thread-Topic: Cubic and LTE (was RE: some editorial comments on draft-ietf-tcpm-cubic-02)
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Cc: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
Subject: [tcpm] Cubic and LTE (was RE: some editorial comments on draft-ietf-tcpm-cubic-02)
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Hi

A comment on section 4.3 "Difficult environments"
Based on LTE system simulation experience, I don't find Cubic to be worse t=
han e.g. Reno, we see occasional issues with HyStart but that is another ma=
tter. With the use of proper AQM (or ECN) I can see that Cubic is actually =
quite well behaved.
The root problem is that LTE networks seem to be bufferbloated today, AQMs =
are in general not deployed or enabled. The outcome is that transport proto=
cols with Cubic or Reno, or any other other loss based congestion control a=
lgorithm for that matter, can be claimed to operate in congestion avoidance=
. The results can be RTT in the order of seconds which may or may not cause=
 problems with RTO and/or large burst losses at handover depending on the d=
eployment of packet forwarding in LTE networks.
So at least my conclusion, based on my own experience is that Cubic is not =
the big problem here.

Regards
Ingemar

>=20
> Message: 1
> Date: Tue, 13 Sep 2016 19:02:26 +0000
> From: Greg White <g.white@CableLabs.com>
>=20
>=20
>=20
> Comment 2:  Section 4.3 reads:
>=20
> 4.3.  Difficult Environments
>=20
>    CUBIC is designed to remedy the poor performance of TCP in fast long-
>    distance networks.  It is not designed for wireless networks.
>=20
> Since a large proportion of TCP flows today cross a wireless link (WiFi o=
r LTE),
> this statement and the heading seem kind of anachronistic.   Perhaps it
> would be better to describe what some of the concerns might be with
> operation across today?s networks that are likely to contain wireless lin=
ks (if
> there are any), or state what properties of a wireless network would caus=
e
> problems for CUBIC.
>=20
> ************************************


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Archived-At: <https://mailarchive.ietf.org/arch/msg/tcpm/vS5fbPoZs5Y7t_IIMAwhpUVe3xY>
Subject: Re: [tcpm] Cubic and LTE (was RE: some editorial comments on draft-ietf-tcpm-cubic-02)
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Thanks, Ingemar, for your comments.

Lisong


On 9/14/2016 2:35 AM, Ingemar Johansson S wrote:
> Hi
>
> A comment on section 4.3 "Difficult environments"
> Based on LTE system simulation experience, I don't find Cubic to be worse than e.g. Reno, we see occasional issues with HyStart but that is another matter. With the use of proper AQM (or ECN) I can see that Cubic is actually quite well behaved.
> The root problem is that LTE networks seem to be bufferbloated today, AQMs are in general not deployed or enabled. The outcome is that transport protocols with Cubic or Reno, or any other other loss based congestion control algorithm for that matter, can be claimed to operate in congestion avoidance. The results can be RTT in the order of seconds which may or may not cause problems with RTO and/or large burst losses at handover depending on the deployment of packet forwarding in LTE networks.
> So at least my conclusion, based on my own experience is that Cubic is not the big problem here.
>
> Regards
> Ingemar
>
>> Message: 1
>> Date: Tue, 13 Sep 2016 19:02:26 +0000
>> From: Greg White <g.white@CableLabs.com>
>>
>>
>>
>> Comment 2:  Section 4.3 reads:
>>
>> 4.3.  Difficult Environments
>>
>>     CUBIC is designed to remedy the poor performance of TCP in fast long-
>>     distance networks.  It is not designed for wireless networks.
>>
>> Since a large proportion of TCP flows today cross a wireless link (WiFi or LTE),
>> this statement and the heading seem kind of anachronistic.   Perhaps it
>> would be better to describe what some of the concerns might be with
>> operation across today?s networks that are likely to contain wireless links (if
>> there are any), or state what properties of a wireless network would cause
>> problems for CUBIC.
>>
>> ************************************
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Cc: tcpm IETF list <tcpm@ietf.org>, "draft-ietf-tcpm-cubic@ietf.org" <draft-ietf-tcpm-cubic@ietf.org>
Subject: Re: [tcpm] some editorial comments on draft-ietf-tcpm-cubic-02
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Thanks, Yuchung. I think you are right that we should change it to "Must 
Not".

Lisong



On 9/13/2016 11:19 PM, Yuchung Cheng wrote:
>
>
> On Tue, Sep 13, 2016 at 7:26 PM, Lisong Xu <xu@unl.edu 
> <mailto:xu@unl.edu>> wrote:
>
>     Thanks, Yuchung!  If the idle time is very short, it might be ok
>     to include the idle time?
>     Lisong
>
> Sure but to me a SHOULD means there is some legit cases where we would 
> include idle time. Since any system has finite timer granularity, the 
> case you mentioned does not violate MUST imo.
>
>
>
>
>     On 9/13/2016 4:39 PM, Yuchung Cheng wrote:
>
>         On Tue, Sep 13, 2016 at 12:02 PM, Greg White
>         <g.white@cablelabs.com <mailto:g.white@cablelabs.com>> wrote:
>
>             Seeing Marceloâ€™s comments prompted me to read the draft as
>             well.  I have just a few comments, and also I found a
>             couple of typos.
>
>
>
>
>
>             Comment 1:  Section 4.8 contains a lower case â€œshould
>             notâ€�.  This seems like it should be a â€œSHOULD NOTâ€�.
>
>
>
>                 In cases of idle periods, t in Eq. 1 SHOULD NOT
>             include the
>
>                 idle time; otherwise, W_cubic(t) might be very high
>             after restarting
>
>                 from a long idle time.
>
>         why SHOULD vs MUST? I can't think of any corner case that we'd
>         want to
>         include the idle time.
>
>
>
>
>
>
>
>             Comment 2:  Section 4.3 reads:
>
>
>
>             4.3.  Difficult Environments
>
>
>
>                 CUBIC is designed to remedy the poor performance of
>             TCP in fast long-
>
>                 distance networks.  It is not designed for wireless
>             networks.
>
>
>
>             Since a large proportion of TCP flows today cross a
>             wireless link (WiFi or LTE), this statement and the
>             heading seem kind of anachronistic.   Perhaps it would be
>             better to describe what some of the concerns might be with
>             operation across todayâ€™s networks that are likely to
>             contain wireless links (if there are any), or state what
>             properties of a wireless network would cause problems for
>             CUBIC.
>
>
>
>
>
>
>
>             Comment 3: Section 4.5 states:
>
>
>
>                 In case that there is congestion collapse, CUBIC
>             behaves likely
>
>                 standard TCP ...
>
>
>
>             I believe that the whole point of congestion avoidance
>             algorithms such as CUBIC is to prevent congestion
>             collapse.  (Also note the typo â€œlikelyâ€� in that
>             sentence.)  Maybe this would be better written as:
>
>
>
>                 With regard to the potential of causing congestion
>             collapse,
>
>                 CUBIC behaves like standard TCP ...
>
>
>
>
>
>
>
>             Typo:
>
>
>
>             Section 3.2, first sentence reads:
>
>
>
>                When receiving an ACK in congestion avoidance, we first
>             check whether
>
>                 the protocol is in the TCP region or not.
>
>
>
>             I believe it should be:
>
>
>
>                 When receiving an ACK in congestion avoidance, we
>             first check whether
>
>                 the protocol is in the TCP **friendly** region or not.
>
>
>
>
>
>
>
>             Best Regards,
>
>             Greg
>
>
>
>
>
>
>             _______________________________________________
>             tcpm mailing list
>             tcpm@ietf.org <mailto:tcpm@ietf.org>
>             https://www.ietf.org/mailman/listinfo/tcpm
>             <https://www.ietf.org/mailman/listinfo/tcpm>
>
>         _______________________________________________
>         tcpm mailing list
>         tcpm@ietf.org <mailto:tcpm@ietf.org>
>         https://www.ietf.org/mailman/listinfo/tcpm
>         <https://www.ietf.org/mailman/listinfo/tcpm>
>
>
>


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    <p>Thanks, Yuchung. I think you are right that we should change it
      to "Must Not".</p>
    <p>Lisong</p>
    <p><br>
    </p>
    <br>
    <div class="moz-cite-prefix">On 9/13/2016 11:19 PM, Yuchung Cheng
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAK6E8=ckOM+70QThq2Y+h1ER15nSnshh69nXKUBQoJDoLpnY0g@mail.gmail.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html; charset=utf-8">
      <div dir="ltr"><br>
        <div class="gmail_extra"><br>
          <div class="gmail_quote">On Tue, Sep 13, 2016 at 7:26 PM,
            Lisong Xu <span dir="ltr">&lt;<a moz-do-not-send="true"
                href="mailto:xu@unl.edu" target="_blank">xu@unl.edu</a>&gt;</span>
            wrote:<br>
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">Thanks,
              Yuchung!Â  If the idle time is very short, it might be ok
              to include the idle time?<span class="HOEnZb"><font
                  color="#888888"><br>
                  Lisong</font></span></blockquote>
            <div>Sure but to me a SHOULD means there is some legit cases
              where we would include idle time. Since any system has
              finite timer granularity, the case you mentioned does not
              violate MUST imo.Â </div>
            <div><br>
            </div>
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">
              <div class="HOEnZb">
                <div class="h5"><br>
                  <br>
                  <br>
                  On 9/13/2016 4:39 PM, Yuchung Cheng wrote:<br>
                  <blockquote class="gmail_quote" style="margin:0 0 0
                    .8ex;border-left:1px #ccc solid;padding-left:1ex">
                    On Tue, Sep 13, 2016 at 12:02 PM, Greg White &lt;<a
                      moz-do-not-send="true"
                      href="mailto:g.white@cablelabs.com"
                      target="_blank">g.white@cablelabs.com</a>&gt;
                    wrote:<br>
                    <blockquote class="gmail_quote" style="margin:0 0 0
                      .8ex;border-left:1px #ccc solid;padding-left:1ex">
                      Seeing Marceloâ€™s comments prompted me to read the
                      draft as well.Â  I have just a few comments, and
                      also I found a couple of typos.<br>
                      <br>
                      <br>
                      <br>
                      <br>
                      <br>
                      Comment 1:Â  Section 4.8 contains a lower case
                      â€œshould notâ€�.Â  This seems like it should be a
                      â€œSHOULD NOTâ€�.<br>
                      <br>
                      <br>
                      <br>
                      Â  Â  In cases of idle periods, t in Eq. 1 SHOULD
                      NOT include the<br>
                      <br>
                      Â  Â  idle time; otherwise, W_cubic(t) might be very
                      high after restarting<br>
                      <br>
                      Â  Â  from a long idle time.<br>
                    </blockquote>
                    why SHOULD vs MUST? I can't think of any corner case
                    that we'd want to<br>
                    include the idle time.<br>
                    <br>
                    <blockquote class="gmail_quote" style="margin:0 0 0
                      .8ex;border-left:1px #ccc solid;padding-left:1ex">
                      <br>
                      <br>
                      <br>
                      <br>
                      <br>
                      <br>
                      Comment 2:Â  Section 4.3 reads:<br>
                      <br>
                      <br>
                      <br>
                      4.3.Â  Difficult Environments<br>
                      <br>
                      <br>
                      <br>
                      Â  Â  CUBIC is designed to remedy the poor
                      performance of TCP in fast long-<br>
                      <br>
                      Â  Â  distance networks.Â  It is not designed for
                      wireless networks.<br>
                      <br>
                      <br>
                      <br>
                      Since a large proportion of TCP flows today cross
                      a wireless link (WiFi or LTE), this statement and
                      the heading seem kind of anachronistic.Â  Â Perhaps
                      it would be better to describe what some of the
                      concerns might be with operation across todayâ€™s
                      networks that are likely to contain wireless links
                      (if there are any), or state what properties of a
                      wireless network would cause problems for CUBIC.<br>
                      <br>
                      <br>
                      <br>
                      <br>
                      <br>
                      <br>
                      <br>
                      Comment 3: Section 4.5 states:<br>
                      <br>
                      <br>
                      <br>
                      Â  Â  In case that there is congestion collapse,
                      CUBIC behaves likely<br>
                      <br>
                      Â  Â  standard TCP ...<br>
                      <br>
                      <br>
                      <br>
                      I believe that the whole point of congestion
                      avoidance algorithms such as CUBIC is to prevent
                      congestion collapse.Â  (Also note the typo â€œlikelyâ€�
                      in that sentence.)Â  Maybe this would be better
                      written as:<br>
                      <br>
                      <br>
                      <br>
                      Â  Â  With regard to the potential of causing
                      congestion collapse,<br>
                      <br>
                      Â  Â  CUBIC behaves like standard TCP ...<br>
                      <br>
                      <br>
                      <br>
                      <br>
                      <br>
                      <br>
                      <br>
                      Typo:<br>
                      <br>
                      <br>
                      <br>
                      Section 3.2, first sentence reads:<br>
                      <br>
                      <br>
                      <br>
                      Â  Â When receiving an ACK in congestion avoidance,
                      we first check whether<br>
                      <br>
                      Â  Â  the protocol is in the TCP region or not.<br>
                      <br>
                      <br>
                      <br>
                      I believe it should be:<br>
                      <br>
                      <br>
                      <br>
                      Â  Â  When receiving an ACK in congestion avoidance,
                      we first check whether<br>
                      <br>
                      Â  Â  the protocol is in the TCP **friendly** region
                      or not.<br>
                      <br>
                      <br>
                      <br>
                      <br>
                      <br>
                      <br>
                      <br>
                      Best Regards,<br>
                      <br>
                      Greg<br>
                      <br>
                      <br>
                      <br>
                      <br>
                      <br>
                      <br>
                      ______________________________<wbr>_________________<br>
                      tcpm mailing list<br>
                      <a moz-do-not-send="true"
                        href="mailto:tcpm@ietf.org" target="_blank">tcpm@ietf.org</a><br>
                      <a moz-do-not-send="true"
                        href="https://www.ietf.org/mailman/listinfo/tcpm"
                        rel="noreferrer" target="_blank">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</a><br>
                      <br>
                    </blockquote>
                    ______________________________<wbr>_________________<br>
                    tcpm mailing list<br>
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                      href="mailto:tcpm@ietf.org" target="_blank">tcpm@ietf.org</a><br>
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                  </blockquote>
                  <br>
                </div>
              </div>
            </blockquote>
          </div>
          <br>
        </div>
      </div>
    </blockquote>
    <br>
  </body>
</html>

--------------03CD75BB95749C0D51A00803--


From nobody Sat Sep 17 21:55:06 2016
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-rack-00.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 of the IETF.

        Title           : RACK: a time-based fast loss detection algorithm for TCP
        Authors         : Yuchung Cheng
                          Neal Cardwell
	Filename        : draft-ietf-tcpm-rack-00.txt
	Pages           : 14
	Date            : 2016-09-17

Abstract:
   This document presents a new TCP loss detection algorithm called RACK
   ("Recent ACKnowledgment").  RACK uses the notion of time, instead of
   packet or sequence counts, to detect losses, for modern TCP
   implementations that can support per-packet timestamps and the
   selective acknowledgment (SACK) option.  It is intended to replace
   the conventional DUPACK threshold approach and its variants, as well
   as other nonstandard approaches.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-tcpm-rack-00


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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To: tcpm@ietf.org
From: Wesley Eddy <wes@mti-systems.com>
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Subject: [tcpm] 793bis: proposed multihoming clarification from 1122
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A couple of places in RFC 1122 discuss TCP behavior for source address 
selection on multihomed hosts.

Section 4.2.4.4 has non-requirements language in the implementation 
section that nebulously says "the networking software will choose an 
appropriate local IP address".  The actual requirements-language "MUST" 
on how this works is in 4.2.3.7 that describes what happens more explicitly.

In 793bis, my proposal is to condense material from these 2 different 
sections together more clearly, into:

          For an active OPEN call, a specified "local IP address"
          parameter will be used for opening the connection.  If the
          parameter is unspecified, TCP MUST ask the IP layer to select
          a local IP address before sending the (first) SYN.  See the
          function GET_SRCADDR() in Section 3.3.4.3 of RFC1122.

I think that's basically just an editorial change, but wanted to make 
sure the WG sees it.  My plan is to include this (or whatever improved 
version the WG suggests) in the next revision of 793bis.

Any feedback is appreciated.



From nobody Wed Sep 28 09:59:57 2016
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References: <fdd40c6e-f4b4-a11c-a633-4e27337ecc5e@mti-systems.com>
From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] 793bis: proposed multihoming clarification from 1122
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Hi, Wes,

Do we also need language that says that if the parameter IS specified,
it MUST NOT be overwritten (e.g., by the process below)?

I'm not sure if anyone does this, but I'm increasingly seeing
implementation discussions where developers do virtually anything that
isn't explicitly prohibited.

Joe


On 9/28/2016 9:11 AM, Wesley Eddy wrote:
> A couple of places in RFC 1122 discuss TCP behavior for source address
> selection on multihomed hosts.
>
> Section 4.2.4.4 has non-requirements language in the implementation
> section that nebulously says "the networking software will choose an
> appropriate local IP address".  The actual requirements-language
> "MUST" on how this works is in 4.2.3.7 that describes what happens
> more explicitly.
>
> In 793bis, my proposal is to condense material from these 2 different
> sections together more clearly, into:
>
>          For an active OPEN call, a specified "local IP address"
>          parameter will be used for opening the connection.  If the
>          parameter is unspecified, TCP MUST ask the IP layer to select
>          a local IP address before sending the (first) SYN.  See the
>          function GET_SRCADDR() in Section 3.3.4.3 of RFC1122.
>
> I think that's basically just an editorial change, but wanted to make
> sure the WG sees it.  My plan is to include this (or whatever improved
> version the WG suggests) in the next revision of 793bis.
>
> Any feedback is appreciated.
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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To: Joe Touch <touch@isi.edu>, tcpm@ietf.org
References: <fdd40c6e-f4b4-a11c-a633-4e27337ecc5e@mti-systems.com> <529aa92e-4039-800e-308c-d181e2c59ffc@isi.edu>
From: Wesley Eddy <wes@mti-systems.com>
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Date: Wed, 28 Sep 2016 13:10:57 -0400
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Subject: Re: [tcpm] 793bis: proposed multihoming clarification from 1122
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We could change "will be used" to "MUST be used".


On 9/28/2016 12:57 PM, Joe Touch wrote:
> Hi, Wes,
>
> Do we also need language that says that if the parameter IS specified,
> it MUST NOT be overwritten (e.g., by the process below)?
>
> I'm not sure if anyone does this, but I'm increasingly seeing
> implementation discussions where developers do virtually anything that
> isn't explicitly prohibited.
>
> Joe
>
>
> On 9/28/2016 9:11 AM, Wesley Eddy wrote:
>> A couple of places in RFC 1122 discuss TCP behavior for source address
>> selection on multihomed hosts.
>>
>> Section 4.2.4.4 has non-requirements language in the implementation
>> section that nebulously says "the networking software will choose an
>> appropriate local IP address".  The actual requirements-language
>> "MUST" on how this works is in 4.2.3.7 that describes what happens
>> more explicitly.
>>
>> In 793bis, my proposal is to condense material from these 2 different
>> sections together more clearly, into:
>>
>>           For an active OPEN call, a specified "local IP address"
>>           parameter will be used for opening the connection.  If the
>>           parameter is unspecified, TCP MUST ask the IP layer to select
>>           a local IP address before sending the (first) SYN.  See the
>>           function GET_SRCADDR() in Section 3.3.4.3 of RFC1122.
>>
>> I think that's basically just an editorial change, but wanted to make
>> sure the WG sees it.  My plan is to include this (or whatever improved
>> version the WG suggests) in the next revision of 793bis.
>>
>> Any feedback is appreciated.
>>
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm


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To: Wesley Eddy <wes@mti-systems.com>, tcpm@ietf.org
References: <fdd40c6e-f4b4-a11c-a633-4e27337ecc5e@mti-systems.com> <529aa92e-4039-800e-308c-d181e2c59ffc@isi.edu> <c460c109-5e22-8470-5f5a-de350cc67b93@mti-systems.com>
From: Joe Touch <touch@isi.edu>
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Date: Wed, 28 Sep 2016 10:20:03 -0700
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Subject: Re: [tcpm] 793bis: proposed multihoming clarification from 1122
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Yes - that does it. ;-)


On 9/28/2016 10:10 AM, Wesley Eddy wrote:
> We could change "will be used" to "MUST be used".
>
>
> On 9/28/2016 12:57 PM, Joe Touch wrote:
>> Hi, Wes,
>>
>> Do we also need language that says that if the parameter IS specified,
>> it MUST NOT be overwritten (e.g., by the process below)?
>>
>> I'm not sure if anyone does this, but I'm increasingly seeing
>> implementation discussions where developers do virtually anything that
>> isn't explicitly prohibited.
>>
>> Joe
>>
>>
>> On 9/28/2016 9:11 AM, Wesley Eddy wrote:
>>> A couple of places in RFC 1122 discuss TCP behavior for source address
>>> selection on multihomed hosts.
>>>
>>> Section 4.2.4.4 has non-requirements language in the implementation
>>> section that nebulously says "the networking software will choose an
>>> appropriate local IP address".  The actual requirements-language
>>> "MUST" on how this works is in 4.2.3.7 that describes what happens
>>> more explicitly.
>>>
>>> In 793bis, my proposal is to condense material from these 2 different
>>> sections together more clearly, into:
>>>
>>>           For an active OPEN call, a specified "local IP address"
>>>           parameter will be used for opening the connection.  If the
>>>           parameter is unspecified, TCP MUST ask the IP layer to select
>>>           a local IP address before sending the (first) SYN.  See the
>>>           function GET_SRCADDR() in Section 3.3.4.3 of RFC1122.
>>>
>>> I think that's basically just an editorial change, but wanted to make
>>> sure the WG sees it.  My plan is to include this (or whatever improved
>>> version the WG suggests) in the next revision of 793bis.
>>>
>>> Any feedback is appreciated.
>>>
>>>
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm


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To: tcpm IETF list <tcpm@ietf.org>, draft-ietf-tcpm-cubic@ietf.org
From: marcelo bagnulo braun <marcelo@it.uc3m.es>
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Subject: [tcpm] Another question about cubic
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Hi,

Does cubic assumes that there is a different phase when the connection 
starts? i.e. slow start? It may be worthwhile stating this explicitly?

In the current draft, it always reffers to a packet loss event, but I 
couldnt find how is the packet loss detected i.e. via duplicated ACKs or 
via timeout. Is the reaction of cubic the same in both cases? Does it 
reduces the window to 1 MSS and enters in slow start if the loss is 
detected via a timeout?

Sorry if this is already in the draft and i missed it.

Regards, marcelo



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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] Another question about cubic
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On 9/28/2016 11:13 PM, marcelo bagnulo braun wrote:
> Hi,
>
> Does cubic assumes that there is a different phase when the connection
> starts? i.e. slow start? It may be worthwhile stating this explicitly?
>
> In the current draft, it always reffers to a packet loss event, but I
> couldnt find how is the packet loss detected i.e. via duplicated ACKs
> or via timeout. 

A duplicate ACK does not necessarily imply a loss - the Internet is not
guaranteed to avoid duplication.

Joe


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Subject: Re: [tcpm] Another question about cubic
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> On Sep 29, 2016, at 9:49 AM, Joe Touch <touch@ISI.EDU> wrote:
>=20
> On 9/28/2016 11:13 PM, marcelo bagnulo braun wrote:
>> Hi,
>>=20
>> Does cubic assumes that there is a different phase when the =
connection
>> starts? i.e. slow start? It may be worthwhile stating this =
explicitly?
>>=20
>> In the current draft, it always reffers to a packet loss event, but I
>> couldnt find how is the packet loss detected i.e. via duplicated ACKs
>> or via timeout.=20
>=20
> A duplicate ACK does not necessarily imply a loss - the Internet is =
not
> guaranteed to avoid duplication.

More commonly, reordering of packets will cause dup acks to be =
generated.
That=E2=80=99s why implementations require multiple duplicate ACKs =
before inferring
that there was a packet loss.
			-David Borman

>=20
> Joe
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Thu Sep 29 08:24:59 2016
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Subject: Re: [tcpm] Another question about cubic
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El 29/09/16 a las 16:49, Joe Touch escribió:
>
> On 9/28/2016 11:13 PM, marcelo bagnulo braun wrote:
>> Hi,
>>
>> Does cubic assumes that there is a different phase when the connection
>> starts? i.e. slow start? It may be worthwhile stating this explicitly?
>>
>> In the current draft, it always reffers to a packet loss event, but I
>> couldnt find how is the packet loss detected i.e. via duplicated ACKs
>> or via timeout.
> A duplicate ACK does not necessarily imply a loss - the Internet is not
> guaranteed to avoid duplication.

:-)
I know that much.

Clearly, i failed to formulate my question properly. Let me try again.

RFC5681 defines two different behaviour when a "loss" is detected/inferred.

If the loss is detected by a timeout, the cwnd is set to 1 MSS

If the loss is detected via a number of dup ACKS, the cwnd is set to 
about half its value (omitting the specific details, can be found in 
fast retransmit and fast recovery, section 3.2 of RFC5681)

Now, draft-ietf-tcpm-cubic mentions generically a packet loss event 
(unless i missed something), so i was wondering if it means a packet 
loss detected by the means of dup acks or a packet loss detected by the 
means of a timeout.
I assume it is based on dup acks, but i couldnt find it stated 
explicitly. I understand the the reaction of a cubic sender to a timeout 
is still the same as the one described in rfc5681, i.e. cwnd is set to 1 
mss, but i wanted to confirm that (and possibly suggest that it should 
be mentioned explicitly in the document)

Hope this clarifies and apologies for the poorly phrased question before.
Regards, marcelo



> Joe
>
>


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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] Another question about cubic
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On 9/29/2016 8:24 AM, marcelo bagnulo wrote:
> El 29/09/16 a las 16:49, Joe Touch escribió:
>>
>> On 9/28/2016 11:13 PM, marcelo bagnulo braun wrote:
>>> Hi,
>>>
>>> Does cubic assumes that there is a different phase when the connection
>>> starts? i.e. slow start? It may be worthwhile stating this explicitly?
>>>
>>> In the current draft, it always reffers to a packet loss event, but I
>>> couldnt find how is the packet loss detected i.e. via duplicated ACKs
>>> or via timeout.
>> A duplicate ACK does not necessarily imply a loss - the Internet is not
>> guaranteed to avoid duplication.
>
> :-)
> I know that much.
>
> Clearly, i failed to formulate my question properly. Let me try again.
>
> RFC5681 defines two different behaviour when a "loss" is
> detected/inferred.
>
> If the loss is detected by a timeout, the cwnd is set to 1 MSS
>
> If the loss is detected via a number of dup ACKS, the cwnd is set to
> about half its value (omitting the specific details, can be found in
> fast retransmit and fast recovery, section 3.2 of RFC5681)

This is part of the point, though - these two indications are not quite
the same, and might not be treated the same.

I appreciate your question, but I'm pointing out that there might be two
answers.

Joe


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From: marcelo bagnulo braun <marcelo@it.uc3m.es>
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Subject: Re: [tcpm] Another question about cubic
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El 29/09/16 a las 18:59, Joe Touch escribió:
>
> On 9/29/2016 8:24 AM, marcelo bagnulo wrote:
>> El 29/09/16 a las 16:49, Joe Touch escribió:
>>> On 9/28/2016 11:13 PM, marcelo bagnulo braun wrote:
>>>> Hi,
>>>>
>>>> Does cubic assumes that there is a different phase when the connection
>>>> starts? i.e. slow start? It may be worthwhile stating this explicitly?
>>>>
>>>> In the current draft, it always reffers to a packet loss event, but I
>>>> couldnt find how is the packet loss detected i.e. via duplicated ACKs
>>>> or via timeout.
>>> A duplicate ACK does not necessarily imply a loss - the Internet is not
>>> guaranteed to avoid duplication.
>> :-)
>> I know that much.
>>
>> Clearly, i failed to formulate my question properly. Let me try again.
>>
>> RFC5681 defines two different behaviour when a "loss" is
>> detected/inferred.
>>
>> If the loss is detected by a timeout, the cwnd is set to 1 MSS
>>
>> If the loss is detected via a number of dup ACKS, the cwnd is set to
>> about half its value (omitting the specific details, can be found in
>> fast retransmit and fast recovery, section 3.2 of RFC5681)
> This is part of the point, though - these two indications are not quite
> the same, and might not be treated the same.
>
> I appreciate your question, but I'm pointing out that there might be two
> answers.

Right, agree and that i why i think that simply stating packet loss in 
the cubic draft without indicating how it is detected, may be less 
specified than what is needed.
So i think we are aligned.


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


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To: marcelo bagnulo braun <marcelo@it.uc3m.es>, tcpm@ietf.org
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] Another question about cubic
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On 9/29/2016 10:21 AM, marcelo bagnulo braun wrote:
>> I appreciate your question, but I'm pointing out that there might be two
>> answers.
>
> Right, agree and that i why i think that simply stating packet loss in
> the cubic draft without indicating how it is detected, may be less
> specified than what is needed.
> So i think we are aligned.

Agreed - a protocol should describe behaviors based on directly observed
events, not inferences.

I.e., it can say why three dupacks is treated as inferred loss, but it
needs to go back to the observable events.

Joe

--------------8AE378B913AB68591D8F6760
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    <br>
    <div class="moz-cite-prefix">On 9/29/2016 10:21 AM, marcelo bagnulo
      braun wrote:<br>
    </div>
    <blockquote
      cite="mid:6a7dc05c-4e33-bc4b-58e7-f0de92defcfa@it.uc3m.es"
      type="cite">
      <blockquote type="cite" style="color: #000000;">I appreciate your
        question, but I'm pointing out that there might be two
        <br>
        answers.
        <br>
      </blockquote>
      <br>
      Right, agree and that i why i think that simply stating packet
      loss in the cubic draft without indicating how it is detected, may
      be less specified than what is needed.
      <br>
      So i think we are aligned.
      <br>
    </blockquote>
    <br>
    Agreed - a protocol should describe behaviors based on directly
    observed events, not inferences.<br>
    <br>
    I.e., it can say why three dupacks is treated as inferred loss, but
    it needs to go back to the observable events.<br>
    <br>
    Joe<br>
  </body>
</html>

--------------8AE378B913AB68591D8F6760--


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From: marcelo bagnulo braun <marcelo@it.uc3m.es>
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Subject: [tcpm] Fwd: I-D Action: draft-bagnulo-tcpm-generalized-ecn-00.txt
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Hi,

After the presentation and discussion in Berlin, we have submitted a new 
version of the draft that includes the details of how to add ECN support 
to TCP control packets and retransmissions.

Comments are welcome.

Regards, marcelo




-------- Mensaje reenviado --------
Asunto: 	I-D Action: draft-bagnulo-tcpm-generalized-ecn-00.txt
Fecha: 	Thu, 29 Sep 2016 22:55:36 -0700
De: 	internet-drafts@ietf.org
Responder a: 	internet-drafts@ietf.org
Para: 	i-d-announce@ietf.org



A New Internet-Draft is available from the on-line Internet-Drafts directories.


         Title           : Adding Explicit Congestion Notification (ECN) to TCP control packets and TCP retransmissions
         Authors         : Marcelo Bagnulo
                           Bob Briscoe
	Filename        : draft-bagnulo-tcpm-generalized-ecn-00.txt
	Pages           : 23
	Date            : 2016-09-29

Abstract:
    This document describes an experimental modification to ECN to allow
    the use of ECN to the following TCP packets: SYNs, Pure ACKs, Window
    probes, FINs, RSTs and retransmissions.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-bagnulo-tcpm-generalized-ecn/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-bagnulo-tcpm-generalized-ecn-00


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: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
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Subject: [tcpm] comments for draft-ietf-tcpm-rack-00.txt
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--001a1144065abdd23b053db46d69
Content-Type: text/plain; charset=UTF-8

Hello,

Thanks for an interesting proposal.
I've read the draft-ietf-tcpm-rack-00.txt and have the following comments
on the draft.
Please point out if I miss something.

1: I think the draft should mention how to control pipe size during loss
recovery.
    When RACK identifies lost packets, how it reduces cwnd?

2: Similar to 1:, doesn't RACK have a certain back-off mechanism? I
couldn't find it in the draft.
    It is nice that it can find the lost of retransmitted packets, but does
it retransmits them again with the same settings?
    It seems to me that RACK might become aggressive under heavy
congestions as it keeps retransmitting packets with short intervals while
other TCPs wait for RTO.

3: I am not very sure if using the minimum value in all RTT measurements
for RACK.min_RTT is a really good option.
    If RTT is significantly increased all of sudden, it might cause lots of
spurious retransmits unless reordering window is dynamically adjusted.

4: It is a bit confusing to me that the abstract says "modern TCP
implementations that can support per-packet timestamps.." because I thought
it was referring timestamp option. But, RACK actually doesn't rely on it. I
might prefer to say something like keep tracking transmit time per packet.

5: I guess you might want to add explanations of "RACK packet" and
"RACK.ack_ts" in Section 4. I think their definitions are not very clear.

Thanks,
--
Yoshi

--001a1144065abdd23b053db46d69
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hello,<br><br>Thanks for an interesting proposal.<br>I&#39=
;ve read the draft-ietf-tcpm-rack-00.txt and have the following comments on=
 the draft.<br>Please point out if I miss something.<br><br>1: I think the =
draft should mention how to control pipe size during loss recovery.<br>=C2=
=A0 =C2=A0 When RACK identifies lost packets, how it reduces cwnd? <br><br>=
2: Similar to 1:, doesn&#39;t RACK have a certain back-off mechanism? I cou=
ldn&#39;t find it in the draft.<br>=C2=A0 =C2=A0 It is nice that it can fin=
d the lost of retransmitted packets, but does it retransmits them again wit=
h the same settings?<br>=C2=A0 =C2=A0 It seems to me that RACK might become=
 aggressive under heavy congestions as it keeps retransmitting packets with=
 short intervals while other TCPs wait for RTO.<br><br>3: I am not very sur=
e if using the minimum value in all RTT measurements for RACK.min_RTT is a =
really good option.<br>=C2=A0 =C2=A0 If RTT is significantly increased all =
of sudden, it might cause lots of spurious retransmits unless reordering wi=
ndow is dynamically adjusted.<div><br></div><div>4: It is a bit confusing t=
o me that the abstract says &quot;modern TCP implementations that can suppo=
rt per-packet timestamps..&quot; because I thought it was referring timesta=
mp option. But, RACK actually doesn&#39;t rely on it. I might prefer to say=
 something like keep tracking transmit time per packet.=C2=A0</div><div><br=
></div><div>5: I guess you might want to add explanations of &quot;RACK pac=
ket&quot; and &quot;RACK.ack_ts&quot; in Section 4. I think their definitio=
ns are not very clear.</div><div><br></div><div>Thanks,</div><div>--</div><=
div>Yoshi</div><div><br></div></div>

--001a1144065abdd23b053db46d69--


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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
Date: Fri, 30 Sep 2016 01:06:06 -0700
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Cc: Kobby Carmona <kobby.Carmona@qlogic.com>, David Borman <dab@weston.borman.com>, "tcpm@ietf.org" <tcpm@ietf.org>
Subject: Re: [tcpm] Possible deadlock scenario with retransmission on both sides at the same time
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--94eb2c041f22ff9945053db5104d
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Hello,

I'm not very sure how many folks are interested in this, but anyway...
I have tested this with packetdrill on linux, netbsd, freebsd and openbsd.

As far as I tested, linux works just like Neal described.
On the other hand, I found a minor issue in others' behavior.
In their tcp input functions, they haven't checked sequence number when
they update ack value while it is checked later in the code. Because of
this, it behaves like the followings.

00:42:58.395100 192.0.2.1.54613 > 192.168.0.1.8080: Flags [S],  seq 0, win
20000, options [mss 1000], length 0
00:42:58.395120 192.168.0.1.8080 > 192.0.2.1.54613: Flags [S.], seq
4183682776, ack 1, win 65535, length 0
00:43:01.422728 192.168.0.1.8080 > 192.0.2.1.54613: Flags [S.], seq
4183682776, ack 1, win 65535, length 0
00:43:01.422801 192.0.2.1.54613 > 192.168.0.1.8080: Flags [.],  seq 1, ack
1, win 20000, length 0
00:43:02.443166 192.0.2.1.54613 > 192.168.0.1.8080: Flags [.],  seq
2001:3001, ack 1, win 20000, length 1000
00:43:02.468148 192.168.0.1.8080 > 192.0.2.1.54613: Flags [.],  seq 1, ack
1, win 65535, length 0
00:43:04.471409 192.168.0.1.8080 > 192.0.2.1.54613: Flags [P.], seq 1:1001,
ack 1, win 65535, length 1000
00:43:06.531462 192.0.2.1.54613 > 192.168.0.1.8080: Flags [.],  seq 1:1001,
ack 1001, win 20000, length 1000
00:43:06.555492 192.168.0.1.8080 > 192.0.2.1.54613: Flags [.],  seq
1001:2001, ack 1001, win 65000, length 1000
00:43:06.555973 192.168.0.1.8080 > 192.0.2.1.54613: Flags [P.], seq
2001:3001, ack 1001, win 65000, length 1000
00:43:07.056276 192.0.2.1.54613 > 192.168.0.1.8080: Flags [.],  seq
1001:2001, ack 1001, win 20000, length 1000
00:43:07.084910 192.168.0.1.8080 > 192.0.2.1.54613: Flags [.],  seq 3001,
ack 3001, win 63000, length 0
00:43:07.585918 192.0.2.1.54613 > 192.168.0.1.8080: Flags [.],  seq
1001:2001, ack 2001, win 20000, length 1000
00:43:07.607117 192.168.0.1.8080 > 192.0.2.1.54613: Flags [.],  seq 3001,
ack 3001, win 63000, length 0
00:43:08.109788 192.0.2.1.54613 > 192.168.0.1.8080: Flags [.],  seq 2001,
ack 2001, win 20000, length 0
00:43:13.908734 192.168.0.1.8080 > 192.0.2.1.54613: Flags [P.], seq
2001:3001, ack 3001, win 63000, length 1000

In this dump file, src addr 192.0.2.1 are the packets generated by
packetdrill and src addr 192.168.0.1 are the packets from OSes. While the
ack values in packets at 00:43:07.585918 and 00:43:13.908734 should be
ignored, it is actually used to update snd_nxt.
As the result, we saw seq 2001:3001 was retransmited after timeout instead
of 1001:2001.
(But, it also means this logic won't create the deadlock situation Kobby
mentioned)

When I added seq num check in updating ack value in FreeBSD code, it works
just like expected.
--
Yoshi


On Tue, Aug 23, 2016 at 12:06 AM, Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
wrote:

> Hi Neal,
>
> Oh. I see. Thanks for the explanation.
> I'd like to think about it a bit more.
> Thanks,
> --
> Yoshi
>
>
> On Sun, Aug 21, 2016 at 12:44 PM, Neal Cardwell <ncardwell@google.com>
> wrote:
>
>> I am pretty sure Linux does not have the issue Kobby pointed out in this
>> thread.
>>
>> At a high level Linux should be OK because it follows the principle
>> David Borman laid out in his August 16 email: "ACK-only packets should
>> be sent with the largest in-window sequence number that has ever been
>> sent."
>>
>> Linux obeys that principle by using tp->snd_nxt to store the largest
>> sequence number that has ever been sent, and having
>> tcp_acceptable_seq() use tp->snd_nxt but clamp the outgoing sequence
>> number to make sure it is in-window. To be able to do this, in Linux,
>> the sender does not rewind tp->snd_nxt on retransmissions.
>>
>> neal
>>
>> On Sun, Aug 21, 2016 at 1:25 PM, Yoshifumi Nishida
>> <nishida@sfc.wide.ad.jp> wrote:
>> > Hi Neal,
>> >
>> > Thanks for the info.
>> > So, it seems to me that the linux code has the issue Kobby pointed out.
>> > Or, am I missing something?
>> > --
>> > Yoshi
>> >
>> >
>> > On Mon, Aug 15, 2016 at 6:18 PM, Neal Cardwell <ncardwell@google.com>
>> wrote:
>> >>
>> >> On Mon, Aug 15, 2016 at 6:39 PM, Yoshifumi Nishida
>> >> <nishida@sfc.wide.ad.jp> wrote:
>> >> > Hello,
>> >> > I personally think this is an interesting corner case for discussion.
>> >> > It looks a minor one, but I'm not not very sure if we can leave it
>> for
>> >> > each
>> >> > implementation.
>> >> > I also guess a question would be if the BSD's fix is the best way for
>> >> > the
>> >> > issue.
>> >>
>> >> Yes, I agree this is an interesting case for discussion.
>> >>
>> >> FWIW, as a point of comparison for discussion, Linux's approach is a
>> >> little different: in Linux, the sender does not rewind SND.NXT on
>> >> retransmissions (RTO or Fast Recovery). Then the sender usually uses
>> >> SND.NXT for the seq field of outgoing pure ACKs. I say "usually"
>> >> because the Linux code has some code to deal with the case where the
>> >> receiver has withdrawn the receive window, so that SND.NXT is now
>> >> beyond the receive window. The code tcp_send_ack() uses to pick a seq
>> >> for outgoing pure ACKs looks like:
>> >>
>> >> /* SND.NXT, if window was not shrunk.
>> >>  * If window has been shrunk, what should we make? It is not clear at
>> >> all.
>> >>  * Using SND.UNA we will fail to open window, SND.NXT is out of
>> >> window. :-(
>> >>  * Anything in between SND.UNA...SND.UNA+SND.WND also can be already
>> >>  * invalid. OK, let's make this for now:
>> >>  */
>> >> static inline __u32 tcp_acceptable_seq(const struct sock *sk)
>> >> {
>> >>         const struct tcp_sock *tp = tcp_sk(sk);
>> >>
>> >>         if (!before(tcp_wnd_end(tp), tp->snd_nxt))
>> >>                 return tp->snd_nxt;
>> >>         else
>> >>                 return tcp_wnd_end(tp);
>> >> }
>> >>
>> >> neal
>> >
>> >
>>
>
>

--94eb2c041f22ff9945053db5104d
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hello,<div><br><div>I&#39;m not very sure how many folks a=
re interested in this, but anyway...</div><div>I have tested this with pack=
etdrill on linux, netbsd, freebsd and openbsd.</div><div><br></div><div>As =
far as I tested, linux works just like Neal described.=C2=A0</div><div>On t=
he other hand, I found a minor issue in others&#39; behavior.=C2=A0</div><d=
iv>In their tcp input functions, they haven&#39;t checked sequence number w=
hen they update ack value while it is checked later in the code. Because of=
 this, it behaves like the followings.</div><div><br></div><div><div><font =
size=3D"1">00:42:58.395100 192.0.2.1.54613 &gt; 192.168.0.1.8080: Flags [S]=
, =C2=A0seq 0, win 20000, options [mss 1000], length 0</font></div><div><fo=
nt size=3D"1">00:42:58.395120 192.168.0.1.8080 &gt; 192.0.2.1.54613: Flags =
[S.], seq 4183682776, ack 1, win 65535, length 0</font></div><div><font siz=
e=3D"1">00:43:01.422728 192.168.0.1.8080 &gt; 192.0.2.1.54613: Flags [S.], =
seq 4183682776, ack 1, win 65535, length 0</font></div><div><font size=3D"1=
">00:43:01.422801 192.0.2.1.54613 &gt; 192.168.0.1.8080: Flags [.], =C2=A0s=
eq 1, ack 1, win 20000, length 0</font></div><div><font size=3D"1">00:43:02=
.443166 192.0.2.1.54613 &gt; 192.168.0.1.8080: Flags [.], =C2=A0seq 2001:30=
01, ack 1, win 20000, length 1000</font></div><div><font size=3D"1">00:43:0=
2.468148 192.168.0.1.8080 &gt; 192.0.2.1.54613: Flags [.], =C2=A0seq 1, ack=
 1, win 65535, length 0</font></div><div><font size=3D"1">00:43:04.471409 1=
92.168.0.1.8080 &gt; 192.0.2.1.54613: Flags [P.], seq 1:1001, ack 1, win 65=
535, length 1000</font></div><div><font size=3D"1">00:43:06.531462 192.0.2.=
1.54613 &gt; 192.168.0.1.8080: Flags [.], =C2=A0seq 1:1001, ack 1001, win 2=
0000, length 1000</font></div><div><font size=3D"1">00:43:06.555492 192.168=
.0.1.8080 &gt; 192.0.2.1.54613: Flags [.], =C2=A0seq 1001:2001, ack 1001, w=
in 65000, length 1000</font></div><div><font size=3D"1">00:43:06.555973 192=
.168.0.1.8080 &gt; 192.0.2.1.54613: Flags [P.], seq 2001:3001, ack 1001, wi=
n 65000, length 1000</font></div><div><font size=3D"1">00:43:07.056276 192.=
0.2.1.54613 &gt; 192.168.0.1.8080: Flags [.], =C2=A0seq 1001:2001, ack 1001=
, win 20000, length 1000</font></div><div><font size=3D"1">00:43:07.084910 =
192.168.0.1.8080 &gt; 192.0.2.1.54613: Flags [.], =C2=A0seq 3001, ack 3001,=
 win 63000, length 0</font></div><div><font size=3D"1">00:43:07.585918 192.=
0.2.1.54613 &gt; 192.168.0.1.8080: Flags [.], =C2=A0seq 1001:2001, ack 2001=
, win 20000, length 1000</font></div><div><font size=3D"1">00:43:07.607117 =
192.168.0.1.8080 &gt; 192.0.2.1.54613: Flags [.], =C2=A0seq 3001, ack 3001,=
 win 63000, length 0</font></div><div><font size=3D"1">00:43:08.109788 192.=
0.2.1.54613 &gt; 192.168.0.1.8080: Flags [.], =C2=A0seq 2001, ack 2001, win=
 20000, length 0</font></div><div><font size=3D"1">00:43:13.908734 192.168.=
0.1.8080 &gt; 192.0.2.1.54613: Flags [P.], seq 2001:3001, ack 3001, win 630=
00, length 1000</font></div></div><div><br></div><div>In this dump file, sr=
c addr 192.0.2.1 are the packets generated by packetdrill and src addr 192.=
168.0.1 are the packets from OSes. While the ack values in packets at=C2=A0=
<span style=3D"font-size:x-small">00:43:07.585918 </span>and<span style=3D"=
font-size:x-small">=C2=A0</span><span style=3D"font-size:x-small">00:43:13.=
908734 </span>should be ignored, it is actually used to update snd_nxt.=C2=
=A0</div><div>As the result, we saw seq 2001:3001 was retransmited after ti=
meout instead of 1001:2001.</div><div>(But, it also means this logic won&#3=
9;t create the deadlock situation Kobby mentioned)</div><div><br></div><div=
>When I added seq num check in updating ack value in FreeBSD code, it works=
 just like expected.</div><div>--</div><div>Yoshi</div><div><br></div><div>=
<div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tue, Aug 23, 2=
016 at 12:06 AM, Yoshifumi Nishida <span dir=3D"ltr">&lt;<a href=3D"mailto:=
nishida@sfc.wide.ad.jp" target=3D"_blank">nishida@sfc.wide.ad.jp</a>&gt;</s=
pan> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0p=
x 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb=
(204,204,204);padding-left:1ex"><div dir=3D"ltr">Hi Neal,<div><br></div><di=
v>Oh. I see. Thanks for the explanation.</div><div>I&#39;d like to think ab=
out it a bit more.</div><div>Thanks,</div><div>--<br></div><div>Yoshi<div><=
div class=3D"gmail-m_3426810027158076340h5"><br><div class=3D"gmail_extra">=
<br><div class=3D"gmail_quote">On Sun, Aug 21, 2016 at 12:44 PM, Neal Cardw=
ell <span dir=3D"ltr">&lt;<a href=3D"mailto:ncardwell@google.com" target=3D=
"_blank">ncardwell@google.com</a>&gt;</span> wrote:<br><blockquote class=3D=
"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;borde=
r-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex">I a=
m pretty sure Linux does not have the issue Kobby pointed out in this threa=
d.<br>
<br>
At a high level Linux should be OK because it follows the principle<br>
David Borman laid out in his August 16 email: &quot;ACK-only packets should=
<br>
be sent with the largest in-window sequence number that has ever been<br>
sent.&quot;<br>
<br>
Linux obeys that principle by using tp-&gt;snd_nxt to store the largest<br>
sequence number that has ever been sent, and having<br>
tcp_acceptable_seq() use tp-&gt;snd_nxt but clamp the outgoing sequence<br>
number to make sure it is in-window. To be able to do this, in Linux,<br>
the sender does not rewind tp-&gt;snd_nxt on retransmissions.<br>
<br>
neal<br>
<br>
On Sun, Aug 21, 2016 at 1:25 PM, Yoshifumi Nishida<br>
<div><div>&lt;<a href=3D"mailto:nishida@sfc.wide.ad.jp" target=3D"_blank">n=
ishida@sfc.wide.ad.jp</a>&gt; wrote:<br>
&gt; Hi Neal,<br>
&gt;<br>
&gt; Thanks for the info.<br>
&gt; So, it seems to me that the linux code has the issue Kobby pointed out=
.<br>
&gt; Or, am I missing something?<br>
&gt; --<br>
&gt; Yoshi<br>
&gt;<br>
&gt;<br>
&gt; On Mon, Aug 15, 2016 at 6:18 PM, Neal Cardwell &lt;<a href=3D"mailto:n=
cardwell@google.com" target=3D"_blank">ncardwell@google.com</a>&gt; wrote:<=
br>
&gt;&gt;<br>
&gt;&gt; On Mon, Aug 15, 2016 at 6:39 PM, Yoshifumi Nishida<br>
&gt;&gt; &lt;<a href=3D"mailto:nishida@sfc.wide.ad.jp" target=3D"_blank">ni=
shida@sfc.wide.ad.jp</a>&gt; wrote:<br>
&gt;&gt; &gt; Hello,<br>
&gt;&gt; &gt; I personally think this is an interesting corner case for dis=
cussion.<br>
&gt;&gt; &gt; It looks a minor one, but I&#39;m not not very sure if we can=
 leave it for<br>
&gt;&gt; &gt; each<br>
&gt;&gt; &gt; implementation.<br>
&gt;&gt; &gt; I also guess a question would be if the BSD&#39;s fix is the =
best way for<br>
&gt;&gt; &gt; the<br>
&gt;&gt; &gt; issue.<br>
&gt;&gt;<br>
&gt;&gt; Yes, I agree this is an interesting case for discussion.<br>
&gt;&gt;<br>
&gt;&gt; FWIW, as a point of comparison for discussion, Linux&#39;s approac=
h is a<br>
&gt;&gt; little different: in Linux, the sender does not rewind SND.NXT on<=
br>
&gt;&gt; retransmissions (RTO or Fast Recovery). Then the sender usually us=
es<br>
&gt;&gt; SND.NXT for the seq field of outgoing pure ACKs. I say &quot;usual=
ly&quot;<br>
&gt;&gt; because the Linux code has some code to deal with the case where t=
he<br>
&gt;&gt; receiver has withdrawn the receive window, so that SND.NXT is now<=
br>
&gt;&gt; beyond the receive window. The code tcp_send_ack() uses to pick a =
seq<br>
&gt;&gt; for outgoing pure ACKs looks like:<br>
&gt;&gt;<br>
&gt;&gt; /* SND.NXT, if window was not shrunk.<br>
&gt;&gt;=C2=A0 * If window has been shrunk, what should we make? It is not =
clear at<br>
&gt;&gt; all.<br>
&gt;&gt;=C2=A0 * Using SND.UNA we will fail to open window, SND.NXT is out =
of<br>
&gt;&gt; window. :-(<br>
&gt;&gt;=C2=A0 * Anything in between SND.UNA...SND.UNA+SND.WND also can be =
already<br>
&gt;&gt;=C2=A0 * invalid. OK, let&#39;s make this for now:<br>
&gt;&gt;=C2=A0 */<br>
&gt;&gt; static inline __u32 tcp_acceptable_seq(const struct sock *sk)<br>
&gt;&gt; {<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0const struct tcp_sock *tp =3D tcp=
_sk(sk);<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0if (!before(tcp_wnd_end(tp), tp-&=
gt;snd_nxt))<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0retur=
n tp-&gt;snd_nxt;<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0else<br>
&gt;&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0retur=
n tcp_wnd_end(tp);<br>
&gt;&gt; }<br>
&gt;&gt;<br>
&gt;&gt; neal<br>
&gt;<br>
&gt;<br>
</div></div></blockquote></div><br></div></div></div></div></div>
</blockquote></div><br></div></div></div></div>

--94eb2c041f22ff9945053db5104d--


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An update to a 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: 1
Length of Session(s):  2.5 Hours
Number of Attendees: 80
Conflicts to Avoid: 
 First Priority: tsvwg tsvarea taps mptcp tcpinc httpbis iccrg quic plus
 Second Priority: rmcat rtcweb aqm maprg 
 Third Priority: ippm teas


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


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Subject: Re: [tcpm] Another question about cubic
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Thanks, Marcelo!

We will revise the draft to more clearly explain the protocol behavior 
according to your discussion with Joe yesterday.

Thanks

Lisong


On 9/29/2016 1:13 AM, marcelo bagnulo braun wrote:
> Hi,
>
> Does cubic assumes that there is a different phase when the connection 
> starts? i.e. slow start? It may be worthwhile stating this explicitly?
>
> In the current draft, it always reffers to a packet loss event, but I 
> couldnt find how is the packet loss detected i.e. via duplicated ACKs 
> or via timeout. Is the reaction of cubic the same in both cases? Does 
> it reduces the window to 1 MSS and enters in slow start if the loss is 
> detected via a timeout?
>
> Sorry if this is already in the draft and i missed it.
>
> Regards, marcelo
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: Yuchung Cheng <ycheng@google.com>
Date: Fri, 30 Sep 2016 12:22:15 -0700
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Subject: Re: [tcpm] Another question about cubic
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A small suggestion: the loss event may imply one single packet drop.
But AFAIK about Cubic, N>0 packet drops in one round trip is 1 loss
event, not N loss events. And 3 packet drops in round trip 1 plus 6
packet drops in round trip 2 mean 2 loss events. A fast recovery can
span across many round trips if retransmission is lost again.

It might be worth clarifying that otherwise the (incorrect)
implementation will be a lot slower.

On Fri, Sep 30, 2016 at 11:11 AM, Lisong Xu <xu@unl.edu> wrote:
> Thanks, Marcelo!
>
> We will revise the draft to more clearly explain the protocol behavior
> according to your discussion with Joe yesterday.
>
> Thanks
>
> Lisong
>
>
>
> On 9/29/2016 1:13 AM, marcelo bagnulo braun wrote:
>>
>> Hi,
>>
>> Does cubic assumes that there is a different phase when the connection
>> starts? i.e. slow start? It may be worthwhile stating this explicitly?
>>
>> In the current draft, it always reffers to a packet loss event, but I
>> couldnt find how is the packet loss detected i.e. via duplicated ACKs or via
>> timeout. Is the reaction of cubic the same in both cases? Does it reduces
>> the window to 1 MSS and enters in slow start if the loss is detected via a
>> timeout?
>>
>> Sorry if this is already in the draft and i missed it.
>>
>> Regards, marcelo
>>
>>
>> _______________________________________________
>> 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


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Subject: Re: [tcpm] Another question about cubic
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Hi Marcelo,

Please see my replies below. Thanks


On 9/29/2016 1:13 AM, marcelo bagnulo braun wrote:
> Hi,
>
> Does cubic assumes that there is a different phase when the connection 
> starts? i.e. slow start? It may be worthwhile stating this explicitly?

Yes, there is still the slow start phase.

>
> In the current draft, it always reffers to a packet loss event, but I 
> couldnt find how is the packet loss detected i.e. via duplicated ACKs 
> or via timeout. Is the reaction of cubic the same in both cases? Does 
> it reduces the window to 1 MSS and enters in slow start if the loss is 
> detected via a timeout?
>
Yes, it refers to a packet loss event inferred by dup acks. In this 
case, both the slow start threshold (ssthresh) and the congestion window 
set (cwnd) are set to cwnd*(1-beta), and then CUBIC increases cwnd 
according to a cubic function described in the draft.

In case of a timeout, ssthresh is set to cwnd*(1-beta), and cwnd is set 
to 1 (CUBIC does not change this value).

Thanks
Lisong

> Sorry if this is already in the draft and i missed it.
>
> Regards, marcelo
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
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From: Yuchung Cheng <ycheng@google.com>
Date: Fri, 30 Sep 2016 15:31:22 -0700
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Subject: [tcpm] Fwd: Returned mail: see transcript for details
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btw, the emails of some authors are invalid. please update.


---------- Forwarded message ----------
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Subject: Returned mail: see transcript for details
To: ycheng@google.com


The original message was received at Fri, 30 Sep 2016 12:23:17 -0700 (PDT)
from vmwexchts01-prd.hq.netapp.com [10.122.105.12]

   ----- The following addresses had permanent fatal errors -----
"|/usr/local/bin/mreject srichard"
    (reason: 1)
    (expanded from: srichard)
"|/usr/local/bin/mreject alexandz"
    (reason: 1)
    (expanded from: alexandz)

   ----- Transcript of session follows -----
451 4.0.0 putbody: write error: Broken pipe
sh: /usr/local/bin/mreject: not found
554 5.3.0 unknown mailer error 1
sh: /usr/local/bin/mreject: not found
554 5.3.0 unknown mailer error 1

Final-Recipient: RFC822; rs@netapp.com
X-Actual-Recipient: X-Unix; |/usr/local/bin/mreject srichard
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Status: 5.0.0
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Final-Recipient: RFC822; alexander.zimmermann@netapp.com
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---------- Forwarded message ----------
From: Yuchung Cheng <ycheng@google.com>
To: Lisong Xu <xu@unl.edu>
Cc: marcelo bagnulo braun <marcelo@it.uc3m.es>, tcpm IETF list
<tcpm@ietf.org>, <draft-ietf-tcpm-cubic@ietf.org>
Date: Fri, 30 Sep 2016 12:22:15 -0700
Subject: Re: [tcpm] Another question about cubic
A small suggestion: the loss event may imply one single packet drop.
But AFAIK about Cubic, N>0 packet drops in one round trip is 1 loss
event, not N loss events. And 3 packet drops in round trip 1 plus 6
packet drops in round trip 2 mean 2 loss events. A fast recovery can
span across many round trips if retransmission is lost again.

It might be worth clarifying that otherwise the (incorrect)
implementation will be a lot slower.

On Fri, Sep 30, 2016 at 11:11 AM, Lisong Xu <xu@unl.edu> wrote:
> Thanks, Marcelo!
>
> We will revise the draft to more clearly explain the protocol behavior
> according to your discussion with Joe yesterday.
>
> Thanks
>
> Lisong
>
>
>
> On 9/29/2016 1:13 AM, marcelo bagnulo braun wrote:
>>
>> Hi,
>>
>> Does cubic assumes that there is a different phase when the connection
>> starts? i.e. slow start? It may be worthwhile stating this explicitly?
>>
>> In the current draft, it always reffers to a packet loss event, but I
>> couldnt find how is the packet loss detected i.e. via duplicated ACKs or via
>> timeout. Is the reaction of cubic the same in both cases? Does it reduces
>> the window to 1 MSS and enters in slow start if the loss is detected via a
>> timeout?
>>
>> Sorry if this is already in the draft and i missed it.
>>
>> Regards, marcelo
>>
>>
>> _______________________________________________
>> 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


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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
Date: Fri, 30 Sep 2016 16:13:12 -0700
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To: Yuchung Cheng <ycheng@google.com>
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>
Subject: Re: [tcpm] Fwd: Returned mail: see transcript for details
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--94eb2c041f2201fdef053dc1bd9f
Content-Type: text/plain; charset=UTF-8

Hmm. a simple work around might be to not include draft-ietf-tcpm-cubic@
ietf.org for the time being and wait for updating authors' mail addresses
in a new version.
--
Yoshi

On Fri, Sep 30, 2016 at 3:31 PM, Yuchung Cheng <ycheng@google.com> wrote:

> btw, the emails of some authors are invalid. please update.
>
>
> ---------- Forwarded message ----------
> From: Mail Delivery Subsystem <MAILER-DAEMON@netapp.com>
> Date: Fri, Sep 30, 2016 at 12:23 PM
> Subject: Returned mail: see transcript for details
> To: ycheng@google.com
>
>
> The original message was received at Fri, 30 Sep 2016 12:23:17 -0700 (PDT)
> from vmwexchts01-prd.hq.netapp.com [10.122.105.12]
>
>    ----- The following addresses had permanent fatal errors -----
> "|/usr/local/bin/mreject srichard"
>     (reason: 1)
>     (expanded from: srichard)
> "|/usr/local/bin/mreject alexandz"
>     (reason: 1)
>     (expanded from: alexandz)
>
>    ----- Transcript of session follows -----
> 451 4.0.0 putbody: write error: Broken pipe
> sh: /usr/local/bin/mreject: not found
> 554 5.3.0 unknown mailer error 1
> sh: /usr/local/bin/mreject: not found
> 554 5.3.0 unknown mailer error 1
>
> Final-Recipient: RFC822; rs@netapp.com
> X-Actual-Recipient: X-Unix; |/usr/local/bin/mreject srichard
> Action: failed
> Status: 5.0.0
> Diagnostic-Code: X-Unix; 1
> Last-Attempt-Date: Fri, 30 Sep 2016 12:23:17 -0700 (PDT)
>
> Final-Recipient: RFC822; alexander.zimmermann@netapp.com
> X-Actual-Recipient: X-Unix; |/usr/local/bin/mreject alexandz
> Action: failed
> Status: 5.0.0
> Diagnostic-Code: X-Unix; 1
> Last-Attempt-Date: Fri, 30 Sep 2016 12:23:17 -0700 (PDT)
>
>
> ---------- Forwarded message ----------
> From: Yuchung Cheng <ycheng@google.com>
> To: Lisong Xu <xu@unl.edu>
> Cc: marcelo bagnulo braun <marcelo@it.uc3m.es>, tcpm IETF list
> <tcpm@ietf.org>, <draft-ietf-tcpm-cubic@ietf.org>
> Date: Fri, 30 Sep 2016 12:22:15 -0700
> Subject: Re: [tcpm] Another question about cubic
> A small suggestion: the loss event may imply one single packet drop.
> But AFAIK about Cubic, N>0 packet drops in one round trip is 1 loss
> event, not N loss events. And 3 packet drops in round trip 1 plus 6
> packet drops in round trip 2 mean 2 loss events. A fast recovery can
> span across many round trips if retransmission is lost again.
>
> It might be worth clarifying that otherwise the (incorrect)
> implementation will be a lot slower.
>
> On Fri, Sep 30, 2016 at 11:11 AM, Lisong Xu <xu@unl.edu> wrote:
> > Thanks, Marcelo!
> >
> > We will revise the draft to more clearly explain the protocol behavior
> > according to your discussion with Joe yesterday.
> >
> > Thanks
> >
> > Lisong
> >
> >
> >
> > On 9/29/2016 1:13 AM, marcelo bagnulo braun wrote:
> >>
> >> Hi,
> >>
> >> Does cubic assumes that there is a different phase when the connection
> >> starts? i.e. slow start? It may be worthwhile stating this explicitly?
> >>
> >> In the current draft, it always reffers to a packet loss event, but I
> >> couldnt find how is the packet loss detected i.e. via duplicated ACKs
> or via
> >> timeout. Is the reaction of cubic the same in both cases? Does it
> reduces
> >> the window to 1 MSS and enters in slow start if the loss is detected
> via a
> >> timeout?
> >>
> >> Sorry if this is already in the draft and i missed it.
> >>
> >> Regards, marcelo
> >>
> >>
> >> _______________________________________________
> >> 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
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

--94eb2c041f2201fdef053dc1bd9f
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hmm. a simple work around might be to not include=C2=A0<a =
href=3D"mailto:draft-ietf-tcpm-cubic@ietf.org" target=3D"_blank">draft-ietf=
-tcpm-cubic@<wbr>ietf.org</a>=C2=A0for the time being and wait for updating=
 authors&#39; mail addresses in a new version.<div><div>--</div><div>Yoshi<=
br><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Sep 30=
, 2016 at 3:31 PM, Yuchung Cheng <span dir=3D"ltr">&lt;<a href=3D"mailto:yc=
heng@google.com" target=3D"_blank">ycheng@google.com</a>&gt;</span> wrote:<=
br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bord=
er-left:1px solid rgb(204,204,204);padding-left:1ex">btw, the emails of som=
e authors are invalid. please update.<br>
<br>
<br>
---------- Forwarded message ----------<br>
From: Mail Delivery Subsystem &lt;<a href=3D"mailto:MAILER-DAEMON@netapp.co=
m" target=3D"_blank">MAILER-DAEMON@netapp.com</a>&gt;<br>
Date: Fri, Sep 30, 2016 at 12:23 PM<br>
Subject: Returned mail: see transcript for details<br>
To: <a href=3D"mailto:ycheng@google.com" target=3D"_blank">ycheng@google.co=
m</a><br>
<br>
<br>
The original message was received at Fri, 30 Sep 2016 12:23:17 -0700 (PDT)<=
br>
from <a href=3D"http://vmwexchts01-prd.hq.netapp.com" rel=3D"noreferrer" ta=
rget=3D"_blank">vmwexchts01-prd.hq.netapp.com</a> [10.122.105.12]<br>
<br>
=C2=A0 =C2=A0----- The following addresses had permanent fatal errors -----=
<br>
&quot;|/usr/local/bin/mreject srichard&quot;<br>
=C2=A0 =C2=A0 (reason: 1)<br>
=C2=A0 =C2=A0 (expanded from: srichard)<br>
&quot;|/usr/local/bin/mreject alexandz&quot;<br>
=C2=A0 =C2=A0 (reason: 1)<br>
=C2=A0 =C2=A0 (expanded from: alexandz)<br>
<br>
=C2=A0 =C2=A0----- Transcript of session follows -----<br>
451 4.0.0 putbody: write error: Broken pipe<br>
sh: /usr/local/bin/mreject: not found<br>
554 5.3.0 unknown mailer error 1<br>
sh: /usr/local/bin/mreject: not found<br>
554 5.3.0 unknown mailer error 1<br>
<br>
Final-Recipient: RFC822; <a href=3D"mailto:rs@netapp.com" target=3D"_blank"=
>rs@netapp.com</a><br>
X-Actual-Recipient: X-Unix; |/usr/local/bin/mreject srichard<br>
Action: failed<br>
Status: 5.0.0<br>
Diagnostic-Code: X-Unix; 1<br>
Last-Attempt-Date: Fri, 30 Sep 2016 12:23:17 -0700 (PDT)<br>
<br>
Final-Recipient: RFC822; <a href=3D"mailto:alexander.zimmermann@netapp.com"=
 target=3D"_blank">alexander.zimmermann@netapp.co<wbr>m</a><br>
X-Actual-Recipient: X-Unix; |/usr/local/bin/mreject alexandz<br>
Action: failed<br>
Status: 5.0.0<br>
Diagnostic-Code: X-Unix; 1<br>
Last-Attempt-Date: Fri, 30 Sep 2016 12:23:17 -0700 (PDT)<br>
<br>
<br>
---------- Forwarded message ----------<br>
From: Yuchung Cheng &lt;<a href=3D"mailto:ycheng@google.com" target=3D"_bla=
nk">ycheng@google.com</a>&gt;<br>
To: Lisong Xu &lt;<a href=3D"mailto:xu@unl.edu" target=3D"_blank">xu@unl.ed=
u</a>&gt;<br>
Cc: marcelo bagnulo braun &lt;<a href=3D"mailto:marcelo@it.uc3m.es" target=
=3D"_blank">marcelo@it.uc3m.es</a>&gt;, tcpm IETF list<br>
&lt;<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a>&gt=
;, &lt;<a href=3D"mailto:draft-ietf-tcpm-cubic@ietf.org" target=3D"_blank">=
draft-ietf-tcpm-cubic@ietf.or<wbr>g</a>&gt;<br>
Date: Fri, 30 Sep 2016 12:22:15 -0700<br>
Subject: Re: [tcpm] Another question about cubic<br>
A small suggestion: the loss event may imply one single packet drop.<br>
But AFAIK about Cubic, N&gt;0 packet drops in one round trip is 1 loss<br>
event, not N loss events. And 3 packet drops in round trip 1 plus 6<br>
packet drops in round trip 2 mean 2 loss events. A fast recovery can<br>
span across many round trips if retransmission is lost again.<br>
<br>
It might be worth clarifying that otherwise the (incorrect)<br>
implementation will be a lot slower.<br>
<br>
On Fri, Sep 30, 2016 at 11:11 AM, Lisong Xu &lt;<a href=3D"mailto:xu@unl.ed=
u" target=3D"_blank">xu@unl.edu</a>&gt; wrote:<br>
&gt; Thanks, Marcelo!<br>
&gt;<br>
&gt; We will revise the draft to more clearly explain the protocol behavior=
<br>
&gt; according to your discussion with Joe yesterday.<br>
&gt;<br>
&gt; Thanks<br>
&gt;<br>
&gt; Lisong<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On 9/29/2016 1:13 AM, marcelo bagnulo braun wrote:<br>
&gt;&gt;<br>
&gt;&gt; Hi,<br>
&gt;&gt;<br>
&gt;&gt; Does cubic assumes that there is a different phase when the connec=
tion<br>
&gt;&gt; starts? i.e. slow start? It may be worthwhile stating this explici=
tly?<br>
&gt;&gt;<br>
&gt;&gt; In the current draft, it always reffers to a packet loss event, bu=
t I<br>
&gt;&gt; couldnt find how is the packet loss detected i.e. via duplicated A=
CKs or via<br>
&gt;&gt; timeout. Is the reaction of cubic the same in both cases? Does it =
reduces<br>
&gt;&gt; the window to 1 MSS and enters in slow start if the loss is detect=
ed via a<br>
&gt;&gt; timeout?<br>
&gt;&gt;<br>
&gt;&gt; Sorry if this is already in the draft and i missed it.<br>
&gt;&gt;<br>
&gt;&gt; Regards, marcelo<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; ______________________________<wbr>_________________<br>
&gt;&gt; tcpm mailing list<br>
&gt;&gt; <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</=
a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"nore=
ferrer" target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</=
a><br>
&gt;<br>
&gt;<br>
&gt; ______________________________<wbr>_________________<br>
&gt; tcpm mailing list<br>
&gt; <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><b=
r>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferr=
er" target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/tcpm</a><b=
r>
<br>
______________________________<wbr>_________________<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/l<wbr>istinfo/tcpm</a><br>
</blockquote></div><br></div></div></div></div>

--94eb2c041f2201fdef053dc1bd9f--


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To: Yuchung Cheng <ycheng@google.com>
References: <2b315b93-3121-8d86-a5b2-6428844f1dc2@it.uc3m.es> <bcfbf09f-8612-0898-e976-0845c90e00a5@unl.edu> <CAK6E8=fVvCwf4_DDe5u6UN2L5EybA2n-ARx8-h0FKe7dAHsRNQ@mail.gmail.com>
From: Lisong Xu <xu@unl.edu>
Organization: University of Nebraska-Lincoln
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Date: Fri, 30 Sep 2016 22:00:31 -0500
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Cc: tcpm IETF list <tcpm@ietf.org>, draft-ietf-tcpm-cubic@ietf.org
Subject: Re: [tcpm] Another question about cubic
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Hi Yuchung,

A quick clarification about your example "3 packet drops in round trip 1 plus 6
packet drops in round trip 2 mean 2 loss events."

If the 6 packet drops happen during the fast recovery of the 3 packet drops, TCP stays in the fast recovery until all lost packets are successfully recovered. In this case (assuming no retransmissions are lost), ssthresh is reduced only once, and there is only one loss event. Right?

Thanks
Lisong


On 9/30/2016 2:22 PM, Yuchung Cheng wrote:
> A small suggestion: the loss event may imply one single packet drop.
> But AFAIK about Cubic, N>0 packet drops in one round trip is 1 loss
> event, not N loss events. And 3 packet drops in round trip 1 plus 6
> packet drops in round trip 2 mean 2 loss events. A fast recovery can
> span across many round trips if retransmission is lost again.
>
> It might be worth clarifying that otherwise the (incorrect)
> implementation will be a lot slower.
>
> On Fri, Sep 30, 2016 at 11:11 AM, Lisong Xu <xu@unl.edu> wrote:
>> Thanks, Marcelo!
>>
>> We will revise the draft to more clearly explain the protocol behavior
>> according to your discussion with Joe yesterday.
>>
>> Thanks
>>
>> Lisong
>>
>>
>>
>> On 9/29/2016 1:13 AM, marcelo bagnulo braun wrote:
>>> Hi,
>>>
>>> Does cubic assumes that there is a different phase when the connection
>>> starts? i.e. slow start? It may be worthwhile stating this explicitly?
>>>
>>> In the current draft, it always reffers to a packet loss event, but I
>>> couldnt find how is the packet loss detected i.e. via duplicated ACKs or via
>>> timeout. Is the reaction of cubic the same in both cases? Does it reduces
>>> the window to 1 MSS and enters in slow start if the loss is detected via a
>>> timeout?
>>>
>>> Sorry if this is already in the draft and i missed it.
>>>
>>> Regards, marcelo
>>>
>>>
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