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From: Per Hurtig <per.hurtig@kau.se>
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Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-rtorestart-09.txt
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Hi,

if there are no objections we=E2=80=99d go for =E2=80=9Ctimeout =
duration=E2=80=9D?


Cheers,
Per
> On 31 Oct 2015, at 01:49, Scharf, Michael (Michael) =
<michael.scharf@alcatel-lucent.com> wrote:
>=20
> Maybe the authors could propose an alternative phrasing on the mailing =
list, picking one of the potential alternative wordings, and update the =
draft if there is no objection?
>=20
> Since this is in the abstract, this seems to me better than waiting =
for the RFC editor.
>=20
> Michael
>=20
> ________________________________________
> Von: Anna Brunstrom [anna.brunstrom@kau.se]
> Gesendet: Freitag, 23. Oktober 2015 23:49
> An: Scharf, Michael (Michael); spencerdawkins.ietf@gmail.com
> Cc: tcpm@ietf.org
> Betreff: Re: [tcpm] I-D Action: draft-ietf-tcpm-rtorestart-09.txt
>=20
> Hi Michael,
>=20
> I am not a native speaker either, but I agree with you that "timeout
> duration" or "timeout value" sounds more clear.
>=20
> BR,
> Anna
>=20
> On 2015-10-23 19:38, Scharf, Michael (Michael) wrote:
>> Sorry for speaking up late... Just to cross-check with others. Is the =
new wording in the abstract...
>>=20
>>    The modification, RTO Restart (RTOR), allows the
>>    transport to restart its retransmission timer using a smaller =
delay,
>>    so that the effective RTO becomes more aggressive in situations =
where
>>    fast retransmit cannot be used.  This enables faster loss =
detection
>>    and recovery for connections that are short-lived or application-
>>    limited.
>>=20
>> ... indeed clear to everybody? To me (as a non-native speaker) this =
use of "delay" in the context of the RTO is a bit confusing.
>>=20
>> To me, a wording like replacing "delay" by "timeout duration", =
"timeout value", etc. would sound more familiar. I'd rather use "delay" =
as a measure between packet (re) transmissions, etc.
>>=20
>> For instance, to me the following wording would fit a bit better ...
>>=20
>>    The modification, RTO Restart (RTOR), allows the
>>    transport to restart its retransmission timer using a smaller =
timeout duration,
>>    so that the effective RTO becomes more aggressive in situations =
where
>>    fast retransmit cannot be used.  This smaller delay before the =
retransmission enables faster loss detection
>>    and recovery for connections that are short-lived or application-
>>    limited.
>>=20
>> But I am not a native speaker. Any thoughts?
>>=20
>> (I guess the RFC editor could just review that.)
>>=20
>> Michael
>>=20
>>=20
>>> -----Original Message-----
>>> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Per Hurtig
>>> Sent: Wednesday, October 21, 2015 9:55 AM
>>> To: tcpm@ietf.org Extensions
>>> Cc: bclaise@cisco.com; barryleiba@computer.org
>>> Subject: Re: [tcpm] I-D Action: draft-ietf-tcpm-rtorestart-09.txt
>>>=20
>>> Hi,
>>>=20
>>> the new draft on RTO restart address the following comments from the
>>> IESG (thanks for the feedback):
>>>=20
>>> o  Clarified, in the abstract, that the modified restart causes a
>>> smaller retransmission delay in total.
>>>=20
>>> o  Clarified, in the introduction, that the fast retransmit =
algorithm
>>> may cause retransmissions upon
>>>     receiving duplicate acknowledgments, not that it unconditionally
>>> does so.
>>>=20
>>> o  Changed wording from "to proposed standard" to "to the standards
>>> track".
>>>=20
>>> o  Changed algorithm description so that a TCP sender MUST track the
>>> time elapsed since the
>>>     transmission of the earliest outstanding segment. This was not
>>> explicitly stated in previous
>>>     versions of the draft.
>>>=20
>>>=20
>>> Cheers,
>>> Per
>>>=20
>>>> On 20 Oct 2015, at 21:39, internet-drafts@ietf.org wrote:
>>>>=20
>>>>=20
>>>> 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
>>> Working Group of the IETF.
>>>>        Title           : TCP and SCTP RTO Restart
>>>>        Authors         : Per Hurtig
>>>>                          Anna Brunstrom
>>>>                          Andreas Petlund
>>>>                          Michael Welzl
>>>>    Filename        : draft-ietf-tcpm-rtorestart-09.txt
>>>>    Pages           : 17
>>>>    Date            : 2015-10-20
>>>>=20
>>>> Abstract:
>>>>   This document describes a modified sender-side algorithm for
>>> managing
>>>>   the TCP and SCTP retransmission timers that provides faster loss
>>>>   recovery when there is a small amount of outstanding data for a
>>>>   connection.  The modification, RTO Restart (RTOR), allows the
>>>>   transport to restart its retransmission timer using a smaller
>>> delay,
>>>>   so that the effective RTO becomes more aggressive in situations
>>> where
>>>>   fast retransmit cannot be used.  This enables faster loss =
detection
>>>>   and recovery for connections that are short-lived or application-
>>>>   limited.
>>>>=20
>>>>=20
>>>> The IETF datatracker status page for this draft is:
>>>> https://datatracker.ietf.org/doc/draft-ietf-tcpm-rtorestart/
>>>>=20
>>>> There's also a htmlized version available at:
>>>> https://tools.ietf.org/html/draft-ietf-tcpm-rtorestart-09
>>>>=20
>>>> A diff from the previous version is available at:
>>>> https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-rtorestart-09
>>>>=20
>>>>=20
>>>> 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.
>>>>=20
>>>> Internet-Drafts are also available by anonymous FTP at:
>>>> ftp://ftp.ietf.org/internet-drafts/
>>>>=20
>>>> _______________________________________________
>>>> 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
>=20
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From nobody Sun Nov  1 15:38:03 2015
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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 Working Group of the IETF.

        Title           : Datacenter TCP (DCTCP): TCP Congestion Control for Datacenters
        Authors         : Stephen Bensley
                          Lars Eggert
                          Dave Thaler
                          Praveen Balasubramanian
                          Glenn Judd
	Filename        : draft-ietf-tcpm-dctcp-01.txt
	Pages           : 14
	Date            : 2015-11-01

Abstract:
   This informational memo describes Datacenter TCP (DCTCP), an
   improvement to TCP congestion control for datacenter traffic.  DCTCP
   uses improved Explicit Congestion Notification (ECN) processing to
   estimate the fraction of bytes that encounter congestion, rather than
   simply detecting that some congestion has occurred.  DCTCP then
   scales the TCP congestion window based on this estimate.  This method
   achieves high burst tolerance, low latency, and high throughput with
   shallow-buffered switches.  This memo also discusses deployment
   issues related to the coexistence of DCTCP and conventional TCP, the
   lack of a negotiating mechanism between sender and receiver, and
   presents some possible mitigations.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-01

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-dctcp-01


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

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


From nobody Sun Nov  1 16:50:14 2015
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Subject: [tcpm] [Errata Verified] RFC7413 (4238)
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The following errata report has been verified for RFC7413,
"TCP Fast Open". 

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=7413&eid=4238

--------------------------------------
Status: Verified
Type: Technical

Reported by: Matthew Luckie <mjl@caida.org>
Date Reported: 2015-01-22
Verified by: Martin Stiemerling (IESG)

Section: 4.1.1

Original Text
-------------
   Kind            1 byte: value = 34
   Length          1 byte: range 6 to 18 (bytes); limited by
                           remaining space in the options field.
                           The number MUST be even.
   Cookie          0, or 4 to 16 bytes (Length - 2)

Corrected Text
--------------
   Kind            1 byte: value = 34
   Length          1 byte: range 2 to 18 (bytes); limited by
                           remaining space in the options field.
                           The number MUST be even.
   Cookie          0, or 4 to 16 bytes in length (Length - 2)

Notes
-----
A Nil cookie is a fast open option with no cookie value.  A length range of 6 to 18 bytes excludes a Nil cookie.

--------------------------------------
RFC7413 (draft-ietf-tcpm-fastopen-10)
--------------------------------------
Title               : TCP Fast Open
Publication Date    : December 2014
Author(s)           : Y. Cheng, J. Chu, S. Radhakrishnan, A. Jain
Category            : EXPERIMENTAL
Source              : TCP Maintenance and Minor Extensions
Area                : Transport
Stream              : IETF
Verifying Party     : IESG


From nobody Sun Nov  1 23:51:11 2015
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Subject: [tcpm] slides for Yokohama meeting
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Hello presenters for Yokohama meeting.

Please make sure to send your slide by ** Wednesday (11/4) noon ** to chairs.

We appreciate your cooperation!

Thanks,
--
Yoshi


From nobody Mon Nov  2 08:59:07 2015
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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 Working Group of the IETF.

        Title           : TCP Extended Data Offset Option
        Authors         : Joe Touch
                          Wesley M. Eddy
	Filename        : draft-ietf-tcpm-tcp-edo-04.txt
	Pages           : 23
	Date            : 2015-11-02

Abstract:
   TCP segments include a Data Offset field to indicate space for TCP
   options but the size of the field can limit the space available for
   complex options such as SACK and Multipath TCP and can limit the
   combination of such options supported in a single connection. This
   document updates RFC 793 with an optional TCP extension to that
   space to support the use of multiple large options. It also explains
   why the initial SYN of a connection cannot be extending a single
   segment.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-tcpm-tcp-edo-04

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-tcp-edo-04


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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Thread-Topic: [tcpm] [e2e] TCP HyStart patch deployment
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From: Veaceslav ROMAN <Veaceslav.Roman@orange.md>
To: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, Neal Cardwell <ncardwell@google.com>
Thread-Topic: [tcpm] [e2e] TCP HyStart patch deployment
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Date: Tue, 3 Nov 2015 19:05:36 +0000
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, Piers O'Hanlon <p.ohanlon@gmail.com>, Sangtae Ha <sangtae.ha@gmail.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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To: Bob Briscoe <ietf@bobbriscoe.net>, "iccrg@irtf.org" <iccrg@irtf.org>, "tcpm@ietf.org" <tcpm@ietf.org>, "tsvwg@ietf.org" <tsvwg@ietf.org>
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To: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, "iccrg@irtf.org" <iccrg@irtf.org>, "tcpm@ietf.org" <tcpm@ietf.org>, "tsvwg@ietf.org" <tsvwg@ietf.org>
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Subject: Re: [tcpm] [tsvwg] FW: New Version Notification for draft-johansson-cc-for-4g-5g-00.txt
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Ingemar,

On 04/11/15 13:27, Ingemar Johansson S wrote:
>
> Hi
>
> And thanks for the review, comments/answers/questions inline below
>
> /Ingemar
>
> *From:*Bob Briscoe [mailto:ietf@bobbriscoe.net]
> *Sent:* den 29 oktober 2015 18:54
> *To:* Ingemar Johansson S; iccrg@irtf.org; tcpm@ietf.org; tsvwg@ietf.org
> *Subject:* Re: [tsvwg] FW: New Version Notification for 
> draft-johansson-cc-for-4g-5g-00.txt
>
> Ingemar,
>
> As promised, here's my comments. They are a mix of suggested 
> improvements to the draft, suggested improvements to 5G, or requests 
> for clarification. I also agree with all Kevin Smith's comments, so I 
> didn't repeat any.
> [Sorry for delay - I had to find time to re-type after losing my 
> review during a power outage]
>
> *_Context questions._
> *Q1. Is this exercise treating the 5G specs solely in read-only mode? 
> Or, if some useful design suggestions arise, is there any chance that 
> the 3GPP might take them on board, given 5G is not fully baked yet?
>
> [IJ] Guess it depends, need to say that I probably the small cog in 
> the wheel. I would say that a lot of the standards work in IETF has 
> been picked up by 3GPP and, I cannot say what will happen in the 5G 
> context.
>
> It's great that you have taken this initiative. Does the 3GPP ever 
> formally ask the IETF for comments on its ideas for each new 
> generation of radio? Given the layers above can be thought of as the 
> 'customers' of the 3GPP, and these 'customer' layers are pretty much 
> all maintained by the IETF, it would seem essential (and courteous) 
> for the 3GPP to ask for customer comment, not least to co-ordinate 
> with whatever future plans the IETF might have for the higher layers.
>
> [IJ] What you say makes much sense. I guess there is some natural 
> leakage between 3GPP and IETF as many of the IETFers are also active 
> in 3GPP standardization. I donâ€™t have full insight in the formal 
> relation between 3GPP and IETF in these areas, not sure if that need 
> to be better, but if that is the case, then it is probably a question 
> for the area directors. Need to say that even though the formal 
> channels are scarce it does probably not mean that 3GPP ignores IETF 
> or what happens above IP in general. For instance the recent work 
> around TCP RACK raises the question if it is necessary that LTE 
> default radio bearers ensure in order delivery.
>
I suspect that 3GPP radio resource control people do not cross-fertilize 
much with the IETF.
So maybe it is as much a question of how to co-ordinate between these 
radio people and L4, which I would imagine is as much a problem within 
3GPP as between 3GPP and IETF.
>
>
>
> Q2. What are your plans for this draft? Are you asking for adoption? 
> And if so, in which WG or RG? ICCRG looks the most appropriate, given 
> the charters of all the IETF cc WGs (TSVWG, TCPM, RMCAT, etc) are 
> scoped on just a subset of your doc.
>
> [IJ] First objective is to get an increased interest in congestion 
> control algorithm work in the IETF. Traditionally this kind of work 
> seem to be more in the terms of conference papers in e.g. Sigcomm, the 
> benefit with having such work in IETF is that input from people with a 
> wider expertise area (including 3GPP experitise) can give algorithm 
> proposals that work well for a wider set of access types. The work 
> around Cubic is a first good step as is the work around DCTCP. What I 
> wonder is if it feasible to take this a bit further ? I would too 
> believe that the current doc fits better in ICCRG.  The algorithm 
> examples are just.. examples. More fruitful work is probably to 
> document different access types and from there derive a set of best 
> current practices for congestion control algorithms
>
I'm sure the iccrg chairs would like this to happen. And initiatives 
like yours to come with proposals to propose specific code to take 
account of L2 I think generate the right level of dialogue.
>
>
>
> *_Section by Section Review_
> *
> *All sections
> *
> The document generally talks about the LTE protocol stack as if data 
> is travelling down it. Are you implying that most sources of buffering 
> delay are at the sending end, and the feedback end is pretty 
> unencumbered by delays? I think it would be useful to explicitly say 
> something about this (whether true or not), to explain why only one 
> direction of travel is discussed in the doc.
>
> [IJ] The feedback path is encumbered by delays, it is perhaps not that 
> clear but section 2.3.2 implies that delay (variation) in uplink will 
> affect the ACKs for downlink data traffic. That said, the difference 
> in DL/UL traffic is today something like 10:1 and that is probably one 
> reason why one can get the feeling that the draft is slanted towards 
> downlink data traffic, this may however change with time as more 
> machine type traffic may enter the system.
>
I was thinking more in terms of up the stack vs. down the stack, not 
uplink/downlink. I.e. for both cases:
* downlink direction: are there any buffering concerns coming up the 
stack at the UE, or on the feedback back-channel from the UE?
* uplink direction: are there any buffering concerns coming up the stack 
at the eNodeB, or on the feedback back-channel from the eNodeB?

For instance, in the UE, perhaps one app blocking the read of another 
due to the way {3-5}G multiplexes different streams into the same PDCP 
frames?
>
>
>
> *2.1 PDCP layer
> *
> PDCP also ensures that all
>    packets are delivered in order up to higher layers.
> I think this means "in the order sent into the link (PDCP) layer". 
> This ought to be clarified, otherwise, when writing in the context of 
> a layer 4 audience, it might be assumed this means in the order sent 
> by the original transport layer source.
>
> [IJ] Yes, you are right , reordering can still occur e.g. in the 
> wireless backhaul
>
>
>
> keep the amount of data in flight as small as possible, without 
> sacrificing throughput.
> Is this just a rather convoluted way of saying "keep the RTT as low as 
> possible"? (which in turn implies keeping queuing delay as low as 
> possible assuming you cannot influence the physical path.) Or were you 
> trying to make a more subtle point? - if so, it was lost on me.
>
> [IJ] No, it is exactly as you describe below
>
>
>
> Reliable delivery at handover may however be turned off or is simply 
> not implemented,
> It would be useful if you could give a feel for roughly how often (in 
> %) these cases hold.
>
> [IJ] I donâ€™t believe that it is possible to give a figure as this is 
> vendor specific.
>
>
>
> *2.2 RLC layer*
>
> role ... is to ensure that packets are delivered in order up to the 
> higher layers
> Eh? The PDCP section just said it did ordering. And the rest of the 
> RLC section talks exclusively about buffering delays necessary while 
> filling transport blocks. If ordering is the role of both layers, 
> surely one will have nothing to do once the other has got everything 
> in order?
>
> [IJ] RLC handles ordering due to various MAC layer failures. PDCP 
> handles ordering  when handover occurs
>
>
>
> varying size of the available transport
> Nit: Given 'transport' means e2e for the audience of this draft, pls 
> use bearer or somehow disambiguate this word.
>
> [IJ] OK
>
>
>
>
> Gap#1: In the draft, it seems like transport blocks arrive by magic, 
> to be filled by data buffered in the RLC layer. To understand the 
> delay compromises here, I think the draft needs to explain what 
> determines how much transport block capacity each flow is given, and 
> how they can influence it. In 3G, I believe that the transport layer 
> could ask the radio resource controller to change radio resource 
> allocation. Is this the same in 4G & 5G? Does the backlog of the 
> buffer into the RLC layer have any automatic influence over allocation 
> of available resources? I assume traffic class can also be used to 
> influence allocation.
>
> [IJ] Will updated with more detail
>
>
>
> Gap#2: In a 3G stack, the RLC layer multiplexes all the active 
> application streams into MAC layer transport block by dicing up 
> buffered packets and packing the pieces into each transport block. 
> Then, at the other end of the link, each piece has to be held until 
> the rest of the packet arrives. This sacrifices delay for bandwidth 
> efficiency. Is this still done in 4G and 5G? If so, it would be good 
> to write about mux delay in the draft too.
>
> [IJ] OK, will add more info
>
>
>
> *2.3 MAC layer
> *
> the maximum number of retransmissions is configurable
>
> Wouldn't it be better to set a limit on the /time/ spent 
> retransmitting, so it would automatically give up with less 
> retransmissions for longer transmission distances?
> [IJ] Possible, 4G specifies a limit to the number of retransmissions, 
> in practice it can be translated to time as each retransmission is 8ms 
> later.
>
I didn't realise it was independent of link-RTT. Then the more basic 
question is why is it independent of link-RTT? Surely on a short link it 
can assume that it needs to retransmit more quickly than on a long link?

You can see that my question was assuming that retransmission happened 
more often on a short link, then I was saying it should still be allowed 
the same amount of time to keep retrying (so on a short link it would be 
allowed to retransmit more times before giving up, but the max delay 
would be the same).

8ms! I was assuming it would be rather much less than that. Surely a 
link-layer ACK could never take that long if the frame was going to get 
through.
>
>
> s/only checks for the presence on download data only at regular intervals/
>  /only checks for the presence of download data at regular intervals/
> [OK]
>
>
> *2.3.2 Uplink scheduling
> *
> The scheduling request does not indicate how many bytes there are in 
> the uplink queue.
> Seems like a bad omission. Is there a reason? If there are less bytes 
> queued than the base station's grant, surely it would be useful for 
> the base station to know, so that it can grant more to something else 
> while the previous short transmission is completing.
> [IJ] It is a design choice. A buffer status report is more bulky, a 
> scheduling request is just one bit.
>
>
> You say that when HARQ breaks up a sequence of ACKs, it can also 
> trigger "coalescing issues". I can understand that it could cause ACK 
> compression, but are you implying something coalesces the ACKs or the 
> subsequent data? What does the coalescing? Similarly, in the last para 
> of 2.3.2 you say "Reduced ACKs can unfortunately cause coalescing." 
> and again I'm not sure what is coalescing what here.
>
> [IJ] What I can see in simulations is that the ACK compression leads 
> to coalescing which further can increase jitter.
>
I still don't understand. The question was "What is coalescing what?"
>
>
> There could be a positive side to the interaction between ACK clocking 
> and link aggregation, at least for long-running flows. This has been 
> observed in 802.11n WiFi where the buffering and aggregation of 
> multiple ACKs into one transmission grant tend to lead the sender to 
> bunch multiple data segments so that they all nicely fill one 
> transmission grant in the other direction [Showail14]. As long as the 
> transmission grant does not wait for more data (which unfortunately I 
> don't think is the case in 3G/4G/5G), this could be good for bandwidth 
> utilisation of elephants and latency of mice, including when they are 
> mixed.
>
> [Showail14] Showail, A., Jamshaid, K. & Shihada, B., "WQM: An 
> Aggregation-aware Queue Management Scheme for IEEE 802.11N Based 
> Networks," In: Proceedings of the 2014 ACM SIGCOMM Workshop on 
> Capacity Sharing Workshop CSWS '14 pp.15-20 ACM (2014)
> <http://dl.acm.org/citation.cfm?id=2630097>
> [IJ] Interesting ,will look at it, not sure though that it will work 
> that well with 4G (cant say anything about 5G), in 4G the transmission 
> grants can vary quite a lot over short time spans.
>
>
> *3.  4G and 5G evolution
> *
> s/a explicit/an explicit/
>
> *4.  Requirements for improved performance
> *
> s/bufferbloat/buffer/
>
> *5.  Congestion control examples
> *
> Regarding Hybrid Cubic using OWD measurements, I have seen a paper 
> (under submission) about getting delay-based congestion controls (e.g. 
> LEDBAT or CAIA Delay Gradient) to interwork with AQMs with various low 
> delay targets. It's not in the context of cellular networks, but I'll 
> try to remember to point it out once it gets published.
>
> [IJ] Sounds interesting , looking forward to see the paper.
>
>
>
> *5.1 HyStartRestart
> *
> s/algorithm used/algorithm is used/
>
> In the new code, it would be useful to explain why the second test 
> before doubling ssthresh, ie, the test for how long it has been since 
> the last hyrestart. Also, in the header definitions, if you intend 
> there to be constraints on the relation between N_RTT_HYRESTART and 
> N_RTT_LOW (e.g.  N_RTT_HYRESTART > N_RTT_LOW) you ought to say, 
> because people will try other values.
>
> Rather than the rather arbitrary wait before an arbitrary doubling of 
> ssthresh, wouldn't it be better to continually increase ssthresh (not 
> necessarily linearly) the longer the RTT has remained only just higher 
> than the min RTT?
>
> [IJ] Yes, could be an option
>
>
>
> (BTW, surely the main problem with this modification to HyStartRestart 
> is that is requires a number of round trips to reduce delay - a 
> sort-of oxymoron.)
>
> [IJ] Yes, have tried some more experiments with this algorithm. And I 
> am getting more hesitant. The problem is that when I add more noise 
> (modeling of small objects traffic) into the system simulation, then I 
> also get more delay jitter and the delay detection algorithm in 
> HyStart is notoriously sensitive to delay jitter.
>
>
>
> *5.2.  Hybrid Cubic*
>
>     Furthermore it is assumed than the timestamp clock frequency in
>     sender and receiver are identical or that the sender can infer the
>     timestamp clock frequency of the receiver and recompute timestamp
>     values based on this information.
>
> Recently on tcpm I sent you a pointer to the chirping paper Mirja & I 
> did, which also made this assumption. Partly prompted by that, Richard 
> Scheffenegger tried to get people interested in standardising use of 
> the timestamp option on a SYN to negotiate stuff like timer 
> resolution. I don't think it went anywhere. Do you think we ought to 
> revitalise that? You mention inference - have you tried that - are 
> there cases where it doesn't work?
>
> [IJ] Yes, actually wondered what happened to Richards draft. I only 
> briefly tried out the inference algorithm with a TCP LEDBAT 
> implementation 
> (https://github.com/silviov/TCP-LEDBAT/blob/master/src/tcp_ledbat.c). 
> I recall that it took a while to converge. Also I am not sure here. Is 
> Hz in e.g a linux stack fixed. ?
>
I believe so. But I don't know about embedded systems etc.

Cheers



Bob
>
>
>
>
> That's it. Thanks again for the draft. V useful.
>
>
>
> Bob
>
>
> On 15/10/15 06:08, Ingemar Johansson S wrote:
>
>     Hi
>
>     A new IETF draft is submitted in response to general discussion that for instance TCP Cubic is not an appropriate congestion control algorithm for LTE.
>
>     The draft briefly outlines typical 4G access behavior on the PDCP, RLC and MAC  layers that can have an impact on transport protocol performance. The draft also outlines two relatively simple modifications to the Cubic congestion control.
>
>     /Ingemar
>
>     -----Original Message-----
>
>     From:internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>  [mailto:internet-drafts@ietf.org]
>
>     Sent: den 14 oktober 2015 20:17
>
>     To: Ingemar Johansson S
>
>     Subject: New Version Notification for draft-johansson-cc-for-4g-5g-00.txt
>
>     A new version of I-D, draft-johansson-cc-for-4g-5g-00.txt
>
>     has been successfully submitted by Ingemar Johansson and posted to the IETF repository.
>
>     Name:           draft-johansson-cc-for-4g-5g
>
>     Revision:       00
>
>     Title:          Congestion control for 4G and 5G access
>
>     Document date:  2015-10-14
>
>     Group:          Individual Submission
>
>     Pages:          14
>
>     URL:https://www.ietf.org/internet-drafts/draft-johansson-cc-for-4g-5g-00.txt
>
>     Status:https://datatracker.ietf.org/doc/draft-johansson-cc-for-4g-5g/
>
>     Htmlized:https://tools.ietf.org/html/draft-johansson-cc-for-4g-5g-00
>
>     Abstract:
>
>         This memo outlines the challenge that 4G and 5G access brings for
>
>         transport protocol congestion control and also outlines a few simple
>
>         examples that can improve transport protocol congestion control
>
>         performance in 4G and 5G access.
>
>                                                                                        
>
>     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.
>
>     The IETF Secretariat
>
>
>
>
> -- 
> ________________________________________________________________
> Bob Briscoehttp://bobbriscoe.net/

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


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    Ingemar,<br>
    <br>
    <div class="moz-cite-prefix">On 04/11/15 13:27, Ingemar Johansson S
      wrote:<br>
    </div>
    <blockquote
cite="mid:81564C0D7D4D2A4B9A86C8C7404A13DA34EB1658@ESESSMB205.ericsson.se"
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      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">Hi<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">And thanks for the review,
            comments/answers/questions inline below<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">/Ingemar<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>Â </o:p></span></p>
        <div style="border:none;border-left:solid blue 1.5pt;padding:0cm
          0cm 0cm 4.0pt">
          <div>
            <div style="border:none;border-top:solid #B5C4DF
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              <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext"
                    lang="EN-US">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext"
                  lang="EN-US"> Bob Briscoe [<a class="moz-txt-link-freetext" href="mailto:ietf@bobbriscoe.net">mailto:ietf@bobbriscoe.net</a>]
                  <br>
                  <b>Sent:</b> den 29 oktober 2015 18:54<br>
                  <b>To:</b> Ingemar Johansson S; <a class="moz-txt-link-abbreviated" href="mailto:iccrg@irtf.org">iccrg@irtf.org</a>;
                  <a class="moz-txt-link-abbreviated" href="mailto:tcpm@ietf.org">tcpm@ietf.org</a>; <a class="moz-txt-link-abbreviated" href="mailto:tsvwg@ietf.org">tsvwg@ietf.org</a><br>
                  <b>Subject:</b> Re: [tsvwg] FW: New Version
                  Notification for draft-johansson-cc-for-4g-5g-00.txt<o:p></o:p></span></p>
            </div>
          </div>
          <p class="MsoNormal"><o:p>Â </o:p></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt">Ingemar,<br>
            <br>
            As promised, here's my comments. They are a mix of suggested
            improvements to the draft, suggested improvements to 5G, or
            requests for clarification. I also agree with all Kevin
            Smith's comments, so I didn't repeat any.<br>
            [Sorry for delay - I had to find time to re-type after
            losing my review during a power outage]<br>
            <br>
            <b><u>Context questions.</u><br>
            </b>Q1. Is this exercise treating the 5G specs solely in
            read-only mode? Or, if some useful design suggestions arise,
            is there any chance that the 3GPP might take them on board,
            given 5G is not fully baked yet?<span style="color:#1F497D"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] Guess it depends, need to say that I
              probably the small cog in the wheel. I would say that a
              lot of the standards work in IETF has been picked up by
              3GPP and, I cannot say what will happen in the 5G context.
            </span><span lang="EN-US"><br>
              <br>
              It's great that you have taken this initiative. </span>Does
            the 3GPP ever formally ask the IETF for comments on its
            ideas for each new generation of radio? Given the layers
            above can be thought of as the 'customers' of the 3GPP, and
            these 'customer' layers are pretty much all maintained by
            the IETF, it would seem essential (and courteous) for the
            3GPP to ask for customer comment, not least to co-ordinate
            with whatever future plans the IETF might have for the
            higher layers.<span style="color:#1F497D"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] What you say makes much sense. I guess
              there is some natural leakage between 3GPP and IETF as
              many of the IETFers are also active in 3GPP
              standardization. I donâ€™t have full insight in the formal
              relation between 3GPP and IETF in these areas, not sure if
              that need to be better, but if that is the case, then it
              is probably a question for the area directors. Need to say
              that even though the formal channels are scarce it does
              probably not mean that 3GPP ignores IETF or what happens
              above IP in general. For instance the recent work around
              TCP RACK raises the question if it is necessary that LTE
              default radio bearers ensure in order delivery. </span></p>
        </div>
      </div>
    </blockquote>
    I suspect that 3GPP radio resource control people do not
    cross-fertilize much with the IETF.<br>
    So maybe it is as much a question of how to co-ordinate between
    these radio people and L4, which I would imagine is as much a
    problem within 3GPP as between 3GPP and IETF.<br>
    <blockquote
cite="mid:81564C0D7D4D2A4B9A86C8C7404A13DA34EB1658@ESESSMB205.ericsson.se"
      type="cite">
      <div class="WordSection1">
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          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              <br>
              Q2. What are your plans for this draft? </span>Are you
            asking for adoption? And if so, in which WG or RG? ICCRG
            looks the most appropriate, given the charters of all the
            IETF cc WGs (TSVWG, TCPM, RMCAT, etc) are scoped on just a
            subset of your doc.<span style="color:#1F497D"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] First objective is to get an increased
              interest in congestion control algorithm work in the IETF.
              Traditionally this kind of work seem to be more in the
              terms of conference papers in e.g. Sigcomm, the benefit
              with having such work in IETF is that input from people
              with a wider expertise area (including 3GPP experitise)
              can give algorithm proposals that work well for a wider
              set of access types. The work around Cubic is a first good
              step as is the work around DCTCP. What I wonder is if it
              feasible to take this a bit further ? I would too believe
              that the current doc fits better in ICCRG. Â The algorithm
              examples are just.. examples. More fruitful work is
              probably to document different access types and from there
              derive a set of best current practices for congestion
              control algorithms</span></p>
        </div>
      </div>
    </blockquote>
    I'm sure the iccrg chairs would like this to happen. And initiatives
    like yours to come with proposals to propose specific code to take
    account of L2 I think generate the right level of dialogue.<br>
    <blockquote
cite="mid:81564C0D7D4D2A4B9A86C8C7404A13DA34EB1658@ESESSMB205.ericsson.se"
      type="cite">
      <div class="WordSection1">
        <div style="border:none;border-left:solid blue 1.5pt;padding:0cm
          0cm 0cm 4.0pt">
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              <br>
              <b><u>Section by Section Review</u><br>
              </b><br>
              <b>All sections<br>
              </b><br>
              The document generally talks about the LTE protocol stack
              as if data is travelling down it.
            </span>Are you implying that most sources of buffering delay
            are at the sending end, and the feedback end is pretty
            unencumbered by delays? I think it would be useful to
            explicitly say something about this (whether true or not),
            to explain why only one direction of travel is discussed in
            the doc.<span style="color:#1F497D"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] The feedback path is encumbered by
              delays, it is perhaps not that clear but section 2.3.2
              implies that delay (variation) in uplink will affect the
              ACKs for downlink data traffic. That said, the difference
              in DL/UL traffic is today something like 10:1 and that is
              probably one reason why one can get the feeling that the
              draft is slanted towards downlink data traffic, this may
              however change with time as more machine type traffic may
              enter the system.</span></p>
        </div>
      </div>
    </blockquote>
    I was thinking more in terms of up the stack vs. down the stack, not
    uplink/downlink. I.e. for both cases:<br>
    * downlink direction: are there any buffering concerns coming up the
    stack at the UE, or on the feedback back-channel from the UE?<br>
    * uplink direction: are there any buffering concerns coming up the
    stack at the eNodeB, or on the feedback back-channel from the
    eNodeB?<br>
    <br>
    For instance, in the UE, perhaps one app blocking the read of
    another due to the way {3-5}G multiplexes different streams into the
    same PDCP frames?<br>
    <blockquote
cite="mid:81564C0D7D4D2A4B9A86C8C7404A13DA34EB1658@ESESSMB205.ericsson.se"
      type="cite">
      <div class="WordSection1">
        <div style="border:none;border-left:solid blue 1.5pt;padding:0cm
          0cm 0cm 4.0pt">
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              <br>
              <b>2.1 PDCP layer<br>
              </b><br>
              Â Â Â  </span><tt><span style="font-size:10.0pt"
                lang="EN-US">PDCP also ensures that all</span></tt><span
              style="font-size:10.0pt;font-family:&quot;Courier
              New&quot;" lang="EN-US"><br>
              <tt>Â Â  packets are delivered in order up to higher layers.</tt><br>
            </span>I think this means "in the order sent into the link
            (PDCP) layer". This ought to be clarified, otherwise, when
            writing in the context of a layer 4 audience, it might be
            assumed this means in the order sent by the original
            transport layer source.
            <span style="color:#1F497D"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] Yes, you are right , reordering can
              still occur e.g. in the wireless backhaul<o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              <br>
            </span><tt><span style="font-size:10.0pt" lang="EN-US">Â Â 
                keep the amount of data in flight as small as possible,
                without sacrificing throughput.</span></tt><span
              style="font-size:10.0pt;font-family:&quot;Courier
              New&quot;" lang="EN-US"><br>
            </span>Is this just a rather convoluted way of saying "keep
            the RTT as low as possible"? (which in turn implies keeping
            queuing delay as low as possible assuming you cannot
            influence the physical path.) Or were you trying to make a
            more subtle point? - if so, it was lost on me.<span
              style="color:#1F497D"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] No, it is exactly as you describe below<o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              <br>
            </span><tt><span style="font-size:10.0pt" lang="EN-US">Â Â 
                Reliable delivery at handover may however be turned off
                or is simply not implemented,</span></tt><span
              style="font-size:10.0pt;font-family:&quot;Courier
              New&quot;" lang="EN-US"><br>
            </span><span lang="EN-US">It would be useful if you could
              give a feel for roughly how often (in %) these cases hold.</span><span
              style="color:#1F497D" lang="EN-US"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] I donâ€™t believe that it is possible to
              give a figure as this is vendor specific.<o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              <br>
              <b>2.2 RLC layer</b><br>
              <br>
            </span><tt><span style="font-size:10.0pt" lang="EN-US">Â Â 
                role ... is to ensure that packets are delivered in
                order up to the higher layers</span></tt><span
              style="font-size:10.0pt;font-family:&quot;Courier
              New&quot;" lang="EN-US"><br>
            </span><span lang="EN-US">Eh? </span>The PDCP section just
            said it did ordering. And the rest of the RLC section talks
            exclusively about buffering delays necessary while filling
            transport blocks. If ordering is the role of both layers,
            surely one will have nothing to do once the other has got
            everything in order?<span style="color:#1F497D"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] RLC handles ordering due to various MAC
              layer failures. PDCP handles ordering Â when handover
              occurs<o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              <br>
            </span><tt><span style="font-size:10.0pt" lang="EN-US">Â Â 
                varying size of the available transport</span></tt><span
              style="font-size:10.0pt;font-family:&quot;Courier
              New&quot;" lang="EN-US"><br>
            </span><span lang="EN-US">Nit: Given 'transport' means e2e
              for the audience of this draft, pls use bearer or somehow
              disambiguate this word.</span><span style="color:#1F497D"
              lang="EN-US"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] OK<o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              <br>
              <br>
              Gap#1: In the draft, it seems like transport blocks arrive
              by magic, to be filled by data buffered in the RLC layer.
            </span>To understand the delay compromises here, I think the
            draft needs to explain what determines how much transport
            block capacity each flow is given, and how they can
            influence it. In 3G, I believe that the transport layer
            could ask the radio resource controller to change radio
            resource allocation. Is this the same in 4G &amp; 5G? Does
            the backlog of the buffer into the RLC layer have any
            automatic influence over allocation of available resources?
            I assume traffic class can also be used to influence
            allocation.<span style="color:#1F497D"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] Will updated with more detail<o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              <br>
              Gap#2: In a 3G stack, the RLC layer multiplexes all the
              active application streams into MAC layer transport block
              by dicing up buffered packets and packing the pieces into
              each transport block.
            </span>Then, at the other end of the link, each piece has to
            be held until the rest of the packet arrives. This
            sacrifices delay for bandwidth efficiency. Is this still
            done in 4G and 5G? If so, it would be good to write about
            mux delay in the draft too.<span style="color:#1F497D"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] OK, will add more info<o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              <br>
              <b>2.3 MAC layer<br>
              </b><br>
            </span><tt><span style="font-size:10.0pt" lang="EN-US">Â Â 
                the maximum number of retransmissions is configurable</span></tt><span
              style="font-size:10.0pt;font-family:&quot;Courier
              New&quot;" lang="EN-US"><br>
            </span><span lang="EN-US"><br>
              Wouldn't it be better to set a limit on the <i>time</i>
              spent retransmitting, so it would automatically give up
              with less retransmissions for longer transmission
              distances?<br>
            </span><span style="color:#1F497D" lang="EN-US">[IJ]
              Possible, 4G specifies a limit to the number of
              retransmissions, in practice it can be translated to time
              as each retransmission is 8ms later.<o:p></o:p></span></p>
        </div>
      </div>
    </blockquote>
    I didn't realise it was independent of link-RTT. Then the more basic
    question is why is it independent of link-RTT? Surely on a short
    link it can assume that it needs to retransmit more quickly than on
    a long link?<br>
    <br>
    You can see that my question was assuming that retransmission
    happened more often on a short link, then I was saying it should
    still be allowed the same amount of time to keep retrying (so on a
    short link it would be allowed to retransmit more times before
    giving up, but the max delay would be the same).<br>
    <br>
    8ms! I was assuming it would be rather much less than that. Surely a
    link-layer ACK could never take that long if the frame was going to
    get through.<br>
    <blockquote
cite="mid:81564C0D7D4D2A4B9A86C8C7404A13DA34EB1658@ESESSMB205.ericsson.se"
      type="cite">
      <div class="WordSection1">
        <div style="border:none;border-left:solid blue 1.5pt;padding:0cm
          0cm 0cm 4.0pt">
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US"><o:p>Â </o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              s/only checks for the presence on download data only at
              regular intervals/<br>
              Â /only checks for the presence of download data at regular
              intervals/<br>
            </span><span style="color:#1F497D" lang="EN-US">[OK]<o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US"><o:p>Â </o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              <b>2.3.2 Uplink scheduling<br>
              </b><br>
            </span><tt><span style="font-size:10.0pt" lang="EN-US">Â Â 
                The scheduling request does not indicate how many bytes
                there are in the uplink queue.</span></tt><span
              style="font-size:10.0pt;font-family:&quot;Courier
              New&quot;" lang="EN-US"><br>
            </span>Seems like a bad omission. Is there a reason? If
            there are less bytes queued than the base station's grant,
            surely it would be useful for the base station to know, so
            that it can grant more to something else while the previous
            short transmission is completing.<br>
            <span style="color:#1F497D" lang="EN-US">[IJ] It is a design
              choice. A buffer status report is more bulky, a scheduling
              request is just one bit.<o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US"><o:p>Â </o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              You say that when HARQ breaks up a sequence of ACKs, it
              can also trigger "coalescing issues".
            </span>I can understand that it could cause ACK compression,
            but are you implying something coalesces the ACKs or the
            subsequent data? What does the coalescing?
            <span lang="EN-US">Similarly, in the last para of 2.3.2 you
              say "Reduced ACKs can unfortunately cause coalescing." and
              again I'm not sure what is coalescing what here.</span><span
              style="color:#1F497D" lang="EN-US"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] What I can see in simulations is that
              the ACK compression leads to coalescing which further can
              increase jitter.</span></p>
        </div>
      </div>
    </blockquote>
    I still don't understand. The question was "What is coalescing
    what?"<br>
    <blockquote
cite="mid:81564C0D7D4D2A4B9A86C8C7404A13DA34EB1658@ESESSMB205.ericsson.se"
      type="cite">
      <div class="WordSection1">
        <div style="border:none;border-left:solid blue 1.5pt;padding:0cm
          0cm 0cm 4.0pt">
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              There could be a positive side to the interaction between
              ACK clocking and link aggregation, at least for
              long-running flows.
            </span>This has been observed in 802.11n WiFi where the
            buffering and aggregation of multiple ACKs into one
            transmission grant tend to lead the sender to bunch multiple
            data segments so that they all nicely fill one transmission
            grant in the other direction [Showail14]. <span
              lang="EN-US">As long as the transmission grant does not
              wait for more data (which unfortunately I don't think is
              the case in 3G/4G/5G), this could be good for bandwidth
              utilisation of elephants and latency of mice, including
              when they are mixed.<br>
              <br>
              [Showail14] Showail, A., Jamshaid, K. &amp; Shihada, B.,
              "WQM: An Aggregation-aware Queue Management Scheme for
              IEEE 802.11N Based Networks," In: Proceedings of the 2014
              ACM SIGCOMM Workshop on Capacity Sharing Workshop CSWS '14
              pp.15-20 ACM (2014)<br>
              &lt;</span><a moz-do-not-send="true"
              href="http://dl.acm.org/citation.cfm?id=2630097"><span
                lang="EN-US">http://dl.acm.org/citation.cfm?id=2630097</span></a><span
              lang="EN-US">&gt;<br>
            </span><span style="color:#1F497D" lang="EN-US">[IJ]
              Interesting ,will look at it, not sure though that it will
              work that well with 4G (cant say anything about 5G), in 4G
              the transmission grants can vary quite a lot over short
              time spans.<o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US"><o:p>Â </o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
            </span><b>3.Â  4G and 5G evolution<br>
            </b><br>
            s/a explicit/an explicit/<br>
            <br>
            <b>4.Â  Requirements for improved performance<br>
            </b><br>
            s/bufferbloat/buffer/<br>
            <br>
            <b>5.Â  Congestion control examples<br>
            </b><br>
            Regarding Hybrid Cubic using OWD measurements, I have seen a
            paper (under submission) about getting delay-based
            congestion controls (e.g. LEDBAT or CAIA Delay Gradient) to
            interwork with AQMs with various low delay targets. It's not
            in the context of cellular networks, but I'll try to
            remember to point it out once it gets published.<span
              style="color:#1F497D"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] Sounds interesting , looking forward to
              see the paper.<o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              <br>
              <b>5.1 HyStartRestart<br>
              </b><br>
              s/algorithm used/algorithm is used/<br>
              <br>
              In the new code, it would be useful to explain why the
              second test before doubling ssthresh, ie, the test for how
              long it has been since the last hyrestart.
            </span>Also, in the header definitions, if you intend there
            to be constraints on the relation between N_RTT_HYRESTART
            and N_RTT_LOW (e.g.Â  N_RTT_HYRESTART &gt; N_RTT_LOW) you
            ought to say, because people will try other values.<br>
            <br>
            Rather than the rather arbitrary wait before an arbitrary
            doubling of ssthresh, wouldn't it be better to continually
            increase ssthresh (not necessarily linearly) the longer the
            RTT has remained only just higher than the min RTT?<span
              style="color:#1F497D"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] Yes, could be an option<o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              <br>
              (BTW, surely the main problem with this modification to
              HyStartRestart is that is requires a number of round trips
              to reduce delay - a sort-of oxymoron.)</span><span
              style="color:#1F497D" lang="EN-US"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] Yes, have tried some more experiments
              with this algorithm. And I am getting more hesitant. The
              problem is that when I add more noise (modeling of small
              objects traffic) into the system simulation, then I also
              get more delay jitter and the delay detection algorithm in
              HyStart is notoriously sensitive to delay jitter.<o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              <br>
            </span><b>5.2.Â  Hybrid Cubic</b><span style="color:#1F497D"
              lang="EN-US"><o:p></o:p></span></p>
          <pre>Â Â  Furthermore it is assumed than the timestamp clock frequency in<o:p></o:p></pre>
          <pre>Â Â  sender and receiver are identical or that the sender can infer the<o:p></o:p></pre>
          <pre>Â Â  timestamp clock frequency of the receiver and recompute timestamp<o:p></o:p></pre>
          <pre>Â Â  values based on this information.<o:p></o:p></pre>
          <p class="MsoNormal" style="margin-bottom:12.0pt">Recently on
            tcpm I sent you a pointer to the chirping paper Mirja &amp;
            I did, which also made this assumption. Partly prompted by
            that, Richard Scheffenegger tried to get people interested
            in standardising use of the timestamp option on a SYN to
            negotiate stuff like timer resolution. I don't think it went
            anywhere. Do you think we ought to revitalise that? You
            mention inference - have you tried that - are there cases
            where it doesn't work?<span style="color:#1F497D"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">[IJ] Yes, actually wondered what happened to
              Richards draft. I only briefly tried out the inference
              algorithm with a TCP LEDBAT implementation (</span><span
style="font-size:9.0pt;font-family:Consolas;color:#969896;background:white"
              lang="EN-US"><a moz-do-not-send="true"
href="https://github.com/silviov/TCP-LEDBAT/blob/master/src/tcp_ledbat.c">https://github.com/silviov/TCP-LEDBAT/blob/master/src/tcp_ledbat.c</a></span><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">). I recall that it took a while to converge.
              Also I am not sure here. Is Hz in e.g a linux stack fixed.
              ?</span></p>
        </div>
      </div>
    </blockquote>
    I believe so. But I don't know about embedded systems etc.<br>
    <br>
    Cheers<br>
    <br>
    <br>
    <br>
    Bob<br>
    <blockquote
cite="mid:81564C0D7D4D2A4B9A86C8C7404A13DA34EB1658@ESESSMB205.ericsson.se"
      type="cite">
      <div class="WordSection1">
        <div style="border:none;border-left:solid blue 1.5pt;padding:0cm
          0cm 0cm 4.0pt">
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US"><o:p></o:p></span></p>
          <p class="MsoNormal" style="margin-bottom:12.0pt"><span
              lang="EN-US"><br>
              <br>
              <br>
              That's it. Thanks again for the draft. V useful.<br>
              <br>
              <br>
              <br>
              Bob<br>
              <br>
              <br>
              <o:p></o:p></span></p>
          <div>
            <p class="MsoNormal"><span lang="EN-US">On 15/10/15 06:08,
                Ingemar Johansson S </span>
              wrote:<o:p></o:p></p>
          </div>
          <blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
            <pre>Hi<o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>A new IETF draft is submitted in response to general discussion that for instance TCP Cubic is not an appropriate congestion control algorithm for LTE. <o:p></o:p></pre>
            <pre>The draft briefly outlines typical 4G access behavior on the PDCP, RLC and MACÂ  layers that can have an impact on transport protocol performance. The draft also outlines two relatively simple modifications to the Cubic congestion control.<o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>/IngemarÂ  <o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>-----Original Message-----<o:p></o:p></pre>
            <pre>From: <a moz-do-not-send="true" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a> [<a moz-do-not-send="true" href="mailto:internet-drafts@ietf.org">mailto:internet-drafts@ietf.org</a>] <o:p></o:p></pre>
            <pre>Sent: den 14 oktober 2015 20:17<o:p></o:p></pre>
            <pre>To: Ingemar Johansson S<o:p></o:p></pre>
            <pre>Subject: New Version Notification for draft-johansson-cc-for-4g-5g-00.txt<o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>A new version of I-D, draft-johansson-cc-for-4g-5g-00.txt<o:p></o:p></pre>
            <pre>has been successfully submitted by Ingemar Johansson and posted to the IETF repository.<o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>Name:Â Â Â Â Â Â Â Â Â Â  draft-johansson-cc-for-4g-5g<o:p></o:p></pre>
            <pre>Revision:Â Â Â Â Â Â  00<o:p></o:p></pre>
            <pre>Title:Â Â Â Â Â Â Â Â Â  Congestion control for 4G and 5G access<o:p></o:p></pre>
            <pre>Document date:Â  2015-10-14<o:p></o:p></pre>
            <pre>Group:Â Â Â Â Â Â Â Â Â  Individual Submission<o:p></o:p></pre>
            <pre>Pages:Â Â Â Â Â Â Â Â Â  14<o:p></o:p></pre>
            <pre>URL:Â Â Â Â Â Â Â Â Â Â Â  <a moz-do-not-send="true" href="https://www.ietf.org/internet-drafts/draft-johansson-cc-for-4g-5g-00.txt">https://www.ietf.org/internet-drafts/draft-johansson-cc-for-4g-5g-00.txt</a><o:p></o:p></pre>
            <pre>Status:Â Â Â Â Â Â Â Â  <a moz-do-not-send="true" href="https://datatracker.ietf.org/doc/draft-johansson-cc-for-4g-5g/">https://datatracker.ietf.org/doc/draft-johansson-cc-for-4g-5g/</a><o:p></o:p></pre>
            <pre>Htmlized:Â Â Â Â Â Â  <a moz-do-not-send="true" href="https://tools.ietf.org/html/draft-johansson-cc-for-4g-5g-00">https://tools.ietf.org/html/draft-johansson-cc-for-4g-5g-00</a><o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>Abstract:<o:p></o:p></pre>
            <pre>Â Â  This memo outlines the challenge that 4G and 5G access brings for<o:p></o:p></pre>
            <pre>Â Â  transport protocol congestion control and also outlines a few simple<o:p></o:p></pre>
            <pre>Â Â  examples that can improve transport protocol congestion control<o:p></o:p></pre>
            <pre>Â Â  performance in 4G and 5G access.<o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â  <o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>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.<o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
            <pre>The IETF Secretariat<o:p></o:p></pre>
            <pre><o:p>Â </o:p></pre>
          </blockquote>
          <p class="MsoNormal"><br>
            <br>
            <br>
            <o:p></o:p></p>
          <pre>-- <o:p></o:p></pre>
          <pre>________________________________________________________________<o:p></o:p></pre>
          <pre>Bob BriscoeÂ Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â  <a moz-do-not-send="true" href="http://bobbriscoe.net/">http://bobbriscoe.net/</a><o:p></o:p></pre>
        </div>
      </div>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
________________________________________________________________
Bob Briscoe                               <a class="moz-txt-link-freetext" href="http://bobbriscoe.net/">http://bobbriscoe.net/</a></pre>
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From: "Eggert, Lars" <lars@netapp.com>
To: tcpm IETF list <tcpm@ietf.org>
Thread-Topic: Papers I mentioned in the session
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Subject: [tcpm] Papers I mentioned in the session
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This is the paper I mentioned in the discussion for =
draft-kitamura-tcp-sharp-close:

	The TIME_WAIT state in TCP and Its Effect on Busy Servers
	http://www.isi.edu/touch/pubs/infocomm99/

This is the Ensemble-TCP paper related to RFC2140 (mentioned in the =
discussion for draft-welzl-tcpm-tcb-sharing):

	Effects of Ensemble-TCP
	https://www.isi.edu/~johnh/PAPERS/Eggert00a/

Finally, this is the RTT study from IMC, mentioned when discussing =
draft-nishida-tcpm-apaws:

	Timeouts: Beware Surprisingly High Delay
	=
http://www.cs.umd.edu/~ramapad/docs/timeouts_imc_15_camera_ready.pdf

Lars

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Subject: [tcpm] Fwd: Re: Promises to review draft-...-tcpm-accurate-ecn
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tcpm note-takers, and the list,

Pls shout if I missed anyone, or if you were scratching your head rather 
than putting up your hand.


Bob

-------- Forwarded Message --------
Subject: 	Re: Promises to review draft-...-tcpm-accurate-ecn
Date: 	Thu, 5 Nov 2015 10:18:40 +0900
From: 	Pasi Sarolahti <pasi.sarolahti@iki.fi>
To: 	Bob Briscoe <research@bobbriscoe.net>
CC: 	Mirja Kühlewind <mirja.kuehlewind@tik.ee.ethz.ch>, Richard 
Scheffenegger <rs@netapp.com>, Yoshifumi Nishida 
<nishida@sfc.wide.ad.jp>, Michael Scharf 
<Michael.SCHARF@alcatel-lucent.com>



Thanks, very helpful! (you could also post it to list, so people know they are tagged :-)

- Pasi


On 05 Nov 2015, at 10:14, Bob Briscoe<research@bobbriscoe.net>  wrote:

> Mirja, Richard,
>
> I noted the following promises to review AccECN (I'll check this gets into the minutes):
>
> Gorry Fairhurst
> Praveen Balasubramian
> Koen De Schepper
> Brian Trammell
> Tommy Pauly
> Karen Nielsen?
>
> Promise to test:
> CableLabs guy (sorry don't know his name)
>
>
> Bob
>
>
> --
> ________________________________________________________________
> Bob Briscoehttp://bobbriscoe.net/
>




-- 
________________________________________________________________
Bob Briscoehttp://bobbriscoe.net/


--------------070802030604090208000104
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<html>
  <head>
    <meta http-equiv="content-type" content="text/html;
      charset=windows-1252">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    tcpm note-takers, and the list,<br>
    <br>
    Pls shout if I missed anyone, or if you were scratching your head
    rather than putting up your hand.<br>
    <br>
    <br>
    Bob<br>
    <div class="moz-forward-container"><br>
      -------- Forwarded Message --------
      <table class="moz-email-headers-table" border="0" cellpadding="0"
        cellspacing="0">
        <tbody>
          <tr>
            <th nowrap="nowrap" valign="BASELINE" align="RIGHT">Subject:
            </th>
            <td>Re: Promises to review draft-...-tcpm-accurate-ecn</td>
          </tr>
          <tr>
            <th nowrap="nowrap" valign="BASELINE" align="RIGHT">Date: </th>
            <td>Thu, 5 Nov 2015 10:18:40 +0900</td>
          </tr>
          <tr>
            <th nowrap="nowrap" valign="BASELINE" align="RIGHT">From: </th>
            <td>Pasi Sarolahti <a class="moz-txt-link-rfc2396E"
                href="mailto:pasi.sarolahti@iki.fi">&lt;pasi.sarolahti@iki.fi&gt;</a></td>
          </tr>
          <tr>
            <th nowrap="nowrap" valign="BASELINE" align="RIGHT">To: </th>
            <td>Bob Briscoe <a class="moz-txt-link-rfc2396E"
                href="mailto:research@bobbriscoe.net">&lt;research@bobbriscoe.net&gt;</a></td>
          </tr>
          <tr>
            <th nowrap="nowrap" valign="BASELINE" align="RIGHT">CC: </th>
            <td>Mirja Kühlewind <a class="moz-txt-link-rfc2396E"
                href="mailto:mirja.kuehlewind@tik.ee.ethz.ch">&lt;mirja.kuehlewind@tik.ee.ethz.ch&gt;</a>,
              Richard Scheffenegger <a class="moz-txt-link-rfc2396E"
                href="mailto:rs@netapp.com">&lt;rs@netapp.com&gt;</a>,
              Yoshifumi Nishida <a class="moz-txt-link-rfc2396E"
                href="mailto:nishida@sfc.wide.ad.jp">&lt;nishida@sfc.wide.ad.jp&gt;</a>,
              Michael Scharf <a class="moz-txt-link-rfc2396E"
                href="mailto:Michael.SCHARF@alcatel-lucent.com">&lt;Michael.SCHARF@alcatel-lucent.com&gt;</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>Thanks, very helpful! (you could also post it to list, so people know they are tagged :-)

- Pasi


On 05 Nov 2015, at 10:14, Bob Briscoe <a class="moz-txt-link-rfc2396E" href="mailto:research@bobbriscoe.net">&lt;research@bobbriscoe.net&gt;</a> wrote:

&gt; Mirja, Richard,
&gt; 
&gt; I noted the following promises to review AccECN (I'll check this gets into the minutes):
&gt; 
&gt; Gorry Fairhurst
&gt; Praveen Balasubramian
&gt; Koen De Schepper
&gt; Brian Trammell
&gt; Tommy Pauly
&gt; Karen Nielsen?
&gt; 
&gt; Promise to test:
&gt; CableLabs guy (sorry don't know his name)
&gt; 
&gt; 
&gt; Bob
&gt; 
&gt; 
&gt; -- 
&gt; ________________________________________________________________
&gt; Bob Briscoe                               <a class="moz-txt-link-freetext" href="http://bobbriscoe.net/">http://bobbriscoe.net/</a>
&gt; 

</pre>
      <br>
    </div>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
________________________________________________________________
Bob Briscoe                               <a class="moz-txt-link-freetext" href="http://bobbriscoe.net/">http://bobbriscoe.net/</a></pre>
  </body>
</html>

--------------070802030604090208000104--


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Date: Thu, 5 Nov 2015 14:12:37 +0900
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From: Yuchung Cheng <ycheng@google.com>
To: Bob Briscoe <ietf@bobbriscoe.net>
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Cc: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>, "tsvwg@ietf.org" <tsvwg@ietf.org>, "iccrg@irtf.org" <iccrg@irtf.org>
Subject: Re: [tcpm] [tsvwg] FW: New Version Notification for draft-johansson-cc-for-4g-5g-00.txt
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I have a question re: how much buffer or backlog is needed to prevent the
schedular from reducing the rate or even increase the rate. I think this
key why cubic or bufferbloat cc wins bc other queue controlling cc would
lose out.

For example if the scheduler needs 1mb backlog to maintain at 40mbps. A
dctcp or vegas keeping the queue well below will get say 5mbps.

The draft mentions about this but the advise is a bit vague for cc designer=
.
On Nov 5, 2015 7:27 AM, "Bob Briscoe" <ietf@bobbriscoe.net> wrote:

> Ingemar,
>
> On 04/11/15 13:27, Ingemar Johansson S wrote:
>
> Hi
>
>
>
> And thanks for the review, comments/answers/questions inline below
>
>
>
> /Ingemar
>
>
>
> *From:* Bob Briscoe [mailto:ietf@bobbriscoe.net <ietf@bobbriscoe.net>]
> *Sent:* den 29 oktober 2015 18:54
> *To:* Ingemar Johansson S; iccrg@irtf.org; tcpm@ietf.org; tsvwg@ietf.org
> *Subject:* Re: [tsvwg] FW: New Version Notification for
> draft-johansson-cc-for-4g-5g-00.txt
>
>
>
> Ingemar,
>
> As promised, here's my comments. They are a mix of suggested improvements
> to the draft, suggested improvements to 5G, or requests for clarification=
.
> I also agree with all Kevin Smith's comments, so I didn't repeat any.
> [Sorry for delay - I had to find time to re-type after losing my review
> during a power outage]
>
>
> *Context questions. *Q1. Is this exercise treating the 5G specs solely in
> read-only mode? Or, if some useful design suggestions arise, is there any
> chance that the 3GPP might take them on board, given 5G is not fully bake=
d
> yet?
>
> [IJ] Guess it depends, need to say that I probably the small cog in the
> wheel. I would say that a lot of the standards work in IETF has been pick=
ed
> up by 3GPP and, I cannot say what will happen in the 5G context.
>
> It's great that you have taken this initiative. Does the 3GPP ever
> formally ask the IETF for comments on its ideas for each new generation o=
f
> radio? Given the layers above can be thought of as the 'customers' of the
> 3GPP, and these 'customer' layers are pretty much all maintained by the
> IETF, it would seem essential (and courteous) for the 3GPP to ask for
> customer comment, not least to co-ordinate with whatever future plans the
> IETF might have for the higher layers.
>
> [IJ] What you say makes much sense. I guess there is some natural leakage
> between 3GPP and IETF as many of the IETFers are also active in 3GPP
> standardization. I don=E2=80=99t have full insight in the formal relation=
 between
> 3GPP and IETF in these areas, not sure if that need to be better, but if
> that is the case, then it is probably a question for the area directors.
> Need to say that even though the formal channels are scarce it does
> probably not mean that 3GPP ignores IETF or what happens above IP in
> general. For instance the recent work around TCP RACK raises the question
> if it is necessary that LTE default radio bearers ensure in order deliver=
y.
>
> I suspect that 3GPP radio resource control people do not cross-fertilize
> much with the IETF.
> So maybe it is as much a question of how to co-ordinate between these
> radio people and L4, which I would imagine is as much a problem within 3G=
PP
> as between 3GPP and IETF.
>
>
>
> Q2. What are your plans for this draft? Are you asking for adoption? And
> if so, in which WG or RG? ICCRG looks the most appropriate, given the
> charters of all the IETF cc WGs (TSVWG, TCPM, RMCAT, etc) are scoped on
> just a subset of your doc.
>
> [IJ] First objective is to get an increased interest in congestion contro=
l
> algorithm work in the IETF. Traditionally this kind of work seem to be mo=
re
> in the terms of conference papers in e.g. Sigcomm, the benefit with havin=
g
> such work in IETF is that input from people with a wider expertise area
> (including 3GPP experitise) can give algorithm proposals that work well f=
or
> a wider set of access types. The work around Cubic is a first good step a=
s
> is the work around DCTCP. What I wonder is if it feasible to take this a
> bit further ? I would too believe that the current doc fits better in
> ICCRG.  The algorithm examples are just.. examples. More fruitful work is
> probably to document different access types and from there derive a set o=
f
> best current practices for congestion control algorithms
>
> I'm sure the iccrg chairs would like this to happen. And initiatives like
> yours to come with proposals to propose specific code to take account of =
L2
> I think generate the right level of dialogue.
>
>
>
>
> *Section by Section Review *
>
> *All sections *
> The document generally talks about the LTE protocol stack as if data is
> travelling down it. Are you implying that most sources of buffering delay
> are at the sending end, and the feedback end is pretty unencumbered by
> delays? I think it would be useful to explicitly say something about this
> (whether true or not), to explain why only one direction of travel is
> discussed in the doc.
>
> [IJ] The feedback path is encumbered by delays, it is perhaps not that
> clear but section 2.3.2 implies that delay (variation) in uplink will
> affect the ACKs for downlink data traffic. That said, the difference in
> DL/UL traffic is today something like 10:1 and that is probably one reaso=
n
> why one can get the feeling that the draft is slanted towards downlink da=
ta
> traffic, this may however change with time as more machine type traffic m=
ay
> enter the system.
>
> I was thinking more in terms of up the stack vs. down the stack, not
> uplink/downlink. I.e. for both cases:
> * downlink direction: are there any buffering concerns coming up the stac=
k
> at the UE, or on the feedback back-channel from the UE?
> * uplink direction: are there any buffering concerns coming up the stack
> at the eNodeB, or on the feedback back-channel from the eNodeB?
>
> For instance, in the UE, perhaps one app blocking the read of another due
> to the way {3-5}G multiplexes different streams into the same PDCP frames=
?
>
>
>
>
> *2.1 PDCP layer *
>     PDCP also ensures that all
>    packets are delivered in order up to higher layers.
> I think this means "in the order sent into the link (PDCP) layer". This
> ought to be clarified, otherwise, when writing in the context of a layer =
4
> audience, it might be assumed this means in the order sent by the origina=
l
> transport layer source.
>
> [IJ] Yes, you are right , reordering can still occur e.g. in the wireless
> backhaul
>
>
>
>    keep the amount of data in flight as small as possible, without
> sacrificing throughput.
> Is this just a rather convoluted way of saying "keep the RTT as low as
> possible"? (which in turn implies keeping queuing delay as low as possibl=
e
> assuming you cannot influence the physical path.) Or were you trying to
> make a more subtle point? - if so, it was lost on me.
>
> [IJ] No, it is exactly as you describe below
>
>
>
>    Reliable delivery at handover may however be turned off or is simply
> not implemented,
> It would be useful if you could give a feel for roughly how often (in %)
> these cases hold.
>
> [IJ] I don=E2=80=99t believe that it is possible to give a figure as this=
 is
> vendor specific.
>
>
>
> *2.2 RLC layer*
>
>    role ... is to ensure that packets are delivered in order up to the
> higher layers
> Eh? The PDCP section just said it did ordering. And the rest of the RLC
> section talks exclusively about buffering delays necessary while filling
> transport blocks. If ordering is the role of both layers, surely one will
> have nothing to do once the other has got everything in order?
>
> [IJ] RLC handles ordering due to various MAC layer failures. PDCP handles
> ordering  when handover occurs
>
>
>
>    varying size of the available transport
> Nit: Given 'transport' means e2e for the audience of this draft, pls use
> bearer or somehow disambiguate this word.
>
> [IJ] OK
>
>
>
>
> Gap#1: In the draft, it seems like transport blocks arrive by magic, to b=
e
> filled by data buffered in the RLC layer. To understand the delay
> compromises here, I think the draft needs to explain what determines how
> much transport block capacity each flow is given, and how they can
> influence it. In 3G, I believe that the transport layer could ask the rad=
io
> resource controller to change radio resource allocation. Is this the same
> in 4G & 5G? Does the backlog of the buffer into the RLC layer have any
> automatic influence over allocation of available resources? I assume
> traffic class can also be used to influence allocation.
>
> [IJ] Will updated with more detail
>
>
>
> Gap#2: In a 3G stack, the RLC layer multiplexes all the active applicatio=
n
> streams into MAC layer transport block by dicing up buffered packets and
> packing the pieces into each transport block. Then, at the other end of
> the link, each piece has to be held until the rest of the packet arrives.
> This sacrifices delay for bandwidth efficiency. Is this still done in 4G
> and 5G? If so, it would be good to write about mux delay in the draft too=
.
>
> [IJ] OK, will add more info
>
>
>
>
> *2.3 MAC layer *
>    the maximum number of retransmissions is configurable
>
> Wouldn't it be better to set a limit on the *time* spent retransmitting,
> so it would automatically give up with less retransmissions for longer
> transmission distances?
> [IJ] Possible, 4G specifies a limit to the number of retransmissions, in
> practice it can be translated to time as each retransmission is 8ms later=
.
>
> I didn't realise it was independent of link-RTT. Then the more basic
> question is why is it independent of link-RTT? Surely on a short link it
> can assume that it needs to retransmit more quickly than on a long link?
>
> You can see that my question was assuming that retransmission happened
> more often on a short link, then I was saying it should still be allowed
> the same amount of time to keep retrying (so on a short link it would be
> allowed to retransmit more times before giving up, but the max delay woul=
d
> be the same).
>
> 8ms! I was assuming it would be rather much less than that. Surely a
> link-layer ACK could never take that long if the frame was going to get
> through.
>
>
>
>
> s/only checks for the presence on download data only at regular intervals=
/
>  /only checks for the presence of download data at regular intervals/
> [OK]
>
>
>
>
>
> *2.3.2 Uplink scheduling *
>    The scheduling request does not indicate how many bytes there are in
> the uplink queue.
> Seems like a bad omission. Is there a reason? If there are less bytes
> queued than the base station's grant, surely it would be useful for the
> base station to know, so that it can grant more to something else while t=
he
> previous short transmission is completing.
> [IJ] It is a design choice. A buffer status report is more bulky, a
> scheduling request is just one bit.
>
>
>
>
> You say that when HARQ breaks up a sequence of ACKs, it can also trigger
> "coalescing issues". I can understand that it could cause ACK
> compression, but are you implying something coalesces the ACKs or the
> subsequent data? What does the coalescing? Similarly, in the last para of
> 2.3.2 you say "Reduced ACKs can unfortunately cause coalescing." and agai=
n
> I'm not sure what is coalescing what here.
>
> [IJ] What I can see in simulations is that the ACK compression leads to
> coalescing which further can increase jitter.
>
> I still don't understand. The question was "What is coalescing what?"
>
>
> There could be a positive side to the interaction between ACK clocking an=
d
> link aggregation, at least for long-running flows. This has been observed
> in 802.11n WiFi where the buffering and aggregation of multiple ACKs into
> one transmission grant tend to lead the sender to bunch multiple data
> segments so that they all nicely fill one transmission grant in the other
> direction [Showail14]. As long as the transmission grant does not wait
> for more data (which unfortunately I don't think is the case in 3G/4G/5G)=
,
> this could be good for bandwidth utilisation of elephants and latency of
> mice, including when they are mixed.
>
> [Showail14] Showail, A., Jamshaid, K. & Shihada, B., "WQM: An
> Aggregation-aware Queue Management Scheme for IEEE 802.11N Based Networks=
,"
> In: Proceedings of the 2014 ACM SIGCOMM Workshop on Capacity Sharing
> Workshop CSWS '14 pp.15-20 ACM (2014)
> <http://dl.acm.org/citation.cfm?id=3D2630097>
> [IJ] Interesting ,will look at it, not sure though that it will work that
> well with 4G (cant say anything about 5G), in 4G the transmission grants
> can vary quite a lot over short time spans.
>
>
>
>
>
> *3.  4G and 5G evolution *
> s/a explicit/an explicit/
>
>
> *4.  Requirements for improved performance *
> s/bufferbloat/buffer/
>
>
> *5.  Congestion control examples *
> Regarding Hybrid Cubic using OWD measurements, I have seen a paper (under
> submission) about getting delay-based congestion controls (e.g. LEDBAT or
> CAIA Delay Gradient) to interwork with AQMs with various low delay target=
s.
> It's not in the context of cellular networks, but I'll try to remember to
> point it out once it gets published.
>
> [IJ] Sounds interesting , looking forward to see the paper.
>
>
>
>
> *5.1 HyStartRestart *
> s/algorithm used/algorithm is used/
>
> In the new code, it would be useful to explain why the second test before
> doubling ssthresh, ie, the test for how long it has been since the last
> hyrestart. Also, in the header definitions, if you intend there to be
> constraints on the relation between N_RTT_HYRESTART and N_RTT_LOW (e.g.
> N_RTT_HYRESTART > N_RTT_LOW) you ought to say, because people will try
> other values.
>
> Rather than the rather arbitrary wait before an arbitrary doubling of
> ssthresh, wouldn't it be better to continually increase ssthresh (not
> necessarily linearly) the longer the RTT has remained only just higher th=
an
> the min RTT?
>
> [IJ] Yes, could be an option
>
>
>
> (BTW, surely the main problem with this modification to HyStartRestart is
> that is requires a number of round trips to reduce delay - a sort-of
> oxymoron.)
>
> [IJ] Yes, have tried some more experiments with this algorithm. And I am
> getting more hesitant. The problem is that when I add more noise (modelin=
g
> of small objects traffic) into the system simulation, then I also get mor=
e
> delay jitter and the delay detection algorithm in HyStart is notoriously
> sensitive to delay jitter.
>
>
>
> *5.2.  Hybrid Cubic*
>
>    Furthermore it is assumed than the timestamp clock frequency in
>
>    sender and receiver are identical or that the sender can infer the
>
>    timestamp clock frequency of the receiver and recompute timestamp
>
>    values based on this information.
>
> Recently on tcpm I sent you a pointer to the chirping paper Mirja & I did=
,
> which also made this assumption. Partly prompted by that, Richard
> Scheffenegger tried to get people interested in standardising use of the
> timestamp option on a SYN to negotiate stuff like timer resolution. I don=
't
> think it went anywhere. Do you think we ought to revitalise that? You
> mention inference - have you tried that - are there cases where it doesn'=
t
> work?
>
> [IJ] Yes, actually wondered what happened to Richards draft. I only
> briefly tried out the inference algorithm with a TCP LEDBAT implementatio=
n (
> https://github.com/silviov/TCP-LEDBAT/blob/master/src/tcp_ledbat.c). I
> recall that it took a while to converge. Also I am not sure here. Is Hz i=
n
> e.g a linux stack fixed. ?
>
> I believe so. But I don't know about embedded systems etc.
>
> Cheers
>
>
>
> Bob
>
>
>
>
> That's it. Thanks again for the draft. V useful.
>
>
>
> Bob
>
>
> On 15/10/15 06:08, Ingemar Johansson S wrote:
>
> Hi
>
>
>
> A new IETF draft is submitted in response to general discussion that for =
instance TCP Cubic is not an appropriate congestion control algorithm for L=
TE.
>
> The draft briefly outlines typical 4G access behavior on the PDCP, RLC an=
d MAC  layers that can have an impact on transport protocol performance. Th=
e draft also outlines two relatively simple modifications to the Cubic cong=
estion control.
>
>
>
> /Ingemar
>
>
>
> -----Original Message-----
>
> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org <internet=
-drafts@ietf.org>]
>
> Sent: den 14 oktober 2015 20:17
>
> To: Ingemar Johansson S
>
> Subject: New Version Notification for draft-johansson-cc-for-4g-5g-00.txt
>
>
>
>
>
> A new version of I-D, draft-johansson-cc-for-4g-5g-00.txt
>
> has been successfully submitted by Ingemar Johansson and posted to the IE=
TF repository.
>
>
>
> Name:           draft-johansson-cc-for-4g-5g
>
> Revision:       00
>
> Title:          Congestion control for 4G and 5G access
>
> Document date:  2015-10-14
>
> Group:          Individual Submission
>
> Pages:          14
>
> URL:            https://www.ietf.org/internet-drafts/draft-johansson-cc-f=
or-4g-5g-00.txt
>
> Status:         https://datatracker.ietf.org/doc/draft-johansson-cc-for-4=
g-5g/
>
> Htmlized:       https://tools.ietf.org/html/draft-johansson-cc-for-4g-5g-=
00
>
>
>
>
>
> Abstract:
>
>    This memo outlines the challenge that 4G and 5G access brings for
>
>    transport protocol congestion control and also outlines a few simple
>
>    examples that can improve transport protocol congestion control
>
>    performance in 4G and 5G access.
>
>
>
>
>
>
>
>
>
>
>
> Please note that it may take a couple of minutes from the time of submiss=
ion until the htmlized version and diff are available at tools.ietf.org.
>
>
>
> The IETF Secretariat
>
>
>
>
>
>
> --
>
> ________________________________________________________________
>
> Bob Briscoe                               http://bobbriscoe.net/
>
>
> --
> ________________________________________________________________
> Bob Briscoe                               http://bobbriscoe.net/
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>
>

--001a113ee6a4fefe3f0523c42c11
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<p dir=3D"ltr">I have a question re: how much buffer or backlog is needed t=
o prevent the schedular from reducing the rate or even increase the rate. I=
 think this key why cubic or bufferbloat cc wins bc other queue controlling=
 cc would lose out.</p>
<p dir=3D"ltr">For example if the scheduler needs 1mb backlog to maintain a=
t 40mbps. A dctcp or vegas keeping the queue well below will get say 5mbps.=
 </p>
<p dir=3D"ltr">The draft mentions about this but the advise is a bit vague =
for cc designer.</p>
<div class=3D"gmail_quote">On Nov 5, 2015 7:27 AM, &quot;Bob Briscoe&quot; =
&lt;<a href=3D"mailto:ietf@bobbriscoe.net">ietf@bobbriscoe.net</a>&gt; wrot=
e:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF">
    Ingemar,<br>
    <br>
    <div>On 04/11/15 13:27, Ingemar Johansson S
      wrote:<br>
    </div>
    <blockquote type=3D"cite">
     =20
     =20
     =20
      <div>
        <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d" lang=3D"EN-US">Hi=
<u></u><u></u></span></p>
        <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d" lang=3D"EN-US"><u=
></u>=C2=A0<u></u></span></p>
        <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d" lang=3D"EN-US">An=
d thanks for the review,
            comments/answers/questions inline below<u></u><u></u></span></p=
>
        <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d" lang=3D"EN-US"><u=
></u>=C2=A0<u></u></span></p>
        <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d" lang=3D"EN-US">/I=
ngemar<u></u><u></u></span></p>
        <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d" lang=3D"EN-US"><u=
></u>=C2=A0<u></u></span></p>
        <div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm =
0cm 0cm 4.0pt">
          <div>
            <div style=3D"border:none;border-top:solid #b5c4df 1.0pt;paddin=
g:3.0pt 0cm 0cm 0cm">
              <p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;fon=
t-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext" lang=
=3D"EN-US">From:</span></b><span style=3D"font-size:10.0pt;font-family:&quo=
t;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext" lang=3D"EN-US"> Bob=
 Briscoe [<a href=3D"mailto:ietf@bobbriscoe.net" target=3D"_blank">mailto:i=
etf@bobbriscoe.net</a>]
                  <br>
                  <b>Sent:</b> den 29 oktober 2015 18:54<br>
                  <b>To:</b> Ingemar Johansson S; <a href=3D"mailto:iccrg@i=
rtf.org" target=3D"_blank">iccrg@irtf.org</a>;
                  <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@i=
etf.org</a>; <a href=3D"mailto:tsvwg@ietf.org" target=3D"_blank">tsvwg@ietf=
.org</a><br>
                  <b>Subject:</b> Re: [tsvwg] FW: New Version
                  Notification for draft-johansson-cc-for-4g-5g-00.txt<u></=
u><u></u></span></p>
            </div>
          </div>
          <p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Ingemar,<br=
>
            <br>
            As promised, here&#39;s my comments. They are a mix of suggeste=
d
            improvements to the draft, suggested improvements to 5G, or
            requests for clarification. I also agree with all Kevin
            Smith&#39;s comments, so I didn&#39;t repeat any.<br>
            [Sorry for delay - I had to find time to re-type after
            losing my review during a power outage]<br>
            <br>
            <b><u>Context questions.</u><br>
            </b>Q1. Is this exercise treating the 5G specs solely in
            read-only mode? Or, if some useful design suggestions arise,
            is there any chance that the 3GPP might take them on board,
            given 5G is not fully baked yet?<span style=3D"color:#1f497d"><=
u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] Guess it depends, need to say that I
              probably the small cog in the wheel. I would say that a
              lot of the standards work in IETF has been picked up by
              3GPP and, I cannot say what will happen in the 5G context.
            </span><span lang=3D"EN-US"><br>
              <br>
              It&#39;s great that you have taken this initiative. </span>Do=
es
            the 3GPP ever formally ask the IETF for comments on its
            ideas for each new generation of radio? Given the layers
            above can be thought of as the &#39;customers&#39; of the 3GPP,=
 and
            these &#39;customer&#39; layers are pretty much all maintained =
by
            the IETF, it would seem essential (and courteous) for the
            3GPP to ask for customer comment, not least to co-ordinate
            with whatever future plans the IETF might have for the
            higher layers.<span style=3D"color:#1f497d"><u></u><u></u></spa=
n></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] What you say makes much sense. I guess
              there is some natural leakage between 3GPP and IETF as
              many of the IETFers are also active in 3GPP
              standardization. I don=E2=80=99t have full insight in the for=
mal
              relation between 3GPP and IETF in these areas, not sure if
              that need to be better, but if that is the case, then it
              is probably a question for the area directors. Need to say
              that even though the formal channels are scarce it does
              probably not mean that 3GPP ignores IETF or what happens
              above IP in general. For instance the recent work around
              TCP RACK raises the question if it is necessary that LTE
              default radio bearers ensure in order delivery. </span></p>
        </div>
      </div>
    </blockquote>
    I suspect that 3GPP radio resource control people do not
    cross-fertilize much with the IETF.<br>
    So maybe it is as much a question of how to co-ordinate between
    these radio people and L4, which I would imagine is as much a
    problem within 3GPP as between 3GPP and IETF.<br>
    <blockquote type=3D"cite">
      <div>
        <div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm =
0cm 0cm 4.0pt">
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              Q2. What are your plans for this draft? </span>Are you
            asking for adoption? And if so, in which WG or RG? ICCRG
            looks the most appropriate, given the charters of all the
            IETF cc WGs (TSVWG, TCPM, RMCAT, etc) are scoped on just a
            subset of your doc.<span style=3D"color:#1f497d"><u></u><u></u>=
</span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] First objective is to get an increased
              interest in congestion control algorithm work in the IETF.
              Traditionally this kind of work seem to be more in the
              terms of conference papers in e.g. Sigcomm, the benefit
              with having such work in IETF is that input from people
              with a wider expertise area (including 3GPP experitise)
              can give algorithm proposals that work well for a wider
              set of access types. The work around Cubic is a first good
              step as is the work around DCTCP. What I wonder is if it
              feasible to take this a bit further ? I would too believe
              that the current doc fits better in ICCRG.=C2=A0 The algorith=
m
              examples are just.. examples. More fruitful work is
              probably to document different access types and from there
              derive a set of best current practices for congestion
              control algorithms</span></p>
        </div>
      </div>
    </blockquote>
    I&#39;m sure the iccrg chairs would like this to happen. And initiative=
s
    like yours to come with proposals to propose specific code to take
    account of L2 I think generate the right level of dialogue.<br>
    <blockquote type=3D"cite">
      <div>
        <div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm =
0cm 0cm 4.0pt">
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              <b><u>Section by Section Review</u><br>
              </b><br>
              <b>All sections<br>
              </b><br>
              The document generally talks about the LTE protocol stack
              as if data is travelling down it.
            </span>Are you implying that most sources of buffering delay
            are at the sending end, and the feedback end is pretty
            unencumbered by delays? I think it would be useful to
            explicitly say something about this (whether true or not),
            to explain why only one direction of travel is discussed in
            the doc.<span style=3D"color:#1f497d"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] The feedback path is encumbered by
              delays, it is perhaps not that clear but section 2.3.2
              implies that delay (variation) in uplink will affect the
              ACKs for downlink data traffic. That said, the difference
              in DL/UL traffic is today something like 10:1 and that is
              probably one reason why one can get the feeling that the
              draft is slanted towards downlink data traffic, this may
              however change with time as more machine type traffic may
              enter the system.</span></p>
        </div>
      </div>
    </blockquote>
    I was thinking more in terms of up the stack vs. down the stack, not
    uplink/downlink. I.e. for both cases:<br>
    * downlink direction: are there any buffering concerns coming up the
    stack at the UE, or on the feedback back-channel from the UE?<br>
    * uplink direction: are there any buffering concerns coming up the
    stack at the eNodeB, or on the feedback back-channel from the
    eNodeB?<br>
    <br>
    For instance, in the UE, perhaps one app blocking the read of
    another due to the way {3-5}G multiplexes different streams into the
    same PDCP frames?<br>
    <blockquote type=3D"cite">
      <div>
        <div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm =
0cm 0cm 4.0pt">
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              <b>2.1 PDCP layer<br>
              </b><br>
              =C2=A0=C2=A0=C2=A0 </span><tt><span style=3D"font-size:10.0pt=
" lang=3D"EN-US">PDCP also ensures that all</span></tt><span lang=3D"EN-US"=
><br>
              <tt>=C2=A0=C2=A0 packets are delivered in order up to higher =
layers.</tt><br>
            </span>I think this means &quot;in the order sent into the link
            (PDCP) layer&quot;. This ought to be clarified, otherwise, when
            writing in the context of a layer 4 audience, it might be
            assumed this means in the order sent by the original
            transport layer source.
            <span style=3D"color:#1f497d"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] Yes, you are right , reordering can
              still occur e.g. in the wireless backhaul<u></u><u></u></span=
></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
            </span><tt><span style=3D"font-size:10.0pt" lang=3D"EN-US">=C2=
=A0=C2=A0
                keep the amount of data in flight as small as possible,
                without sacrificing throughput.</span></tt><span lang=3D"EN=
-US"><br>
            </span>Is this just a rather convoluted way of saying &quot;kee=
p
            the RTT as low as possible&quot;? (which in turn implies keepin=
g
            queuing delay as low as possible assuming you cannot
            influence the physical path.) Or were you trying to make a
            more subtle point? - if so, it was lost on me.<span style=3D"co=
lor:#1f497d"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] No, it is exactly as you describe below=
<u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
            </span><tt><span style=3D"font-size:10.0pt" lang=3D"EN-US">=C2=
=A0=C2=A0
                Reliable delivery at handover may however be turned off
                or is simply not implemented,</span></tt><span lang=3D"EN-U=
S"><br>
            </span><span lang=3D"EN-US">It would be useful if you could
              give a feel for roughly how often (in %) these cases hold.</s=
pan><span style=3D"color:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] I don=E2=80=99t believe that it is poss=
ible to
              give a figure as this is vendor specific.<u></u><u></u></span=
></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              <b>2.2 RLC layer</b><br>
              <br>
            </span><tt><span style=3D"font-size:10.0pt" lang=3D"EN-US">=C2=
=A0=C2=A0
                role ... is to ensure that packets are delivered in
                order up to the higher layers</span></tt><span lang=3D"EN-U=
S"><br>
            </span><span lang=3D"EN-US">Eh? </span>The PDCP section just
            said it did ordering. And the rest of the RLC section talks
            exclusively about buffering delays necessary while filling
            transport blocks. If ordering is the role of both layers,
            surely one will have nothing to do once the other has got
            everything in order?<span style=3D"color:#1f497d"><u></u><u></u=
></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] RLC handles ordering due to various MAC
              layer failures. PDCP handles ordering =C2=A0when handover
              occurs<u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
            </span><tt><span style=3D"font-size:10.0pt" lang=3D"EN-US">=C2=
=A0=C2=A0
                varying size of the available transport</span></tt><span la=
ng=3D"EN-US"><br>
            </span><span lang=3D"EN-US">Nit: Given &#39;transport&#39; mean=
s e2e
              for the audience of this draft, pls use bearer or somehow
              disambiguate this word.</span><span style=3D"color:#1f497d" l=
ang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] OK<u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              <br>
              Gap#1: In the draft, it seems like transport blocks arrive
              by magic, to be filled by data buffered in the RLC layer.
            </span>To understand the delay compromises here, I think the
            draft needs to explain what determines how much transport
            block capacity each flow is given, and how they can
            influence it. In 3G, I believe that the transport layer
            could ask the radio resource controller to change radio
            resource allocation. Is this the same in 4G &amp; 5G? Does
            the backlog of the buffer into the RLC layer have any
            automatic influence over allocation of available resources?
            I assume traffic class can also be used to influence
            allocation.<span style=3D"color:#1f497d"><u></u><u></u></span><=
/p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] Will updated with more detail<u></u><u>=
</u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              Gap#2: In a 3G stack, the RLC layer multiplexes all the
              active application streams into MAC layer transport block
              by dicing up buffered packets and packing the pieces into
              each transport block.
            </span>Then, at the other end of the link, each piece has to
            be held until the rest of the packet arrives. This
            sacrifices delay for bandwidth efficiency. Is this still
            done in 4G and 5G? If so, it would be good to write about
            mux delay in the draft too.<span style=3D"color:#1f497d"><u></u=
><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] OK, will add more info<u></u><u></u></s=
pan></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              <b>2.3 MAC layer<br>
              </b><br>
            </span><tt><span style=3D"font-size:10.0pt" lang=3D"EN-US">=C2=
=A0=C2=A0
                the maximum number of retransmissions is configurable</span=
></tt><span lang=3D"EN-US"><br>
            </span><span lang=3D"EN-US"><br>
              Wouldn&#39;t it be better to set a limit on the <i>time</i>
              spent retransmitting, so it would automatically give up
              with less retransmissions for longer transmission
              distances?<br>
            </span><span style=3D"color:#1f497d" lang=3D"EN-US">[IJ]
              Possible, 4G specifies a limit to the number of
              retransmissions, in practice it can be translated to time
              as each retransmission is 8ms later.<u></u><u></u></span></p>
        </div>
      </div>
    </blockquote>
    I didn&#39;t realise it was independent of link-RTT. Then the more basi=
c
    question is why is it independent of link-RTT? Surely on a short
    link it can assume that it needs to retransmit more quickly than on
    a long link?<br>
    <br>
    You can see that my question was assuming that retransmission
    happened more often on a short link, then I was saying it should
    still be allowed the same amount of time to keep retrying (so on a
    short link it would be allowed to retransmit more times before
    giving up, but the max delay would be the same).<br>
    <br>
    8ms! I was assuming it would be rather much less than that. Surely a
    link-layer ACK could never take that long if the frame was going to
    get through.<br>
    <blockquote type=3D"cite">
      <div>
        <div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm =
0cm 0cm 4.0pt">
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              s/only checks for the presence on download data only at
              regular intervals/<br>
              =C2=A0/only checks for the presence of download data at regul=
ar
              intervals/<br>
            </span><span style=3D"color:#1f497d" lang=3D"EN-US">[OK]<u></u>=
<u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <b>2.3.2 Uplink scheduling<br>
              </b><br>
            </span><tt><span style=3D"font-size:10.0pt" lang=3D"EN-US">=C2=
=A0=C2=A0
                The scheduling request does not indicate how many bytes
                there are in the uplink queue.</span></tt><span lang=3D"EN-=
US"><br>
            </span>Seems like a bad omission. Is there a reason? If
            there are less bytes queued than the base station&#39;s grant,
            surely it would be useful for the base station to know, so
            that it can grant more to something else while the previous
            short transmission is completing.<br>
            <span style=3D"color:#1f497d" lang=3D"EN-US">[IJ] It is a desig=
n
              choice. A buffer status report is more bulky, a scheduling
              request is just one bit.<u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              You say that when HARQ breaks up a sequence of ACKs, it
              can also trigger &quot;coalescing issues&quot;.
            </span>I can understand that it could cause ACK compression,
            but are you implying something coalesces the ACKs or the
            subsequent data? What does the coalescing?
            <span lang=3D"EN-US">Similarly, in the last para of 2.3.2 you
              say &quot;Reduced ACKs can unfortunately cause coalescing.&qu=
ot; and
              again I&#39;m not sure what is coalescing what here.</span><s=
pan style=3D"color:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] What I can see in simulations is that
              the ACK compression leads to coalescing which further can
              increase jitter.</span></p>
        </div>
      </div>
    </blockquote>
    I still don&#39;t understand. The question was &quot;What is coalescing
    what?&quot;<br>
    <blockquote type=3D"cite">
      <div>
        <div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm =
0cm 0cm 4.0pt">
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              There could be a positive side to the interaction between
              ACK clocking and link aggregation, at least for
              long-running flows.
            </span>This has been observed in 802.11n WiFi where the
            buffering and aggregation of multiple ACKs into one
            transmission grant tend to lead the sender to bunch multiple
            data segments so that they all nicely fill one transmission
            grant in the other direction [Showail14]. <span lang=3D"EN-US">=
As long as the transmission grant does not
              wait for more data (which unfortunately I don&#39;t think is
              the case in 3G/4G/5G), this could be good for bandwidth
              utilisation of elephants and latency of mice, including
              when they are mixed.<br>
              <br>
              [Showail14] Showail, A., Jamshaid, K. &amp; Shihada, B.,
              &quot;WQM: An Aggregation-aware Queue Management Scheme for
              IEEE 802.11N Based Networks,&quot; In: Proceedings of the 201=
4
              ACM SIGCOMM Workshop on Capacity Sharing Workshop CSWS &#39;1=
4
              pp.15-20 ACM (2014)<br>
              &lt;</span><a href=3D"http://dl.acm.org/citation.cfm?id=3D263=
0097" target=3D"_blank"><span lang=3D"EN-US">http://dl.acm.org/citation.cfm=
?id=3D2630097</span></a><span lang=3D"EN-US">&gt;<br>
            </span><span style=3D"color:#1f497d" lang=3D"EN-US">[IJ]
              Interesting ,will look at it, not sure though that it will
              work that well with 4G (cant say anything about 5G), in 4G
              the transmission grants can vary quite a lot over short
              time spans.<u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
            </span><b>3.=C2=A0 4G and 5G evolution<br>
            </b><br>
            s/a explicit/an explicit/<br>
            <br>
            <b>4.=C2=A0 Requirements for improved performance<br>
            </b><br>
            s/bufferbloat/buffer/<br>
            <br>
            <b>5.=C2=A0 Congestion control examples<br>
            </b><br>
            Regarding Hybrid Cubic using OWD measurements, I have seen a
            paper (under submission) about getting delay-based
            congestion controls (e.g. LEDBAT or CAIA Delay Gradient) to
            interwork with AQMs with various low delay targets. It&#39;s no=
t
            in the context of cellular networks, but I&#39;ll try to
            remember to point it out once it gets published.<span style=3D"=
color:#1f497d"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] Sounds interesting , looking forward to
              see the paper.<u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              <b>5.1 HyStartRestart<br>
              </b><br>
              s/algorithm used/algorithm is used/<br>
              <br>
              In the new code, it would be useful to explain why the
              second test before doubling ssthresh, ie, the test for how
              long it has been since the last hyrestart.
            </span>Also, in the header definitions, if you intend there
            to be constraints on the relation between N_RTT_HYRESTART
            and N_RTT_LOW (e.g.=C2=A0 N_RTT_HYRESTART &gt; N_RTT_LOW) you
            ought to say, because people will try other values.<br>
            <br>
            Rather than the rather arbitrary wait before an arbitrary
            doubling of ssthresh, wouldn&#39;t it be better to continually
            increase ssthresh (not necessarily linearly) the longer the
            RTT has remained only just higher than the min RTT?<span style=
=3D"color:#1f497d"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] Yes, could be an option<u></u><u></u></=
span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              (BTW, surely the main problem with this modification to
              HyStartRestart is that is requires a number of round trips
              to reduce delay - a sort-of oxymoron.)</span><span style=3D"c=
olor:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] Yes, have tried some more experiments
              with this algorithm. And I am getting more hesitant. The
              problem is that when I add more noise (modeling of small
              objects traffic) into the system simulation, then I also
              get more delay jitter and the delay detection algorithm in
              HyStart is notoriously sensitive to delay jitter.<u></u><u></=
u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
            </span><b>5.2.=C2=A0 Hybrid Cubic</b><span style=3D"color:#1f49=
7d" lang=3D"EN-US"><u></u><u></u></span></p>
          <pre>=C2=A0=C2=A0 Furthermore it is assumed than the timestamp cl=
ock frequency in<u></u><u></u></pre>
          <pre>=C2=A0=C2=A0 sender and receiver are identical or that the s=
ender can infer the<u></u><u></u></pre>
          <pre>=C2=A0=C2=A0 timestamp clock frequency of the receiver and r=
ecompute timestamp<u></u><u></u></pre>
          <pre>=C2=A0=C2=A0 values based on this information.<u></u><u></u>=
</pre>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Recently on
            tcpm I sent you a pointer to the chirping paper Mirja &amp;
            I did, which also made this assumption. Partly prompted by
            that, Richard Scheffenegger tried to get people interested
            in standardising use of the timestamp option on a SYN to
            negotiate stuff like timer resolution. I don&#39;t think it wen=
t
            anywhere. Do you think we ought to revitalise that? You
            mention inference - have you tried that - are there cases
            where it doesn&#39;t work?<span style=3D"color:#1f497d"><u></u>=
<u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] Yes, actually wondered what happened to
              Richards draft. I only briefly tried out the inference
              algorithm with a TCP LEDBAT implementation (</span><span styl=
e=3D"font-size:9.0pt;font-family:Consolas;color:#969896;background:white" l=
ang=3D"EN-US"><a href=3D"https://github.com/silviov/TCP-LEDBAT/blob/master/=
src/tcp_ledbat.c" target=3D"_blank">https://github.com/silviov/TCP-LEDBAT/b=
lob/master/src/tcp_ledbat.c</a></span><span style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d" lang=3D"EN=
-US">). I recall that it took a while to converge.
              Also I am not sure here. Is Hz in e.g a linux stack fixed.
              ?</span></p>
        </div>
      </div>
    </blockquote>
    I believe so. But I don&#39;t know about embedded systems etc.<br>
    <br>
    Cheers<br>
    <br>
    <br>
    <br>
    Bob<br>
    <blockquote type=3D"cite">
      <div>
        <div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm =
0cm 0cm 4.0pt">
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              <br>
              That&#39;s it. Thanks again for the draft. V useful.<br>
              <br>
              <br>
              <br>
              Bob<br>
              <br>
              <br>
              <u></u><u></u></span></p>
          <div>
            <p class=3D"MsoNormal"><span lang=3D"EN-US">On 15/10/15 06:08,
                Ingemar Johansson S </span>
              wrote:<u></u><u></u></p>
          </div>
          <blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
            <pre>Hi<u></u><u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>A new IETF draft is submitted in response to general discu=
ssion that for instance TCP Cubic is not an appropriate congestion control =
algorithm for LTE. <u></u><u></u></pre>
            <pre>The draft briefly outlines typical 4G access behavior on t=
he PDCP, RLC and MAC=C2=A0 layers that can have an impact on transport prot=
ocol performance. The draft also outlines two relatively simple modificatio=
ns to the Cubic congestion control.<u></u><u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>/Ingemar=C2=A0 <u></u><u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>-----Original Message-----<u></u><u></u></pre>
            <pre>From: <a href=3D"mailto:internet-drafts@ietf.org" target=
=3D"_blank">internet-drafts@ietf.org</a> [<a href=3D"mailto:internet-drafts=
@ietf.org" target=3D"_blank">mailto:internet-drafts@ietf.org</a>] <u></u><u=
></u></pre>
            <pre>Sent: den 14 oktober 2015 20:17<u></u><u></u></pre>
            <pre>To: Ingemar Johansson S<u></u><u></u></pre>
            <pre>Subject: New Version Notification for draft-johansson-cc-f=
or-4g-5g-00.txt<u></u><u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>A new version of I-D, draft-johansson-cc-for-4g-5g-00.txt<=
u></u><u></u></pre>
            <pre>has been successfully submitted by Ingemar Johansson and p=
osted to the IETF repository.<u></u><u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>Name:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0 draft-johansson-cc-for-4g-5g<u></u><u></u></pre>
            <pre>Revision:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 00<u></u><u>=
</u></pre>
            <pre>Title:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0 Congestion control for 4G and 5G access<u></u><u></u></pre>
            <pre>Document date:=C2=A0 2015-10-14<u></u><u></u></pre>
            <pre>Group:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0 Individual Submission<u></u><u></u></pre>
            <pre>Pages:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0 14<u></u><u></u></pre>
            <pre>URL:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0 <a href=3D"https://www.ietf.org/internet-drafts/draft-johansso=
n-cc-for-4g-5g-00.txt" target=3D"_blank">https://www.ietf.org/internet-draf=
ts/draft-johansson-cc-for-4g-5g-00.txt</a><u></u><u></u></pre>
            <pre>Status:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a=
 href=3D"https://datatracker.ietf.org/doc/draft-johansson-cc-for-4g-5g/" ta=
rget=3D"_blank">https://datatracker.ietf.org/doc/draft-johansson-cc-for-4g-=
5g/</a><u></u><u></u></pre>
            <pre>Htmlized:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a href=3D"h=
ttps://tools.ietf.org/html/draft-johansson-cc-for-4g-5g-00" target=3D"_blan=
k">https://tools.ietf.org/html/draft-johansson-cc-for-4g-5g-00</a><u></u><u=
></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>Abstract:<u></u><u></u></pre>
            <pre>=C2=A0=C2=A0 This memo outlines the challenge that 4G and =
5G access brings for<u></u><u></u></pre>
            <pre>=C2=A0=C2=A0 transport protocol congestion control and als=
o outlines a few simple<u></u><u></u></pre>
            <pre>=C2=A0=C2=A0 examples that can improve transport protocol =
congestion control<u></u><u></u></pre>
            <pre>=C2=A0=C2=A0 performance in 4G and 5G access.<u></u><u></u=
></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <u></u><u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>Please note that it may take a couple of minutes from the =
time of submission until the htmlized version and diff are available at <a =
href=3D"http://tools.ietf.org" target=3D"_blank">tools.ietf.org</a>.<u></u>=
<u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>The IETF Secretariat<u></u><u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
          </blockquote>
          <p class=3D"MsoNormal"><br>
            <br>
            <br>
            <u></u><u></u></p>
          <pre>-- <u></u><u></u></pre>
          <pre>____________________________________________________________=
____<u></u><u></u></pre>
          <pre>Bob Briscoe=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a href=3D"http:/=
/bobbriscoe.net/" target=3D"_blank">http://bobbriscoe.net/</a><u></u><u></u=
></pre>
        </div>
      </div>
    </blockquote>
    <br>
    <pre cols=3D"72">--=20
________________________________________________________________
Bob Briscoe                               <a href=3D"http://bobbriscoe.net/=
" target=3D"_blank">http://bobbriscoe.net/</a></pre>
  </div>

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Date: Thu, 5 Nov 2015 14:26:40 +0900
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Cc: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>, "tsvwg@ietf.org" <tsvwg@ietf.org>, "iccrg@irtf.org" <iccrg@irtf.org>
Subject: Re: [tcpm] [tsvwg] FW: New Version Notification for draft-johansson-cc-for-4g-5g-00.txt
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I also support this doc to be adopted by iccrg or tsvwg
On Nov 5, 2015 7:27 AM, "Bob Briscoe" <ietf@bobbriscoe.net> wrote:

> Ingemar,
>
> On 04/11/15 13:27, Ingemar Johansson S wrote:
>
> Hi
>
>
>
> And thanks for the review, comments/answers/questions inline below
>
>
>
> /Ingemar
>
>
>
> *From:* Bob Briscoe [mailto:ietf@bobbriscoe.net <ietf@bobbriscoe.net>]
> *Sent:* den 29 oktober 2015 18:54
> *To:* Ingemar Johansson S; iccrg@irtf.org; tcpm@ietf.org; tsvwg@ietf.org
> *Subject:* Re: [tsvwg] FW: New Version Notification for
> draft-johansson-cc-for-4g-5g-00.txt
>
>
>
> Ingemar,
>
> As promised, here's my comments. They are a mix of suggested improvements
> to the draft, suggested improvements to 5G, or requests for clarification=
.
> I also agree with all Kevin Smith's comments, so I didn't repeat any.
> [Sorry for delay - I had to find time to re-type after losing my review
> during a power outage]
>
>
> *Context questions. *Q1. Is this exercise treating the 5G specs solely in
> read-only mode? Or, if some useful design suggestions arise, is there any
> chance that the 3GPP might take them on board, given 5G is not fully bake=
d
> yet?
>
> [IJ] Guess it depends, need to say that I probably the small cog in the
> wheel. I would say that a lot of the standards work in IETF has been pick=
ed
> up by 3GPP and, I cannot say what will happen in the 5G context.
>
> It's great that you have taken this initiative. Does the 3GPP ever
> formally ask the IETF for comments on its ideas for each new generation o=
f
> radio? Given the layers above can be thought of as the 'customers' of the
> 3GPP, and these 'customer' layers are pretty much all maintained by the
> IETF, it would seem essential (and courteous) for the 3GPP to ask for
> customer comment, not least to co-ordinate with whatever future plans the
> IETF might have for the higher layers.
>
> [IJ] What you say makes much sense. I guess there is some natural leakage
> between 3GPP and IETF as many of the IETFers are also active in 3GPP
> standardization. I don=E2=80=99t have full insight in the formal relation=
 between
> 3GPP and IETF in these areas, not sure if that need to be better, but if
> that is the case, then it is probably a question for the area directors.
> Need to say that even though the formal channels are scarce it does
> probably not mean that 3GPP ignores IETF or what happens above IP in
> general. For instance the recent work around TCP RACK raises the question
> if it is necessary that LTE default radio bearers ensure in order deliver=
y.
>
> I suspect that 3GPP radio resource control people do not cross-fertilize
> much with the IETF.
> So maybe it is as much a question of how to co-ordinate between these
> radio people and L4, which I would imagine is as much a problem within 3G=
PP
> as between 3GPP and IETF.
>
>
>
> Q2. What are your plans for this draft? Are you asking for adoption? And
> if so, in which WG or RG? ICCRG looks the most appropriate, given the
> charters of all the IETF cc WGs (TSVWG, TCPM, RMCAT, etc) are scoped on
> just a subset of your doc.
>
> [IJ] First objective is to get an increased interest in congestion contro=
l
> algorithm work in the IETF. Traditionally this kind of work seem to be mo=
re
> in the terms of conference papers in e.g. Sigcomm, the benefit with havin=
g
> such work in IETF is that input from people with a wider expertise area
> (including 3GPP experitise) can give algorithm proposals that work well f=
or
> a wider set of access types. The work around Cubic is a first good step a=
s
> is the work around DCTCP. What I wonder is if it feasible to take this a
> bit further ? I would too believe that the current doc fits better in
> ICCRG.  The algorithm examples are just.. examples. More fruitful work is
> probably to document different access types and from there derive a set o=
f
> best current practices for congestion control algorithms
>
> I'm sure the iccrg chairs would like this to happen. And initiatives like
> yours to come with proposals to propose specific code to take account of =
L2
> I think generate the right level of dialogue.
>
>
>
>
> *Section by Section Review *
>
> *All sections *
> The document generally talks about the LTE protocol stack as if data is
> travelling down it. Are you implying that most sources of buffering delay
> are at the sending end, and the feedback end is pretty unencumbered by
> delays? I think it would be useful to explicitly say something about this
> (whether true or not), to explain why only one direction of travel is
> discussed in the doc.
>
> [IJ] The feedback path is encumbered by delays, it is perhaps not that
> clear but section 2.3.2 implies that delay (variation) in uplink will
> affect the ACKs for downlink data traffic. That said, the difference in
> DL/UL traffic is today something like 10:1 and that is probably one reaso=
n
> why one can get the feeling that the draft is slanted towards downlink da=
ta
> traffic, this may however change with time as more machine type traffic m=
ay
> enter the system.
>
> I was thinking more in terms of up the stack vs. down the stack, not
> uplink/downlink. I.e. for both cases:
> * downlink direction: are there any buffering concerns coming up the stac=
k
> at the UE, or on the feedback back-channel from the UE?
> * uplink direction: are there any buffering concerns coming up the stack
> at the eNodeB, or on the feedback back-channel from the eNodeB?
>
> For instance, in the UE, perhaps one app blocking the read of another due
> to the way {3-5}G multiplexes different streams into the same PDCP frames=
?
>
>
>
>
> *2.1 PDCP layer *
>     PDCP also ensures that all
>    packets are delivered in order up to higher layers.
> I think this means "in the order sent into the link (PDCP) layer". This
> ought to be clarified, otherwise, when writing in the context of a layer =
4
> audience, it might be assumed this means in the order sent by the origina=
l
> transport layer source.
>
> [IJ] Yes, you are right , reordering can still occur e.g. in the wireless
> backhaul
>
>
>
>    keep the amount of data in flight as small as possible, without
> sacrificing throughput.
> Is this just a rather convoluted way of saying "keep the RTT as low as
> possible"? (which in turn implies keeping queuing delay as low as possibl=
e
> assuming you cannot influence the physical path.) Or were you trying to
> make a more subtle point? - if so, it was lost on me.
>
> [IJ] No, it is exactly as you describe below
>
>
>
>    Reliable delivery at handover may however be turned off or is simply
> not implemented,
> It would be useful if you could give a feel for roughly how often (in %)
> these cases hold.
>
> [IJ] I don=E2=80=99t believe that it is possible to give a figure as this=
 is
> vendor specific.
>
>
>
> *2.2 RLC layer*
>
>    role ... is to ensure that packets are delivered in order up to the
> higher layers
> Eh? The PDCP section just said it did ordering. And the rest of the RLC
> section talks exclusively about buffering delays necessary while filling
> transport blocks. If ordering is the role of both layers, surely one will
> have nothing to do once the other has got everything in order?
>
> [IJ] RLC handles ordering due to various MAC layer failures. PDCP handles
> ordering  when handover occurs
>
>
>
>    varying size of the available transport
> Nit: Given 'transport' means e2e for the audience of this draft, pls use
> bearer or somehow disambiguate this word.
>
> [IJ] OK
>
>
>
>
> Gap#1: In the draft, it seems like transport blocks arrive by magic, to b=
e
> filled by data buffered in the RLC layer. To understand the delay
> compromises here, I think the draft needs to explain what determines how
> much transport block capacity each flow is given, and how they can
> influence it. In 3G, I believe that the transport layer could ask the rad=
io
> resource controller to change radio resource allocation. Is this the same
> in 4G & 5G? Does the backlog of the buffer into the RLC layer have any
> automatic influence over allocation of available resources? I assume
> traffic class can also be used to influence allocation.
>
> [IJ] Will updated with more detail
>
>
>
> Gap#2: In a 3G stack, the RLC layer multiplexes all the active applicatio=
n
> streams into MAC layer transport block by dicing up buffered packets and
> packing the pieces into each transport block. Then, at the other end of
> the link, each piece has to be held until the rest of the packet arrives.
> This sacrifices delay for bandwidth efficiency. Is this still done in 4G
> and 5G? If so, it would be good to write about mux delay in the draft too=
.
>
> [IJ] OK, will add more info
>
>
>
>
> *2.3 MAC layer *
>    the maximum number of retransmissions is configurable
>
> Wouldn't it be better to set a limit on the *time* spent retransmitting,
> so it would automatically give up with less retransmissions for longer
> transmission distances?
> [IJ] Possible, 4G specifies a limit to the number of retransmissions, in
> practice it can be translated to time as each retransmission is 8ms later=
.
>
> I didn't realise it was independent of link-RTT. Then the more basic
> question is why is it independent of link-RTT? Surely on a short link it
> can assume that it needs to retransmit more quickly than on a long link?
>
> You can see that my question was assuming that retransmission happened
> more often on a short link, then I was saying it should still be allowed
> the same amount of time to keep retrying (so on a short link it would be
> allowed to retransmit more times before giving up, but the max delay woul=
d
> be the same).
>
> 8ms! I was assuming it would be rather much less than that. Surely a
> link-layer ACK could never take that long if the frame was going to get
> through.
>
>
>
>
> s/only checks for the presence on download data only at regular intervals=
/
>  /only checks for the presence of download data at regular intervals/
> [OK]
>
>
>
>
>
> *2.3.2 Uplink scheduling *
>    The scheduling request does not indicate how many bytes there are in
> the uplink queue.
> Seems like a bad omission. Is there a reason? If there are less bytes
> queued than the base station's grant, surely it would be useful for the
> base station to know, so that it can grant more to something else while t=
he
> previous short transmission is completing.
> [IJ] It is a design choice. A buffer status report is more bulky, a
> scheduling request is just one bit.
>
>
>
>
> You say that when HARQ breaks up a sequence of ACKs, it can also trigger
> "coalescing issues". I can understand that it could cause ACK
> compression, but are you implying something coalesces the ACKs or the
> subsequent data? What does the coalescing? Similarly, in the last para of
> 2.3.2 you say "Reduced ACKs can unfortunately cause coalescing." and agai=
n
> I'm not sure what is coalescing what here.
>
> [IJ] What I can see in simulations is that the ACK compression leads to
> coalescing which further can increase jitter.
>
> I still don't understand. The question was "What is coalescing what?"
>
>
> There could be a positive side to the interaction between ACK clocking an=
d
> link aggregation, at least for long-running flows. This has been observed
> in 802.11n WiFi where the buffering and aggregation of multiple ACKs into
> one transmission grant tend to lead the sender to bunch multiple data
> segments so that they all nicely fill one transmission grant in the other
> direction [Showail14]. As long as the transmission grant does not wait
> for more data (which unfortunately I don't think is the case in 3G/4G/5G)=
,
> this could be good for bandwidth utilisation of elephants and latency of
> mice, including when they are mixed.
>
> [Showail14] Showail, A., Jamshaid, K. & Shihada, B., "WQM: An
> Aggregation-aware Queue Management Scheme for IEEE 802.11N Based Networks=
,"
> In: Proceedings of the 2014 ACM SIGCOMM Workshop on Capacity Sharing
> Workshop CSWS '14 pp.15-20 ACM (2014)
> <http://dl.acm.org/citation.cfm?id=3D2630097>
> [IJ] Interesting ,will look at it, not sure though that it will work that
> well with 4G (cant say anything about 5G), in 4G the transmission grants
> can vary quite a lot over short time spans.
>
>
>
>
>
> *3.  4G and 5G evolution *
> s/a explicit/an explicit/
>
>
> *4.  Requirements for improved performance *
> s/bufferbloat/buffer/
>
>
> *5.  Congestion control examples *
> Regarding Hybrid Cubic using OWD measurements, I have seen a paper (under
> submission) about getting delay-based congestion controls (e.g. LEDBAT or
> CAIA Delay Gradient) to interwork with AQMs with various low delay target=
s.
> It's not in the context of cellular networks, but I'll try to remember to
> point it out once it gets published.
>
> [IJ] Sounds interesting , looking forward to see the paper.
>
>
>
>
> *5.1 HyStartRestart *
> s/algorithm used/algorithm is used/
>
> In the new code, it would be useful to explain why the second test before
> doubling ssthresh, ie, the test for how long it has been since the last
> hyrestart. Also, in the header definitions, if you intend there to be
> constraints on the relation between N_RTT_HYRESTART and N_RTT_LOW (e.g.
> N_RTT_HYRESTART > N_RTT_LOW) you ought to say, because people will try
> other values.
>
> Rather than the rather arbitrary wait before an arbitrary doubling of
> ssthresh, wouldn't it be better to continually increase ssthresh (not
> necessarily linearly) the longer the RTT has remained only just higher th=
an
> the min RTT?
>
> [IJ] Yes, could be an option
>
>
>
> (BTW, surely the main problem with this modification to HyStartRestart is
> that is requires a number of round trips to reduce delay - a sort-of
> oxymoron.)
>
> [IJ] Yes, have tried some more experiments with this algorithm. And I am
> getting more hesitant. The problem is that when I add more noise (modelin=
g
> of small objects traffic) into the system simulation, then I also get mor=
e
> delay jitter and the delay detection algorithm in HyStart is notoriously
> sensitive to delay jitter.
>
>
>
> *5.2.  Hybrid Cubic*
>
>    Furthermore it is assumed than the timestamp clock frequency in
>
>    sender and receiver are identical or that the sender can infer the
>
>    timestamp clock frequency of the receiver and recompute timestamp
>
>    values based on this information.
>
> Recently on tcpm I sent you a pointer to the chirping paper Mirja & I did=
,
> which also made this assumption. Partly prompted by that, Richard
> Scheffenegger tried to get people interested in standardising use of the
> timestamp option on a SYN to negotiate stuff like timer resolution. I don=
't
> think it went anywhere. Do you think we ought to revitalise that? You
> mention inference - have you tried that - are there cases where it doesn'=
t
> work?
>
> [IJ] Yes, actually wondered what happened to Richards draft. I only
> briefly tried out the inference algorithm with a TCP LEDBAT implementatio=
n (
> https://github.com/silviov/TCP-LEDBAT/blob/master/src/tcp_ledbat.c). I
> recall that it took a while to converge. Also I am not sure here. Is Hz i=
n
> e.g a linux stack fixed. ?
>
> I believe so. But I don't know about embedded systems etc.
>
> Cheers
>
>
>
> Bob
>
>
>
>
> That's it. Thanks again for the draft. V useful.
>
>
>
> Bob
>
>
> On 15/10/15 06:08, Ingemar Johansson S wrote:
>
> Hi
>
>
>
> A new IETF draft is submitted in response to general discussion that for =
instance TCP Cubic is not an appropriate congestion control algorithm for L=
TE.
>
> The draft briefly outlines typical 4G access behavior on the PDCP, RLC an=
d MAC  layers that can have an impact on transport protocol performance. Th=
e draft also outlines two relatively simple modifications to the Cubic cong=
estion control.
>
>
>
> /Ingemar
>
>
>
> -----Original Message-----
>
> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org <internet=
-drafts@ietf.org>]
>
> Sent: den 14 oktober 2015 20:17
>
> To: Ingemar Johansson S
>
> Subject: New Version Notification for draft-johansson-cc-for-4g-5g-00.txt
>
>
>
>
>
> A new version of I-D, draft-johansson-cc-for-4g-5g-00.txt
>
> has been successfully submitted by Ingemar Johansson and posted to the IE=
TF repository.
>
>
>
> Name:           draft-johansson-cc-for-4g-5g
>
> Revision:       00
>
> Title:          Congestion control for 4G and 5G access
>
> Document date:  2015-10-14
>
> Group:          Individual Submission
>
> Pages:          14
>
> URL:            https://www.ietf.org/internet-drafts/draft-johansson-cc-f=
or-4g-5g-00.txt
>
> Status:         https://datatracker.ietf.org/doc/draft-johansson-cc-for-4=
g-5g/
>
> Htmlized:       https://tools.ietf.org/html/draft-johansson-cc-for-4g-5g-=
00
>
>
>
>
>
> Abstract:
>
>    This memo outlines the challenge that 4G and 5G access brings for
>
>    transport protocol congestion control and also outlines a few simple
>
>    examples that can improve transport protocol congestion control
>
>    performance in 4G and 5G access.
>
>
>
>
>
>
>
>
>
>
>
> Please note that it may take a couple of minutes from the time of submiss=
ion until the htmlized version and diff are available at tools.ietf.org.
>
>
>
> The IETF Secretariat
>
>
>
>
>
>
> --
>
> ________________________________________________________________
>
> Bob Briscoe                               http://bobbriscoe.net/
>
>
> --
> ________________________________________________________________
> Bob Briscoe                               http://bobbriscoe.net/
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>
>

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<p dir=3D"ltr">I also support this doc to be adopted by iccrg or tsvwg</p>
<div class=3D"gmail_quote">On Nov 5, 2015 7:27 AM, &quot;Bob Briscoe&quot; =
&lt;<a href=3D"mailto:ietf@bobbriscoe.net">ietf@bobbriscoe.net</a>&gt; wrot=
e:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF">
    Ingemar,<br>
    <br>
    <div>On 04/11/15 13:27, Ingemar Johansson S
      wrote:<br>
    </div>
    <blockquote type=3D"cite">
     =20
     =20
     =20
      <div>
        <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d" lang=3D"EN-US">Hi=
<u></u><u></u></span></p>
        <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d" lang=3D"EN-US"><u=
></u>=C2=A0<u></u></span></p>
        <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d" lang=3D"EN-US">An=
d thanks for the review,
            comments/answers/questions inline below<u></u><u></u></span></p=
>
        <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d" lang=3D"EN-US"><u=
></u>=C2=A0<u></u></span></p>
        <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d" lang=3D"EN-US">/I=
ngemar<u></u><u></u></span></p>
        <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d" lang=3D"EN-US"><u=
></u>=C2=A0<u></u></span></p>
        <div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm =
0cm 0cm 4.0pt">
          <div>
            <div style=3D"border:none;border-top:solid #b5c4df 1.0pt;paddin=
g:3.0pt 0cm 0cm 0cm">
              <p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;fon=
t-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext" lang=
=3D"EN-US">From:</span></b><span style=3D"font-size:10.0pt;font-family:&quo=
t;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext" lang=3D"EN-US"> Bob=
 Briscoe [<a href=3D"mailto:ietf@bobbriscoe.net" target=3D"_blank">mailto:i=
etf@bobbriscoe.net</a>]
                  <br>
                  <b>Sent:</b> den 29 oktober 2015 18:54<br>
                  <b>To:</b> Ingemar Johansson S; <a href=3D"mailto:iccrg@i=
rtf.org" target=3D"_blank">iccrg@irtf.org</a>;
                  <a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@i=
etf.org</a>; <a href=3D"mailto:tsvwg@ietf.org" target=3D"_blank">tsvwg@ietf=
.org</a><br>
                  <b>Subject:</b> Re: [tsvwg] FW: New Version
                  Notification for draft-johansson-cc-for-4g-5g-00.txt<u></=
u><u></u></span></p>
            </div>
          </div>
          <p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Ingemar,<br=
>
            <br>
            As promised, here&#39;s my comments. They are a mix of suggeste=
d
            improvements to the draft, suggested improvements to 5G, or
            requests for clarification. I also agree with all Kevin
            Smith&#39;s comments, so I didn&#39;t repeat any.<br>
            [Sorry for delay - I had to find time to re-type after
            losing my review during a power outage]<br>
            <br>
            <b><u>Context questions.</u><br>
            </b>Q1. Is this exercise treating the 5G specs solely in
            read-only mode? Or, if some useful design suggestions arise,
            is there any chance that the 3GPP might take them on board,
            given 5G is not fully baked yet?<span style=3D"color:#1f497d"><=
u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] Guess it depends, need to say that I
              probably the small cog in the wheel. I would say that a
              lot of the standards work in IETF has been picked up by
              3GPP and, I cannot say what will happen in the 5G context.
            </span><span lang=3D"EN-US"><br>
              <br>
              It&#39;s great that you have taken this initiative. </span>Do=
es
            the 3GPP ever formally ask the IETF for comments on its
            ideas for each new generation of radio? Given the layers
            above can be thought of as the &#39;customers&#39; of the 3GPP,=
 and
            these &#39;customer&#39; layers are pretty much all maintained =
by
            the IETF, it would seem essential (and courteous) for the
            3GPP to ask for customer comment, not least to co-ordinate
            with whatever future plans the IETF might have for the
            higher layers.<span style=3D"color:#1f497d"><u></u><u></u></spa=
n></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] What you say makes much sense. I guess
              there is some natural leakage between 3GPP and IETF as
              many of the IETFers are also active in 3GPP
              standardization. I don=E2=80=99t have full insight in the for=
mal
              relation between 3GPP and IETF in these areas, not sure if
              that need to be better, but if that is the case, then it
              is probably a question for the area directors. Need to say
              that even though the formal channels are scarce it does
              probably not mean that 3GPP ignores IETF or what happens
              above IP in general. For instance the recent work around
              TCP RACK raises the question if it is necessary that LTE
              default radio bearers ensure in order delivery. </span></p>
        </div>
      </div>
    </blockquote>
    I suspect that 3GPP radio resource control people do not
    cross-fertilize much with the IETF.<br>
    So maybe it is as much a question of how to co-ordinate between
    these radio people and L4, which I would imagine is as much a
    problem within 3GPP as between 3GPP and IETF.<br>
    <blockquote type=3D"cite">
      <div>
        <div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm =
0cm 0cm 4.0pt">
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              Q2. What are your plans for this draft? </span>Are you
            asking for adoption? And if so, in which WG or RG? ICCRG
            looks the most appropriate, given the charters of all the
            IETF cc WGs (TSVWG, TCPM, RMCAT, etc) are scoped on just a
            subset of your doc.<span style=3D"color:#1f497d"><u></u><u></u>=
</span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] First objective is to get an increased
              interest in congestion control algorithm work in the IETF.
              Traditionally this kind of work seem to be more in the
              terms of conference papers in e.g. Sigcomm, the benefit
              with having such work in IETF is that input from people
              with a wider expertise area (including 3GPP experitise)
              can give algorithm proposals that work well for a wider
              set of access types. The work around Cubic is a first good
              step as is the work around DCTCP. What I wonder is if it
              feasible to take this a bit further ? I would too believe
              that the current doc fits better in ICCRG.=C2=A0 The algorith=
m
              examples are just.. examples. More fruitful work is
              probably to document different access types and from there
              derive a set of best current practices for congestion
              control algorithms</span></p>
        </div>
      </div>
    </blockquote>
    I&#39;m sure the iccrg chairs would like this to happen. And initiative=
s
    like yours to come with proposals to propose specific code to take
    account of L2 I think generate the right level of dialogue.<br>
    <blockquote type=3D"cite">
      <div>
        <div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm =
0cm 0cm 4.0pt">
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              <b><u>Section by Section Review</u><br>
              </b><br>
              <b>All sections<br>
              </b><br>
              The document generally talks about the LTE protocol stack
              as if data is travelling down it.
            </span>Are you implying that most sources of buffering delay
            are at the sending end, and the feedback end is pretty
            unencumbered by delays? I think it would be useful to
            explicitly say something about this (whether true or not),
            to explain why only one direction of travel is discussed in
            the doc.<span style=3D"color:#1f497d"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] The feedback path is encumbered by
              delays, it is perhaps not that clear but section 2.3.2
              implies that delay (variation) in uplink will affect the
              ACKs for downlink data traffic. That said, the difference
              in DL/UL traffic is today something like 10:1 and that is
              probably one reason why one can get the feeling that the
              draft is slanted towards downlink data traffic, this may
              however change with time as more machine type traffic may
              enter the system.</span></p>
        </div>
      </div>
    </blockquote>
    I was thinking more in terms of up the stack vs. down the stack, not
    uplink/downlink. I.e. for both cases:<br>
    * downlink direction: are there any buffering concerns coming up the
    stack at the UE, or on the feedback back-channel from the UE?<br>
    * uplink direction: are there any buffering concerns coming up the
    stack at the eNodeB, or on the feedback back-channel from the
    eNodeB?<br>
    <br>
    For instance, in the UE, perhaps one app blocking the read of
    another due to the way {3-5}G multiplexes different streams into the
    same PDCP frames?<br>
    <blockquote type=3D"cite">
      <div>
        <div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm =
0cm 0cm 4.0pt">
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              <b>2.1 PDCP layer<br>
              </b><br>
              =C2=A0=C2=A0=C2=A0 </span><tt><span style=3D"font-size:10.0pt=
" lang=3D"EN-US">PDCP also ensures that all</span></tt><span lang=3D"EN-US"=
><br>
              <tt>=C2=A0=C2=A0 packets are delivered in order up to higher =
layers.</tt><br>
            </span>I think this means &quot;in the order sent into the link
            (PDCP) layer&quot;. This ought to be clarified, otherwise, when
            writing in the context of a layer 4 audience, it might be
            assumed this means in the order sent by the original
            transport layer source.
            <span style=3D"color:#1f497d"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] Yes, you are right , reordering can
              still occur e.g. in the wireless backhaul<u></u><u></u></span=
></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
            </span><tt><span style=3D"font-size:10.0pt" lang=3D"EN-US">=C2=
=A0=C2=A0
                keep the amount of data in flight as small as possible,
                without sacrificing throughput.</span></tt><span lang=3D"EN=
-US"><br>
            </span>Is this just a rather convoluted way of saying &quot;kee=
p
            the RTT as low as possible&quot;? (which in turn implies keepin=
g
            queuing delay as low as possible assuming you cannot
            influence the physical path.) Or were you trying to make a
            more subtle point? - if so, it was lost on me.<span style=3D"co=
lor:#1f497d"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] No, it is exactly as you describe below=
<u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
            </span><tt><span style=3D"font-size:10.0pt" lang=3D"EN-US">=C2=
=A0=C2=A0
                Reliable delivery at handover may however be turned off
                or is simply not implemented,</span></tt><span lang=3D"EN-U=
S"><br>
            </span><span lang=3D"EN-US">It would be useful if you could
              give a feel for roughly how often (in %) these cases hold.</s=
pan><span style=3D"color:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] I don=E2=80=99t believe that it is poss=
ible to
              give a figure as this is vendor specific.<u></u><u></u></span=
></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              <b>2.2 RLC layer</b><br>
              <br>
            </span><tt><span style=3D"font-size:10.0pt" lang=3D"EN-US">=C2=
=A0=C2=A0
                role ... is to ensure that packets are delivered in
                order up to the higher layers</span></tt><span lang=3D"EN-U=
S"><br>
            </span><span lang=3D"EN-US">Eh? </span>The PDCP section just
            said it did ordering. And the rest of the RLC section talks
            exclusively about buffering delays necessary while filling
            transport blocks. If ordering is the role of both layers,
            surely one will have nothing to do once the other has got
            everything in order?<span style=3D"color:#1f497d"><u></u><u></u=
></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] RLC handles ordering due to various MAC
              layer failures. PDCP handles ordering =C2=A0when handover
              occurs<u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
            </span><tt><span style=3D"font-size:10.0pt" lang=3D"EN-US">=C2=
=A0=C2=A0
                varying size of the available transport</span></tt><span la=
ng=3D"EN-US"><br>
            </span><span lang=3D"EN-US">Nit: Given &#39;transport&#39; mean=
s e2e
              for the audience of this draft, pls use bearer or somehow
              disambiguate this word.</span><span style=3D"color:#1f497d" l=
ang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] OK<u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              <br>
              Gap#1: In the draft, it seems like transport blocks arrive
              by magic, to be filled by data buffered in the RLC layer.
            </span>To understand the delay compromises here, I think the
            draft needs to explain what determines how much transport
            block capacity each flow is given, and how they can
            influence it. In 3G, I believe that the transport layer
            could ask the radio resource controller to change radio
            resource allocation. Is this the same in 4G &amp; 5G? Does
            the backlog of the buffer into the RLC layer have any
            automatic influence over allocation of available resources?
            I assume traffic class can also be used to influence
            allocation.<span style=3D"color:#1f497d"><u></u><u></u></span><=
/p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] Will updated with more detail<u></u><u>=
</u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              Gap#2: In a 3G stack, the RLC layer multiplexes all the
              active application streams into MAC layer transport block
              by dicing up buffered packets and packing the pieces into
              each transport block.
            </span>Then, at the other end of the link, each piece has to
            be held until the rest of the packet arrives. This
            sacrifices delay for bandwidth efficiency. Is this still
            done in 4G and 5G? If so, it would be good to write about
            mux delay in the draft too.<span style=3D"color:#1f497d"><u></u=
><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] OK, will add more info<u></u><u></u></s=
pan></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              <b>2.3 MAC layer<br>
              </b><br>
            </span><tt><span style=3D"font-size:10.0pt" lang=3D"EN-US">=C2=
=A0=C2=A0
                the maximum number of retransmissions is configurable</span=
></tt><span lang=3D"EN-US"><br>
            </span><span lang=3D"EN-US"><br>
              Wouldn&#39;t it be better to set a limit on the <i>time</i>
              spent retransmitting, so it would automatically give up
              with less retransmissions for longer transmission
              distances?<br>
            </span><span style=3D"color:#1f497d" lang=3D"EN-US">[IJ]
              Possible, 4G specifies a limit to the number of
              retransmissions, in practice it can be translated to time
              as each retransmission is 8ms later.<u></u><u></u></span></p>
        </div>
      </div>
    </blockquote>
    I didn&#39;t realise it was independent of link-RTT. Then the more basi=
c
    question is why is it independent of link-RTT? Surely on a short
    link it can assume that it needs to retransmit more quickly than on
    a long link?<br>
    <br>
    You can see that my question was assuming that retransmission
    happened more often on a short link, then I was saying it should
    still be allowed the same amount of time to keep retrying (so on a
    short link it would be allowed to retransmit more times before
    giving up, but the max delay would be the same).<br>
    <br>
    8ms! I was assuming it would be rather much less than that. Surely a
    link-layer ACK could never take that long if the frame was going to
    get through.<br>
    <blockquote type=3D"cite">
      <div>
        <div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm =
0cm 0cm 4.0pt">
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              s/only checks for the presence on download data only at
              regular intervals/<br>
              =C2=A0/only checks for the presence of download data at regul=
ar
              intervals/<br>
            </span><span style=3D"color:#1f497d" lang=3D"EN-US">[OK]<u></u>=
<u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <b>2.3.2 Uplink scheduling<br>
              </b><br>
            </span><tt><span style=3D"font-size:10.0pt" lang=3D"EN-US">=C2=
=A0=C2=A0
                The scheduling request does not indicate how many bytes
                there are in the uplink queue.</span></tt><span lang=3D"EN-=
US"><br>
            </span>Seems like a bad omission. Is there a reason? If
            there are less bytes queued than the base station&#39;s grant,
            surely it would be useful for the base station to know, so
            that it can grant more to something else while the previous
            short transmission is completing.<br>
            <span style=3D"color:#1f497d" lang=3D"EN-US">[IJ] It is a desig=
n
              choice. A buffer status report is more bulky, a scheduling
              request is just one bit.<u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              You say that when HARQ breaks up a sequence of ACKs, it
              can also trigger &quot;coalescing issues&quot;.
            </span>I can understand that it could cause ACK compression,
            but are you implying something coalesces the ACKs or the
            subsequent data? What does the coalescing?
            <span lang=3D"EN-US">Similarly, in the last para of 2.3.2 you
              say &quot;Reduced ACKs can unfortunately cause coalescing.&qu=
ot; and
              again I&#39;m not sure what is coalescing what here.</span><s=
pan style=3D"color:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] What I can see in simulations is that
              the ACK compression leads to coalescing which further can
              increase jitter.</span></p>
        </div>
      </div>
    </blockquote>
    I still don&#39;t understand. The question was &quot;What is coalescing
    what?&quot;<br>
    <blockquote type=3D"cite">
      <div>
        <div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm =
0cm 0cm 4.0pt">
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              There could be a positive side to the interaction between
              ACK clocking and link aggregation, at least for
              long-running flows.
            </span>This has been observed in 802.11n WiFi where the
            buffering and aggregation of multiple ACKs into one
            transmission grant tend to lead the sender to bunch multiple
            data segments so that they all nicely fill one transmission
            grant in the other direction [Showail14]. <span lang=3D"EN-US">=
As long as the transmission grant does not
              wait for more data (which unfortunately I don&#39;t think is
              the case in 3G/4G/5G), this could be good for bandwidth
              utilisation of elephants and latency of mice, including
              when they are mixed.<br>
              <br>
              [Showail14] Showail, A., Jamshaid, K. &amp; Shihada, B.,
              &quot;WQM: An Aggregation-aware Queue Management Scheme for
              IEEE 802.11N Based Networks,&quot; In: Proceedings of the 201=
4
              ACM SIGCOMM Workshop on Capacity Sharing Workshop CSWS &#39;1=
4
              pp.15-20 ACM (2014)<br>
              &lt;</span><a href=3D"http://dl.acm.org/citation.cfm?id=3D263=
0097" target=3D"_blank"><span lang=3D"EN-US">http://dl.acm.org/citation.cfm=
?id=3D2630097</span></a><span lang=3D"EN-US">&gt;<br>
            </span><span style=3D"color:#1f497d" lang=3D"EN-US">[IJ]
              Interesting ,will look at it, not sure though that it will
              work that well with 4G (cant say anything about 5G), in 4G
              the transmission grants can vary quite a lot over short
              time spans.<u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
            </span><b>3.=C2=A0 4G and 5G evolution<br>
            </b><br>
            s/a explicit/an explicit/<br>
            <br>
            <b>4.=C2=A0 Requirements for improved performance<br>
            </b><br>
            s/bufferbloat/buffer/<br>
            <br>
            <b>5.=C2=A0 Congestion control examples<br>
            </b><br>
            Regarding Hybrid Cubic using OWD measurements, I have seen a
            paper (under submission) about getting delay-based
            congestion controls (e.g. LEDBAT or CAIA Delay Gradient) to
            interwork with AQMs with various low delay targets. It&#39;s no=
t
            in the context of cellular networks, but I&#39;ll try to
            remember to point it out once it gets published.<span style=3D"=
color:#1f497d"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] Sounds interesting , looking forward to
              see the paper.<u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              <b>5.1 HyStartRestart<br>
              </b><br>
              s/algorithm used/algorithm is used/<br>
              <br>
              In the new code, it would be useful to explain why the
              second test before doubling ssthresh, ie, the test for how
              long it has been since the last hyrestart.
            </span>Also, in the header definitions, if you intend there
            to be constraints on the relation between N_RTT_HYRESTART
            and N_RTT_LOW (e.g.=C2=A0 N_RTT_HYRESTART &gt; N_RTT_LOW) you
            ought to say, because people will try other values.<br>
            <br>
            Rather than the rather arbitrary wait before an arbitrary
            doubling of ssthresh, wouldn&#39;t it be better to continually
            increase ssthresh (not necessarily linearly) the longer the
            RTT has remained only just higher than the min RTT?<span style=
=3D"color:#1f497d"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] Yes, could be an option<u></u><u></u></=
span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              (BTW, surely the main problem with this modification to
              HyStartRestart is that is requires a number of round trips
              to reduce delay - a sort-of oxymoron.)</span><span style=3D"c=
olor:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] Yes, have tried some more experiments
              with this algorithm. And I am getting more hesitant. The
              problem is that when I add more noise (modeling of small
              objects traffic) into the system simulation, then I also
              get more delay jitter and the delay detection algorithm in
              HyStart is notoriously sensitive to delay jitter.<u></u><u></=
u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
            </span><b>5.2.=C2=A0 Hybrid Cubic</b><span style=3D"color:#1f49=
7d" lang=3D"EN-US"><u></u><u></u></span></p>
          <pre>=C2=A0=C2=A0 Furthermore it is assumed than the timestamp cl=
ock frequency in<u></u><u></u></pre>
          <pre>=C2=A0=C2=A0 sender and receiver are identical or that the s=
ender can infer the<u></u><u></u></pre>
          <pre>=C2=A0=C2=A0 timestamp clock frequency of the receiver and r=
ecompute timestamp<u></u><u></u></pre>
          <pre>=C2=A0=C2=A0 values based on this information.<u></u><u></u>=
</pre>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Recently on
            tcpm I sent you a pointer to the chirping paper Mirja &amp;
            I did, which also made this assumption. Partly prompted by
            that, Richard Scheffenegger tried to get people interested
            in standardising use of the timestamp option on a SYN to
            negotiate stuff like timer resolution. I don&#39;t think it wen=
t
            anywhere. Do you think we ought to revitalise that? You
            mention inference - have you tried that - are there cases
            where it doesn&#39;t work?<span style=3D"color:#1f497d"><u></u>=
<u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US">[IJ] Yes, actually wondered what happened to
              Richards draft. I only briefly tried out the inference
              algorithm with a TCP LEDBAT implementation (</span><span styl=
e=3D"font-size:9.0pt;font-family:Consolas;color:#969896;background:white" l=
ang=3D"EN-US"><a href=3D"https://github.com/silviov/TCP-LEDBAT/blob/master/=
src/tcp_ledbat.c" target=3D"_blank">https://github.com/silviov/TCP-LEDBAT/b=
lob/master/src/tcp_ledbat.c</a></span><span style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d" lang=3D"EN=
-US">). I recall that it took a while to converge.
              Also I am not sure here. Is Hz in e.g a linux stack fixed.
              ?</span></p>
        </div>
      </div>
    </blockquote>
    I believe so. But I don&#39;t know about embedded systems etc.<br>
    <br>
    Cheers<br>
    <br>
    <br>
    <br>
    Bob<br>
    <blockquote type=3D"cite">
      <div>
        <div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm =
0cm 0cm 4.0pt">
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:#1f497d" lang=3D"EN-US"><u></u><u></u></span></p>
          <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=
=3D"EN-US"><br>
              <br>
              <br>
              That&#39;s it. Thanks again for the draft. V useful.<br>
              <br>
              <br>
              <br>
              Bob<br>
              <br>
              <br>
              <u></u><u></u></span></p>
          <div>
            <p class=3D"MsoNormal"><span lang=3D"EN-US">On 15/10/15 06:08,
                Ingemar Johansson S </span>
              wrote:<u></u><u></u></p>
          </div>
          <blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
            <pre>Hi<u></u><u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>A new IETF draft is submitted in response to general discu=
ssion that for instance TCP Cubic is not an appropriate congestion control =
algorithm for LTE. <u></u><u></u></pre>
            <pre>The draft briefly outlines typical 4G access behavior on t=
he PDCP, RLC and MAC=C2=A0 layers that can have an impact on transport prot=
ocol performance. The draft also outlines two relatively simple modificatio=
ns to the Cubic congestion control.<u></u><u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>/Ingemar=C2=A0 <u></u><u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>-----Original Message-----<u></u><u></u></pre>
            <pre>From: <a href=3D"mailto:internet-drafts@ietf.org" target=
=3D"_blank">internet-drafts@ietf.org</a> [<a href=3D"mailto:internet-drafts=
@ietf.org" target=3D"_blank">mailto:internet-drafts@ietf.org</a>] <u></u><u=
></u></pre>
            <pre>Sent: den 14 oktober 2015 20:17<u></u><u></u></pre>
            <pre>To: Ingemar Johansson S<u></u><u></u></pre>
            <pre>Subject: New Version Notification for draft-johansson-cc-f=
or-4g-5g-00.txt<u></u><u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>A new version of I-D, draft-johansson-cc-for-4g-5g-00.txt<=
u></u><u></u></pre>
            <pre>has been successfully submitted by Ingemar Johansson and p=
osted to the IETF repository.<u></u><u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>Name:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0 draft-johansson-cc-for-4g-5g<u></u><u></u></pre>
            <pre>Revision:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 00<u></u><u>=
</u></pre>
            <pre>Title:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0 Congestion control for 4G and 5G access<u></u><u></u></pre>
            <pre>Document date:=C2=A0 2015-10-14<u></u><u></u></pre>
            <pre>Group:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0 Individual Submission<u></u><u></u></pre>
            <pre>Pages:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0 14<u></u><u></u></pre>
            <pre>URL:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0 <a href=3D"https://www.ietf.org/internet-drafts/draft-johansso=
n-cc-for-4g-5g-00.txt" target=3D"_blank">https://www.ietf.org/internet-draf=
ts/draft-johansson-cc-for-4g-5g-00.txt</a><u></u><u></u></pre>
            <pre>Status:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a=
 href=3D"https://datatracker.ietf.org/doc/draft-johansson-cc-for-4g-5g/" ta=
rget=3D"_blank">https://datatracker.ietf.org/doc/draft-johansson-cc-for-4g-=
5g/</a><u></u><u></u></pre>
            <pre>Htmlized:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a href=3D"h=
ttps://tools.ietf.org/html/draft-johansson-cc-for-4g-5g-00" target=3D"_blan=
k">https://tools.ietf.org/html/draft-johansson-cc-for-4g-5g-00</a><u></u><u=
></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>Abstract:<u></u><u></u></pre>
            <pre>=C2=A0=C2=A0 This memo outlines the challenge that 4G and =
5G access brings for<u></u><u></u></pre>
            <pre>=C2=A0=C2=A0 transport protocol congestion control and als=
o outlines a few simple<u></u><u></u></pre>
            <pre>=C2=A0=C2=A0 examples that can improve transport protocol =
congestion control<u></u><u></u></pre>
            <pre>=C2=A0=C2=A0 performance in 4G and 5G access.<u></u><u></u=
></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <u></u><u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>Please note that it may take a couple of minutes from the =
time of submission until the htmlized version and diff are available at <a =
href=3D"http://tools.ietf.org" target=3D"_blank">tools.ietf.org</a>.<u></u>=
<u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
            <pre>The IETF Secretariat<u></u><u></u></pre>
            <pre><u></u>=C2=A0<u></u></pre>
          </blockquote>
          <p class=3D"MsoNormal"><br>
            <br>
            <br>
            <u></u><u></u></p>
          <pre>-- <u></u><u></u></pre>
          <pre>____________________________________________________________=
____<u></u><u></u></pre>
          <pre>Bob Briscoe=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a href=3D"http:/=
/bobbriscoe.net/" target=3D"_blank">http://bobbriscoe.net/</a><u></u><u></u=
></pre>
        </div>
      </div>
    </blockquote>
    <br>
    <pre cols=3D"72">--=20
________________________________________________________________
Bob Briscoe                               <a href=3D"http://bobbriscoe.net/=
" target=3D"_blank">http://bobbriscoe.net/</a></pre>
  </div>

<br>_______________________________________________<br>
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	charset=windows-1252

Ingemar,

We believe this is a useful document, agree that it's a good match for =
ICCRG, and would be happy with ICCRG being the home of this draft.

Thanks,

Michael & David (as ICCRG chairs)


On 04 Nov 2015, at 22:27, Ingemar Johansson S =
<ingemar.s.johansson@ericsson.com> wrote:

> Hi
> =20
> And thanks for the review, comments/answers/questions inline below
> =20
> /Ingemar
> =20
> From: Bob Briscoe [mailto:ietf@bobbriscoe.net]=20
> Sent: den 29 oktober 2015 18:54
> To: Ingemar Johansson S; iccrg@irtf.org; tcpm@ietf.org; tsvwg@ietf.org
> Subject: Re: [tsvwg] FW: New Version Notification for =
draft-johansson-cc-for-4g-5g-00.txt
> =20
> Ingemar,
>=20
> As promised, here's my comments. They are a mix of suggested =
improvements to the draft, suggested improvements to 5G, or requests for =
clarification. I also agree with all Kevin Smith's comments, so I didn't =
repeat any.
> [Sorry for delay - I had to find time to re-type after losing my =
review during a power outage]
>=20
> Context questions.
> Q1. Is this exercise treating the 5G specs solely in read-only mode? =
Or, if some useful design suggestions arise, is there any chance that =
the 3GPP might take them on board, given 5G is not fully baked yet?
>=20
> [IJ] Guess it depends, need to say that I probably the small cog in =
the wheel. I would say that a lot of the standards work in IETF has been =
picked up by 3GPP and, I cannot say what will happen in the 5G context.=20=

>=20
> It's great that you have taken this initiative. Does the 3GPP ever =
formally ask the IETF for comments on its ideas for each new generation =
of radio? Given the layers above can be thought of as the 'customers' of =
the 3GPP, and these 'customer' layers are pretty much all maintained by =
the IETF, it would seem essential (and courteous) for the 3GPP to ask =
for customer comment, not least to co-ordinate with whatever future =
plans the IETF might have for the higher layers.
>=20
> [IJ] What you say makes much sense. I guess there is some natural =
leakage between 3GPP and IETF as many of the IETFers are also active in =
3GPP standardization. I don=92t have full insight in the formal relation =
between 3GPP and IETF in these areas, not sure if that need to be =
better, but if that is the case, then it is probably a question for the =
area directors. Need to say that even though the formal channels are =
scarce it does probably not mean that 3GPP ignores IETF or what happens =
above IP in general. For instance the recent work around TCP RACK raises =
the question if it is necessary that LTE default radio bearers ensure in =
order delivery.
>=20
>=20
>=20
> Q2. What are your plans for this draft? Are you asking for adoption? =
And if so, in which WG or RG? ICCRG looks the most appropriate, given =
the charters of all the IETF cc WGs (TSVWG, TCPM, RMCAT, etc) are scoped =
on just a subset of your doc.
>=20
> [IJ] First objective is to get an increased interest in congestion =
control algorithm work in the IETF. Traditionally this kind of work seem =
to be more in the terms of conference papers in e.g. Sigcomm, the =
benefit with having such work in IETF is that input from people with a =
wider expertise area (including 3GPP experitise) can give algorithm =
proposals that work well for a wider set of access types. The work =
around Cubic is a first good step as is the work around DCTCP. What I =
wonder is if it feasible to take this a bit further ? I would too =
believe that the current doc fits better in ICCRG.  The algorithm =
examples are just.. examples. More fruitful work is probably to document =
different access types and from there derive a set of best current =
practices for congestion control algorithms
>=20
>=20
>=20
> Section by Section Review
>=20
> All sections
>=20
> The document generally talks about the LTE protocol stack as if data =
is travelling down it. Are you implying that most sources of buffering =
delay are at the sending end, and the feedback end is pretty =
unencumbered by delays? I think it would be useful to explicitly say =
something about this (whether true or not), to explain why only one =
direction of travel is discussed in the doc.
>=20
> [IJ] The feedback path is encumbered by delays, it is perhaps not that =
clear but section 2.3.2 implies that delay (variation) in uplink will =
affect the ACKs for downlink data traffic. That said, the difference in =
DL/UL traffic is today something like 10:1 and that is probably one =
reason why one can get the feeling that the draft is slanted towards =
downlink data traffic, this may however change with time as more machine =
type traffic may enter the system.
>=20
>=20
>=20
> 2.1 PDCP layer
>=20
>     PDCP also ensures that all
>    packets are delivered in order up to higher layers.
> I think this means "in the order sent into the link (PDCP) layer". =
This ought to be clarified, otherwise, when writing in the context of a =
layer 4 audience, it might be assumed this means in the order sent by =
the original transport layer source.
>=20
> [IJ] Yes, you are right , reordering can still occur e.g. in the =
wireless backhaul
>=20
>=20
>=20
>    keep the amount of data in flight as small as possible, without =
sacrificing throughput.
> Is this just a rather convoluted way of saying "keep the RTT as low as =
possible"? (which in turn implies keeping queuing delay as low as =
possible assuming you cannot influence the physical path.) Or were you =
trying to make a more subtle point? - if so, it was lost on me.
>=20
> [IJ] No, it is exactly as you describe below
>=20
>=20
>=20
>    Reliable delivery at handover may however be turned off or is =
simply not implemented,
> It would be useful if you could give a feel for roughly how often (in =
%) these cases hold.
>=20
> [IJ] I don=92t believe that it is possible to give a figure as this is =
vendor specific.
>=20
>=20
>=20
> 2.2 RLC layer
>=20
>    role ... is to ensure that packets are delivered in order up to the =
higher layers
> Eh? The PDCP section just said it did ordering. And the rest of the =
RLC section talks exclusively about buffering delays necessary while =
filling transport blocks. If ordering is the role of both layers, surely =
one will have nothing to do once the other has got everything in order?
>=20
> [IJ] RLC handles ordering due to various MAC layer failures. PDCP =
handles ordering  when handover occurs
>=20
>=20
>=20
>    varying size of the available transport
> Nit: Given 'transport' means e2e for the audience of this draft, pls =
use bearer or somehow disambiguate this word.
>=20
> [IJ] OK
>=20
>=20
>=20
>=20
> Gap#1: In the draft, it seems like transport blocks arrive by magic, =
to be filled by data buffered in the RLC layer. To understand the delay =
compromises here, I think the draft needs to explain what determines how =
much transport block capacity each flow is given, and how they can =
influence it. In 3G, I believe that the transport layer could ask the =
radio resource controller to change radio resource allocation. Is this =
the same in 4G & 5G? Does the backlog of the buffer into the RLC layer =
have any automatic influence over allocation of available resources? I =
assume traffic class can also be used to influence allocation.
>=20
> [IJ] Will updated with more detail
>=20
>=20
>=20
> Gap#2: In a 3G stack, the RLC layer multiplexes all the active =
application streams into MAC layer transport block by dicing up buffered =
packets and packing the pieces into each transport block. Then, at the =
other end of the link, each piece has to be held until the rest of the =
packet arrives. This sacrifices delay for bandwidth efficiency. Is this =
still done in 4G and 5G? If so, it would be good to write about mux =
delay in the draft too.
>=20
> [IJ] OK, will add more info
>=20
>=20
>=20
> 2.3 MAC layer
>=20
>    the maximum number of retransmissions is configurable
>=20
> Wouldn't it be better to set a limit on the time spent retransmitting, =
so it would automatically give up with less retransmissions for longer =
transmission distances?
> [IJ] Possible, 4G specifies a limit to the number of retransmissions, =
in practice it can be translated to time as each retransmission is 8ms =
later.
>=20
> =20
>=20
>=20
> s/only checks for the presence on download data only at regular =
intervals/
>  /only checks for the presence of download data at regular intervals/
> [OK]
>=20
> =20
>=20
>=20
> 2.3.2 Uplink scheduling
>=20
>    The scheduling request does not indicate how many bytes there are =
in the uplink queue.
> Seems like a bad omission. Is there a reason? If there are less bytes =
queued than the base station's grant, surely it would be useful for the =
base station to know, so that it can grant more to something else while =
the previous short transmission is completing.
> [IJ] It is a design choice. A buffer status report is more bulky, a =
scheduling request is just one bit.
>=20
> =20
>=20
>=20
> You say that when HARQ breaks up a sequence of ACKs, it can also =
trigger "coalescing issues". I can understand that it could cause ACK =
compression, but are you implying something coalesces the ACKs or the =
subsequent data? What does the coalescing? Similarly, in the last para =
of 2.3.2 you say "Reduced ACKs can unfortunately cause coalescing." and =
again I'm not sure what is coalescing what here.
>=20
> [IJ] What I can see in simulations is that the ACK compression leads =
to coalescing which further can increase jitter.
>=20
>=20
> There could be a positive side to the interaction between ACK clocking =
and link aggregation, at least for long-running flows. This has been =
observed in 802.11n WiFi where the buffering and aggregation of multiple =
ACKs into one transmission grant tend to lead the sender to bunch =
multiple data segments so that they all nicely fill one transmission =
grant in the other direction [Showail14]. As long as the transmission =
grant does not wait for more data (which unfortunately I don't think is =
the case in 3G/4G/5G), this could be good for bandwidth utilisation of =
elephants and latency of mice, including when they are mixed.
>=20
> [Showail14] Showail, A., Jamshaid, K. & Shihada, B., "WQM: An =
Aggregation-aware Queue Management Scheme for IEEE 802.11N Based =
Networks," In: Proceedings of the 2014 ACM SIGCOMM Workshop on Capacity =
Sharing Workshop CSWS '14 pp.15-20 ACM (2014)
> <http://dl.acm.org/citation.cfm?id=3D2630097>
> [IJ] Interesting ,will look at it, not sure though that it will work =
that well with 4G (cant say anything about 5G), in 4G the transmission =
grants can vary quite a lot over short time spans.
>=20
> =20
>=20
>=20
> 3.  4G and 5G evolution
>=20
> s/a explicit/an explicit/
>=20
> 4.  Requirements for improved performance
>=20
> s/bufferbloat/buffer/
>=20
> 5.  Congestion control examples
>=20
> Regarding Hybrid Cubic using OWD measurements, I have seen a paper =
(under submission) about getting delay-based congestion controls (e.g. =
LEDBAT or CAIA Delay Gradient) to interwork with AQMs with various low =
delay targets. It's not in the context of cellular networks, but I'll =
try to remember to point it out once it gets published.
>=20
> [IJ] Sounds interesting , looking forward to see the paper.
>=20
>=20
>=20
> 5.1 HyStartRestart
>=20
> s/algorithm used/algorithm is used/
>=20
> In the new code, it would be useful to explain why the second test =
before doubling ssthresh, ie, the test for how long it has been since =
the last hyrestart. Also, in the header definitions, if you intend there =
to be constraints on the relation between N_RTT_HYRESTART and N_RTT_LOW =
(e.g.  N_RTT_HYRESTART > N_RTT_LOW) you ought to say, because people =
will try other values.
>=20
> Rather than the rather arbitrary wait before an arbitrary doubling of =
ssthresh, wouldn't it be better to continually increase ssthresh (not =
necessarily linearly) the longer the RTT has remained only just higher =
than the min RTT?
>=20
> [IJ] Yes, could be an option
>=20
>=20
>=20
> (BTW, surely the main problem with this modification to HyStartRestart =
is that is requires a number of round trips to reduce delay - a sort-of =
oxymoron.)
>=20
> [IJ] Yes, have tried some more experiments with this algorithm. And I =
am getting more hesitant. The problem is that when I add more noise =
(modeling of small objects traffic) into the system simulation, then I =
also get more delay jitter and the delay detection algorithm in HyStart =
is notoriously sensitive to delay jitter.
>=20
>=20
>=20
> 5.2.  Hybrid Cubic
>=20
>    Furthermore it is assumed than the timestamp clock frequency in
>    sender and receiver are identical or that the sender can infer the
>    timestamp clock frequency of the receiver and recompute timestamp
>    values based on this information.
> Recently on tcpm I sent you a pointer to the chirping paper Mirja & I =
did, which also made this assumption. Partly prompted by that, Richard =
Scheffenegger tried to get people interested in standardising use of the =
timestamp option on a SYN to negotiate stuff like timer resolution. I =
don't think it went anywhere. Do you think we ought to revitalise that? =
You mention inference - have you tried that - are there cases where it =
doesn't work?
>=20
> [IJ] Yes, actually wondered what happened to Richards draft. I only =
briefly tried out the inference algorithm with a TCP LEDBAT =
implementation =
(https://github.com/silviov/TCP-LEDBAT/blob/master/src/tcp_ledbat.c). I =
recall that it took a while to converge. Also I am not sure here. Is Hz =
in e.g a linux stack fixed. ?
>=20
>=20
>=20
>=20
> That's it. Thanks again for the draft. V useful.
>=20
>=20
>=20
> Bob
>=20
>=20
>=20
> On 15/10/15 06:08, Ingemar Johansson S wrote:
> Hi
> =20
> A new IETF draft is submitted in response to general discussion that =
for instance TCP Cubic is not an appropriate congestion control =
algorithm for LTE.=20
> The draft briefly outlines typical 4G access behavior on the PDCP, RLC =
and MAC  layers that can have an impact on transport protocol =
performance. The draft also outlines two relatively simple modifications =
to the Cubic congestion control.
> =20
> /Ingemar =20
> =20
> -----Original Message-----
> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]=20
> Sent: den 14 oktober 2015 20:17
> To: Ingemar Johansson S
> Subject: New Version Notification for =
draft-johansson-cc-for-4g-5g-00.txt
> =20
> =20
> A new version of I-D, draft-johansson-cc-for-4g-5g-00.txt
> has been successfully submitted by Ingemar Johansson and posted to the =
IETF repository.
> =20
> Name:           draft-johansson-cc-for-4g-5g
> Revision:       00
> Title:          Congestion control for 4G and 5G access
> Document date:  2015-10-14
> Group:          Individual Submission
> Pages:          14
> URL:            =
https://www.ietf.org/internet-drafts/draft-johansson-cc-for-4g-5g-00.txt
> Status:         =
https://datatracker.ietf.org/doc/draft-johansson-cc-for-4g-5g/
> Htmlized:       =
https://tools.ietf.org/html/draft-johansson-cc-for-4g-5g-00
> =20
> =20
> Abstract:
>    This memo outlines the challenge that 4G and 5G access brings for
>    transport protocol congestion control and also outlines a few =
simple
>    examples that can improve transport protocol congestion control
>    performance in 4G and 5G access.
> =20
> =20
>                                                                        =
          =20
> =20
> =20
> 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.
> =20
> The IETF Secretariat
> =20
>=20
>=20
>=20
> --=20
> ________________________________________________________________
> Bob Briscoe                               http://bobbriscoe.net/
> _______________________________________________
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;"><div>Ingemar,</div><div><br></div><div>We believe =
this is a useful document, agree that it's a good match for ICCRG, and =
would be happy with ICCRG being the home of this =
draft.</div><div><br></div><div>Thanks,</div><div><br></div><div>Michael =
&amp; David (as ICCRG chairs)</div><br><div><div><br></div><div>On 04 =
Nov 2015, at 22:27, Ingemar Johansson S &lt;<a =
href=3D"mailto:ingemar.s.johansson@ericsson.com">ingemar.s.johansson@erics=
son.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div =
bgcolor=3D"white" lang=3D"SV" link=3D"blue" vlink=3D"purple" =
style=3D"font-family: Helvetica; font-size: 13px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: auto; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; widows: auto; word-spacing: =
0px; -webkit-text-stroke-width: 0px;"><div class=3D"WordSection1" =
style=3D"page: WordSection1;"><div style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125);">Hi<o:p></o:p></span></div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;"><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, =
125);">&nbsp;</span></div><div style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125);">And thanks for the review, =
comments/answers/questions inline below<o:p></o:p></span></div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;"><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, =
125);">&nbsp;</span></div><div style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, =
125);">/Ingemar<o:p></o:p></span></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125);">&nbsp;</span></div><div =
style=3D"border-style: none none none solid; border-left-color: blue; =
border-left-width: 1.5pt; padding: 0cm 0cm 0cm 4pt;"><div><div =
style=3D"border-style: solid none none; border-top-color: rgb(181, 196, =
223); border-top-width: 1pt; padding: 3pt 0cm 0cm;"><div style=3D"margin: =
0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', =
serif;"><b><span lang=3D"EN-US" style=3D"font-size: 10pt; font-family: =
Tahoma, sans-serif; color: windowtext;">From:</span></b><span =
lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Tahoma, =
sans-serif; color: windowtext;"><span =
class=3D"Apple-converted-space">&nbsp;</span>Bob Briscoe [<a =
href=3D"mailto:ietf@bobbriscoe.net" style=3D"color: purple; =
text-decoration: underline;">mailto:ietf@bobbriscoe.net</a>]<span =
class=3D"Apple-converted-space">&nbsp;</span><br><b>Sent:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>den 29 oktober 2015 =
18:54<br><b>To:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>Ingemar Johansson S;<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:iccrg@irtf.org" style=3D"color: purple; text-decoration: =
underline;">iccrg@irtf.org</a>;<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:tcpm@ietf.org" style=3D"color: purple; text-decoration: =
underline;">tcpm@ietf.org</a>;<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:tsvwg@ietf.org" style=3D"color: purple; text-decoration: =
underline;">tsvwg@ietf.org</a><br><b>Subject:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>Re: [tsvwg] FW: New Version =
Notification for =
draft-johansson-cc-for-4g-5g-00.txt<o:p></o:p></span></div></div></div><di=
v style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: =
'Times New Roman', serif;"><o:p>&nbsp;</o:p></div><p class=3D"MsoNormal" =
style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New =
Roman', serif;">Ingemar,<br><br>As promised, here's my comments. They =
are a mix of suggested improvements to the draft, suggested improvements =
to 5G, or requests for clarification. I also agree with all Kevin =
Smith's comments, so I didn't repeat any.<br>[Sorry for delay - I had to =
find time to re-type after losing my review during a power =
outage]<br><br><b><u>Context questions.</u><br></b>Q1. Is this exercise =
treating the 5G specs solely in read-only mode? Or, if some useful =
design suggestions arise, is there any chance that the 3GPP might take =
them on board, given 5G is not fully baked yet?<span style=3D"color: =
rgb(31, 73, 125);"><o:p></o:p></span></p><p class=3D"MsoNormal" =
style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New =
Roman', serif;"><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125);">[IJ] Guess =
it depends, need to say that I probably the small cog in the wheel. I =
would say that a lot of the standards work in IETF has been picked up by =
3GPP and, I cannot say what will happen in the 5G context.<span =
class=3D"Apple-converted-space">&nbsp;</span></span><span =
lang=3D"EN-US"><br><br>It's great that you have taken this =
initiative.<span class=3D"Apple-converted-space">&nbsp;</span></span>Does =
the 3GPP ever formally ask the IETF for comments on its ideas for each =
new generation of radio? Given the layers above can be thought of as the =
'customers' of the 3GPP, and these 'customer' layers are pretty much all =
maintained by the IETF, it would seem essential (and courteous) for the =
3GPP to ask for customer comment, not least to co-ordinate with whatever =
future plans the IETF might have for the higher layers.<span =
style=3D"color: rgb(31, 73, 125);"><o:p></o:p></span></p><p =
class=3D"MsoNormal" style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; =
font-family: 'Times New Roman', serif;"><span lang=3D"EN-US" =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125);">[IJ] What you say makes much sense. I guess there is =
some natural leakage between 3GPP and IETF as many of the IETFers are =
also active in 3GPP standardization. I don=92t have full insight in the =
formal relation between 3GPP and IETF in these areas, not sure if that =
need to be better, but if that is the case, then it is probably a =
question for the area directors. Need to say that even though the formal =
channels are scarce it does probably not mean that 3GPP ignores IETF or =
what happens above IP in general. For instance the recent work around =
TCP RACK raises the question if it is necessary that LTE default radio =
bearers ensure in order delivery.<o:p></o:p></span></p><p =
class=3D"MsoNormal" style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; =
font-family: 'Times New Roman', serif;"><span lang=3D"EN-US"><br><br>Q2. =
What are your plans for this draft?<span =
class=3D"Apple-converted-space">&nbsp;</span></span>Are you asking for =
adoption? And if so, in which WG or RG? ICCRG looks the most =
appropriate, given the charters of all the IETF cc WGs (TSVWG, TCPM, =
RMCAT, etc) are scoped on just a subset of your doc.<span style=3D"color: =
rgb(31, 73, 125);"><o:p></o:p></span></p><p class=3D"MsoNormal" =
style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New =
Roman', serif;"><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125);">[IJ] First =
objective is to get an increased interest in congestion control =
algorithm work in the IETF. Traditionally this kind of work seem to be =
more in the terms of conference papers in e.g. Sigcomm, the benefit with =
having such work in IETF is that input from people with a wider =
expertise area (including 3GPP experitise) can give algorithm proposals =
that work well for a wider set of access types. The work around Cubic is =
a first good step as is the work around DCTCP. What I wonder is if it =
feasible to take this a bit further ? I would too believe that the =
current doc fits better in ICCRG. &nbsp;The algorithm examples are =
just.. examples. More fruitful work is probably to document different =
access types and from there derive a set of best current practices for =
congestion control algorithms<o:p></o:p></span></p><p class=3D"MsoNormal" =
style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New =
Roman', serif;"><span lang=3D"EN-US"><br><br><b><u>Section by Section =
Review</u><br></b><br><b>All sections<br></b><br>The document generally =
talks about the LTE protocol stack as if data is travelling down =
it.<span class=3D"Apple-converted-space">&nbsp;</span></span>Are you =
implying that most sources of buffering delay are at the sending end, =
and the feedback end is pretty unencumbered by delays? I think it would =
be useful to explicitly say something about this (whether true or not), =
to explain why only one direction of travel is discussed in the =
doc.<span style=3D"color: rgb(31, 73, 125);"><o:p></o:p></span></p><p =
class=3D"MsoNormal" style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; =
font-family: 'Times New Roman', serif;"><span lang=3D"EN-US" =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125);">[IJ] The feedback path is encumbered by delays, it is =
perhaps not that clear but section 2.3.2 implies that delay (variation) =
in uplink will affect the ACKs for downlink data traffic. That said, the =
difference in DL/UL traffic is today something like 10:1 and that is =
probably one reason why one can get the feeling that the draft is =
slanted towards downlink data traffic, this may however change with time =
as more machine type traffic may enter the =
system.<o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: 0cm =
0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US"><br><br><b>2.1 PDCP =
layer<br></b><br>&nbsp;&nbsp;&nbsp;<span =
class=3D"Apple-converted-space">&nbsp;</span></span><tt =
style=3D"font-family: 'Courier New';"><span lang=3D"EN-US" =
style=3D"font-size: 10pt;">PDCP also ensures that all</span></tt><span =
lang=3D"EN-US" style=3D"font-size: 10pt; font-family: 'Courier =
New';"><br><tt style=3D"font-family: 'Courier New';">&nbsp;&nbsp; =
packets are delivered in order up to higher layers.</tt><br></span>I =
think this means "in the order sent into the link (PDCP) layer". This =
ought to be clarified, otherwise, when writing in the context of a layer =
4 audience, it might be assumed this means in the order sent by the =
original transport layer source.<span style=3D"color: rgb(31, 73, =
125);"><o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: 0cm =
0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125);">[IJ] Yes, you are right , =
reordering can still occur e.g. in the wireless =
backhaul<o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: =
0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', =
serif;"><span lang=3D"EN-US"><br><br></span><tt style=3D"font-family: =
'Courier New';"><span lang=3D"EN-US" style=3D"font-size: =
10pt;">&nbsp;&nbsp; keep the amount of data in flight as small as =
possible, without sacrificing throughput.</span></tt><span lang=3D"EN-US" =
style=3D"font-size: 10pt; font-family: 'Courier New';"><br></span>Is =
this just a rather convoluted way of saying "keep the RTT as low as =
possible"? (which in turn implies keeping queuing delay as low as =
possible assuming you cannot influence the physical path.) Or were you =
trying to make a more subtle point? - if so, it was lost on me.<span =
style=3D"color: rgb(31, 73, 125);"><o:p></o:p></span></p><p =
class=3D"MsoNormal" style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; =
font-family: 'Times New Roman', serif;"><span lang=3D"EN-US" =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125);">[IJ] No, it is exactly as you describe =
below<o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: 0cm =
0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US"><br><br></span><tt style=3D"font-family: 'Courier =
New';"><span lang=3D"EN-US" style=3D"font-size: 10pt;">&nbsp;&nbsp; =
Reliable delivery at handover may however be turned off or is simply not =
implemented,</span></tt><span lang=3D"EN-US" style=3D"font-size: 10pt; =
font-family: 'Courier New';"><br></span><span lang=3D"EN-US">It would be =
useful if you could give a feel for roughly how often (in %) these cases =
hold.</span><span lang=3D"EN-US" style=3D"color: rgb(31, 73, =
125);"><o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: 0cm =
0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125);">[IJ] I don=92t believe that it is =
possible to give a figure as this is vendor =
specific.<o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: =
0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', =
serif;"><span lang=3D"EN-US"><br><br><b>2.2 RLC =
layer</b><br><br></span><tt style=3D"font-family: 'Courier New';"><span =
lang=3D"EN-US" style=3D"font-size: 10pt;">&nbsp;&nbsp; role ... is to =
ensure that packets are delivered in order up to the higher =
layers</span></tt><span lang=3D"EN-US" style=3D"font-size: 10pt; =
font-family: 'Courier New';"><br></span><span lang=3D"EN-US">Eh?<span =
class=3D"Apple-converted-space">&nbsp;</span></span>The PDCP section =
just said it did ordering. And the rest of the RLC section talks =
exclusively about buffering delays necessary while filling transport =
blocks. If ordering is the role of both layers, surely one will have =
nothing to do once the other has got everything in order?<span =
style=3D"color: rgb(31, 73, 125);"><o:p></o:p></span></p><p =
class=3D"MsoNormal" style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; =
font-family: 'Times New Roman', serif;"><span lang=3D"EN-US" =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125);">[IJ] RLC handles ordering due to various MAC layer =
failures. PDCP handles ordering &nbsp;when handover =
occurs<o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: 0cm =
0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US"><br><br></span><tt style=3D"font-family: 'Courier =
New';"><span lang=3D"EN-US" style=3D"font-size: 10pt;">&nbsp;&nbsp; =
varying size of the available transport</span></tt><span lang=3D"EN-US" =
style=3D"font-size: 10pt; font-family: 'Courier New';"><br></span><span =
lang=3D"EN-US">Nit: Given 'transport' means e2e for the audience of this =
draft, pls use bearer or somehow disambiguate this word.</span><span =
lang=3D"EN-US" style=3D"color: rgb(31, 73, =
125);"><o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: 0cm =
0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125);">[IJ] OK<o:p></o:p></span></p><p =
class=3D"MsoNormal" style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; =
font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US"><br><br><br>Gap#1: In the draft, it seems like transport =
blocks arrive by magic, to be filled by data buffered in the RLC =
layer.<span class=3D"Apple-converted-space">&nbsp;</span></span>To =
understand the delay compromises here, I think the draft needs to =
explain what determines how much transport block capacity each flow is =
given, and how they can influence it. In 3G, I believe that the =
transport layer could ask the radio resource controller to change radio =
resource allocation. Is this the same in 4G &amp; 5G? Does the backlog =
of the buffer into the RLC layer have any automatic influence over =
allocation of available resources? I assume traffic class can also be =
used to influence allocation.<span style=3D"color: rgb(31, 73, =
125);"><o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: 0cm =
0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125);">[IJ] Will updated with more =
detail<o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: 0cm =
0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US"><br><br>Gap#2: In a 3G stack, the RLC layer multiplexes =
all the active application streams into MAC layer transport block by =
dicing up buffered packets and packing the pieces into each transport =
block.<span class=3D"Apple-converted-space">&nbsp;</span></span>Then, at =
the other end of the link, each piece has to be held until the rest of =
the packet arrives. This sacrifices delay for bandwidth efficiency. Is =
this still done in 4G and 5G? If so, it would be good to write about mux =
delay in the draft too.<span style=3D"color: rgb(31, 73, =
125);"><o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: 0cm =
0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125);">[IJ] OK, will add more =
info<o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: 0cm =
0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US"><br><br><b>2.3 MAC layer<br></b><br></span><tt =
style=3D"font-family: 'Courier New';"><span lang=3D"EN-US" =
style=3D"font-size: 10pt;">&nbsp;&nbsp; the maximum number of =
retransmissions is configurable</span></tt><span lang=3D"EN-US" =
style=3D"font-size: 10pt; font-family: 'Courier New';"><br></span><span =
lang=3D"EN-US"><br>Wouldn't it be better to set a limit on the<span =
class=3D"Apple-converted-space">&nbsp;</span><i>time</i><span =
class=3D"Apple-converted-space">&nbsp;</span>spent retransmitting, so it =
would automatically give up with less retransmissions for longer =
transmission distances?<br></span><span lang=3D"EN-US" style=3D"color: =
rgb(31, 73, 125);">[IJ] Possible, 4G specifies a limit to the number of =
retransmissions, in practice it can be translated to time as each =
retransmission is 8ms later.<o:p></o:p></span></p><p class=3D"MsoNormal" =
style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New =
Roman', serif;"><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, =
125);">&nbsp;</span></p><p class=3D"MsoNormal" style=3D"margin: 0cm 0cm =
12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US"><br>s/only checks for the presence on download data only =
at regular intervals/<br>&nbsp;/only checks for the presence of download =
data at regular intervals/<br></span><span lang=3D"EN-US" style=3D"color: =
rgb(31, 73, 125);">[OK]<o:p></o:p></span></p><p class=3D"MsoNormal" =
style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New =
Roman', serif;"><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, =
125);">&nbsp;</span></p><p class=3D"MsoNormal" style=3D"margin: 0cm 0cm =
12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US"><br><b>2.3.2 Uplink scheduling<br></b><br></span><tt =
style=3D"font-family: 'Courier New';"><span lang=3D"EN-US" =
style=3D"font-size: 10pt;">&nbsp;&nbsp; The scheduling request does not =
indicate how many bytes there are in the uplink queue.</span></tt><span =
lang=3D"EN-US" style=3D"font-size: 10pt; font-family: 'Courier =
New';"><br></span>Seems like a bad omission. Is there a reason? If there =
are less bytes queued than the base station's grant, surely it would be =
useful for the base station to know, so that it can grant more to =
something else while the previous short transmission is =
completing.<br><span lang=3D"EN-US" style=3D"color: rgb(31, 73, =
125);">[IJ] It is a design choice. A buffer status report is more bulky, =
a scheduling request is just one bit.<o:p></o:p></span></p><p =
class=3D"MsoNormal" style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; =
font-family: 'Times New Roman', serif;"><span lang=3D"EN-US" =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125);">&nbsp;</span></p><p class=3D"MsoNormal" =
style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New =
Roman', serif;"><span lang=3D"EN-US"><br>You say that when HARQ breaks =
up a sequence of ACKs, it can also trigger "coalescing issues".<span =
class=3D"Apple-converted-space">&nbsp;</span></span>I can understand =
that it could cause ACK compression, but are you implying something =
coalesces the ACKs or the subsequent data? What does the =
coalescing?<span class=3D"Apple-converted-space">&nbsp;</span><span =
lang=3D"EN-US">Similarly, in the last para of 2.3.2 you say "Reduced =
ACKs can unfortunately cause coalescing." and again I'm not sure what is =
coalescing what here.</span><span lang=3D"EN-US" style=3D"color: rgb(31, =
73, 125);"><o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: =
0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', =
serif;"><span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125);">[IJ] What I can see in =
simulations is that the ACK compression leads to coalescing which =
further can increase jitter.<o:p></o:p></span></p><p class=3D"MsoNormal" =
style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New =
Roman', serif;"><span lang=3D"EN-US"><br>There could be a positive side =
to the interaction between ACK clocking and link aggregation, at least =
for long-running flows.<span =
class=3D"Apple-converted-space">&nbsp;</span></span>This has been =
observed in 802.11n WiFi where the buffering and aggregation of multiple =
ACKs into one transmission grant tend to lead the sender to bunch =
multiple data segments so that they all nicely fill one transmission =
grant in the other direction [Showail14].<span =
class=3D"Apple-converted-space">&nbsp;</span><span lang=3D"EN-US">As =
long as the transmission grant does not wait for more data (which =
unfortunately I don't think is the case in 3G/4G/5G), this could be good =
for bandwidth utilisation of elephants and latency of mice, including =
when they are mixed.<br><br>[Showail14] Showail, A., Jamshaid, K. &amp; =
Shihada, B., "WQM: An Aggregation-aware Queue Management Scheme for IEEE =
802.11N Based Networks," In: Proceedings of the 2014 ACM SIGCOMM =
Workshop on Capacity Sharing Workshop CSWS '14 pp.15-20 ACM =
(2014)<br>&lt;</span><a href=3D"http://dl.acm.org/citation.cfm?id=3D263009=
7" style=3D"color: purple; text-decoration: underline;"><span =
lang=3D"EN-US">http://dl.acm.org/citation.cfm?id=3D2630097</span></a><span=
 lang=3D"EN-US">&gt;<br></span><span lang=3D"EN-US" style=3D"color: =
rgb(31, 73, 125);">[IJ] Interesting ,will look at it, not sure though =
that it will work that well with 4G (cant say anything about 5G), in 4G =
the transmission grants can vary quite a lot over short time =
spans.<o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: 0cm =
0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125);">&nbsp;</span></p><p =
class=3D"MsoNormal" style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; =
font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US"><br></span><b>3.&nbsp; 4G and 5G evolution<br></b><br>s/a =
explicit/an explicit/<br><br><b>4.&nbsp; Requirements for improved =
performance<br></b><br>s/bufferbloat/buffer/<br><br><b>5.&nbsp; =
Congestion control examples<br></b><br>Regarding Hybrid Cubic using OWD =
measurements, I have seen a paper (under submission) about getting =
delay-based congestion controls (e.g. LEDBAT or CAIA Delay Gradient) to =
interwork with AQMs with various low delay targets. It's not in the =
context of cellular networks, but I'll try to remember to point it out =
once it gets published.<span style=3D"color: rgb(31, 73, =
125);"><o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: 0cm =
0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125);">[IJ] Sounds interesting , looking =
forward to see the paper.<o:p></o:p></span></p><p class=3D"MsoNormal" =
style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New =
Roman', serif;"><span lang=3D"EN-US"><br><br><b>5.1 =
HyStartRestart<br></b><br>s/algorithm used/algorithm is used/<br><br>In =
the new code, it would be useful to explain why the second test before =
doubling ssthresh, ie, the test for how long it has been since the last =
hyrestart.<span class=3D"Apple-converted-space">&nbsp;</span></span>Also, =
in the header definitions, if you intend there to be constraints on the =
relation between N_RTT_HYRESTART and N_RTT_LOW (e.g.&nbsp; =
N_RTT_HYRESTART &gt; N_RTT_LOW) you ought to say, because people will =
try other values.<br><br>Rather than the rather arbitrary wait before an =
arbitrary doubling of ssthresh, wouldn't it be better to continually =
increase ssthresh (not necessarily linearly) the longer the RTT has =
remained only just higher than the min RTT?<span style=3D"color: rgb(31, =
73, 125);"><o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: =
0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', =
serif;"><span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125);">[IJ] Yes, could be an =
option<o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: 0cm =
0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US"><br><br>(BTW, surely the main problem with this =
modification to HyStartRestart is that is requires a number of round =
trips to reduce delay - a sort-of oxymoron.)</span><span lang=3D"EN-US" =
style=3D"color: rgb(31, 73, 125);"><o:p></o:p></span></p><p =
class=3D"MsoNormal" style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; =
font-family: 'Times New Roman', serif;"><span lang=3D"EN-US" =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125);">[IJ] Yes, have tried some more experiments with this =
algorithm. And I am getting more hesitant. The problem is that when I =
add more noise (modeling of small objects traffic) into the system =
simulation, then I also get more delay jitter and the delay detection =
algorithm in HyStart is notoriously sensitive to delay =
jitter.<o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: 0cm =
0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US"><br><br></span><b>5.2.&nbsp; Hybrid Cubic</b><span =
lang=3D"EN-US" style=3D"color: rgb(31, 73, =
125);"><o:p></o:p></span></p><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier New';">&nbsp;&nbsp; Furthermore =
it is assumed than the timestamp clock frequency in<o:p></o:p></pre><pre =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 10pt; font-family: =
'Courier New';">&nbsp;&nbsp; sender and receiver are identical or that =
the sender can infer the<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 10pt; font-family: 'Courier New';">&nbsp;&nbsp; =
timestamp clock frequency of the receiver and recompute =
timestamp<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier New';">&nbsp;&nbsp; values based =
on this information.<o:p></o:p></pre><p class=3D"MsoNormal" =
style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New =
Roman', serif;">Recently on tcpm I sent you a pointer to the chirping =
paper Mirja &amp; I did, which also made this assumption. Partly =
prompted by that, Richard Scheffenegger tried to get people interested =
in standardising use of the timestamp option on a SYN to negotiate stuff =
like timer resolution. I don't think it went anywhere. Do you think we =
ought to revitalise that? You mention inference - have you tried that - =
are there cases where it doesn't work?<span style=3D"color: rgb(31, 73, =
125);"><o:p></o:p></span></p><p class=3D"MsoNormal" style=3D"margin: 0cm =
0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif;"><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125);">[IJ] Yes, actually wondered what =
happened to Richards draft. I only briefly tried out the inference =
algorithm with a TCP LEDBAT implementation (</span><span lang=3D"EN-US" =
style=3D"font-size: 9pt; font-family: Consolas; color: rgb(150, 152, =
150); background-color: white; background-position: initial initial; =
background-repeat: initial initial;"><a =
href=3D"https://github.com/silviov/TCP-LEDBAT/blob/master/src/tcp_ledbat.c=
" style=3D"color: purple; text-decoration: =
underline;">https://github.com/silviov/TCP-LEDBAT/blob/master/src/tcp_ledb=
at.c</a></span><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125);">). I recall =
that it took a while to converge. Also I am not sure here. Is Hz in e.g =
a linux stack fixed. ?<o:p></o:p></span></p><p class=3D"MsoNormal" =
style=3D"margin: 0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New =
Roman', serif;"><span lang=3D"EN-US"><br><br><br>That's it. Thanks again =
for the draft. V =
useful.<br><br><br><br>Bob<br><br><br><o:p></o:p></span></p><div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif;"><span lang=3D"EN-US">On 15/10/15 06:08, Ingemar =
Johansson S<span =
class=3D"Apple-converted-space">&nbsp;</span></span>wrote:<o:p></o:p></div=
></div><blockquote style=3D"margin-top: 5pt; margin-bottom: 5pt;"><pre =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 10pt; font-family: =
'Courier New';">Hi<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 10pt; font-family: 'Courier =
New';"><o:p>&nbsp;</o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier New';">A new IETF draft is =
submitted in response to general discussion that for instance TCP Cubic =
is not an appropriate congestion control algorithm for LTE. =
<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; font-size: =
10pt; font-family: 'Courier New';">The draft briefly outlines typical 4G =
access behavior on the PDCP, RLC and MAC&nbsp; layers that can have an =
impact on transport protocol performance. The draft also outlines two =
relatively simple modifications to the Cubic congestion =
control.<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier =
New';"><o:p>&nbsp;</o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier New';">/Ingemar&nbsp; =
<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; font-size: =
10pt; font-family: 'Courier New';"><o:p>&nbsp;</o:p></pre><pre =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 10pt; font-family: =
'Courier New';">-----Original Message-----<o:p></o:p></pre><pre =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 10pt; font-family: =
'Courier New';">From: <a href=3D"mailto:internet-drafts@ietf.org" =
style=3D"color: purple; text-decoration: =
underline;">internet-drafts@ietf.org</a> [<a =
href=3D"mailto:internet-drafts@ietf.org" style=3D"color: purple; =
text-decoration: underline;">mailto:internet-drafts@ietf.org</a>] =
<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; font-size: =
10pt; font-family: 'Courier New';">Sent: den 14 oktober 2015 =
20:17<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; font-size: =
10pt; font-family: 'Courier New';">To: Ingemar Johansson =
S<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; font-size: =
10pt; font-family: 'Courier New';">Subject: New Version Notification for =
draft-johansson-cc-for-4g-5g-00.txt<o:p></o:p></pre><pre style=3D"margin: =
0cm 0cm 0.0001pt; font-size: 10pt; font-family: 'Courier =
New';"><o:p>&nbsp;</o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier =
New';"><o:p>&nbsp;</o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier New';">A new version of I-D, =
draft-johansson-cc-for-4g-5g-00.txt<o:p></o:p></pre><pre style=3D"margin: =
0cm 0cm 0.0001pt; font-size: 10pt; font-family: 'Courier New';">has been =
successfully submitted by Ingemar Johansson and posted to the IETF =
repository.<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier =
New';"><o:p>&nbsp;</o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier =
New';">Name:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-johansson-cc-for-4g-5g<o:p></o:p></pre><pre style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 10pt; font-family: 'Courier =
New';">Revision:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
00<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; font-size: =
10pt; font-family: 'Courier =
New';">Title:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Congestion control for 4G and 5G access<o:p></o:p></pre><pre =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 10pt; font-family: =
'Courier New';">Document date:&nbsp; 2015-10-14<o:p></o:p></pre><pre =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 10pt; font-family: =
'Courier =
New';">Group:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Individual Submission<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 10pt; font-family: 'Courier =
New';">Pages:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
14<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; font-size: =
10pt; font-family: 'Courier =
New';">URL:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; <a =
href=3D"https://www.ietf.org/internet-drafts/draft-johansson-cc-for-4g-5g-=
00.txt" style=3D"color: purple; text-decoration: =
underline;">https://www.ietf.org/internet-drafts/draft-johansson-cc-for-4g=
-5g-00.txt</a><o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier =
New';">Status:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a =
href=3D"https://datatracker.ietf.org/doc/draft-johansson-cc-for-4g-5g/" =
style=3D"color: purple; text-decoration: =
underline;">https://datatracker.ietf.org/doc/draft-johansson-cc-for-4g-5g/=
</a><o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; font-size: =
10pt; font-family: 'Courier =
New';">Htmlized:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a =
href=3D"https://tools.ietf.org/html/draft-johansson-cc-for-4g-5g-00" =
style=3D"color: purple; text-decoration: =
underline;">https://tools.ietf.org/html/draft-johansson-cc-for-4g-5g-00</a=
><o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; font-size: =
10pt; font-family: 'Courier New';"><o:p>&nbsp;</o:p></pre><pre =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 10pt; font-family: =
'Courier New';"><o:p>&nbsp;</o:p></pre><pre style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 10pt; font-family: 'Courier =
New';">Abstract:<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier New';">&nbsp;&nbsp; This memo =
outlines the challenge that 4G and 5G access brings =
for<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; font-size: =
10pt; font-family: 'Courier New';">&nbsp;&nbsp; transport protocol =
congestion control and also outlines a few simple<o:p></o:p></pre><pre =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 10pt; font-family: =
'Courier New';">&nbsp;&nbsp; examples that can improve transport =
protocol congestion control<o:p></o:p></pre><pre style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 10pt; font-family: 'Courier New';">&nbsp;&nbsp; =
performance in 4G and 5G access.<o:p></o:p></pre><pre style=3D"margin: =
0cm 0cm 0.0001pt; font-size: 10pt; font-family: 'Courier =
New';"><o:p>&nbsp;</o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier =
New';"><o:p>&nbsp;</o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier =
New';">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <o:p></o:p></pre><pre =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 10pt; font-family: =
'Courier New';"><o:p>&nbsp;</o:p></pre><pre style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 10pt; font-family: 'Courier =
New';"><o:p>&nbsp;</o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier New';">Please note that it may =
take a couple of minutes from the time of submission until the htmlized =
version and diff are available at <a href=3D"http://tools.ietf.org/" =
style=3D"color: purple; text-decoration: =
underline;">tools.ietf.org</a>.<o:p></o:p></pre><pre style=3D"margin: =
0cm 0cm 0.0001pt; font-size: 10pt; font-family: 'Courier =
New';"><o:p>&nbsp;</o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier New';">The IETF =
Secretariat<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 10pt; font-family: 'Courier =
New';"><o:p>&nbsp;</o:p></pre></blockquote><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', =
serif;"><br><br><br><o:p></o:p></div><pre style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 10pt; font-family: 'Courier New';">-- =
<o:p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; font-size: =
10pt; font-family: 'Courier =
New';">________________________________________________________________<o:=
p></o:p></pre><pre style=3D"margin: 0cm 0cm 0.0001pt; font-size: 10pt; =
font-family: 'Courier New';">Bob =
Briscoe&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D"http://bobbriscoe.net/"=
 style=3D"color: purple; text-decoration: =
underline;">http://bobbriscoe.net/</a><o:p></o:p></pre></div></div>_______=
________________________________________<br>tcpm mailing list<br><a =
href=3D"mailto:tcpm@ietf.org" style=3D"color: purple; text-decoration: =
underline;">tcpm@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/tcpm" style=3D"color: =
purple; text-decoration: =
underline;">https://www.ietf.org/mailman/listinfo/tcpm</a><br></div></bloc=
kquote></div><br></body></html>=

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Subject: [tcpm] A clarification about draft-welzl-tcpm-tcb-sharing
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Hi all,

I just watched the video of TCPM and noticed a minor confusion at the =
end about what the plan really is - because draft-welzl-tcpm-tcb-sharing =
just contains a table contrasting RFC 2140 to implementations, and are =
we aiming at Informational or a recommendation?

Our plan is to update RFC 2140 to bring it in line with today's reality =
- what is currently implemented. I think Informational is still fine as =
status (but open to debate if people feel otherwise).

Why didn't we write RFC2140bis but a separate document? Because one of =
the chairs suggested this as a way forward, rather than doing a full =
first version of RFC2140bis, given that we believed that just showing =
this table and opening it for discussion is the best way to get this =
started.

Cheers,
Michael


From nobody Thu Nov  5 04:34:36 2015
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Subject: [tcpm] I-D Action: draft-ietf-tcpm-rtorestart-10.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 Working Group of the IETF.

        Title           : TCP and SCTP RTO Restart
        Authors         : Per Hurtig
                          Anna Brunstrom
                          Andreas Petlund
                          Michael Welzl
	Filename        : draft-ietf-tcpm-rtorestart-10.txt
	Pages           : 17
	Date            : 2015-11-05

Abstract:
   This document describes a modified sender-side algorithm for managing
   the TCP and SCTP retransmission timers that provides faster loss
   recovery when there is a small amount of outstanding data for a
   connection.  The modification, RTO Restart (RTOR), allows the
   transport to restart its retransmission timer using a smaller timeout
   duration, so that the effective RTO becomes more aggressive in
   situations where fast retransmit cannot be used.  This enables faster
   loss detection and recovery for connections that are short-lived or
   application-limited.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-tcpm-rtorestart-10

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tcpm-rtorestart-10


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

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


From nobody Thu Nov  5 04:41:16 2015
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From: Per Hurtig <per.hurtig@kau.se>
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Hi all,

as no objections were made to the suggested change in the abstract, =
we=E2=80=99ve now submitted a new version of the draft.


That is, changed from draft-ietf-...-09 to -10

   o  Changed wording in abstract, from "delay" to "timeout duration=E2=80=
=9D.


Cheers,
Per Hurtig
> On 05 Nov 2015, at 13:34, internet-drafts@ietf.org wrote:
>=20
>=20
> 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 =
Working Group of the IETF.
>=20
>        Title           : TCP and SCTP RTO Restart
>        Authors         : Per Hurtig
>                          Anna Brunstrom
>                          Andreas Petlund
>                          Michael Welzl
> 	Filename        : draft-ietf-tcpm-rtorestart-10.txt
> 	Pages           : 17
> 	Date            : 2015-11-05
>=20
> Abstract:
>   This document describes a modified sender-side algorithm for =
managing
>   the TCP and SCTP retransmission timers that provides faster loss
>   recovery when there is a small amount of outstanding data for a
>   connection.  The modification, RTO Restart (RTOR), allows the
>   transport to restart its retransmission timer using a smaller =
timeout
>   duration, so that the effective RTO becomes more aggressive in
>   situations where fast retransmit cannot be used.  This enables =
faster
>   loss detection and recovery for connections that are short-lived or
>   application-limited.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tcpm-rtorestart/
>=20
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-tcpm-rtorestart-10
>=20
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tcpm-rtorestart-10
>=20
>=20
> 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.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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Subject: Re: [tcpm] A clarification about draft-welzl-tcpm-tcb-sharing
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Some data points to the Linux implementation: we've found it more harmful
than useful, and have disabled it at Google (internal/external) for years.

Externally: IP no longer serves a good index. e.g., DHCP, NAT. Therefore
most if not all metrics are brittle.
Even for places where IP indexing still works: reusing ssthresh from TCP
just recovered from timeout will force the new TCP to exit slow start right
away





On Thu, Nov 5, 2015 at 3:18 AM, Michael Welzl <michawe@ifi.uio.no> wrote:

> Hi all,
>
> I just watched the video of TCPM and noticed a minor confusion at the end
> about what the plan really is - because draft-welzl-tcpm-tcb-sharing just
> contains a table contrasting RFC 2140 to implementations, and are we aiming
> at Informational or a recommendation?
>
> Our plan is to update RFC 2140 to bring it in line with today's reality -
> what is currently implemented. I think Informational is still fine as
> status (but open to debate if people feel otherwise).
>
> Why didn't we write RFC2140bis but a separate document? Because one of the
> chairs suggested this as a way forward, rather than doing a full first
> version of RFC2140bis, given that we believed that just showing this table
> and opening it for discussion is the best way to get this started.
>
> Cheers,
> Michael
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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<div dir=3D"ltr">Some data points to the Linux implementation: we&#39;ve fo=
und it more harmful than useful, and have disabled it at Google (internal/e=
xternal) for years.=C2=A0<div><br></div><div>Externally: IP no longer serve=
s a good index. e.g., DHCP, NAT. Therefore most if not all metrics are brit=
tle.</div><div>Even for places where IP indexing still works: reusing ssthr=
esh from TCP just recovered from timeout will force the new TCP to exit slo=
w start right away</div><div><br></div><div>=C2=A0</div><div><br></div><div=
>=C2=A0 =C2=A0 =C2=A0</div><div class=3D"gmail_extra"><br><div class=3D"gma=
il_quote">On Thu, Nov 5, 2015 at 3:18 AM, Michael Welzl <span dir=3D"ltr">&=
lt;<a href=3D"mailto:michawe@ifi.uio.no" target=3D"_blank">michawe@ifi.uio.=
no</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"marg=
in:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi all,<br>
<br>
I just watched the video of TCPM and noticed a minor confusion at the end a=
bout what the plan really is - because draft-welzl-tcpm-tcb-sharing just co=
ntains a table contrasting RFC 2140 to implementations, and are we aiming a=
t Informational or a recommendation?<br>
<br>
Our plan is to update RFC 2140 to bring it in line with today&#39;s reality=
 - what is currently implemented. I think Informational is still fine as st=
atus (but open to debate if people feel otherwise).<br>
<br>
Why didn&#39;t we write RFC2140bis but a separate document? Because one of =
the chairs suggested this as a way forward, rather than doing a full first =
version of RFC2140bis, given that we believed that just showing this table =
and opening it for discussion is the best way to get this started.<br>
<br>
Cheers,<br>
Michael<br>
<br>
_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
</blockquote></div><br></div></div>

--001a114336bc4c18660523cfaa0b--


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Subject: Re: [tcpm] A clarification about draft-welzl-tcpm-tcb-sharing
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On 11/5/2015 10:54 AM, Yuchung Cheng wrote:
...
> Externally: IP no longer serves a good index. e.g., DHCP, NAT.

That's not necessarily true. In both cases, the machines that might
share an IP address often share the dominant aspects of their network
paths - delay, loss, max bandwidth, etc.

Additionally, the shared information is used only for connection
initialization. That shared information doesn't have to be accurate,
only somewhat better than the current default initial values.

> Therefore
> most if not all metrics are brittle.
> Even for places where IP indexing still works: reusing ssthresh from TCP
> just recovered from timeout will force the new TCP to exit slow start
> right away

Agreed; that's one of the issues we're hoping to learn more about and
include in the update. E.g., SSTHRESH from steady-state is a better
predictor of future behavior than an immediate anomalous value - low or
high. That's true for most parameters.

Joe


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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: Michael Welzl <michawe@ifi.uio.no>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: [tcpm] A clarification about draft-welzl-tcpm-tcb-sharing
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Subject: Re: [tcpm] A clarification about draft-welzl-tcpm-tcb-sharing
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> I just watched the video of TCPM and noticed a minor confusion at the
> end about what the plan really is - because draft-welzl-tcpm-tcb-
> sharing just contains a table contrasting RFC 2140 to implementations,
> and are we aiming at Informational or a recommendation?
>=20
> Our plan is to update RFC 2140 to bring it in line with today's reality
> - what is currently implemented. I think Informational is still fine as
> status (but open to debate if people feel otherwise).
>=20
> Why didn't we write RFC2140bis but a separate document? Because one of
> the chairs suggested this as a way forward, rather than doing a full
> first version of RFC2140bis, given that we believed that just showing
> this table and opening it for discussion is the best way to get this
> started.

And this chair somehow believes that having a very short I-D is much better=
 than a presentation w/o any I-D...

Michael


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From: Pasi Sarolahti <pasi.sarolahti@iki.fi>
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Yep, I was only interested to hear about your future plans about the =
document and rfc2140bis work in general. Understood that the current =
version was mainly to support the discussion, which is good.

- Pasi


On 06 Nov 2015, at 07:51, Scharf, Michael (Michael) =
<michael.scharf@alcatel-lucent.com> wrote:

>> I just watched the video of TCPM and noticed a minor confusion at the
>> end about what the plan really is - because draft-welzl-tcpm-tcb-
>> sharing just contains a table contrasting RFC 2140 to =
implementations,
>> and are we aiming at Informational or a recommendation?
>>=20
>> Our plan is to update RFC 2140 to bring it in line with today's =
reality
>> - what is currently implemented. I think Informational is still fine =
as
>> status (but open to debate if people feel otherwise).
>>=20
>> Why didn't we write RFC2140bis but a separate document? Because one =
of
>> the chairs suggested this as a way forward, rather than doing a full
>> first version of RFC2140bis, given that we believed that just showing
>> this table and opening it for discussion is the best way to get this
>> started.
>=20
> And this chair somehow believes that having a very short I-D is much =
better than a presentation w/o any I-D...
>=20
> Michael
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: Per Hurtig <per.hurtig@kau.se>, "tcpm@ietf.org Extensions" <tcpm@ietf.org>
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Michael Welzl <michawe@ifi.uio.no>
Subject: Re: [tcpm] A clarification about draft-welzl-tcpm-tcb-sharing
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On Thu, Nov 5, 2015 at 1:50 PM, Joe Touch <touch@isi.edu> wrote:
>
>
>
> On 11/5/2015 10:54 AM, Yuchung Cheng wrote:
> ...
> > Externally: IP no longer serves a good index. e.g., DHCP, NAT.
>
> That's not necessarily true. In both cases, the machines that might
> share an IP address often share the dominant aspects of their network
> paths - delay, loss, max bandwidth, etc.
>
> Additionally, the shared information is used only for connection
> initialization. That shared information doesn't have to be accurate,
> only somewhat better than the current default initial values.
Intuitively yes. But our tests show little difference in terms of
latency, loss rate, etc with or without the Linux IP metrics cache.
And disabling not only save cache write/read but also skip the pathological
cases.

Today a phone can have multiple public IP addresses and an IP address
can be used for multiple phones simultaneously (with different ports).
The RTT, link
quality, and BW can all be very different. This makes IP a less
reliable index for server cache or CB sharing even only for seeding phase.

I am not claiming it's universally useless. I am presenting a data
point primarily on servers.

>
>
> > Therefore
> > most if not all metrics are brittle.
> > Even for places where IP indexing still works: reusing ssthresh from TCP
> > just recovered from timeout will force the new TCP to exit slow start
> > right away
>
> Agreed; that's one of the issues we're hoping to learn more about and
> include in the update. E.g., SSTHRESH from steady-state is a better
> predictor of future behavior than an immediate anomalous value - low or
> high. That's true for most parameters.
>
> Joe


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Subject: Re: [tcpm] A clarification about draft-welzl-tcpm-tcb-sharing
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> On 06 Nov 2015, at 05:58, Yuchung Cheng <ycheng@google.com> wrote:
>=20
> On Thu, Nov 5, 2015 at 1:50 PM, Joe Touch <touch@isi.edu> wrote:
>>=20
>>=20
>>=20
>> On 11/5/2015 10:54 AM, Yuchung Cheng wrote:
>> ...
>>> Externally: IP no longer serves a good index. e.g., DHCP, NAT.
>>=20
>> That's not necessarily true. In both cases, the machines that might
>> share an IP address often share the dominant aspects of their network
>> paths - delay, loss, max bandwidth, etc.
>>=20
>> Additionally, the shared information is used only for connection
>> initialization. That shared information doesn't have to be accurate,
>> only somewhat better than the current default initial values.
> Intuitively yes. But our tests show little difference in terms of
> latency, loss rate, etc with or without the Linux IP metrics cache.
> And disabling not only save cache write/read but also skip the =
pathological
> cases.
>=20
> Today a phone can have multiple public IP addresses and an IP address
> can be used for multiple phones simultaneously (with different ports).
> The RTT, link
> quality, and BW can all be very different. This makes IP a less
> reliable index for server cache or CB sharing even only for seeding =
phase.
>=20
> I am not claiming it's universally useless. I am presenting a data
> point primarily on servers.

Sounds like at least some text explaining these potential problems with =
relying on the IP address is needed for the planned RFC2140bis. Which I =
think is quite appropriate. Thanks for the data point!

Cheers,
Michael


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From: "Black, David" <david.black@emc.com>
To: David Ros <dros@simula.no>, Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
Thread-Topic: [tsvwg] [tcpm] FW: New Version Notification for draft-johansson-cc-for-4g-5g-00.txt
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Subject: Re: [tcpm] [tsvwg] FW: New Version Notification for draft-johansson-cc-for-4g-5g-00.txt
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--_000_CE03DB3D7B45C245BCA0D243277949362137FE78MX104CL02corpem_
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As a tsvwg chair, I concur that ICCRG would be a good initial home for this=
 draft.

Thanks,
--David

From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of David Ros
Sent: Thursday, November 05, 2015 5:13 AM
To: Ingemar Johansson S
Cc: tcpm@ietf.org; iccrg@irtf.org; tsvwg@ietf.org; Michael Welzl
Subject: Re: [tsvwg] [tcpm] FW: New Version Notification for draft-johansso=
n-cc-for-4g-5g-00.txt

Ingemar,

We believe this is a useful document, agree that it's a good match for ICCR=
G, and would be happy with ICCRG being the home of this draft.

Thanks,

Michael & David (as ICCRG chairs)


On 04 Nov 2015, at 22:27, Ingemar Johansson S <ingemar.s.johansson@ericsson=
.com<mailto:ingemar.s.johansson@ericsson.com>> wrote:


Hi

And thanks for the review, comments/answers/questions inline below

/Ingemar

From: Bob Briscoe [mailto:ietf@bobbriscoe.net]
Sent: den 29 oktober 2015 18:54
To: Ingemar Johansson S; iccrg@irtf.org<mailto:iccrg@irtf.org>; tcpm@ietf.o=
rg<mailto:tcpm@ietf.org>; tsvwg@ietf.org<mailto:tsvwg@ietf.org>
Subject: Re: [tsvwg] FW: New Version Notification for draft-johansson-cc-fo=
r-4g-5g-00.txt

Ingemar,

As promised, here's my comments. They are a mix of suggested improvements t=
o the draft, suggested improvements to 5G, or requests for clarification. I=
 also agree with all Kevin Smith's comments, so I didn't repeat any.
[Sorry for delay - I had to find time to re-type after losing my review dur=
ing a power outage]

Context questions.
Q1. Is this exercise treating the 5G specs solely in read-only mode? Or, if=
 some useful design suggestions arise, is there any chance that the 3GPP mi=
ght take them on board, given 5G is not fully baked yet?
[IJ] Guess it depends, need to say that I probably the small cog in the whe=
el. I would say that a lot of the standards work in IETF has been picked up=
 by 3GPP and, I cannot say what will happen in the 5G context.

It's great that you have taken this initiative. Does the 3GPP ever formally=
 ask the IETF for comments on its ideas for each new generation of radio? G=
iven the layers above can be thought of as the 'customers' of the 3GPP, and=
 these 'customer' layers are pretty much all maintained by the IETF, it wou=
ld seem essential (and courteous) for the 3GPP to ask for customer comment,=
 not least to co-ordinate with whatever future plans the IETF might have fo=
r the higher layers.
[IJ] What you say makes much sense. I guess there is some natural leakage b=
etween 3GPP and IETF as many of the IETFers are also active in 3GPP standar=
dization. I don't have full insight in the formal relation between 3GPP and=
 IETF in these areas, not sure if that need to be better, but if that is th=
e case, then it is probably a question for the area directors. Need to say =
that even though the formal channels are scarce it does probably not mean t=
hat 3GPP ignores IETF or what happens above IP in general. For instance the=
 recent work around TCP RACK raises the question if it is necessary that LT=
E default radio bearers ensure in order delivery.


Q2. What are your plans for this draft? Are you asking for adoption? And if=
 so, in which WG or RG? ICCRG looks the most appropriate, given the charter=
s of all the IETF cc WGs (TSVWG, TCPM, RMCAT, etc) are scoped on just a sub=
set of your doc.
[IJ] First objective is to get an increased interest in congestion control =
algorithm work in the IETF. Traditionally this kind of work seem to be more=
 in the terms of conference papers in e.g. Sigcomm, the benefit with having=
 such work in IETF is that input from people with a wider expertise area (i=
ncluding 3GPP experitise) can give algorithm proposals that work well for a=
 wider set of access types. The work around Cubic is a first good step as i=
s the work around DCTCP. What I wonder is if it feasible to take this a bit=
 further ? I would too believe that the current doc fits better in ICCRG.  =
The algorithm examples are just.. examples. More fruitful work is probably =
to document different access types and from there derive a set of best curr=
ent practices for congestion control algorithms


Section by Section Review

All sections

The document generally talks about the LTE protocol stack as if data is tra=
velling down it. Are you implying that most sources of buffering delay are =
at the sending end, and the feedback end is pretty unencumbered by delays? =
I think it would be useful to explicitly say something about this (whether =
true or not), to explain why only one direction of travel is discussed in t=
he doc.
[IJ] The feedback path is encumbered by delays, it is perhaps not that clea=
r but section 2.3.2 implies that delay (variation) in uplink will affect th=
e ACKs for downlink data traffic. That said, the difference in DL/UL traffi=
c is today something like 10:1 and that is probably one reason why one can =
get the feeling that the draft is slanted towards downlink data traffic, th=
is may however change with time as more machine type traffic may enter the =
system.


2.1 PDCP layer

    PDCP also ensures that all
   packets are delivered in order up to higher layers.
I think this means "in the order sent into the link (PDCP) layer". This oug=
ht to be clarified, otherwise, when writing in the context of a layer 4 aud=
ience, it might be assumed this means in the order sent by the original tra=
nsport layer source.
[IJ] Yes, you are right , reordering can still occur e.g. in the wireless b=
ackhaul


   keep the amount of data in flight as small as possible, without sacrific=
ing throughput.
Is this just a rather convoluted way of saying "keep the RTT as low as poss=
ible"? (which in turn implies keeping queuing delay as low as possible assu=
ming you cannot influence the physical path.) Or were you trying to make a =
more subtle point? - if so, it was lost on me.
[IJ] No, it is exactly as you describe below


   Reliable delivery at handover may however be turned off or is simply not=
 implemented,
It would be useful if you could give a feel for roughly how often (in %) th=
ese cases hold.
[IJ] I don't believe that it is possible to give a figure as this is vendor=
 specific.


2.2 RLC layer

   role ... is to ensure that packets are delivered in order up to the high=
er layers
Eh? The PDCP section just said it did ordering. And the rest of the RLC sec=
tion talks exclusively about buffering delays necessary while filling trans=
port blocks. If ordering is the role of both layers, surely one will have n=
othing to do once the other has got everything in order?
[IJ] RLC handles ordering due to various MAC layer failures. PDCP handles o=
rdering  when handover occurs


   varying size of the available transport
Nit: Given 'transport' means e2e for the audience of this draft, pls use be=
arer or somehow disambiguate this word.
[IJ] OK



Gap#1: In the draft, it seems like transport blocks arrive by magic, to be =
filled by data buffered in the RLC layer. To understand the delay compromis=
es here, I think the draft needs to explain what determines how much transp=
ort block capacity each flow is given, and how they can influence it. In 3G=
, I believe that the transport layer could ask the radio resource controlle=
r to change radio resource allocation. Is this the same in 4G & 5G? Does th=
e backlog of the buffer into the RLC layer have any automatic influence ove=
r allocation of available resources? I assume traffic class can also be use=
d to influence allocation.
[IJ] Will updated with more detail


Gap#2: In a 3G stack, the RLC layer multiplexes all the active application =
streams into MAC layer transport block by dicing up buffered packets and pa=
cking the pieces into each transport block. Then, at the other end of the l=
ink, each piece has to be held until the rest of the packet arrives. This s=
acrifices delay for bandwidth efficiency. Is this still done in 4G and 5G? =
If so, it would be good to write about mux delay in the draft too.
[IJ] OK, will add more info


2.3 MAC layer

   the maximum number of retransmissions is configurable

Wouldn't it be better to set a limit on the time spent retransmitting, so i=
t would automatically give up with less retransmissions for longer transmis=
sion distances?
[IJ] Possible, 4G specifies a limit to the number of retransmissions, in pr=
actice it can be translated to time as each retransmission is 8ms later.


s/only checks for the presence on download data only at regular intervals/
 /only checks for the presence of download data at regular intervals/
[OK]


2.3.2 Uplink scheduling

   The scheduling request does not indicate how many bytes there are in the=
 uplink queue.
Seems like a bad omission. Is there a reason? If there are less bytes queue=
d than the base station's grant, surely it would be useful for the base sta=
tion to know, so that it can grant more to something else while the previou=
s short transmission is completing.
[IJ] It is a design choice. A buffer status report is more bulky, a schedul=
ing request is just one bit.


You say that when HARQ breaks up a sequence of ACKs, it can also trigger "c=
oalescing issues". I can understand that it could cause ACK compression, bu=
t are you implying something coalesces the ACKs or the subsequent data? Wha=
t does the coalescing? Similarly, in the last para of 2.3.2 you say "Reduce=
d ACKs can unfortunately cause coalescing." and again I'm not sure what is =
coalescing what here.
[IJ] What I can see in simulations is that the ACK compression leads to coa=
lescing which further can increase jitter.

There could be a positive side to the interaction between ACK clocking and =
link aggregation, at least for long-running flows. This has been observed i=
n 802.11n WiFi where the buffering and aggregation of multiple ACKs into on=
e transmission grant tend to lead the sender to bunch multiple data segment=
s so that they all nicely fill one transmission grant in the other directio=
n [Showail14]. As long as the transmission grant does not wait for more dat=
a (which unfortunately I don't think is the case in 3G/4G/5G), this could b=
e good for bandwidth utilisation of elephants and latency of mice, includin=
g when they are mixed.

[Showail14] Showail, A., Jamshaid, K. & Shihada, B., "WQM: An Aggregation-a=
ware Queue Management Scheme for IEEE 802.11N Based Networks," In: Proceedi=
ngs of the 2014 ACM SIGCOMM Workshop on Capacity Sharing Workshop CSWS '14 =
pp.15-20 ACM (2014)
<http://dl.acm.org/citation.cfm?id=3D2630097>
[IJ] Interesting ,will look at it, not sure though that it will work that w=
ell with 4G (cant say anything about 5G), in 4G the transmission grants can=
 vary quite a lot over short time spans.


3.  4G and 5G evolution

s/a explicit/an explicit/

4.  Requirements for improved performance

s/bufferbloat/buffer/

5.  Congestion control examples

Regarding Hybrid Cubic using OWD measurements, I have seen a paper (under s=
ubmission) about getting delay-based congestion controls (e.g. LEDBAT or CA=
IA Delay Gradient) to interwork with AQMs with various low delay targets. I=
t's not in the context of cellular networks, but I'll try to remember to po=
int it out once it gets published.
[IJ] Sounds interesting , looking forward to see the paper.


5.1 HyStartRestart

s/algorithm used/algorithm is used/

In the new code, it would be useful to explain why the second test before d=
oubling ssthresh, ie, the test for how long it has been since the last hyre=
start. Also, in the header definitions, if you intend there to be constrain=
ts on the relation between N_RTT_HYRESTART and N_RTT_LOW (e.g.  N_RTT_HYRES=
TART > N_RTT_LOW) you ought to say, because people will try other values.

Rather than the rather arbitrary wait before an arbitrary doubling of ssthr=
esh, wouldn't it be better to continually increase ssthresh (not necessaril=
y linearly) the longer the RTT has remained only just higher than the min R=
TT?
[IJ] Yes, could be an option


(BTW, surely the main problem with this modification to HyStartRestart is t=
hat is requires a number of round trips to reduce delay - a sort-of oxymoro=
n.)
[IJ] Yes, have tried some more experiments with this algorithm. And I am ge=
tting more hesitant. The problem is that when I add more noise (modeling of=
 small objects traffic) into the system simulation, then I also get more de=
lay jitter and the delay detection algorithm in HyStart is notoriously sens=
itive to delay jitter.


5.2.  Hybrid Cubic

   Furthermore it is assumed than the timestamp clock frequency in

   sender and receiver are identical or that the sender can infer the

   timestamp clock frequency of the receiver and recompute timestamp

   values based on this information.
Recently on tcpm I sent you a pointer to the chirping paper Mirja & I did, =
which also made this assumption. Partly prompted by that, Richard Scheffene=
gger tried to get people interested in standardising use of the timestamp o=
ption on a SYN to negotiate stuff like timer resolution. I don't think it w=
ent anywhere. Do you think we ought to revitalise that? You mention inferen=
ce - have you tried that - are there cases where it doesn't work?
[IJ] Yes, actually wondered what happened to Richards draft. I only briefly=
 tried out the inference algorithm with a TCP LEDBAT implementation (https:=
//github.com/silviov/TCP-LEDBAT/blob/master/src/tcp_ledbat.c). I recall tha=
t it took a while to converge. Also I am not sure here. Is Hz in e.g a linu=
x stack fixed. ?



That's it. Thanks again for the draft. V useful.



Bob



On 15/10/15 06:08, Ingemar Johansson S wrote:

Hi



A new IETF draft is submitted in response to general discussion that for in=
stance TCP Cubic is not an appropriate congestion control algorithm for LTE=
.

The draft briefly outlines typical 4G access behavior on the PDCP, RLC and =
MAC  layers that can have an impact on transport protocol performance. The =
draft also outlines two relatively simple modifications to the Cubic conges=
tion control.



/Ingemar



-----Original Message-----

From: internet-drafts@ietf.org<mailto:internet-drafts@ietf.org> [mailto:int=
ernet-drafts@ietf.org]

Sent: den 14 oktober 2015 20:17

To: Ingemar Johansson S

Subject: New Version Notification for draft-johansson-cc-for-4g-5g-00.txt





A new version of I-D, draft-johansson-cc-for-4g-5g-00.txt

has been successfully submitted by Ingemar Johansson and posted to the IETF=
 repository.



Name:           draft-johansson-cc-for-4g-5g

Revision:       00

Title:          Congestion control for 4G and 5G access

Document date:  2015-10-14

Group:          Individual Submission

Pages:          14

URL:            https://www.ietf.org/internet-drafts/draft-johansson-cc-for=
-4g-5g-00.txt

Status:         https://datatracker.ietf.org/doc/draft-johansson-cc-for-4g-=
5g/

Htmlized:       https://tools.ietf.org/html/draft-johansson-cc-for-4g-5g-00





Abstract:

   This memo outlines the challenge that 4G and 5G access brings for

   transport protocol congestion control and also outlines a few simple

   examples that can improve transport protocol congestion control

   performance in 4G and 5G access.











Please note that it may take a couple of minutes from the time of submissio=
n until the htmlized version and diff are available at tools.ietf.org<http:=
//tools.ietf.org/>.



The IETF Secretariat







--

________________________________________________________________

Bob Briscoe                               http://bobbriscoe.net/
_______________________________________________
tcpm mailing list
tcpm@ietf.org<mailto:tcpm@ietf.org>
https://www.ietf.org/mailman/listinfo/tcpm


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</head>
<body lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;;color:black">As a tsvwg chair, I concur t=
hat ICCRG would be a good initial home for this draft.<o:p></o:p></span></p=
>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;;color:black"><o:p>&nbsp;</o:p></span></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;;color:black">Thanks,<br>
--David</span><span style=3D"font-size:11.0pt;font-family:&quot;Courier New=
&quot;,&quot;serif&quot;;color:black"><o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;;color:black"><o:p>&nbsp;</o:p></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> tsvwg [m=
ailto:tsvwg-bounces@ietf.org]
<b>On Behalf Of </b>David Ros<br>
<b>Sent:</b> Thursday, November 05, 2015 5:13 AM<br>
<b>To:</b> Ingemar Johansson S<br>
<b>Cc:</b> tcpm@ietf.org; iccrg@irtf.org; tsvwg@ietf.org; Michael Welzl<br>
<b>Subject:</b> Re: [tsvwg] [tcpm] FW: New Version Notification for draft-j=
ohansson-cc-for-4g-5g-00.txt<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">Ingemar,<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">We believe this is a useful document, agree that it'=
s a good match for ICCRG, and would be happy with ICCRG being the home of t=
his draft.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Thanks,<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Michael &amp; David (as ICCRG chairs)<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">On 04 Nov 2015, at 22:27, Ingemar Johansson S &lt;<a=
 href=3D"mailto:ingemar.s.johansson@ericsson.com">ingemar.s.johansson@erics=
son.com</a>&gt; wrote:<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi</span><span lang=3D"SV=
"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><span lang=
=3D"SV"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">And thanks for the review=
, comments/answers/questions inline below</span><span lang=3D"SV"><o:p></o:=
p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><span lang=
=3D"SV"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">/Ingemar</span><span lang=
=3D"SV"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><span lang=
=3D"SV"><o:p></o:p></span></p>
</div>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<div>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span class=3D"apple-=
converted-space"><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&q=
uot;,&quot;sans-serif&quot;">&nbsp;</span></span><span style=3D"font-size:1=
0.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">Bob
 Briscoe [<a href=3D"mailto:ietf@bobbriscoe.net"><span style=3D"color:purpl=
e">mailto:ietf@bobbriscoe.net</span></a>]<span class=3D"apple-converted-spa=
ce">&nbsp;</span><br>
<b>Sent:</b><span class=3D"apple-converted-space">&nbsp;</span>den 29 oktob=
er 2015 18:54<br>
<b>To:</b><span class=3D"apple-converted-space">&nbsp;</span>Ingemar Johans=
son S;<span class=3D"apple-converted-space">&nbsp;</span><a href=3D"mailto:=
iccrg@irtf.org"><span style=3D"color:purple">iccrg@irtf.org</span></a>;<spa=
n class=3D"apple-converted-space">&nbsp;</span><a href=3D"mailto:tcpm@ietf.=
org"><span style=3D"color:purple">tcpm@ietf.org</span></a>;<span class=3D"a=
pple-converted-space">&nbsp;</span><a href=3D"mailto:tsvwg@ietf.org"><span =
style=3D"color:purple">tsvwg@ietf.org</span></a><br>
<b>Subject:</b><span class=3D"apple-converted-space">&nbsp;</span>Re: [tsvw=
g] FW: New Version Notification for draft-johansson-cc-for-4g-5g-00.txt</sp=
an><span lang=3D"SV"><o:p></o:p></span></p>
</div>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"SV">&nbsp;<o:p></o:p></span></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=3D"SV">Ing=
emar,<br>
<br>
As promised, here's my comments. They are a mix of suggested improvements t=
o the draft, suggested improvements to 5G, or requests for clarification. I=
 also agree with all Kevin Smith's comments, so I didn't repeat any.<br>
[Sorry for delay - I had to find time to re-type after losing my review dur=
ing a power outage]<br>
<br>
<b><u>Context questions.</u><br>
</b>Q1. Is this exercise treating the 5G specs solely in read-only mode? Or=
, if some useful design suggestions arise, is there any chance that the 3GP=
P might take them on board, given 5G is not fully baked yet?<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] Guess it depends, need to say that I probably the small cog in t=
he wheel. I would say that a lot of the standards work in
 IETF has been picked up by 3GPP and, I cannot say what will happen in the =
5G context.<span class=3D"apple-converted-space">&nbsp;</span></span><br>
<br>
It's great that you have taken this initiative.<span class=3D"apple-convert=
ed-space">&nbsp;</span><span lang=3D"SV">Does the 3GPP ever formally ask th=
e IETF for comments on its ideas for each new generation of radio? Given th=
e layers above can be thought of as the 'customers'
 of the 3GPP, and these 'customer' layers are pretty much all maintained by=
 the IETF, it would seem essential (and courteous) for the 3GPP to ask for =
customer comment, not least to co-ordinate with whatever future plans the I=
ETF might have for the higher layers.<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] What you say makes much sense. I guess there is some natural lea=
kage between 3GPP and IETF as many of the IETFers are also
 active in 3GPP standardization. I don&#8217;t have full insight in the for=
mal relation between 3GPP and IETF in these areas, not sure if that need to=
 be better, but if that is the case, then it is probably a question for the=
 area directors. Need to say that even
 though the formal channels are scarce it does probably not mean that 3GPP =
ignores IETF or what happens above IP in general. For instance the recent w=
ork around TCP RACK raises the question if it is necessary that LTE default=
 radio bearers ensure in order delivery.</span><span lang=3D"SV"><o:p></o:p=
></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
Q2. What are your plans for this draft?<span class=3D"apple-converted-space=
">&nbsp;</span><span lang=3D"SV">Are you asking for adoption? And if so, in=
 which WG or RG? ICCRG looks the most appropriate, given the charters of al=
l the IETF cc WGs (TSVWG, TCPM, RMCAT, etc)
 are scoped on just a subset of your doc.<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] First objective is to get an increased interest in congestion co=
ntrol algorithm work in the IETF. Traditionally this kind
 of work seem to be more in the terms of conference papers in e.g. Sigcomm,=
 the benefit with having such work in IETF is that input from people with a=
 wider expertise area (including 3GPP experitise) can give algorithm propos=
als that work well for a wider set
 of access types. The work around Cubic is a first good step as is the work=
 around DCTCP. What I wonder is if it feasible to take this a bit further ?=
 I would too believe that the current doc fits better in ICCRG. &nbsp;The a=
lgorithm examples are just.. examples.
 More fruitful work is probably to document different access types and from=
 there derive a set of best current practices for congestion control algori=
thms</span><span lang=3D"SV"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
<b><u>Section by Section Review</u><br>
</b><br>
<b>All sections<br>
</b><br>
The document generally talks about the LTE protocol stack as if data is tra=
velling down it.<span class=3D"apple-converted-space">&nbsp;</span><span la=
ng=3D"SV">Are you implying that most sources of buffering delay are at the =
sending end, and the feedback end is pretty
 unencumbered by delays? I think it would be useful to explicitly say somet=
hing about this (whether true or not), to explain why only one direction of=
 travel is discussed in the doc.<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] The feedback path is encumbered by delays, it is perhaps not tha=
t clear but section 2.3.2 implies that delay (variation) in
 uplink will affect the ACKs for downlink data traffic. That said, the diff=
erence in DL/UL traffic is today something like 10:1 and that is probably o=
ne reason why one can get the feeling that the draft is slanted towards dow=
nlink data traffic, this may however
 change with time as more machine type traffic may enter the system.</span>=
<span lang=3D"SV"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
<b>2.1 PDCP layer<br>
</b><br>
&nbsp;&nbsp;&nbsp;<span class=3D"apple-converted-space">&nbsp;</span><tt><s=
pan style=3D"font-size:10.0pt">PDCP also ensures that all</span></tt><span =
style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;,&quot;serif&q=
uot;"><br>
<tt>&nbsp;&nbsp; packets are delivered in order up to higher layers.</tt><b=
r>
</span><span lang=3D"SV">I think this means &quot;in the order sent into th=
e link (PDCP) layer&quot;. This ought to be clarified, otherwise, when writ=
ing in the context of a layer 4 audience, it might be assumed this means in=
 the order sent by the original transport layer
 source.<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] Yes, you are right , reordering can still occur e.g. in the wire=
less backhaul</span><span lang=3D"SV"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
<tt><span style=3D"font-size:10.0pt">&nbsp;&nbsp; keep the amount of data i=
n flight as small as possible, without sacrificing throughput.</span></tt><=
span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;,&quot;se=
rif&quot;"><br>
</span><span lang=3D"SV">Is this just a rather convoluted way of saying &qu=
ot;keep the RTT as low as possible&quot;? (which in turn implies keeping qu=
euing delay as low as possible assuming you cannot influence the physical p=
ath.) Or were you trying to make a more subtle
 point? - if so, it was lost on me.<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] No, it is exactly as you describe below</span><span lang=3D"SV">=
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
<tt><span style=3D"font-size:10.0pt">&nbsp;&nbsp; Reliable delivery at hand=
over may however be turned off or is simply not implemented,</span></tt><sp=
an style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;,&quot;seri=
f&quot;"><br>
</span>It would be useful if you could give a feel for roughly how often (i=
n %) these cases hold.<span lang=3D"SV"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] I don&#8217;t believe that it is possible to give a figure as th=
is is vendor specific.</span><span lang=3D"SV"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
<b>2.2 RLC layer</b><br>
<br>
<tt><span style=3D"font-size:10.0pt">&nbsp;&nbsp; role ... is to ensure tha=
t packets are delivered in order up to the higher layers</span></tt><span s=
tyle=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;,&quot;serif&qu=
ot;"><br>
</span>Eh?<span class=3D"apple-converted-space">&nbsp;</span><span lang=3D"=
SV">The PDCP section just said it did ordering. And the rest of the RLC sec=
tion talks exclusively about buffering delays necessary while filling trans=
port blocks. If ordering is the role of both
 layers, surely one will have nothing to do once the other has got everythi=
ng in order?<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] RLC handles ordering due to various MAC layer failures. PDCP han=
dles ordering &nbsp;when handover occurs</span><span lang=3D"SV"><o:p></o:p=
></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
<tt><span style=3D"font-size:10.0pt">&nbsp;&nbsp; varying size of the avail=
able transport</span></tt><span style=3D"font-size:10.0pt;font-family:&quot=
;Courier New&quot;,&quot;serif&quot;"><br>
</span>Nit: Given 'transport' means e2e for the audience of this draft, pls=
 use bearer or somehow disambiguate this word.<span lang=3D"SV"><o:p></o:p>=
</span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] OK</span><span lang=3D"SV"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
<br>
Gap#1: In the draft, it seems like transport blocks arrive by magic, to be =
filled by data buffered in the RLC layer.<span class=3D"apple-converted-spa=
ce">&nbsp;</span><span lang=3D"SV">To understand the delay compromises here=
, I think the draft needs to explain what
 determines how much transport block capacity each flow is given, and how t=
hey can influence it. In 3G, I believe that the transport layer could ask t=
he radio resource controller to change radio resource allocation. Is this t=
he same in 4G &amp; 5G? Does the backlog
 of the buffer into the RLC layer have any automatic influence over allocat=
ion of available resources? I assume traffic class can also be used to infl=
uence allocation.<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] Will updated with more detail</span><span lang=3D"SV"><o:p></o:p=
></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
Gap#2: In a 3G stack, the RLC layer multiplexes all the active application =
streams into MAC layer transport block by dicing up buffered packets and pa=
cking the pieces into each transport block.<span class=3D"apple-converted-s=
pace">&nbsp;</span><span lang=3D"SV">Then,
 at the other end of the link, each piece has to be held until the rest of =
the packet arrives. This sacrifices delay for bandwidth efficiency. Is this=
 still done in 4G and 5G? If so, it would be good to write about mux delay =
in the draft too.<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] OK, will add more info</span><span lang=3D"SV"><o:p></o:p></span=
></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
<b>2.3 MAC layer<br>
</b><br>
<tt><span style=3D"font-size:10.0pt">&nbsp;&nbsp; the maximum number of ret=
ransmissions is configurable</span></tt><span style=3D"font-size:10.0pt;fon=
t-family:&quot;Courier New&quot;,&quot;serif&quot;"><br>
</span><br>
Wouldn't it be better to set a limit on the<span class=3D"apple-converted-s=
pace">&nbsp;</span><i>time</i><span class=3D"apple-converted-space">&nbsp;<=
/span>spent retransmitting, so it would automatically give up with less ret=
ransmissions for longer transmission distances?<br>
<span style=3D"color:#1F497D">[IJ] Possible, 4G specifies a limit to the nu=
mber of retransmissions, in practice it can be translated to time as each r=
etransmission is 8ms later.</span><span lang=3D"SV"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">&nbsp;</span><span lang=3D"SV"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
s/only checks for the presence on download data only at regular intervals/<=
br>
&nbsp;/only checks for the presence of download data at regular intervals/<=
br>
<span style=3D"color:#1F497D">[OK]</span><span lang=3D"SV"><o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">&nbsp;</span><span lang=3D"SV"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<b>2.3.2 Uplink scheduling<br>
</b><br>
<tt><span style=3D"font-size:10.0pt">&nbsp;&nbsp; The scheduling request do=
es not indicate how many bytes there are in the uplink queue.</span></tt><s=
pan style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;,&quot;ser=
if&quot;"><br>
</span><span lang=3D"SV">Seems like a bad omission. Is there a reason? If t=
here are less bytes queued than the base station's grant, surely it would b=
e useful for the base station to know, so that it can grant more to somethi=
ng else while the previous short transmission
 is completing.<br>
</span><span style=3D"color:#1F497D">[IJ] It is a design choice. A buffer s=
tatus report is more bulky, a scheduling request is just one bit.</span><sp=
an lang=3D"SV"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">&nbsp;</span><span lang=3D"SV"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
You say that when HARQ breaks up a sequence of ACKs, it can also trigger &q=
uot;coalescing issues&quot;.<span class=3D"apple-converted-space">&nbsp;</s=
pan><span lang=3D"SV">I can understand that it could cause ACK compression,=
 but are you implying something coalesces the ACKs
 or the subsequent data? What does the coalescing?<span class=3D"apple-conv=
erted-space">&nbsp;</span></span>Similarly, in the last para of 2.3.2 you s=
ay &quot;Reduced ACKs can unfortunately cause coalescing.&quot; and again I=
'm not sure what is coalescing what here.<span lang=3D"SV"><o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] What I can see in simulations is that the ACK compression leads =
to coalescing which further can increase jitter.</span><span lang=3D"SV"><o=
:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
There could be a positive side to the interaction between ACK clocking and =
link aggregation, at least for long-running flows.<span class=3D"apple-conv=
erted-space">&nbsp;</span><span lang=3D"SV">This has been observed in 802.1=
1n WiFi where the buffering and aggregation
 of multiple ACKs into one transmission grant tend to lead the sender to bu=
nch multiple data segments so that they all nicely fill one transmission gr=
ant in the other direction [Showail14].<span class=3D"apple-converted-space=
">&nbsp;</span></span>As long as the transmission
 grant does not wait for more data (which unfortunately I don't think is th=
e case in 3G/4G/5G), this could be good for bandwidth utilisation of elepha=
nts and latency of mice, including when they are mixed.<br>
<br>
[Showail14] Showail, A., Jamshaid, K. &amp; Shihada, B., &quot;WQM: An Aggr=
egation-aware Queue Management Scheme for IEEE 802.11N Based Networks,&quot=
; In: Proceedings of the 2014 ACM SIGCOMM Workshop on Capacity Sharing Work=
shop CSWS '14 pp.15-20 ACM (2014)<br>
&lt;<span lang=3D"SV"><a href=3D"http://dl.acm.org/citation.cfm?id=3D263009=
7"><span lang=3D"EN-US" style=3D"color:purple">http://dl.acm.org/citation.c=
fm?id=3D2630097</span></a></span>&gt;<br>
<span style=3D"color:#1F497D">[IJ] Interesting ,will look at it, not sure t=
hough that it will work that well with 4G (cant say anything about 5G), in =
4G the transmission grants can vary quite a lot over short time spans.</spa=
n><span lang=3D"SV"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">&nbsp;</span><span lang=3D"SV"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<b><span lang=3D"SV">3.&nbsp; 4G and 5G evolution<br>
</span></b><span lang=3D"SV"><br>
s/a explicit/an explicit/<br>
<br>
<b>4.&nbsp; Requirements for improved performance<br>
</b><br>
s/bufferbloat/buffer/<br>
<br>
<b>5.&nbsp; Congestion control examples<br>
</b><br>
Regarding Hybrid Cubic using OWD measurements, I have seen a paper (under s=
ubmission) about getting delay-based congestion controls (e.g. LEDBAT or CA=
IA Delay Gradient) to interwork with AQMs with various low delay targets. I=
t's not in the context of cellular
 networks, but I'll try to remember to point it out once it gets published.=
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] Sounds interesting , looking forward to see the paper.</span><sp=
an lang=3D"SV"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
<b>5.1 HyStartRestart<br>
</b><br>
s/algorithm used/algorithm is used/<br>
<br>
In the new code, it would be useful to explain why the second test before d=
oubling ssthresh, ie, the test for how long it has been since the last hyre=
start.<span class=3D"apple-converted-space">&nbsp;</span><span lang=3D"SV">=
Also, in the header definitions, if you intend
 there to be constraints on the relation between N_RTT_HYRESTART and N_RTT_=
LOW (e.g.&nbsp; N_RTT_HYRESTART &gt; N_RTT_LOW) you ought to say, because p=
eople will try other values.<br>
<br>
Rather than the rather arbitrary wait before an arbitrary doubling of ssthr=
esh, wouldn't it be better to continually increase ssthresh (not necessaril=
y linearly) the longer the RTT has remained only just higher than the min R=
TT?<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] Yes, could be an option</span><span lang=3D"SV"><o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
(BTW, surely the main problem with this modification to HyStartRestart is t=
hat is requires a number of round trips to reduce delay - a sort-of oxymoro=
n.)<span lang=3D"SV"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] Yes, have tried some more experiments with this algorithm. And I=
 am getting more hesitant. The problem is that when I add
 more noise (modeling of small objects traffic) into the system simulation,=
 then I also get more delay jitter and the delay detection algorithm in HyS=
tart is notoriously sensitive to delay jitter.</span><span lang=3D"SV"><o:p=
></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
<b><span lang=3D"SV">5.2.&nbsp; Hybrid Cubic</span></b><span lang=3D"SV"><o=
:p></o:p></span></p>
<pre><span lang=3D"SV">&nbsp;&nbsp; Furthermore it is assumed than the time=
stamp clock frequency in<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;&nbsp; sender and receiver are identical or th=
at the sender can infer the<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;&nbsp; timestamp clock frequency of the receiv=
er and recompute timestamp<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;&nbsp; values based on this information.<o:p><=
/o:p></span></pre>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span lang=3D"SV">Rec=
ently on tcpm I sent you a pointer to the chirping paper Mirja &amp; I did,=
 which also made this assumption. Partly prompted by that, Richard Scheffen=
egger tried to get people interested in standardising
 use of the timestamp option on a SYN to negotiate stuff like timer resolut=
ion. I don't think it went anywhere. Do you think we ought to revitalise th=
at? You mention inference - have you tried that - are there cases where it =
doesn't work?<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">[IJ] Yes, actually wondered what happened to Richards draft. I only b=
riefly tried out the inference algorithm with a TCP LEDBAT
 implementation (</span><span style=3D"font-size:9.0pt;font-family:Consolas=
;color:#969896;background:white"><a href=3D"https://github.com/silviov/TCP-=
LEDBAT/blob/master/src/tcp_ledbat.c"><span style=3D"color:purple">https://g=
ithub.com/silviov/TCP-LEDBAT/blob/master/src/tcp_ledbat.c</span></a></span>=
<span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-=
serif&quot;;color:#1F497D">).
 I recall that it took a while to converge. Also I am not sure here. Is Hz =
in e.g a linux stack fixed. ?</span><span lang=3D"SV"><o:p></o:p></span></p=
>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
<br>
That's it. Thanks again for the draft. V useful.<br>
<br>
<br>
<br>
Bob<br>
<br>
<br>
<br>
<span lang=3D"SV"><o:p></o:p></span></p>
<div>
<div>
<p class=3D"MsoNormal">On 15/10/15 06:08, Ingemar Johansson S<span class=3D=
"apple-converted-space">&nbsp;</span><span lang=3D"SV">wrote:<o:p></o:p></s=
pan></p>
</div>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<pre><span lang=3D"SV">Hi<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;<o:p></o:p></span></pre>
<pre><span lang=3D"SV">A new IETF draft is submitted in response to general=
 discussion that for instance TCP Cubic is not an appropriate congestion co=
ntrol algorithm for LTE. <o:p></o:p></span></pre>
<pre><span lang=3D"SV">The draft briefly outlines typical 4G access behavio=
r on the PDCP, RLC and MAC&nbsp; layers that can have an impact on transpor=
t protocol performance. The draft also outlines two relatively simple modif=
ications to the Cubic congestion control.<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;<o:p></o:p></span></pre>
<pre><span lang=3D"SV">/Ingemar&nbsp; <o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;<o:p></o:p></span></pre>
<pre><span lang=3D"SV">-----Original Message-----<o:p></o:p></span></pre>
<pre><span lang=3D"SV">From: <a href=3D"mailto:internet-drafts@ietf.org"><s=
pan style=3D"color:purple">internet-drafts@ietf.org</span></a> [<a href=3D"=
mailto:internet-drafts@ietf.org"><span style=3D"color:purple">mailto:intern=
et-drafts@ietf.org</span></a>] <o:p></o:p></span></pre>
<pre><span lang=3D"SV">Sent: den 14 oktober 2015 20:17<o:p></o:p></span></p=
re>
<pre><span lang=3D"SV">To: Ingemar Johansson S<o:p></o:p></span></pre>
<pre><span lang=3D"SV">Subject: New Version Notification for draft-johansso=
n-cc-for-4g-5g-00.txt<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;<o:p></o:p></span></pre>
<pre><span lang=3D"SV">A new version of I-D, draft-johansson-cc-for-4g-5g-0=
0.txt<o:p></o:p></span></pre>
<pre><span lang=3D"SV">has been successfully submitted by Ingemar Johansson=
 and posted to the IETF repository.<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;<o:p></o:p></span></pre>
<pre><span lang=3D"SV">Name:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp; draft-johansson-cc-for-4g-5g<o:p></o:p></span></pre>
<pre><span lang=3D"SV">Revision:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 00<o:p=
></o:p></span></pre>
<pre><span lang=3D"SV">Title:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; Congestion control for 4G and 5G access<o:p></o:p></span></pre>
<pre><span lang=3D"SV">Document date:&nbsp; 2015-10-14<o:p></o:p></span></p=
re>
<pre><span lang=3D"SV">Group:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; Individual Submission<o:p></o:p></span></pre>
<pre><span lang=3D"SV">Pages:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; 14<o:p></o:p></span></pre>
<pre><span lang=3D"SV">URL:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp; <a href=3D"https://www.ietf.org/internet-drafts/draft-jo=
hansson-cc-for-4g-5g-00.txt"><span style=3D"color:purple">https://www.ietf.=
org/internet-drafts/draft-johansson-cc-for-4g-5g-00.txt</span></a><o:p></o:=
p></span></pre>
<pre><span lang=3D"SV">Status:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; <a href=3D"https://datatracker.ietf.org/doc/draft-johansson-cc-for-4g-5=
g/"><span style=3D"color:purple">https://datatracker.ietf.org/doc/draft-joh=
ansson-cc-for-4g-5g/</span></a><o:p></o:p></span></pre>
<pre><span lang=3D"SV">Htmlized:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a hre=
f=3D"https://tools.ietf.org/html/draft-johansson-cc-for-4g-5g-00"><span sty=
le=3D"color:purple">https://tools.ietf.org/html/draft-johansson-cc-for-4g-5=
g-00</span></a><o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;<o:p></o:p></span></pre>
<pre><span lang=3D"SV">Abstract:<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;&nbsp; This memo outlines the challenge that 4=
G and 5G access brings for<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;&nbsp; transport protocol congestion control a=
nd also outlines a few simple<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;&nbsp; examples that can improve transport pro=
tocol congestion control<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;&nbsp; performance in 4G and 5G access.<o:p></=
o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <o:p></o:p></sp=
an></pre>
<pre><span lang=3D"SV">&nbsp;<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;<o:p></o:p></span></pre>
<pre><span lang=3D"SV">Please note that it may take a couple of minutes fro=
m the time of submission until the htmlized version and diff are available =
at <a href=3D"http://tools.ietf.org/"><span style=3D"color:purple">tools.ie=
tf.org</span></a>.<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;<o:p></o:p></span></pre>
<pre><span lang=3D"SV">The IETF Secretariat<o:p></o:p></span></pre>
<pre><span lang=3D"SV">&nbsp;<o:p></o:p></span></pre>
</blockquote>
<div>
<p class=3D"MsoNormal"><span lang=3D"SV"><br>
<br>
<br>
<br>
<o:p></o:p></span></p>
</div>
<pre><span lang=3D"SV">-- <o:p></o:p></span></pre>
<pre><span lang=3D"SV">____________________________________________________=
____________<o:p></o:p></span></pre>
<pre><span lang=3D"SV">Bob Briscoe&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D=
"http://bobbriscoe.net/"><span style=3D"color:purple">http://bobbriscoe.net=
/</span></a><o:p></o:p></span></pre>
</div>
<p class=3D"MsoNormal"><span lang=3D"SV" style=3D"font-size:10.0pt;font-fam=
ily:&quot;Helvetica&quot;,&quot;sans-serif&quot;">_________________________=
______________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org"><span style=3D"color:purple">tcpm@ietf.org=
</span></a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm"><span style=3D"color=
:purple">https://www.ietf.org/mailman/listinfo/tcpm</span></a><o:p></o:p></=
span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

--_000_CE03DB3D7B45C245BCA0D243277949362137FE78MX104CL02corpem_--


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Subject: [tcpm] Review of draft-ietf-tcpm-dctcp-01
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Praveen & co-authors,

As promised, here's my review of draft-ietf-tcpm-dctcp-01

_*General Comments*__*
*_
In general, in the comments below I have tried to reduce the multiple 
assumptions about controlled environments down to the one assumption 
that they are controlled by a single authority. I believe DCTCP can work 
without the other assumptions, so the following do not apply in all 
controlled environments:
   * not always low RTT (e.g. inter-DC scenarios, or global private 
networks)
   * not always no risk of traffic attacks (e.g. internally arranged 
attacks)
I may have missed some other occurrences of these assumptions, so pls 
feel free to seek out them all.

Many of the comments argue with your choices of MUST, SHOULD, MAY etc. 
This might seem nit-picking, given the primary purpose is to document 
the algo. However, it would be wrong to require things that aren't 
required or to allow exceptions when exceptions would not be 
interoperable. An alternative approach would be to just remove all 
capitalised language.

_*Section-by-section review.*__*
*_
*Abstract**
*
I think it needs to have an applicability statement in the abstract (and 
introduction) that refers to the deployment and implementation status 
sections at the end. Something like:

"This is an informational specification of the implementation of DCTCP 
in Microsoft Windows Server 2012. It is applicable to deployments in 
controlled environments like data centres but it MUST NOT be deployed 
over the public Internet without additional measures, as detailed in 
sections 4-6. "

*1. Intro**
*s/its many servers/
  /their many servers/

"...limited queue capacities..."
I understand that the motivation has been abbreviated, but I think this 
has lost too much. Perhaps mention memory shared between interfaces, so 
either bloated buffers or too little, making traffic susceptible to 
packet losses?

    [RFC3168] describes a mechanism for using Explicit Congestion
    Notification (ECN) from the switches for early detection of
    congestion, rather than waiting for packet loss to occur.

This isn't correct. 3168 requires ECN marking to occur no earlier than 
drop. It is precisely that 3168 does not allow early marking unless 
there is also early drop that is the problem.

    It is recommended that DCTCP be deployed...

"recommended" is too weak. I suggest you use something like the 
applicability text given above, both here and in the abstract.

*2. Terminology**
*
Suggest you explain that capitalised words are not quite the same as in 
most RFCs:
"Normative language is used to describe how necessary the various 
aspects of the Microsoft implementation are for interoperability, but 
even compliant implementations without the measures in sections 4-6 
would still only be safe to deploy in controlled environments."

*3. DCTCP Algo**
*
The three bullets say no more than "this is normal stuff". Would it be 
better to supplement each sentence with a brief phrase saying what is 
different about each aspect compared to RFC3168?

*3.1 Marking**
*
This needs to say something about how you mark the IP header from L2. I 
suggest you refer to the up-and-forward mode of draft-ietf-tsvwg-ecn-encap.

s/sending rate/
  /link rate/

*3.2 **
*
CURRENT:

    1.  If the CE codepoint is set and DCTCP.CE is false, send an ACK for
        any previously unacknowledged packets and set DCTCP.CE to true.

    2.  If the CE codepoint is not set and DCTCP.CE is true, send an ACK
        for any previously unacknowledged packets and set DCTCP.CE to
        false.

SUGGESTED:

    1.  If the CE codepoint is set and DCTCP.CE is false, set DCTCP.CE to
        true and send an ACK for any previously unacknowledged packets.

    2.  If the CE codepoint is not set and DCTCP.CE is true, set DCTCP.CE
        to false and send an ACK for any previously unacknowledged packets.

     The figure would have to be changed to be consistent too.
RATIONALE:
If the receiver changes DCTCP.CE /after/ sending the ACK like this, it 
delays each signal until the following ACK is sent. That could add a 
delay of anything from 1 to m packets. I've never understood why the 
order of these operations was specified in this way. If there is no 
reason, then I suggest that it is specified in the opposite order, as I 
describe above.

A note could be added to say the Windows Server 12 implementation does 
these steps in reverse order, but the order specified is preferred 
because it reduces signalling delay and has no interoperability issues 
with the original Windows implementation.

    The handling of the "Congestion Window Reduced" (CWR) bit is also
    exactly as per [RFC3168 <https://tools.ietf.org/html/rfc3168>] including [RFC3168-ERRATA3639 
<https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-01#ref-RFC3168-ERRATA3639>].  That is, on
    receipt of a segment with both the CE and CWR bits set, CWR is
    processed first and then ECE is processed.

Even tho this is the receiver section, I suggest:
s/The handling/Receiver handling/

Whatever, this cannot be correct. A DCTCP receiver surely does nothing 
on receipt of CWR, whereas an RFC3168 receiver does something. A DCTCP 
receiver certainly cannot do exactly what RFC3168 says, because that 
says stop setting ECE on receipt of CWR until the next CE arrives, which 
would surely break this feedback protocol.

*3.3 **
*

    DCTCP.Alpha, which is initialized to 1 and MUST be updated
    as follows:

s/MUST/SHOULD/
And add "An alternative congestion avoidance algorithm MAY be used, but 
it MUST lead to flow rate inversely proportional to 1/M".

Rationale: There are many ways to implement interoperable 1/p cc. This 
is just one.

    DCTCP.BytesSent

This variable actually counts bytes received. it's pretty confusing to 
call it BytesSent.

s/TCP SACK options [RFC2018] are ignored./
  /The sender MUST ignore TCP SACK options [RFC2018] for this computation./
RATIONALE:
   a) I think the idea is to ignore CE marks after a gap because, if the 
gap becomes considered a loss, the response to loss will override any 
need to count incoming CE marks for the following round trip. However, 
if the gap is filled before it is considered a loss {Note 1}, then CE 
marks that were ignored because they arrived after a gap will need to be 
"unignored".
   b) We need to make clear that SACK is not ignored altogether, only 
for this computation.

{Note 1}: I can think of two cases:
* Less than three dupacks then the gap is filled
* The gap is filled by a delayed segment so that an earlier 
retransmission was spurious

the sender MUST update cwnd as follows:
       cwnd = cwnd * (1 - DCTCP.Alpha / 2)

s/MUST/SHOULD/
Rationale: Alternative interoperable response functions are possible. 
For instance an algorithm like Relentless TCP that simply reduces cwnd 
by half the size of any CE-marked segment. Then there is no Alpha and no 
g variable.

CURRENT:

    Just as specified in [RFC3168 <https://tools.ietf.org/html/rfc3168>], TCP should not react to congestion
    indications more than once for every window of data.

SUGGESTED:

    Just as specified in [RFC3168 <https://tools.ietf.org/html/rfc3168>], DCTCP does not react to congestion
    indications more than once for every window of data.

I don't know whether you intended this to be an upper-case SHOULD NOT. 
If so, I would argue against this being normative, because it isn't 
necessary for interop; therefore I've suggested avoiding he 'should' word.

    The setting of
    the "Congestion Window Reduced" (CWR) bit is also exactly as per
    [RFC3168 <https://tools.ietf.org/html/rfc3168>].

It is wrong to say "exactly". The sender certainly sets CWR when it does 
a congestion response. But in DCTCP the congestion response occurs every 
RTT, triggered by its own measurement-period logic, whereas in RFC3168 
it is triggered by the arrival of an ECE after the ACK of any previous 
segment it sent with CWR set.

I think it is worth saying that setting CWR is necessary for safety, 
otherwise in the case where there has been a configuration management 
error and the receiver complies with RFC 3168 not DCTCP, it will 
continue sending ECE for ever and starve the session.

It might help to explicitly confirm that:
"Other aspects of TCP congestion avoidance and control such as the slow 
start phase are no different to those in RFC5681."

*3.4 SYN, SYN-ACK, RST**
*
    [RFC3168] requires that a compliant TCP MUST NOT set ECT on SYN or
    SYN-ACK packets.

s/MUST NOT/'MUST NOT'/
Rationale: Best to make it clear this is quoting a normative statement 
in another RFC, because it is going to be contradicted.

CURRENT

    These RFCs, however, are
    intended for general Internet use, and do not directly apply to a
    controlled datacenter environment.
...
    For DCTCP connections, the sender
    SHOULD set ECT for SYN, SYN-ACK and RST packets.

SUGGESTED:

    Also, a SYN is sent before the ECN-capability has been
    negotiated, so if it is CE-marked, a non-ECN TCP server would not
    recognise the marking. RFC 3168 does not specify the ECN field on
    a RST. The security concerns addressed by these RFCs
    might not apply in controlled environments like data centres, and
    it might not be necessary to cater for the presence of non-ECN
    servers.
...

    Therefore, the sender MUST set ECT for SYN/ACK and RST packets and a
    configuration option SHOULD be provided to set ECT for SYNs.

RATIONALE:
    The draft should avoid assuming that security concerns do not apply 
in all controlled environments.
NOTE:
    This might not reflect the way your implementation is coded, so you 
might want to make that clear.

*4. Implementation Issues**
*
CURRENT:

    the implementation MUST choose a suitable
    estimation gain.  [DCTCP10 <https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-01#ref-DCTCP10>] provides a theoretical basis

SUGGEST:

    the implementation will need to choose a suitable
    estimation gain.  [ADCTCP10 <https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-01#ref-DCTCP10>] provides a theoretical basis ...

RATIONALE:
    a) As above, this is only one way to implement the cc, and others 
could well prove to be interoperable when tested.
    b) More particularly, it would be better for an implementation to 
adapt the estimation gain to the RTT, so the spec ought not to imply 
that one value has to be chosen.
    c) The later paper [ADCTCP10] provides a corrected way to determine 
the gain. From memory (I'm offline at the mo) it at least makes 
throughput inversely proportional to 1/RTT, whereas the approach in 
[DCTCP10] made it inversely proportional to 1/(RTT^2). I believe the 
Linux implementation uses the [ADCTCP10] approach.

    The implementation must also decide when to use DCTCP.
...
    likely in the same datacenter network.

The 2nd & 3rd paras are a mix of implementation and deployment issues, 
but mostly deployment, so they might be better moved to the deployment 
issues section.

CURRENT:

    It is RECOMMENDED that an implementation deal with loss episodes in
    the same way as conventional TCP.
...
    DCTCP implementation MAY also allow configuration of resetting the
    value of DCTCP.Alpha as part of processing any loss episodes.

SUGGESTED: Move this whole para to 3.3 Sender Behaviour section, and:

    An implementation MUST deal with loss episodes in
    the same way as conventional TCP.
...
    DCTCP implementation SHOULD also reset the
    value of DCTCP.Alpha as part of processing any loss episodes.
    This aspect of the behaviour MAY be configurable.

RATIONALE:
    a) There would never be a case for not falling back to conventional 
TCP  (or do you have one?), so "RECOMMENDED" is too weak.
    b) I have tried to interpret what you might have meant about 
resetting Alpha. I'm not sure why you think it might need to be 
configurable though?

    ...it is also RECOMMENDED that an implementation allow
    configuration of restarting the congestion window (cwnd) of idle
    DCTCP connections

Where special implementation measures are necessary for low RTT, these 
should be related to the measured RTT, not hard-coded. There is an 
assumption in a number of places  that "data centre" implies low RTT. Of 
course this is true in a single DC, but I believe the the single admin 
aspect of a DC is what characterises DCTCP, not the low RTT aspect. With 
the modifications in [ADCTCP], DCTCP generalises to a network with 
larger RTTs as long as it is still operated by a single admin.

CURRENT:
    [RFC3168] forbids the ECN-marking of pure ACK packets, because of the
    inability of TCP to mitigate ACK-path congestion and protocol-wise
    preferential treatment by routers.  However, dropping pure ACKs -
    rather than ECN marking them - has disadvantages for typical
    datacenter traffic patterns.  Because of the prevalence of bursty
    traffic patterns that feature transient congestion, dropping of ACKs
    causes subsequent retransmissions. It is RECOMMENDED that an
    implementation provide a configuration knob that will cause ECT to 
be set
    on such control packes, which can be used in environments where such
    concerns do not apply.
SUGGESTED:
    [RFC3168] forbids the ECN-marking of pure ACK packets, because of the
    inability of TCP to mitigate ACK-path congestion and the extra
    advantage to injection attackers that ECN is perceived to offer.
    For the latter reason RFC 3168 also forbids ECN-marking of
    retransmissions, window probes and RSTs.

    However, dropping all these control packets -
    rather than ECN marking them - has considerable performance
    disadvantages. It is RECOMMENDED that an
    implementation provide a configuration knob that will cause ECT to 
be set
    on such control packes, which can be used in environments where such
    concerns do not apply.
NOTE:
    This might not reflect the way your implementation is coded, so you 
might want to make that clear.


*5. Deployment Issues**
*

    If
    the traffic in the datacenter is a mix of conventional TCP and DCTCP,
    it is RECOMMENDED that DCTCP traffic be segregated from conventional
    TCP traffic.

In a data centre (or anywhere), is there any case where not segregating 
them would make sense? If not, this needs to be a MUST, not just a 
RECOMMENDED. I can imagine cases where there is no long-running 
conventional TCP or no long-running DCTCP, but that's covered by the 
"if" at the start of the sentence.

You might want to refer to draft-briscoe-aqm-dualq-coupled as another 
segregation approach here too.

    Since DCTCP relies on congestion marking by the switches, DCTCP can
    only be deployed in datacenters where the entire network
    infrastructure supports ECN.

Surely, the "fall-back to conventional TCP on loss" rule means you can 
delete this constraint.

    A
    variant of DCTCP that can be deployed unilaterally and only requires
    standard ECN behavior has been described in [ODCTCP <https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-01#ref-ODCTCP>][BSDCAN],


Which part of "standard ECN behaviour" do you mean? The host part? The 
network part? The receiver part? And "can be deployed unilaterally" 
ought to be in the active voice not the passive, which would help 
resolve the ambiguity too.

I haven't read [ODCTCP] and [BSDCAN] (I'm offline on a plane at the mo). 
Are they similar to "Instant ECN" in [Wu12] listed below? If not, that 
could be referred to as well for a technique that uses the regular ECN 
host behaviour.

*6. Known Issues**
*
First para could refer to appendix A of RFC7560, which describes the 
problem precisely.

A method for improving the fairness of DCTCP has been proposed
    in [ADCTCP <https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-01#ref-ADCTCP>], but requires additional experimental evaluation.

s/fairness/RTT fairness/

(In our experiments, it's much better than without.)

*11. References**
*
I think the following are cited normatively, not just informatively:
RFC5681, RFC5562.

Additional informational Reference:

[Wu12] Wu, H., Ju, J., Lu, G., Guo, C., Xiong, Y. & Zhang, Y., "Tuning 
ECN for Data Center Networks," In: Proceedings of the 8th International 
Conference on Emerging Networking Experiments and Technologies CoNEXT 
'12 pp.25-36 ACM (2012)

That's it. HTH.


Bob

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


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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=utf-8">
    <title>Review of draft-ietf-tcpm-dctcp-01</title>
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    Praveen &amp; co-authors,<br>
    <br>
    As promised, here's my review of draft-ietf-tcpm-dctcp-01<br>
    <br>
    <u><b>General Comments</b></u><u><b><br>
      </b></u><br>
    In general, in the comments below I have tried to reduce the
    multiple assumptions about controlled environments down to the one
    assumption that they are controlled by a single authority. I believe
    DCTCP can work without the other assumptions, so the following do
    not apply in all controlled environments:<br>
    Â  * not always low RTT (e.g. inter-DC scenarios, or global private
    networks)<br>
    Â  * not always no risk of traffic attacks (e.g. internally arranged
    attacks)<br>
    I may have missed some other occurrences of these assumptions, so
    pls feel free to seek out them all.<br>
    <br>
    Many of the comments argue with your choices of MUST, SHOULD, MAY
    etc. This might seem nit-picking, given the primary purpose is to
    document the algo. However, it would be wrong to require things that
    aren't required or to allow exceptions when exceptions would not be
    interoperable. An alternative approach would be to just remove all
    capitalised language.<br>
    <br>
    <u><b>Section-by-section review.</b></u><u><b><br>
      </b></u><br>
    <b>Abstract</b><b><br>
    </b><br>
    I think it needs to have an applicability statement in the abstract
    (and introduction) that refers to the deployment and implementation
    status sections at the end. Something like:<br>
    <br>
    "This is an informational specification of the implementation of
    DCTCP in Microsoft Windows Server 2012. It is applicable to
    deployments in controlled environments like data centres but it MUST
    NOT be deployed over the public Internet without additional
    measures, as detailed in sections 4-6. "<br>
    <br>
    <b>
      1. Intro</b><b><br>
    </b>s/its many servers/<br>
    Â /their many servers/<br>
    <br>
    "...limited queue capacities..." <br>
    I understand that the motivation has been abbreviated, but I think
    this has lost too much. Perhaps mention memory shared between
    interfaces, so either bloated buffers or too little, making traffic
    susceptible to packet losses?<br>
    <br>
    <tt>Â Â  [RFC3168] describes a mechanism for using Explicit Congestion</tt><tt><br>
    </tt><tt>Â Â  Notification (ECN) from the switches for early detection
      of</tt><tt><br>
    </tt><tt>Â Â  congestion, rather than waiting for packet loss to
      occur.</tt><tt><br>
    </tt><br>
    This isn't correct. 3168 requires ECN marking to occur no earlier
    than drop. It is precisely that 3168 does not allow early marking
    unless there is also early drop that is the problem.<br>
    <br>
    <tt>Â Â  It is recommended that DCTCP be deployed...</tt><tt><br>
    </tt><br>
    "recommended" is too weak. I suggest you use something like the
    applicability text given above, both here and in the abstract.<br>
    <br>
    <b>2. Terminology</b><b><br>
    </b><br>
    Suggest you explain that capitalised words are not quite the same as
    in most RFCs:<br>
    "Normative language is used to describe how necessary the various
    aspects of the Microsoft implementation are for interoperability,
    but even compliant implementations without the measures in sections
    4-6 would still only be safe to deploy in controlled environments."<br>
    <br>
    <b>3. DCTCP Algo</b><b><br>
    </b><br>
    The three bullets say no more than "this is normal stuff". Would it
    be better to supplement each sentence with a brief phrase saying
    what is different about each aspect compared to RFC3168?<br>
    <br>
    <b>3.1 Marking</b><b><br>
    </b><br>
    This needs to say something about how you mark the IP header from
    L2. I suggest you refer to the up-and-forward mode of
    draft-ietf-tsvwg-ecn-encap.<br>
    <br>
    s/sending rate/<br>
    Â /link rate/<br>
    <br>
    <b>3.2 </b><b><br>
    </b><br>
    CURRENT:<br>
    <pre class="newpage">   1.  If the CE codepoint is set and DCTCP.CE is false, send an ACK for
       any previously unacknowledged packets and set DCTCP.CE to true.

   2.  If the CE codepoint is not set and DCTCP.CE is true, send an ACK
       for any previously unacknowledged packets and set DCTCP.CE to
       false.
</pre>
    SUGGESTED:<br>
    <pre class="newpage">   1.  If the CE codepoint is set and DCTCP.CE is false, set DCTCP.CE to 
       true and send an ACK for any previously unacknowledged packets.

   2.  If the CE codepoint is not set and DCTCP.CE is true, set DCTCP.CE
       to false and send an ACK for any previously unacknowledged packets.
</pre>
    Â Â Â  The figure would have to be changed to be consistent too.<br>
    RATIONALE:<br>
    If the receiver changes DCTCP.CE /after/ sending the ACK like this,
    it delays each signal until the following ACK is sent. That could
    add a delay of anything from 1 to m packets. I've never understood
    why the order of these operations was specified in this way. If
    there is no reason, then I suggest that it is specified in the
    opposite order, as I describe above. <br>
    <br>
    A note could be added to say the Windows Server 12 implementation
    does these steps in reverse order, but the order specified is
    preferred because it reduces signalling delay and has no
    interoperability issues with the original Windows implementation. <br>
    <br>
    <pre class="newpage">   The handling of the "Congestion Window Reduced" (CWR) bit is also
   exactly as per [<a href="https://tools.ietf.org/html/rfc3168" title="&quot;The Addition of Explicit Congestion Notification (ECN) to IP&quot;">RFC3168</a>] including [<a href="https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-01#ref-RFC3168-ERRATA3639">RFC3168-ERRATA3639</a>].  That is, on
   receipt of a segment with both the CE and CWR bits set, CWR is
   processed first and then ECE is processed.
</pre>
    Even tho this is the receiver section, I suggest:<br>
    s/The handling/Receiver handling/<br>
    <br>
    Whatever, this cannot be correct. A DCTCP receiver surely does
    nothing on receipt of CWR, whereas an RFC3168 receiver does
    something. A DCTCP receiver certainly cannot do exactly what RFC3168
    says, because that says stop setting ECE on receipt of CWR until the
    next CE arrives, which would surely break this feedback protocol.<br>
    <br>
    <b>3.3 </b><b><br>
    </b>
    <pre class="newpage">   DCTCP.Alpha, which is initialized to 1 and MUST be updated
   as follows:</pre>
    s/MUST/SHOULD/<br>
    And add "An alternative congestion avoidance algorithm MAY be used,
    but it MUST lead to flow rate inversely proportional to 1/M".<br>
    <br>
    Rationale: There are many ways to implement interoperable 1/p cc.
    This is just one. <br>
    <pre class="newpage">   DCTCP.BytesSent</pre>
    This variable actually counts bytes received. it's pretty confusing
    to call it BytesSent.<br>
    <br>
    s/TCP SACK options [RFC2018] are ignored./<br>
    Â /The sender MUST ignore TCP SACK options [RFC2018] for this
    computation./<br>
    RATIONALE: <br>
    Â  a) I think the idea is to ignore CE marks after a gap because, if
    the gap becomes considered a loss, the response to loss will
    override any need to count incoming CE marks for the following round
    trip. However, if the gap is filled before it is considered a loss
    {Note 1}, then CE marks that were ignored because they arrived after
    a gap will need to be "unignored". <br>
    Â  b) We need to make clear that SACK is not ignored altogether, only
    for this computation.<br>
    <br>
    {Note 1}: I can think of two cases:<br>
    * Less than three dupacks then the gap is filled<br>
    * The gap is filled by a delayed segment so that an earlier
    retransmission was spurious<br>
    <br>
    <pre class="newpage">the sender MUST update cwnd as follows:
      cwnd = cwnd * (1 - DCTCP.Alpha / 2)
</pre>
    s/MUST/SHOULD/<br>
    Rationale: Alternative interoperable response functions are
    possible. For instance an algorithm like Relentless TCP that simply
    reduces cwnd by half the size of any CE-marked segment. Then there
    is no Alpha and no g variable.<br>
    <br>
    CURRENT:<br>
    <pre class="newpage">   Just as specified in [<a href="https://tools.ietf.org/html/rfc3168" title="&quot;The Addition of Explicit Congestion Notification (ECN) to IP&quot;">RFC3168</a>], TCP should not react to congestion
   indications more than once for every window of data.</pre>
    SUGGESTED:<br>
    <pre class="newpage">   Just as specified in [<a href="https://tools.ietf.org/html/rfc3168" title="&quot;The Addition of Explicit Congestion Notification (ECN) to IP&quot;">RFC3168</a>], DCTCP does not react to congestion
   indications more than once for every window of data.</pre>
    I don't know whether you intended this to be an upper-case SHOULD
    NOT. If so, I would argue against this being normative, because it
    isn't necessary for interop; therefore I've suggested avoiding he
    'should' word.<br>
    <pre class="newpage">   The setting of
   the "Congestion Window Reduced" (CWR) bit is also exactly as per
   [<a href="https://tools.ietf.org/html/rfc3168" title="&quot;The Addition of Explicit Congestion Notification (ECN) to IP&quot;">RFC3168</a>].

</pre>
    It is wrong to say "exactly". The sender certainly sets CWR when it
    does a congestion response. But in DCTCP the congestion response
    occurs every RTT, triggered by its own measurement-period logic,
    whereas in RFC3168 it is triggered by the arrival of an ECE after
    the ACK of any previous segment it sent with CWR set.<br>
    <br>
    I think it is worth saying that setting CWR is necessary for safety,
    otherwise in the case where there has been a configuration
    management error and the receiver complies with RFC 3168 not DCTCP,
    it will continue sending ECE for ever and starve the session.<br>
    <br>
    It might help to explicitly confirm that: <br>
    "Other aspects of TCP congestion avoidance and control such as the
    slow start phase are no different to those in RFC5681."<br>
    <br>
    <b>3.4 SYN, SYN-ACK, RST</b><b><br>
    </b><br>
    <tt>Â Â  [RFC3168] requires that a compliant TCP MUST NOT set ECT on
      SYN or</tt><tt><br>
    </tt><tt>Â Â  SYN-ACK packets.</tt><tt><br>
    </tt><br>
    s/MUST NOT/'MUST NOT'/<br>
    Rationale: Best to make it clear this is quoting a normative
    statement in another RFC, because it is going to be contradicted.<br>
    <br>
    CURRENT<br>
    <pre class="newpage">   These RFCs, however, are
   intended for general Internet use, and do not directly apply to a
   controlled datacenter environment.
...
   For DCTCP connections, the sender
   SHOULD set ECT for SYN, SYN-ACK and RST packets.
</pre>
    SUGGESTED:<br>
    <br>
    <tt>Â Â  Also, a SYN is sent before the ECN-capability has been<br>
      Â Â  negotiated, so if it is CE-marked, a non-ECN TCP server would
      not<br>
      Â Â  recognise the marking. RFC 3168 does not specify the ECN field
      on <br>
      Â Â  a RST. The security concerns addressed by these RFCs <br>
      Â Â  might not apply in controlled environments like data centres,
      and <br>
      Â Â  it might not be necessary to cater for the presence of non-ECN
      <br>
      Â Â  servers.<br>
      ...</tt><br>
    <pre class="newpage">   Therefore, the sender MUST set ECT for SYN/ACK and RST packets and a 
   configuration option SHOULD be provided to set ECT for SYNs.
</pre>
    RATIONALE:<br>
    Â Â  The draft should avoid assuming that security concerns do not
    apply in all controlled environments.<br>
    NOTE:<br>
    Â Â  This might not reflect the way your implementation is coded, so
    you might want to make that clear.<br>
    <br>
    <b>4. Implementation Issues</b><b><br>
    </b><br>
    CURRENT:<br>
    <pre class="newpage">   the implementation MUST choose a suitable
   estimation gain.  [<a href="https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-01#ref-DCTCP10" title="&quot;Data Center TCP (DCTCP)&quot;">DCTCP10</a>] provides a theoretical basis </pre>
    SUGGEST:<br>
    <pre class="newpage">   the implementation will need to choose a suitable
   estimation gain.  [<a href="https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-01#ref-DCTCP10" title="&quot;Data Center TCP (DCTCP)&quot;">ADCTCP10</a>] provides a theoretical basis ...</pre>
    RATIONALE:<br>
    Â Â  a) As above, this is only one way to implement the cc, and others
    could well prove to be interoperable when tested.<br>
    Â Â  b) More particularly, it would be better for an implementation to
    adapt the estimation gain to the RTT, so the spec ought not to imply
    that one value has to be chosen.<br>
    Â Â  c) The later paper [ADCTCP10] provides a corrected way to
    determine the gain. From memory (I'm offline at the mo) it at least
    makes throughput inversely proportional to 1/RTT, whereas the
    approach in [DCTCP10] made it inversely proportional to 1/(RTT^2). I
    believe the Linux implementation uses the [ADCTCP10] approach.<br>
    <br>
    <pre class="newpage">   The implementation must also decide when to use DCTCP.
...
   likely in the same datacenter network.</pre>
    The 2nd &amp; 3rd paras are a mix of implementation and deployment
    issues, but mostly deployment, so they might be better moved to the
    deployment issues section.<br>
    <br>
    CURRENT:<br>
    <pre class="newpage">   It is RECOMMENDED that an implementation deal with loss episodes in
   the same way as conventional TCP.
...
  Â DCTCP implementation MAY also allow configuration of resetting the
   value of DCTCP.Alpha as part of processing any loss episodes.
</pre>
    SUGGESTED: Move this whole para to 3.3 Sender Behaviour section,
    and:<br>
    <pre class="newpage">   An implementation MUST deal with loss episodes in
   the same way as conventional TCP.
...
  Â DCTCP implementation SHOULD also reset the
   value of DCTCP.Alpha as part of processing any loss episodes. 
   This aspect of the behaviour MAY be configurable.</pre>
    RATIONALE:<br>
    Â Â  a) There would never be a case for not falling back to
    conventional TCPÂ  (or do you have one?), so "RECOMMENDED" is too
    weak.<br>
    Â Â  b) I have tried to interpret what you might have meant about
    resetting Alpha. I'm not sure why you think it might need to be
    configurable though?<br>
    <br>
    <pre class="newpage">   ...it is also RECOMMENDED that an implementation allow
   configuration of restarting the congestion window (cwnd) of idle
   DCTCP connections</pre>
    Where special implementation measures are necessary for low RTT,
    these should be related to the measured RTT, not hard-coded. There
    is an assumption in a number of placesÂ  that "data centre" implies
    low RTT. Of course this is true in a single DC, but I believe the
    the single admin aspect of a DC is what characterises DCTCP, not the
    low RTT aspect. With the modifications in [ADCTCP], DCTCP
    generalises to a network with larger RTTs as long as it is stillÂ 
    operated by a single admin. <br>
    <br>
    CURRENT:<br>
    <tt>Â Â  [RFC3168] forbids the ECN-marking of pure ACK packets,
      because of the</tt><tt><br>
    </tt><tt>Â Â  inability of TCP to mitigate ACK-path congestion and
      protocol-wise</tt><tt><br>
    </tt><tt>Â Â  preferential treatment by routers.Â  However, dropping
      pure ACKs -</tt><tt><br>
    </tt><tt>Â Â  rather than ECN marking them - has disadvantages for
      typical</tt><tt><br>
    </tt><tt>Â Â  datacenter traffic patterns.Â  Because of the prevalence
      of bursty</tt><tt><br>
    </tt><tt>Â Â  traffic patterns that feature transient congestion,
      dropping of ACKs</tt><tt><br>
    </tt><tt>Â Â  causes subsequent retransmissions. It is RECOMMENDED
      that an</tt><tt><br>
    </tt><tt>Â Â  implementation provide a configuration knob that will
      cause ECT to be set</tt><tt><br>
    </tt><tt>Â Â  on such control packes, which can be used in
      environments where such </tt><tt><br>
    </tt><tt>Â Â  concerns do not apply.</tt><tt><br>
    </tt>SUGGESTED:<br>
    <tt>Â Â  [RFC3168] forbids the ECN-marking of pure ACK packets,
      because of the</tt><tt><br>
    </tt><tt>Â Â  inability of TCP to mitigate ACK-path congestion and the
    </tt><tt>extra <br>
      Â Â  advantage to injection attackers that ECN is perceived to
      offer. <br>
      Â Â  For the latter reason RFC 3168 also forbids ECN-marking of <br>
      Â Â  retransmissions, window probes and RSTs. <br>
      <br>
      Â Â  However, dropping all these control packets -</tt><tt><br>
    </tt><tt>Â Â  rather than ECN marking them - has considerable
      performance <br>
      Â Â  disadvantages</tt><tt>.Â  </tt><tt>It is RECOMMENDED that an</tt><tt><br>
    </tt><tt>Â Â  implementation provide a configuration knob that will
      cause ECT to be set</tt><tt><br>
    </tt><tt>Â Â  on such control packes, which can be used in
      environments where such </tt><tt><br>
    </tt><tt>Â Â  concerns do not apply.</tt><br>
    NOTE:<br>
    Â Â  This might not reflect the way your implementation is coded, so
    you might want to make that clear.<br>
    <br>
    <br>
    <b>5. Deployment Issues</b><b><br>
    </b>
    <pre class="newpage">   If
   the traffic in the datacenter is a mix of conventional TCP and DCTCP,
  Â it is RECOMMENDED that DCTCP traffic be segregated from conventional
   TCP traffic.</pre>
    In a data centre (or anywhere), is there any case where not
    segregating them would make sense? If not, this needs to be a MUST,
    not just a RECOMMENDED. I can imagine cases where there is no
    long-running conventional TCP or no long-running DCTCP, but that's
    covered by the "if" at the start of the sentence.<br>
    <br>
    You might want to refer to draft-briscoe-aqm-dualq-coupled as
    another segregation approach here too.<br>
    <br>
    <pre class="newpage">   Since DCTCP relies on congestion marking by the switches, DCTCP can
   only be deployed in datacenters where the entire network
   infrastructure supports ECN.</pre>
    Surely, the "fall-back to conventional TCP on loss" rule means you
    can delete this constraint.<br>
    <br>
    <pre class="newpage">   A
   variant of DCTCP that can be deployed unilaterally and only requires
   standard ECN behavior has been described in [<a href="https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-01#ref-ODCTCP" title="&quot;Improving Transmission Performance with One- Sided Datacenter TCP&quot;">ODCTCP</a>][BSDCAN],</pre>
    <br>
    Which part of "standard ECN behaviour" do you mean? The host part?
    The network part? The receiver part? And "can be deployed
    unilaterally" ought to be in the active voice not the passive, which
    would help resolve the ambiguity too.<br>
    <br>
    I haven't read [ODCTCP] and [BSDCAN] (I'm offline on a plane at the
    mo). Are they similar to "Instant ECN" in [Wu12] listed below? If
    not, that could be referred to as well for a technique that uses the
    regular ECN host behaviour.<br>
    <br>
    <b>6. Known Issues</b><b><br>
    </b><br>
    First para could refer to appendix A of RFC7560, which describes the
    problem precisely.<br>
    <pre class="newpage">A method for improving the fairness of DCTCP has been proposed
   in [<a href="https://tools.ietf.org/html/draft-ietf-tcpm-dctcp-01#ref-ADCTCP" title="&quot;Analysis of DCTCP: Stability, Convergence, and Fairness&quot;">ADCTCP</a>], but requires additional experimental evaluation.</pre>
    s/fairness/RTT fairness/<br>
    <br>
    (In our experiments, it's much better than without.)<br>
    <br>
    <b>11. References</b><b><br>
    </b><br>
    I think the following are cited normatively, not just informatively:<br>
    RFC5681, RFC5562.<br>
    <br>
    Additional informational Reference:<br>
    <br>
    [Wu12] Wu, H., Ju, J., Lu, G., Guo, C., Xiong, Y. &amp; Zhang, Y.,
    "Tuning ECN for Data Center Networks," In: Proceedings of the 8th
    International Conference on Emerging Networking Experiments and
    Technologies CoNEXT '12 pp.25-36 ACM (2012)<br>
    <br>
    That's it. HTH.<br>
    <br>
    <br>
    Bob<br>
    <pre class="moz-signature" cols="72">-- 
________________________________________________________________
Bob Briscoe                               <a class="moz-txt-link-freetext" href="http://bobbriscoe.net/">http://bobbriscoe.net/</a></pre>
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From nobody Fri Nov  6 11:05:40 2015
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] A clarification about draft-welzl-tcpm-tcb-sharing
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FWIW, we also need to differentiate the impact of these issues on
temporal vs. ensemble sharing.

On 11/5/2015 11:57 PM, Michael Welzl wrote:
> 
>> On 06 Nov 2015, at 05:58, Yuchung Cheng <ycheng@google.com> wrote:
>>
>> On Thu, Nov 5, 2015 at 1:50 PM, Joe Touch <touch@isi.edu> wrote:
>>>
>>>
>>>
>>> On 11/5/2015 10:54 AM, Yuchung Cheng wrote:
>>> ...
>>>> Externally: IP no longer serves a good index. e.g., DHCP, NAT.
>>>
>>> That's not necessarily true. In both cases, the machines that might
>>> share an IP address often share the dominant aspects of their network
>>> paths - delay, loss, max bandwidth, etc.
>>>
>>> Additionally, the shared information is used only for connection
>>> initialization. That shared information doesn't have to be accurate,
>>> only somewhat better than the current default initial values.
>> Intuitively yes. But our tests show little difference in terms of
>> latency, loss rate, etc with or without the Linux IP metrics cache.
>> And disabling not only save cache write/read but also skip the pathological
>> cases.
>>
>> Today a phone can have multiple public IP addresses and an IP address
>> can be used for multiple phones simultaneously (with different ports).
>> The RTT, link
>> quality, and BW can all be very different. This makes IP a less
>> reliable index for server cache or CB sharing even only for seeding phase.
>>
>> I am not claiming it's universally useless. I am presenting a data
>> point primarily on servers.
> 
> Sounds like at least some text explaining these potential problems with relying on the IP address is needed for the planned RFC2140bis. Which I think is quite appropriate. Thanks for the data point!
> 
> Cheers,
> Michael
> 


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From: Yuchung Cheng <ycheng@google.com>
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Subject: Re: [tcpm] A clarification about draft-welzl-tcpm-tcb-sharing
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On Fri, Nov 6, 2015 at 11:04 AM, Joe Touch <touch@isi.edu> wrote:

> FWIW, we also need to differentiate the impact of these issues on
> temporal vs. ensemble sharing.
>
I also think it's more useful to analyze the impact of existing
implementations, instead of documenting code. the metrics cache isn't hard
to understand at all in Linux and the code changes over time. What we
document today can be obsolete tomorrow, so in the end one still need to
read the code to get the latest info. In other word I think this draft is a
waste of time honestly speaking.


>
> On 11/5/2015 11:57 PM, Michael Welzl wrote:
> >
> >> On 06 Nov 2015, at 05:58, Yuchung Cheng <ycheng@google.com> wrote:
> >>
> >> On Thu, Nov 5, 2015 at 1:50 PM, Joe Touch <touch@isi.edu> wrote:
> >>>
> >>>
> >>>
> >>> On 11/5/2015 10:54 AM, Yuchung Cheng wrote:
> >>> ...
> >>>> Externally: IP no longer serves a good index. e.g., DHCP, NAT.
> >>>
> >>> That's not necessarily true. In both cases, the machines that might
> >>> share an IP address often share the dominant aspects of their network
> >>> paths - delay, loss, max bandwidth, etc.
> >>>
> >>> Additionally, the shared information is used only for connection
> >>> initialization. That shared information doesn't have to be accurate,
> >>> only somewhat better than the current default initial values.
> >> Intuitively yes. But our tests show little difference in terms of
> >> latency, loss rate, etc with or without the Linux IP metrics cache.
> >> And disabling not only save cache write/read but also skip the
> pathological
> >> cases.
> >>
> >> Today a phone can have multiple public IP addresses and an IP address
> >> can be used for multiple phones simultaneously (with different ports).
> >> The RTT, link
> >> quality, and BW can all be very different. This makes IP a less
> >> reliable index for server cache or CB sharing even only for seeding
> phase.
> >>
> >> I am not claiming it's universally useless. I am presenting a data
> >> point primarily on servers.
> >
> > Sounds like at least some text explaining these potential problems with
> relying on the IP address is needed for the planned RFC2140bis. Which I
> think is quite appropriate. Thanks for the data point!
> >
> > Cheers,
> > Michael
> >
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Fri, Nov 6, 2015 at 11:04 AM, Joe Touch <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:touch@isi.edu" target=3D"_blank">touch@isi.edu</a>&gt;</span> w=
rote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex">FWIW, we also need to differentiate=
 the impact of these issues on<br>
temporal vs. ensemble sharing.<br></blockquote><div>I also think it&#39;s m=
ore useful to analyze the impact of existing implementations, instead of do=
cumenting code. the metrics cache isn&#39;t hard to understand at all in Li=
nux and the code changes over time. What we document today can be obsolete =
tomorrow, so in the end one still need to read the code to get the latest i=
nfo. In other word I think this draft is a waste of time honestly speaking.=
</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 =
0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class=3D"HOEnZb"><div class=3D"h5"><br>
On 11/5/2015 11:57 PM, Michael Welzl wrote:<br>
&gt;<br>
&gt;&gt; On 06 Nov 2015, at 05:58, Yuchung Cheng &lt;<a href=3D"mailto:yche=
ng@google.com">ycheng@google.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; On Thu, Nov 5, 2015 at 1:50 PM, Joe Touch &lt;<a href=3D"mailto:to=
uch@isi.edu">touch@isi.edu</a>&gt; wrote:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; On 11/5/2015 10:54 AM, Yuchung Cheng wrote:<br>
&gt;&gt;&gt; ...<br>
&gt;&gt;&gt;&gt; Externally: IP no longer serves a good index. e.g., DHCP, =
NAT.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; That&#39;s not necessarily true. In both cases, the machines t=
hat might<br>
&gt;&gt;&gt; share an IP address often share the dominant aspects of their =
network<br>
&gt;&gt;&gt; paths - delay, loss, max bandwidth, etc.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Additionally, the shared information is used only for connecti=
on<br>
&gt;&gt;&gt; initialization. That shared information doesn&#39;t have to be=
 accurate,<br>
&gt;&gt;&gt; only somewhat better than the current default initial values.<=
br>
&gt;&gt; Intuitively yes. But our tests show little difference in terms of<=
br>
&gt;&gt; latency, loss rate, etc with or without the Linux IP metrics cache=
.<br>
&gt;&gt; And disabling not only save cache write/read but also skip the pat=
hological<br>
&gt;&gt; cases.<br>
&gt;&gt;<br>
&gt;&gt; Today a phone can have multiple public IP addresses and an IP addr=
ess<br>
&gt;&gt; can be used for multiple phones simultaneously (with different por=
ts).<br>
&gt;&gt; The RTT, link<br>
&gt;&gt; quality, and BW can all be very different. This makes IP a less<br=
>
&gt;&gt; reliable index for server cache or CB sharing even only for seedin=
g phase.<br>
&gt;&gt;<br>
&gt;&gt; I am not claiming it&#39;s universally useless. I am presenting a =
data<br>
&gt;&gt; point primarily on servers.<br>
&gt;<br>
&gt; Sounds like at least some text explaining these potential problems wit=
h relying on the IP address is needed for the planned RFC2140bis. Which I t=
hink is quite appropriate. Thanks for the data point!<br>
&gt;<br>
&gt; Cheers,<br>
&gt; Michael<br>
&gt;<br>
</div></div></blockquote></div><br></div></div>

--001a113ef4be1c73e20523e45359--


From nobody Fri Nov  6 12:16:53 2015
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From: Joe Touch <touch@isi.edu>
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Cc: "tcpm@ietf.org Extensions" <tcpm@ietf.org>, Michael Welzl <michawe@ifi.uio.no>, touch@isi.edu
Subject: Re: [tcpm] A clarification about draft-welzl-tcpm-tcb-sharing
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On 11/6/2015 11:33 AM, Yuchung Cheng wrote:
> 
> 
> On Fri, Nov 6, 2015 at 11:04 AM, Joe Touch <touch@isi.edu
> <mailto:touch@isi.edu>> wrote:
> 
>     FWIW, we also need to differentiate the impact of these issues on
>     temporal vs. ensemble sharing.
> 
> I also think it's more useful to analyze the impact of existing
> implementations, instead of documenting code.

Right; this is the IETF, and code documentation isn't the goal.

> the metrics cache isn't
> hard to understand at all in Linux and the code changes over time. What
> we document today can be obsolete tomorrow, so in the end one still need
> to read the code to get the latest info. In other word I think this
> draft is a waste of time honestly speaking.

The draft is the starting point; it is intended to gather information
about the impact of existing implementations, exactly as you describe.

Joe

>  
> 
> 
>     On 11/5/2015 11:57 PM, Michael Welzl wrote:
>     >
>     >> On 06 Nov 2015, at 05:58, Yuchung Cheng <ycheng@google.com
>     <mailto:ycheng@google.com>> wrote:
>     >>
>     >> On Thu, Nov 5, 2015 at 1:50 PM, Joe Touch <touch@isi.edu
>     <mailto:touch@isi.edu>> wrote:
>     >>>
>     >>>
>     >>>
>     >>> On 11/5/2015 10:54 AM, Yuchung Cheng wrote:
>     >>> ...
>     >>>> Externally: IP no longer serves a good index. e.g., DHCP, NAT.
>     >>>
>     >>> That's not necessarily true. In both cases, the machines that might
>     >>> share an IP address often share the dominant aspects of their
>     network
>     >>> paths - delay, loss, max bandwidth, etc.
>     >>>
>     >>> Additionally, the shared information is used only for connection
>     >>> initialization. That shared information doesn't have to be accurate,
>     >>> only somewhat better than the current default initial values.
>     >> Intuitively yes. But our tests show little difference in terms of
>     >> latency, loss rate, etc with or without the Linux IP metrics cache.
>     >> And disabling not only save cache write/read but also skip the
>     pathological
>     >> cases.
>     >>
>     >> Today a phone can have multiple public IP addresses and an IP address
>     >> can be used for multiple phones simultaneously (with different
>     ports).
>     >> The RTT, link
>     >> quality, and BW can all be very different. This makes IP a less
>     >> reliable index for server cache or CB sharing even only for
>     seeding phase.
>     >>
>     >> I am not claiming it's universally useless. I am presenting a data
>     >> point primarily on servers.
>     >
>     > Sounds like at least some text explaining these potential problems
>     with relying on the IP address is needed for the planned RFC2140bis.
>     Which I think is quite appropriate. Thanks for the data point!
>     >
>     > Cheers,
>     > Michael
>     >
> 
> 


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--_000_81564C0D7D4D2A4B9A86C8C7404A13DA34EB59DEESESSMB205erics_--


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From: Karen Elisabeth Egede Nielsen <karen.nielsen@tieto.com>
References: <55C4C4CB.9080302@bobbriscoe.net> <55C4CEEF.1040504@bobbriscoe.net> <338a11745e05dd944c66be76fb925e97@mail.gmail.com> <55D36138.5060106@bobbriscoe.net> <7A2801D5E40DD64A85E38DF22117852C70B38246@wdc1exchmbxp01.hq.corp.viasat.com> <8A5507EB-07A2-46E4-8EA5-E60C1845B131@netapp.com> <CAK6E8=dW4qhP+wmR_YGzL_eV6oheSdneoXtyoXzSMZC4xRQg8A@mail.gmail.com> <7A2801D5E40DD64A85E38DF22117852C70B38ECD@wdc1exchmbxp01.hq.corp.viasat.com> <CAK6E8=cPGiRHw+cQ-jEdwPp+SgsqBUHS_EcYRHy3_WTiZS=R=Q@mail.gmail.com>
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Date: Wed, 11 Nov 2015 16:08:39 +0100
Message-ID: <814ca62506a894c067efd7161e46d0ee@mail.gmail.com>
To: Yuchung Cheng <ycheng@google.com>, "Eggert, Lars" <lars@netapp.com>,  "Zimmermann, Alexander" <Alexander.Zimmermann@netapp.com>, Jana Iyengar <jri@google.com>
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Cc: tcpm IETF list <tcpm@ietf.org>
Subject: Re: [tcpm] Cubic (was Re: New I-D: draft-briscoe-aqm-dualq-coupled-00.txt)
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HI,

I (also) was a bit surprised by the CUBIC fault presented in tcpm as I
would have assumed that CUBIC would be implemented in the framework of
RFC5681 for how to handle idle situations and also possibly in the
framework of RFC7661/2861
(or this mechanism in which ever from it takes in Linux).
Anyway I think that it would be great and best if the CUBIC paper would
describe a CUBIC solution
following these principles, but then I also see that we might derivate
from existing implementations,
so that is of course an issue (Though see below on pacing).

In terms of how to handle CUBIC after idle situation or other where the
CUBIC algorithm need
to be re-calibrated, then I wonder if the situation is so simply that one
would re-calibrate the
CUBIC W(t) formula so that t=3D0 corresponds to re-start of transmission an=
d
K is adjusted so
That W(0) gives the present CWND (whatever it has been left at now) - just
a na=C4=ABve thought ?
Perhaps you have already designed a different solution ?

Something else - more general:

* at the tcpm session, Lars mentioned that also Hystart should be
described in this document. Is that correctly understood ?
Personally I think that sounds very good.

* Pacing has been recognized as a necessity and I believe even more a
necessity when running CUBIC than New Reno,
as CUBIC CWND increase at large RTTs may clock out large bursts. Is it
valid to describe CUBIC CC (as implemented in Linux) without describing
pacing ?

Thanks.

BR, Karen



> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Yuchung Cheng
> Sent: 29. september 2015 01:28
> To: Agarwal, Anil <Anil.Agarwal@viasat.com>
> Cc: Jana Iyengar <jri@google.com>; tcpm IETF list <tcpm@ietf.org>
> Subject: Re: [tcpm] Cubic (was Re: New I-D: draft-briscoe-aqm-dualq-
> coupled-00.txt)
>
> On Mon, Sep 28, 2015 at 3:35 PM, Agarwal, Anil <Anil.Agarwal@viasat.com>
> wrote:
> > Yuchung,
> >
> > You wrote -
> > "If the TCP implementation follows RFC 5681 or RFC 2861 to reduce cwnd
> after idle, the issue is masqueraded".
> >
> > Would it be more appropriate to state that - If the TCP implementation
> > follows RFC 5681 or RFC 2861 to reduce cwnd after idle, the TCP
slow-start
> cwnd increase algorithm should get used, not CUBIC.
> That'd be fine w/ me if the draft considers Cubic is only effective when
cwnd
> >=3D ssthresh.
>
> Jana would present the troubleshooting and results in the next mtg.
> the bug was initially discovered in QUIC (which uses Cubic).
>
> >
> > Thanks,
> > Anil
> >
> > P.S. Sorry about the confusion created by not editing the subject line
> > correctly
> >
> >
> > -----Original Message-----
> > From: Yuchung Cheng [mailto:ycheng@google.com]
> > Sent: Monday, September 28, 2015 4:23 PM
> > To: Eggert, Lars
> > Cc: Agarwal, Anil; tcpm IETF list; Zimmermann, Alexander
> > Subject: Re: [tcpm] Cubic (was Re: New I-D:
> > draft-briscoe-aqm-dualq-coupled-00.txt)
> >
> > On Mon, Sep 28, 2015 at 6:34 AM, Eggert, Lars <lars@netapp.com> wrote:
> >>
> >> Hi,
> >>
> >> On 2015-09-28, at 15:28, Agarwal, Anil <Anil.Agarwal@viasat.com>
wrote:
> >> >
> >> > Question - where is the CUBIC specification documented? How will it
be
> updated? Is there an RFC on CUBIC?
> >>
> >> there is https://urldefense.proofpoint.com/v2/url?u=3Dhttps-
> 3A__datatracker.ietf.org_doc_draft-2Dietf-2Dtcpm-
> 2Dcubic_&d=3DBQIBaQ&c=3Djcv3orpCsv7C4ly8-
> ubDob57ycZ4jvhoYZNDBA06fPk&r=3DFyvaklKYrHaSCPjbBTdviWIW9uSbnxdNSh
> eSGz1Jvq4&m=3DgpOIxBF_PNjCo7ij5xixz0eqOB-
> Xt_ddmiwbeQmW7lI&s=3Db18CZjGX71Lsmel9s4E-8R8aFY1-
> 71uvwem3J6QnMOg&e=3D . The goal was that this would document the Cubic
> code currently implemented by Linux.
> >>
> >> This has been a slow effort, and it's getting slower since both
> >> Richard and Alex probably won't have cycles in the foreseeable future
> >> to work on this. So I think we're looking for someone else to take
> >> over the pen (hi, Yuchung & Google folks :-)
> >>
> > Sorry I won't be able to help co-authoring the draft. But the fix is
not that
> complicated. A new subsection on section 3 should suffice.
> > Something like:
> >
> > 3.x Dealing with application idle
> >
> > Cubic determines how fast to grow the window based on how long ago the
> last loss event was. An oversight in the original design is that idling
periods
> can make Cubic arbitrarily aggressive: after an idle period, the delta
of (t - k)
> as well as the slope of cwnd increase become arbitrarily large in
Section 3.1.
> In some implementation (e.g., Linux) the maximum increase is capped to
be
> as fast as slow start.
> >
> > If the TCP implementation follows RFC 5681 or RFC 2861 to reduce cwnd
> after idle, the issue is masqueraded. But if the TCP implementation does
not
> reduce cwnd after idle (i.e., RFC2861 or RFC5681), Cubic will slow start
> without a reduced cwnd and likely to induce losses quickly.
> >
> > To address this issue Cubic algorithm SHOULD offset the (t - k) in
Section 3.1
> by the amount of idle time. One approach (e.g., Linux) is to record the
send
> time of the last packet, and calculate the idle time when the amount of
> inflight becomes positive again when sending the next packet after idle.
> >
> >
> > These are the Linux patches for reference.
> >
> > https://urldefense.proofpoint.com/v2/url?u=3Dhttps-
> 3A__patchwork.ozlabs.
> > org_patch_518918_&d=3DBQIBaQ&c=3Djcv3orpCsv7C4ly8-
> ubDob57ycZ4jvhoYZNDBA06f
> >
> Pk&r=3DFyvaklKYrHaSCPjbBTdviWIW9uSbnxdNSheSGz1Jvq4&m=3DgpOIxBF_PNj
> Co7ij5xi
> > xz0eqOB-Xt_ddmiwbeQmW7lI&s=3D-
> 40xAglDVsZhZJ3Cd5PncmRKZue4vz4V5sA3iTqYafo
> > &e=3D
> > https://urldefense.proofpoint.com/v2/url?u=3Dhttps-
> 3A__patchwork.ozlabs.
> > org_patch_516122_&d=3DBQIBaQ&c=3Djcv3orpCsv7C4ly8-
> ubDob57ycZ4jvhoYZNDBA06f
> >
> Pk&r=3DFyvaklKYrHaSCPjbBTdviWIW9uSbnxdNSheSGz1Jvq4&m=3DgpOIxBF_PNj
> Co7ij5xi
> > xz0eqOB-Xt_ddmiwbeQmW7lI&s=3DaA3H-aoHtcUBnfA-
> 7s85Nu4oxArwhmM2N8tT1fV9uy0
> > &e=3D
> > https://urldefense.proofpoint.com/v2/url?u=3Dhttps-
> 3A__patchwork.ozlabs.
> > org_patch_516123_&d=3DBQIBaQ&c=3Djcv3orpCsv7C4ly8-
> ubDob57ycZ4jvhoYZNDBA06f
> >
> Pk&r=3DFyvaklKYrHaSCPjbBTdviWIW9uSbnxdNSheSGz1Jvq4&m=3DgpOIxBF_PNj
> Co7ij5xi
> > xz0eqOB-Xt_ddmiwbeQmW7lI&s=3DM-8Jum-
> _SDW0fVKKsSHmUto0Cw3DajRYRhxo0cdjOYw
> > &e=3D
> >
> > HTH
> >
> >
> >> Lars
> >>
> >> _______________________________________________
> >> tcpm mailing list
> >> tcpm@ietf.org
> >> https://urldefense.proofpoint.com/v2/url?u=3Dhttps-
> 3A__www.ietf.org_mai
> >> l
> >> man_listinfo_tcpm&d=3DBQIBaQ&c=3Djcv3orpCsv7C4ly8-
> ubDob57ycZ4jvhoYZNDBA06
> >> f
> >>
> Pk&r=3DFyvaklKYrHaSCPjbBTdviWIW9uSbnxdNSheSGz1Jvq4&m=3DgpOIxBF_PNj
> Co7ij5x
> >> i
> >> xz0eqOB-
> Xt_ddmiwbeQmW7lI&s=3DFbVX6NYOKMRF5mIg0Z5WW4mBHmMudm1VvExLt
> mTkfp
> >> c
> >> &e=3D
> >>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Thu Nov 12 12:31:49 2015
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Subject: [tcpm] TCP API in 793 / 793bis
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I checked out the Yokohama meeting minutes and recording (sorry I missed 
it in person!).

We need to follow-up on the discussion of what to do (or not do) about 
the TCP API in RFC 793bis.  I know Karen had interest in what will 
happen to this, for one.

As I see it, the issue raised on the TAPS list earlier is that the TCP 
API in 793 dates from 1981 and hasn't been updated significantly since 
then.  It is an abstract API, whereas operating systems implement 
concrete APIs based on sockets, primarily, and that have diverged from 
the description in 793 (and currently 793bis).

I think our options are:
1. Leave it alone, modulo any corrections from errata, other RFCs, etc.
2. Take it out
3. Change it in more significant ways

I don't think option 3 is somewhere we should go.  I think that should 
be a separate effort, if there is desire to do that.  My inclination is 
for option 1.

It will be good to hear what others think, and understand the consensus 
on this though.


From nobody Thu Nov 12 13:05:11 2015
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To: Wesley Eddy <wes@mti-systems.com>
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Subject: Re: [tcpm] TCP API in 793 / 793bis
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I think option 1 makes sense. I wouldn't be opposed strongly to option 2,
but the doc seems clear on the "advisory" nature of this API.

On Thu, Nov 12, 2015 at 12:31 PM, Wesley Eddy <wes@mti-systems.com> wrote:

> I checked out the Yokohama meeting minutes and recording (sorry I missed
> it in person!).
>
> We need to follow-up on the discussion of what to do (or not do) about the
> TCP API in RFC 793bis.  I know Karen had interest in what will happen to
> this, for one.
>
> As I see it, the issue raised on the TAPS list earlier is that the TCP API
> in 793 dates from 1981 and hasn't been updated significantly since then.
> It is an abstract API, whereas operating systems implement concrete APIs
> based on sockets, primarily, and that have diverged from the description in
> 793 (and currently 793bis).
>
> I think our options are:
> 1. Leave it alone, modulo any corrections from errata, other RFCs, etc.
> 2. Take it out
> 3. Change it in more significant ways
>
> I don't think option 3 is somewhere we should go.  I think that should be
> a separate effort, if there is desire to do that.  My inclination is for
> option 1.
>
> It will be good to hear what others think, and understand the consensus on
> this though.
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

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<div dir=3D"ltr">I think option 1 makes sense. I wouldn&#39;t be opposed st=
rongly to option 2, but the doc seems clear on the &quot;advisory&quot; nat=
ure of this API.</div><div class=3D"gmail_extra"><br><div class=3D"gmail_qu=
ote">On Thu, Nov 12, 2015 at 12:31 PM, Wesley Eddy <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:wes@mti-systems.com" target=3D"_blank">wes@mti-systems.com<=
/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">I checked out the Y=
okohama meeting minutes and recording (sorry I missed it in person!).<br>
<br>
We need to follow-up on the discussion of what to do (or not do) about the =
TCP API in RFC 793bis.=C2=A0 I know Karen had interest in what will happen =
to this, for one.<br>
<br>
As I see it, the issue raised on the TAPS list earlier is that the TCP API =
in 793 dates from 1981 and hasn&#39;t been updated significantly since then=
.=C2=A0 It is an abstract API, whereas operating systems implement concrete=
 APIs based on sockets, primarily, and that have diverged from the descript=
ion in 793 (and currently 793bis).<br>
<br>
I think our options are:<br>
1. Leave it alone, modulo any corrections from errata, other RFCs, etc.<br>
2. Take it out<br>
3. Change it in more significant ways<br>
<br>
I don&#39;t think option 3 is somewhere we should go.=C2=A0 I think that sh=
ould be a separate effort, if there is desire to do that.=C2=A0 My inclinat=
ion is for option 1.<br>
<br>
It will be good to hear what others think, and understand the consensus on =
this though.<br>
<br>
_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
</blockquote></div><br></div>

--001a113dae8c13888705245e4cf1--


From nobody Thu Nov 12 13:41:00 2015
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] TCP API in 793 / 793bis
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Option 1.

A protocol without an (abstract) API isn't a protocol.

I don't think there's a lot that needs to change to support the
requirements of 793-bis.

Joe

On 11/12/2015 12:31 PM, Wesley Eddy wrote:
> I checked out the Yokohama meeting minutes and recording (sorry I missed
> it in person!).
> 
> We need to follow-up on the discussion of what to do (or not do) about
> the TCP API in RFC 793bis.  I know Karen had interest in what will
> happen to this, for one.
> 
> As I see it, the issue raised on the TAPS list earlier is that the TCP
> API in 793 dates from 1981 and hasn't been updated significantly since
> then.  It is an abstract API, whereas operating systems implement
> concrete APIs based on sockets, primarily, and that have diverged from
> the description in 793 (and currently 793bis).
> 
> I think our options are:
> 1. Leave it alone, modulo any corrections from errata, other RFCs, etc.
> 2. Take it out
> 3. Change it in more significant ways
> 
> I don't think option 3 is somewhere we should go.  I think that should
> be a separate effort, if there is desire to do that.  My inclination is
> for option 1.
> 
> It will be good to hear what others think, and understand the consensus
> on this though.
> 
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Thu Nov 12 13:43:36 2015
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Another vote for Option 1.

On 11/12/15 12:31 PM, Wesley Eddy wrote:
> I checked out the Yokohama meeting minutes and recording (sorry I missed
> it in person!).
>
> We need to follow-up on the discussion of what to do (or not do) about
> the TCP API in RFC 793bis.  I know Karen had interest in what will
> happen to this, for one.
>
> As I see it, the issue raised on the TAPS list earlier is that the TCP
> API in 793 dates from 1981 and hasn't been updated significantly since
> then.  It is an abstract API, whereas operating systems implement
> concrete APIs based on sockets, primarily, and that have diverged from
> the description in 793 (and currently 793bis).
>
> I think our options are:
> 1. Leave it alone, modulo any corrections from errata, other RFCs, etc.
> 2. Take it out
> 3. Change it in more significant ways
>
> I don't think option 3 is somewhere we should go.  I think that should
> be a separate effort, if there is desire to do that.  My inclination is
> for option 1.
>
> It will be good to hear what others think, and understand the consensus
> on this though.
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Thu Nov 12 14:30:14 2015
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Opt 1 vote
On Nov 12, 2015 1:43 PM, "Tim Wicinski" <tjw.ietf@gmail.com> wrote:

>
> Another vote for Option 1.
>
> On 11/12/15 12:31 PM, Wesley Eddy wrote:
>
>> I checked out the Yokohama meeting minutes and recording (sorry I missed
>> it in person!).
>>
>> We need to follow-up on the discussion of what to do (or not do) about
>> the TCP API in RFC 793bis.  I know Karen had interest in what will
>> happen to this, for one.
>>
>> As I see it, the issue raised on the TAPS list earlier is that the TCP
>> API in 793 dates from 1981 and hasn't been updated significantly since
>> then.  It is an abstract API, whereas operating systems implement
>> concrete APIs based on sockets, primarily, and that have diverged from
>> the description in 793 (and currently 793bis).
>>
>> I think our options are:
>> 1. Leave it alone, modulo any corrections from errata, other RFCs, etc.
>> 2. Take it out
>> 3. Change it in more significant ways
>>
>> I don't think option 3 is somewhere we should go.  I think that should
>> be a separate effort, if there is desire to do that.  My inclination is
>> for option 1.
>>
>> It will be good to hear what others think, and understand the consensus
>> on this though.
>>
>> _______________________________________________
>> 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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<p dir=3D"ltr">Opt 1 vote</p>
<div class=3D"gmail_quote">On Nov 12, 2015 1:43 PM, &quot;Tim Wicinski&quot=
; &lt;<a href=3D"mailto:tjw.ietf@gmail.com">tjw.ietf@gmail.com</a>&gt; wrot=
e:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
Another vote for Option 1.<br>
<br>
On 11/12/15 12:31 PM, Wesley Eddy wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I checked out the Yokohama meeting minutes and recording (sorry I missed<br=
>
it in person!).<br>
<br>
We need to follow-up on the discussion of what to do (or not do) about<br>
the TCP API in RFC 793bis.=C2=A0 I know Karen had interest in what will<br>
happen to this, for one.<br>
<br>
As I see it, the issue raised on the TAPS list earlier is that the TCP<br>
API in 793 dates from 1981 and hasn&#39;t been updated significantly since<=
br>
then.=C2=A0 It is an abstract API, whereas operating systems implement<br>
concrete APIs based on sockets, primarily, and that have diverged from<br>
the description in 793 (and currently 793bis).<br>
<br>
I think our options are:<br>
1. Leave it alone, modulo any corrections from errata, other RFCs, etc.<br>
2. Take it out<br>
3. Change it in more significant ways<br>
<br>
I don&#39;t think option 3 is somewhere we should go.=C2=A0 I think that sh=
ould<br>
be a separate effort, if there is desire to do that.=C2=A0 My inclination i=
s<br>
for option 1.<br>
<br>
It will be good to hear what others think, and understand the consensus<br>
on this though.<br>
<br>
_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
</blockquote>
<br>
_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/tcpm</a><br>
</blockquote></div>

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Subject: Re: [tcpm] TCP API in 793 / 793bis
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+1 for opt 1

> On 12. nov. 2015, at 23.30, Yuchung Cheng <ycheng@google.com> wrote:
>=20
> Opt 1 vote
>=20
> On Nov 12, 2015 1:43 PM, "Tim Wicinski" <tjw.ietf@gmail.com =
<mailto:tjw.ietf@gmail.com>> wrote:
>=20
> Another vote for Option 1.
>=20
> On 11/12/15 12:31 PM, Wesley Eddy wrote:
> I checked out the Yokohama meeting minutes and recording (sorry I =
missed
> it in person!).
>=20
> We need to follow-up on the discussion of what to do (or not do) about
> the TCP API in RFC 793bis.  I know Karen had interest in what will
> happen to this, for one.
>=20
> As I see it, the issue raised on the TAPS list earlier is that the TCP
> API in 793 dates from 1981 and hasn't been updated significantly since
> then.  It is an abstract API, whereas operating systems implement
> concrete APIs based on sockets, primarily, and that have diverged from
> the description in 793 (and currently 793bis).
>=20
> I think our options are:
> 1. Leave it alone, modulo any corrections from errata, other RFCs, =
etc.
> 2. Take it out
> 3. Change it in more significant ways
>=20
> I don't think option 3 is somewhere we should go.  I think that should
> be a separate effort, if there is desire to do that.  My inclination =
is
> for option 1.
>=20
> It will be good to hear what others think, and understand the =
consensus
> on this though.
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org <mailto:tcpm@ietf.org>
> https://www.ietf.org/mailman/listinfo/tcpm =
<https://www.ietf.org/mailman/listinfo/tcpm>
>=20
> _______________________________________________
> 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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<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">+1 for opt 1<div class=""><br class=""><div style=""><blockquote type="cite" class=""><div class="">On 12. nov. 2015, at 23.30, Yuchung Cheng &lt;<a href="mailto:ycheng@google.com" class="">ycheng@google.com</a>&gt; wrote:</div><br class="Apple-interchange-newline"><div class=""><p dir="ltr" class="">Opt 1 vote</p>
<div class="gmail_quote">On Nov 12, 2015 1:43 PM, "Tim Wicinski" &lt;<a href="mailto:tjw.ietf@gmail.com" class="">tjw.ietf@gmail.com</a>&gt; wrote:<br type="attribution" class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br class="">
Another vote for Option 1.<br class="">
<br class="">
On 11/12/15 12:31 PM, Wesley Eddy wrote:<br class="">
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I checked out the Yokohama meeting minutes and recording (sorry I missed<br class="">
it in person!).<br class="">
<br class="">
We need to follow-up on the discussion of what to do (or not do) about<br class="">
the TCP API in RFC 793bis.&nbsp; I know Karen had interest in what will<br class="">
happen to this, for one.<br class="">
<br class="">
As I see it, the issue raised on the TAPS list earlier is that the TCP<br class="">
API in 793 dates from 1981 and hasn't been updated significantly since<br class="">
then.&nbsp; It is an abstract API, whereas operating systems implement<br class="">
concrete APIs based on sockets, primarily, and that have diverged from<br class="">
the description in 793 (and currently 793bis).<br class="">
<br class="">
I think our options are:<br class="">
1. Leave it alone, modulo any corrections from errata, other RFCs, etc.<br class="">
2. Take it out<br class="">
3. Change it in more significant ways<br class="">
<br class="">
I don't think option 3 is somewhere we should go.&nbsp; I think that should<br class="">
be a separate effort, if there is desire to do that.&nbsp; My inclination is<br class="">
for option 1.<br class="">
<br class="">
It will be good to hear what others think, and understand the consensus<br class="">
on this though.<br class="">
<br class="">
_______________________________________________<br class="">
tcpm mailing list<br class="">
<a href="mailto:tcpm@ietf.org" target="_blank" class="">tcpm@ietf.org</a><br class="">
<a href="https://www.ietf.org/mailman/listinfo/tcpm" rel="noreferrer" target="_blank" class="">https://www.ietf.org/mailman/listinfo/tcpm</a><br class="">
</blockquote>
<br class="">
_______________________________________________<br class="">
tcpm mailing list<br class="">
<a href="mailto:tcpm@ietf.org" target="_blank" class="">tcpm@ietf.org</a><br class="">
<a href="https://www.ietf.org/mailman/listinfo/tcpm" rel="noreferrer" target="_blank" class="">https://www.ietf.org/mailman/listinfo/tcpm</a><br class="">
</blockquote></div>
_______________________________________________<br class="">tcpm mailing list<br class=""><a href="mailto:tcpm@ietf.org" class="">tcpm@ietf.org</a><br class="">https://www.ietf.org/mailman/listinfo/tcpm<br class=""></div></blockquote></div><br class=""></div></body></html>
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Subject: Re: [tcpm] TCP API in 793 / 793bis
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I agree with option 1 as well as.

I am a little curious as to what the =E2=80=9Cetc.=E2=80=9D might end up in=
cluding.

But let us see.



I agree with option 1 instead of option 3 from the perspective of

achieving something good and valuable with reasonable effort.



We often deal with people referring back to RFC793 API and

having what is in there be correct is important.

That it is not comprehensive is more acceptable.



BR, Karen



*From:* tcpm [mailto:tcpm-bounces@ietf.org] *On Behalf Of *Jana Iyengar
*Sent:* 12. november 2015 22:05
*To:* Wesley Eddy <wes@mti-systems.com>
*Cc:* tcpm@ietf.org
*Subject:* Re: [tcpm] TCP API in 793 / 793bis



I think option 1 makes sense. I wouldn't be opposed strongly to option 2,
but the doc seems clear on the "advisory" nature of this API.



On Thu, Nov 12, 2015 at 12:31 PM, Wesley Eddy <wes@mti-systems.com> wrote:

I checked out the Yokohama meeting minutes and recording (sorry I missed it
in person!).

We need to follow-up on the discussion of what to do (or not do) about the
TCP API in RFC 793bis.  I know Karen had interest in what will happen to
this, for one.

As I see it, the issue raised on the TAPS list earlier is that the TCP API
in 793 dates from 1981 and hasn't been updated significantly since then.
It is an abstract API, whereas operating systems implement concrete APIs
based on sockets, primarily, and that have diverged from the description in
793 (and currently 793bis).

I think our options are:
1. Leave it alone, modulo any corrections from errata, other RFCs, etc.
2. Take it out
3. Change it in more significant ways

I don't think option 3 is somewhere we should go.  I think that should be a
separate effort, if there is desire to do that.  My inclination is for
option 1.

It will be good to hear what others think, and understand the consensus on
this though.

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

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	{mso-style-priority:99;
	color:purple;
	text-decoration:underline;}
span.EmailStyle17
	{mso-style-type:personal-reply;
	font-family:"Calibri",sans-serif;
	color:#1F497D;}
.MsoChpDefault
	{mso-style-type:export-only;
	font-family:"Calibri",sans-serif;
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	margin:3.0cm 2.0cm 3.0cm 2.0cm;}
div.WordSection1
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--></style></head><body lang=3D"DA" link=3D"blue" vlink=3D"purple"><div cla=
ss=3D"WordSection1"><p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"fo=
nt-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1f497d">I =
agree with option 1 as well as. </span></p><p class=3D"MsoNormal"><span lan=
g=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-=
serif;color:#1f497d">I am a little curious as to what the =E2=80=9Cetc.=E2=
=80=9D might end up including.</span></p><p class=3D"MsoNormal"><span lang=
=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-s=
erif;color:#1f497d">But let us see. </span></p><p class=3D"MsoNormal"><span=
 lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,s=
ans-serif;color:#1f497d">=C2=A0</span></p><p class=3D"MsoNormal"><span lang=
=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-s=
erif;color:#1f497d">I agree with option 1 instead of option 3 from the pers=
pective of </span></p><p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"=
font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1f497d">=
achieving something good and valuable with reasonable effort.</span></p><p =
class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,sans-serif;color:#1f497d">=C2=A0</span></p><p class=
=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,sans-serif;color:#1f497d">We often deal with people refe=
rring back to RFC793 API and </span></p><p class=3D"MsoNormal"><span lang=
=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-s=
erif;color:#1f497d">having what is in there be correct is important.</span>=
</p><p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;f=
ont-family:&quot;Calibri&quot;,sans-serif;color:#1f497d">That it is not com=
prehensive is more acceptable.</span></p><p class=3D"MsoNormal"><span lang=
=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-s=
erif;color:#1f497d">=C2=A0</span></p><p class=3D"MsoNormal"><span lang=3D"E=
N-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;=
color:#1f497d">BR, Karen</span></p><p class=3D"MsoNormal"><span lang=3D"EN-=
US" style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;co=
lor:#1f497d">=C2=A0</span></p><div style=3D"border:none;border-left:solid b=
lue 1.5pt;padding:0cm 0cm 0cm 4.0pt"><div><div style=3D"border:none;border-=
top:solid #e1e1e1 1.0pt;padding:3.0pt 0cm 0cm 0cm"><p class=3D"MsoNormal"><=
b><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri&=
quot;,sans-serif">From:</span></b><span lang=3D"EN-US" style=3D"font-size:1=
1.0pt;font-family:&quot;Calibri&quot;,sans-serif"> tcpm [mailto:<a href=3D"=
mailto:tcpm-bounces@ietf.org">tcpm-bounces@ietf.org</a>] <b>On Behalf Of </=
b>Jana Iyengar<br><b>Sent:</b> 12. november 2015 22:05<br><b>To:</b> Wesley=
 Eddy &lt;<a href=3D"mailto:wes@mti-systems.com">wes@mti-systems.com</a>&gt=
;<br><b>Cc:</b> <a href=3D"mailto:tcpm@ietf.org">tcpm@ietf.org</a><br><b>Su=
bject:</b> Re: [tcpm] TCP API in 793 / 793bis</span></p></div></div><p clas=
s=3D"MsoNormal">=C2=A0</p><div><p class=3D"MsoNormal">I think option 1 make=
s sense. I wouldn&#39;t be opposed strongly to option 2, but the doc seems =
clear on the &quot;advisory&quot; nature of this API.</p></div><div><p clas=
s=3D"MsoNormal">=C2=A0</p><div><p class=3D"MsoNormal">On Thu, Nov 12, 2015 =
at 12:31 PM, Wesley Eddy &lt;<a href=3D"mailto:wes@mti-systems.com" target=
=3D"_blank">wes@mti-systems.com</a>&gt; wrote:</p><blockquote style=3D"bord=
er:none;border-left:solid #cccccc 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-le=
ft:4.8pt;margin-right:0cm"><p class=3D"MsoNormal">I checked out the Yokoham=
a meeting minutes and recording (sorry I missed it in person!).<br><br>We n=
eed to follow-up on the discussion of what to do (or not do) about the TCP =
API in RFC 793bis.=C2=A0 I know Karen had interest in what will happen to t=
his, for one.<br><br>As I see it, the issue raised on the TAPS list earlier=
 is that the TCP API in 793 dates from 1981 and hasn&#39;t been updated sig=
nificantly since then.=C2=A0 It is an abstract API, whereas operating syste=
ms implement concrete APIs based on sockets, primarily, and that have diver=
ged from the description in 793 (and currently 793bis).<br><br>I think our =
options are:<br>1. Leave it alone, modulo any corrections from errata, othe=
r RFCs, etc.<br>2. Take it out<br>3. Change it in more significant ways<br>=
<br>I don&#39;t think option 3 is somewhere we should go.=C2=A0 I think tha=
t should be a separate effort, if there is desire to do that.=C2=A0 My incl=
ination is for option 1.<br><br>It will be good to hear what others think, =
and understand the consensus on this though.<br><br>_______________________=
________________________<br>tcpm mailing list<br><a href=3D"mailto:tcpm@iet=
f.org" target=3D"_blank">tcpm@ietf.org</a><br><a href=3D"https://www.ietf.o=
rg/mailman/listinfo/tcpm" target=3D"_blank">https://www.ietf.org/mailman/li=
stinfo/tcpm</a></p></blockquote></div><p class=3D"MsoNormal">=C2=A0</p></di=
v></div></div></body></html>

--001a1136b07a0f74d0052467d7cf--


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From: Pasi Sarolahti <pasi.sarolahti@iki.fi>
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Subject: [tcpm] Adoption of More Accurate ECN Feedback in TCP
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Hi,

In the Yokohama meeting we discussed adoption of "More Accurate ECN =
Feedback in TCP" as Experimental RFC, with =
draft-kuehlewind-tcpm-accurate-ecn-05 serving as the baseline for the =
working group draft. We called a hum in the meeting that very strongly =
supported adoption of the document.

The chairs would now want to confirm the adoption call on the list. =
Please let us know by Friday, November 27 whether you support or object =
adoption of the document.

The proposed milestone is "Submit specification of more accurate ECN =
feedback in TCP to the IESG for publication as an Experimental RFC=94 =97 =
dated for Aug 2016.

- Pasi, Yoshi, Michael


From nobody Mon Nov 16 12:23:40 2015
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From: Pasi Sarolahti <pasi.sarolahti@iki.fi>
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Subject: [tcpm] TCPM notes from IETF-94
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Hi,

The minutes of the Yokohama TCPM meeting are available at =
https://www.ietf.org/proceedings/94/minutes/minutes-94-tcpm. Many thanks =
to Christoph and David for compiling them!

If you have corrections to make, please send them to list, or directly =
to chairs.

- Pasi


From nobody Mon Nov 16 13:24:01 2015
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Subject: Re: [tcpm] Adoption of More Accurate ECN Feedback in TCP
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On Nov 16, 2015, at 12:11, Pasi Sarolahti <pasi.sarolahti@iki.fi> wrote:
>=20
> Hi,
>=20
> In the Yokohama meeting we discussed adoption of "More Accurate ECN =
Feedback in TCP" as Experimental RFC, with =
draft-kuehlewind-tcpm-accurate-ecn-05 serving as the baseline for the =
working group draft. We called a hum in the meeting that very strongly =
supported adoption of the document.
>=20
> The chairs would now want to confirm the adoption call on the list. =
Please let us know by Friday, November 27 whether you support or object =
adoption of the document.
>=20
> The proposed milestone is "Submit specification of more accurate ECN =
feedback in TCP to the IESG for publication as an Experimental RFC=94 =97 =
dated for Aug 2016.

Support.

--
Rui Paulo




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From: Joe Touch <touch@isi.edu>
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Subject: [tcpm] regarding draft-kitamura-tcp-sharp-close
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Hi, all,

This doc was never posted to this list (although it was presented at
Yokohama), so I figured I'd start a thread with some feedback (that IMO
probably should have happened online before the meeting).

Lars already pointed out a paper that Ted Faber and I wrote on this
issue at Infocom 1999. Here's a link to that:
http://www.isi.edu/touch/pubs/infocomm99/

Some other info. below.

Joe

------

	- I don't think the doc is correct on the purpose of SO_LINGER
	See: http://developerweb.net/viewtopic.php?id=2982

	SO_LINGER focuses on half-closed connections, and how long to
	wait to deliver remaining information.TIME-WAIT is a state
	that occurs only after both sides have closed the connection.

	- TIME-WAIT's primary purpose is to avoid the impact of old
	packets from a previous connection corrupting the data
	of a new connection.

		I.e., it's primary purpose is to ensure that *data*
		packets from an old connection are not accepted
		by a new connection that reuses the port numbers.

		The only connection resource that isn't released
		is the socket pair - i.e., the pair of IP addresses
		and port numbers that uniquely identify the connection.
		TIME-WAIT need not consume other resources.

		TIME-WAIT has nothing to do with resending FINs. It
		can be entered only when the closing three-way FIN
		handshake completes.

	- "sharp close" appears to reinvent simultaneous close

		However, this doesn't avoid TIME-WAIT; in current
		TCP, it would require that both sides enter TIME-WAIT.

		The issue is that the port numbers can't be reused
		while there might be old data packets that could still
		arrive (e.g., duplicates from the previous connection
		that are floating around). TCP requires that these old
		packets be discarded when a new connection starts.

		Normally this would happen by the use of the ISN, but
		a quick enough connection might end up wrapping the
		sequence numbers around fast enough that the old data
		could end up in the valid incoming data window.

		As per RFC 6191, timestamps can be used to avoid this
		problem as well.

	- it's not clear how both sides would know to issue the FINs
	concurrently
		i.e., this cannot be achieved by user-level operations
		unless the application coordinates the close in-band;
		even then, TCP might exchange the FIN/FIN-ACK/ACK before
		the other side has a chance to start its FIN anyway.

	- we also already have other ways to deal with TIMEWAIT,
		e.g., RFC 6191, using lower TIME-WAIT timers because
		of known properties of port reuse or the willingness
		to "take a risk" on old data coming in late

----


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From: Veaceslav ROMAN <Veaceslav.Roman@orange.md>
To: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, Neal Cardwell <ncardwell@google.com>
Thread-Topic: [tcpm] [e2e] TCP HyStart patch deployment
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Subject: Re: [tcpm] [e2e] TCP HyStart patch deployment
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To: Veaceslav ROMAN <Veaceslav.Roman@orange.md>
Date: Tue, 17 Nov 2015 14:48:12 -0800
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Subject: Re: [tcpm] [e2e] TCP HyStart patch deployment
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On Tue, 2015-11-17 at 21:04 +0000, Veaceslav ROMAN wrote:
> Hi
> Still experimenting with Hystart in LTE by trying different parameters for HYSTART_DELAY_MIN and HYSTART_DELAY_MAX.
> In an attempt to easy the testing we've modified the 4.1.10 kernel to make these parameters externally modifiable plus an attempt is done to fix what I do perceive as a significand deviation of implementation from the original idea of Hystart, which, in my view, does have a negative impact, especially in networks with high throughput and delay variability, like radio networks. 
> Unfortunately 4.1.10 was very unstable, server frequently freezing and, not completely unsure, whether this is due to the kernel of due to modified code. No kernel development experience here.
> Anyway we decided to move to 3.19.0-33 and we try currently to recompile. 

v-4.1.13 should be stable, 4.1.10 had tcp bugs.



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From: Veaceslav ROMAN <Veaceslav.Roman@orange.md>
To: Eric Dumazet <eric.dumazet@gmail.com>
Thread-Topic: [tcpm] [e2e] TCP HyStart patch deployment
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Date: Wed, 18 Nov 2015 07:37:19 +0000
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Subject: Re: [tcpm] [e2e] TCP HyStart patch deployment
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Archived-At: <http://mailarchive.ietf.org/arch/msg/tcpm/ZSFuITYLwr1JV2tjTCfgYZ3wQ7Y>
Subject: Re: [tcpm] TCPM notes from IETF-94
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Hi,

nit: s/what is write/what is right

Else: the transcription of what David Black said to Naeem Khademi on the =
mic is misleading - it misses key bits that make it seem like a =
different statement than it was:


The minutes say:
***
David Black: RFC3168 says: don't do anything that breaks the Internet. =
If there is a ECE, react to it the same way as with packet-loss.
***

... which makes it sound like this would be his advice and he would =
opposed to experimental alternatives. However, what he really said is =
(transcribed from Meetecho:
=
http://recs.conf.meetecho.com/Playout/watch.jsp?recording=3DIETF94_TCPM&ch=
apter=3Dchapter_1   at approx. 0:31):

***
As one of the authors of RFC 3168: I agree that encouraging Experimental =
for congestion control is a really good idea.
3168 was written for the broad internet, hence contains some very strict =
language that basically said don't do anything that would break things. =
We had people react to congestion if you CE-mark as if you're dropping a =
packet, we said: the only thing we know for certain is safe right now - =
this was a long time ago - was: react to it as you would react to a =
packet drop. I'm not opposed to interesting innovative things being done =
under the guise of experimentation, I wanna provide some history for why =
3168 says what it does.
***

The last sentence matters - it's really quite a different thing.

Cheers,
Michael





> On 16 Nov 2015, at 21:23, Pasi Sarolahti <pasi.sarolahti@iki.fi> =
wrote:
>=20
> Hi,
>=20
> The minutes of the Yokohama TCPM meeting are available at =
https://www.ietf.org/proceedings/94/minutes/minutes-94-tcpm. Many thanks =
to Christoph and David for compiling them!
>=20
> If you have corrections to make, please send them to list, or directly =
to chairs.
>=20
> - Pasi
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Thu Nov 19 07:46:45 2015
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Cc: tcpm@ietf.org, mls.ietf@gmail.com, draft-ietf-tcpm-rtorestart@ietf.org, The IESG <iesg@ietf.org>, tcpm-chairs@ietf.org, rfc-editor@rfc-editor.org
Subject: [tcpm] Document Action: 'TCP and SCTP RTO Restart' to Experimental RFC (draft-ietf-tcpm-rtorestart-10.txt)
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The IESG has approved the following document:
- 'TCP and SCTP RTO Restart'
  (draft-ietf-tcpm-rtorestart-10.txt) as Experimental RFC

This document is the product of the TCP Maintenance and Minor Extensions
Working Group.

The IESG contact persons are Spencer Dawkins and Martin Stiemerling.

A URL of this Internet Draft is:
https://datatracker.ietf.org/doc/draft-ietf-tcpm-rtorestart/





Technical Summary

This document describes a modified sender-side algorithm for managing 
the TCP and SCTP retransmission timers that provides faster loss 
recovery when there is a small amount of outstanding data for a 
connection.  The modification, RTO Restart (RTOR), allows the transport 
to restart its retransmission timer so that the effective RTO becomes 
more aggressive in situations where fast retransmit cannot be used.  
This enables faster loss detection and recovery for connections that are 
short-lived or application-limited.

Working Group Summary

It is the consensus of the TCPM working group to document this 
alternative algorithm, given the potential performance benefit. The work 
has mostly been driven by the authors, but the document has been 
reviewed in detail by several experts and the content has been modified 
accordingly. Performance experiments in simulations and testbeds have 
been performed and published by the authors and the experimental results 
have been reviewed in several TCPM meetings. At the time of writing, 
there is only limited deployment experience. 

Two issues have been discussed extensively in the working group. First, 
any reduction of the retransmission timeout duration inherently comes 
along with a risk of negative impact on TCP performance, e.g. in mobile 
networks with highly variable RTT. The current understanding is that 
this risk is low and that the algorithm is conservative and relatively 
robust, but further experimentation has to confirm this. Second, the 
Linux operation system uses the "Tail Loss Probe" method discussed in 
Section 6, which is similar but more complex. This method was not 
adopted in TCPM since it depends on FACK error recovery method, which 
has not been standardizes so far.

Document Quality

This document was also last called in TSVWG, since it specifies an 
algorithm that can be applied both to TCP and SCTP. As a result of WGLC 
comments the applicability to SCTP has been better explained, including 
the SCTP API. One issue is that TCP and SCTP use slightly different 
terminology for comparable concepts. In order to keep the document 
simple, it was decided not to add another, duplicated description of the 
algorithm using SCTP terminology.

Personnel

The document shepherd is Michael Scharf <michael.scharf@alcatel-
lucent.com>. The responsible Area Director is Martin Stiemerling 
<mls.ietf@gmail.com>.


From nobody Fri Nov 20 06:41:06 2015
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From: SALLANTIN Renaud <renaud.sallantin@thalesaleniaspace.com>
To: "'tcpm@ietf.org'" <tcpm@ietf.org>, BAUDOIN Cedric <cedric.baudoin@thalesaleniaspace.com>
Date: Fri, 20 Nov 2015 15:40:52 +0100
Thread-Topic: How to improve TCP efficiency in a large RTT context?
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--_000_9b39fe7d76ff4baabe4232d2e58833a8THSONEA01HUB02Ponegrp_
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Dear all,
With the emergence of new generation satellite systems, including mega LEO =
constellations as well as very high throughput satellite offering terabit c=
apacity, the problem of the end-to end performances arise again. Indeed, th=
e large RTT (larger than 500ms for geo sat) is drastically downgrading regu=
lar TCP performance, notably during the beginning of the connection. Curren=
tly, proxies such as T-PEPs are used to minimize the RTT impact. But, their=
 consequences in terms of security and mobility are preventing their usage =
in many cases and can't be considered as a sustainable solution.
Without  proxies,  the most intuitive solution is therefore to increase aga=
in the IW. Our studies therefore shown that IW10 is not sufficient and that=
 short-lived connections continue to be the most impacted. But we do believ=
e that an adequate use of Pacing and a larger IW should enable to mitigate =
the Long Fat Network latency. However  this raises a lot of questions that =
we would like to share with you:

=B7         Is there room to increase even more the Initial Window? Several=
 major players already use IW=3D32 or higher (with and without pacing), tha=
t would really help for satellite communications.  But the question here is=
 how to mitigate the burst impact?

=B7         Can we rely on SYN-SYN/ACK RTT measurement to dynamically adapt=
 the slow start behavior (and notably the IW size)  since it's not possible=
 to differentiate delay due to propagation from bufferbloat at this stage ?

=B7         Can we configure pacing to adapt to different networks ? Basica=
lly, 1ms is generally used in SPDY and QUIC, but is there any optimal?  For=
 example, in case of a large RTT (either due to the delay or the congestion=
),  a larger pacing value (such as 5 ms) seems to offer better results.
Finally, based on several discussions we had with some of you, we are consi=
dering writing an Internet Draft on the "Pacing in TCP", and we would like =
to know if there are some interest for this?
Regards,
Renaud and C=E9dric

--_000_9b39fe7d76ff4baabe4232d2e58833a8THSONEA01HUB02Ponegrp_
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=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span lang=3DEN-US=
>Dear all, <o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US>Wi=
th the emergence of new generation satellite systems, including mega LEO co=
nstellations as well as very high throughput satellite offering terabit cap=
acity, the problem of the end-to end performances arise again. Indeed, the =
large RTT (larger than 500ms for geo sat) is drastically downgrading regula=
r TCP performance, notably during the beginning of the connection. Currentl=
y, proxies such as T-PEPs are used to minimize the RTT impact. But, their c=
onsequences in terms of security and mobility are preventing their usage in=
 many cases and can&#8217;t be considered as a sustainable solution.<o:p></=
o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US>Without=A0 proxies,=
=A0 the most intuitive solution is therefore to increase again the IW. Our =
studies therefore shown that IW10 is not sufficient and that short-lived co=
nnections continue to be the most impacted. But we do believe that an adequ=
ate use of Pacing and a larger IW should enable to mitigate the Long Fat Ne=
twork latency. However=A0 this raises a lot of questions that we would like=
 to share with you:<o:p></o:p></span></p><p class=3DMsoListParagraphCxSpFir=
st style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if !supportList=
s]><span lang=3DEN-US style=3D'font-family:Symbol'><span style=3D'mso-list:=
Ignore'>=B7<span style=3D'font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span></span><![endif]><span lang=3DE=
N-US>Is there room to increase even more the Initial Window? Several major =
players already use IW=3D32 or higher (with and without pacing), that would=
 really help for satellite communications.=A0 But the question here is how =
to mitigate the burst impact?<o:p></o:p></span></p><p class=3DMsoListParagr=
aphCxSpMiddle style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if !=
supportLists]><span lang=3DEN-US style=3D'font-family:Symbol'><span style=
=3D'mso-list:Ignore'>=B7<span style=3D'font:7.0pt "Times New Roman"'>&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span></span><![endif]><=
span lang=3DEN-US>Can we rely on SYN-SYN/ACK RTT measurement to dynamically=
 adapt the slow start behavior (and notably the IW size)=A0 since it&#8217;=
s not possible to differentiate delay due to propagation from bufferbloat a=
t this stage ? <o:p></o:p></span></p><p class=3DMsoListParagraphCxSpLast st=
yle=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if !supportLists]><s=
pan lang=3DEN-US style=3D'font-family:Symbol'><span style=3D'mso-list:Ignor=
e'>=B7<span style=3D'font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp; </span></span></span><![endif]><span lang=3DEN-US>=
Can we configure pacing to adapt to different networks ? Basically, 1ms is =
generally used in SPDY and QUIC, but is there any optimal?=A0 For example, =
in case of a large RTT (either due to the delay or the congestion),=A0 a la=
rger pacing value (such as 5 ms) seems to offer better results.<o:p></o:p><=
/span></p><p class=3DMsoNormal><span lang=3DEN-US>Finally, based on several=
 discussions we had with some of you, we are considering writing an Interne=
t Draft on the &#8220;Pacing in TCP&#8221;, and we would like to know if th=
ere are some interest for this?<o:p></o:p></span></p><p class=3DMsoNormal><=
span lang=3DEN-US>Regards, <o:p></o:p></span></p><p class=3DMsoNormal><span=
 lang=3DEN-US>Renaud and C=E9dric</span><o:p></o:p></p></div></body></html>=

--_000_9b39fe7d76ff4baabe4232d2e58833a8THSONEA01HUB02Ponegrp_--


From nobody Fri Nov 20 06:50:41 2015
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Subject: Re: [tcpm] How to improve TCP efficiency in a large RTT context?
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> =C2=B7 Is there room to increase even more the Initial Window? Several
> major players already use IW=3D32 or higher

I'll just flog this again: http://www.icir.org/mallman/pubs/LAJW07/

:)

(I chaired TCPM when that came out.  The IETF subsequently wised up
and found better people.  Lars was the TSV AD at the time.  His
comment on the above when it was presented at the workshop was
something along the lines of: can we really have a TCPM chair who
would propose an arbitrary IW?!  I think he meant it as a joke.  I
took it as a compliment. :) )

But, in all seriousness, we don't need to specify the IW.  Or, how
it gets set.  As long as the rest of CC is in place, there are
consequences for being too aggressive that will keep things
reasonable.  IMHO.  Not to mention that the actual impact of lousy
choices isn't all that bad.  And, of course, if we never had to
discuss the IW again, that'd be cool, too. :)

allman


=2D-
http://www.icir.org/mallman/




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Subject: Re: [tcpm] [e2e] TCP HyStart patch deployment
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On Tue, 2015-11-17 at 21:05 +0000, Veaceslav ROMAN wrote:
...
> The reason for what I believed the call of hystart_update before the hyst=
art_update is a big deviation from the initial logic is as follow.
> Both HYSTART_ACK_TRAIN and  HYSTART_DELAY algorithms in function hystart_=
update are supposed to do some computing starting with the first ACK of the=
 train. The beginning of the train itself is determined by the function hys=
tart_reset. But, because, in the current implementation the hystart_update =
is called for each ACK before the hystart_reset, the first ACK of the train=
 N+1 is still considered in hystart_update as an ACK of the previous train =
N. (in tcp_input.c: tcp_ack() -> tcp_clean_rtx_queue() ->  pkts_acked()=3Db=
ictcp_acked() -> hystart_update();  then  tcp_input.c: tcp_ack() -> tcp_con=
g_avoid() -> cong_avoid=3Dbictcp_cong_avoid() -> hystart_reset(); and, btw,=
 tcp_slow_start() is called next ).
> Therefor HYSTART_ACK_TRAIN, in the current implementation, will not measu=
re, as it was supposed in the original paper, the length of the ACK train, =
but a duration between the 2-nd ACK of the train N and the first ACK of the=
 train N+1. This will usually result in approximately one full RTT. Due to =
protection (now - ca->last_ack) <=3D hystart_ack_delta (2 ms) this would le=
ad:
> - in sub 2 ms RTT networks to always too early exit (in the second round)
>              - in over 2 ms RTT networks to always too late exit (due to =
RTT > RTT/2 while  now - ca->last_ack) <=3D 2 ms when the HOL ACK of N+1 ap=
proaches tail ACK of N to less than 2 ms.
> Means that current implementation of HYSTART_ACK_TRAIN does not work as e=
xpected (did it ever ?).

+1

Would you submit a patch to netdev that fixes this?

It would be nice of you to fix tcp_cdg's HyStart as well. Just move
the call to tcp_cdg_hystart_update() a few lines:

diff --git a/net/ipv4/tcp_cdg.c b/net/ipv4/tcp_cdg.c
index 167b6a3..a02d944 100644
--- a/net/ipv4/tcp_cdg.c
+++ b/net/ipv4/tcp_cdg.c
@@ -264,9 +264,6 @@ static void tcp_cdg_cong_avoid(struct sock *sk,
u32 ack, u32 acked)
        u32 prior_snd_cwnd;
        u32 incr;

-       if (tcp_in_slow_start(tp) && hystart_detect)
-               tcp_cdg_hystart_update(sk);
-
        if (after(ack, ca->rtt_seq) && ca->rtt.v64) {
                s32 grad =3D 0;

@@ -282,6 +279,9 @@ static void tcp_cdg_cong_avoid(struct sock *sk,
u32 ack, u32 acked)
                        return;
        }

+       if (tcp_in_slow_start(tp) && hystart_detect)
+               tcp_cdg_hystart_update(sk);
+
        if (!tcp_is_cwnd_limited(sk)) {
                ca->shadow_wnd =3D min(ca->shadow_wnd, tp->snd_cwnd);
                return;


I set up a server<>client scenario with 1 sec OWD to verify.

Before the change, the last ACK of each round mistakenly gets a train
length of ~1 sec:
2231665.726383: tcp_cdg_cong_avoid: round 0 train_len 0 rtt_seq
1401191661 ack_seq 1401191685 sample_cnt 0
2231665.726408: tcp_cdg_cong_avoid: round 1 train_len 0 rtt_seq
1401204717 ack_seq 1401193133 sample_cnt 0
2231665.726415: tcp_cdg_cong_avoid: round 1 train_len 8 rtt_seq
1401204717 ack_seq 1401194581 sample_cnt 1
2231665.726424: tcp_cdg_cong_avoid: round 1 train_len 16 rtt_seq
1401204717 ack_seq 1401196029 sample_cnt 2
2231665.726429: tcp_cdg_cong_avoid: round 1 train_len 22 rtt_seq
1401204717 ack_seq 1401197477 sample_cnt 3
2231665.726502: tcp_cdg_cong_avoid: round 1 train_len 95 rtt_seq
1401204717 ack_seq 1401198925 sample_cnt 4
2231665.726516: tcp_cdg_cong_avoid: round 1 train_len 108 rtt_seq
1401204717 ack_seq 1401200373 sample_cnt 5
2231665.726521: tcp_cdg_cong_avoid: round 1 train_len 114 rtt_seq
1401204717 ack_seq 1401201821 sample_cnt 6
2231665.726528: tcp_cdg_cong_avoid: round 1 train_len 121 rtt_seq
1401204717 ack_seq 1401203269 sample_cnt 7
2231665.726577: tcp_cdg_cong_avoid: round 1 train_len 167 rtt_seq
1401204717 ack_seq 1401204717 sample_cnt 8
2231666.726340: tcp_cdg_cong_avoid: round 1 train_len 999932 rtt_seq
1401204717 ack_seq 1401206165 sample_cnt 8
2231666.726363: tcp_cdg_cong_avoid: round 2 train_len 0 rtt_seq
1401233677 ack_seq 1401207613 sample_cnt 0
2231666.726369: tcp_cdg_cong_avoid: round 2 train_len 6 rtt_seq
1401233677 ack_seq 1401209061 sample_cnt 1
...

After the change:
2231763.680223: tcp_cdg_cong_avoid: round 1 train_len 0 rtt_seq
4218529922 ack_seq 4218518338 sample_cnt 0
2231763.680232: tcp_cdg_cong_avoid: round 1 train_len 12 rtt_seq
4218529922 ack_seq 4218519786 sample_cnt 1
2231763.680240: tcp_cdg_cong_avoid: round 1 train_len 21 rtt_seq
4218529922 ack_seq 4218521234 sample_cnt 2
2231763.680315: tcp_cdg_cong_avoid: round 1 train_len 96 rtt_seq
4218529922 ack_seq 4218522682 sample_cnt 3
2231763.680328: tcp_cdg_cong_avoid: round 1 train_len 109 rtt_seq
4218529922 ack_seq 4218524130 sample_cnt 4
2231763.680333: tcp_cdg_cong_avoid: round 1 train_len 114 rtt_seq
4218529922 ack_seq 4218525578 sample_cnt 5
2231763.680339: tcp_cdg_cong_avoid: round 1 train_len 119 rtt_seq
4218529922 ack_seq 4218527026 sample_cnt 6
2231763.680345: tcp_cdg_cong_avoid: round 1 train_len 126 rtt_seq
4218529922 ack_seq 4218528474 sample_cnt 7
2231763.680387: tcp_cdg_cong_avoid: round 1 train_len 167 rtt_seq
4218529922 ack_seq 4218529922 sample_cnt 8
2231764.680125: tcp_cdg_cong_avoid: round 2 train_len 0 rtt_seq
4218558882 ack_seq 4218531370 sample_cnt 0
2231764.680151: tcp_cdg_cong_avoid: round 2 train_len 28 rtt_seq
4218558882 ack_seq 4218532818 sample_cnt 1
...


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From nobody Mon Nov 23 03:48:15 2015
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From: "De Schepper, Koen (Koen)" <koen.de_schepper@alcatel-lucent.com>
To: Rui Paulo <rpaulo@apple.com>, Pasi Sarolahti <pasi.sarolahti@iki.fi>
Thread-Topic: [tcpm] Adoption of More Accurate ECN Feedback in TCP
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> -----Original Message-----
> From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Rui Paulo
> Sent: maandag 16 november 2015 22:24
> To: Pasi Sarolahti
> Cc: tcpm@ietf.org Extensions
> Subject: Re: [tcpm] Adoption of More Accurate ECN Feedback in TCP
>=20
> On Nov 16, 2015, at 12:11, Pasi Sarolahti <pasi.sarolahti@iki.fi> wrote:
> >
> > Hi,
> >
> > In the Yokohama meeting we discussed adoption of "More Accurate ECN
> Feedback in TCP" as Experimental RFC, with draft-kuehlewind-tcpm-
> accurate-ecn-05 serving as the baseline for the working group draft. We
> called a hum in the meeting that very strongly supported adoption of the
> document.
> >
> > The chairs would now want to confirm the adoption call on the list.
> Please let us know by Friday, November 27 whether you support or object
> adoption of the document.
> >
> > The proposed milestone is "Submit specification of more accurate ECN
> feedback in TCP to the IESG for publication as an Experimental RFC" -
> dated for Aug 2016.
>=20
> Support.
>=20

+1

Standardization of support for a more frequent (multiple per RTT) ECN=20
feedback is important to further evolve (TCP) congestion control wrt throug=
hput=20
scalability and low latency!

Koen.


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


From nobody Mon Nov 23 08:04:24 2015
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From: Neal Cardwell <ncardwell@google.com>
To: BAUDOIN Cedric <cedric.baudoin@thalesaleniaspace.com>
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Subject: Re: [tcpm] How to improve TCP efficiency in a large RTT context?
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On Mon, Nov 23, 2015 at 5:19 AM, BAUDOIN Cedric
<cedric.baudoin@thalesaleniaspace.com> wrote:
> Basically, we are not trying to reopen the IW debate :) We came to this q=
uestion mostly because
> 1- for long RTT links, the current IW is clearly suboptimal
> 2- pacing techniques brings major improvement in term of burst impact , e=
nabling more aggressive schemes
> Our need is to  enable to better fill the satellite pipe especially for s=
hort TCP flows .

What sorts of IW and pacing rate values are needed to achieve
acceptable performance in the satellite systems you are concerned
about?

>From my perspective, one concern is that using a much bigger IW or
default pacing rate value can create scenarios where either the server
or client is vulnerable to DoS. Even if the larger IW or pacing rate
are only enabled for high RTTs, eg > 500ms. For example, suppose a
server sees a 500ms RTT and decides to use a larger IW or pacing
rate... what if the 500ms was due not to a large propagation delay
from a satellite link, but rather a low-bandwidth, congested, or
bufferbloated bottleneck? Then the higher IW and pacing rate are
pouring gasoline on a fire.

BTW, if you are considering writing an Internet Draft on =E2=80=9CPacing in
TCP", make sure you include the Linux TCP pacing work done by Eric
Dumazet, which is widely deployed (e.g. on google.com and
youtube.com):

  https://lwn.net/Articles/564978/
  https://www.ietf.org/proceedings/88/slides/slides-88-tcpm-9.pdf

neal


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From: SALLANTIN Renaud <renaud.sallantin@thalesaleniaspace.com>
To: Neal Cardwell <ncardwell@google.com>, BAUDOIN Cedric <cedric.baudoin@thalesaleniaspace.com>
Date: Mon, 23 Nov 2015 17:13:09 +0100
Thread-Topic: [tcpm] How to improve TCP efficiency in a large RTT context?
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Subject: Re: [tcpm] How to improve TCP efficiency in a large RTT context?
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From nobody Mon Nov 23 14:34:23 2015
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From: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] How to improve TCP efficiency in a large RTT context?
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On 11/23/2015 8:13 AM, SALLANTIN Renaud wrote:
> I totally agree with the fact that enlarging the IW is dangerous in
> the situation you described. But I don't see what could be the
> repercussions of increasing the pacing rate?
> 
> If the large RTT is due to the propagation delay, and not the
> congestion, you will indeed add an unnecessary delay (depending on your
> IW size and pacing rate). But, in the other case, you will probably
> lower the consequences of the congestion on the segments you sent...

If the large RTT is propagation delay, then increasing the IW is exactly
the right answer, though you do want to pace the data out as well.

2140 TCB parameter reuse ought to fix the RTT issue, as well as pacing
if pacing is implemented and part of the set of parameters reused across
connections.

Joe




> 
> 
> PS: don't worry, we know and appreciate Eric's work on the pacing ;-)
> 
> [@@ THALES ALENIA SPACE INTERNAL @@]
> 
> 
> -----Message d'origine-----
> De : Neal Cardwell [mailto:ncardwell@google.com] 
> EnvoyÃ© : lundi 23 novembre 2015 17:04
> Ã€ : BAUDOIN Cedric
> Cc : mallman@icir.org; SALLANTIN Renaud; tcpm@ietf.org; Yuchung Cheng; Eric Dumazet
> Objet : Re: [tcpm] How to improve TCP efficiency in a large RTT context?
> 
> On Mon, Nov 23, 2015 at 5:19 AM, BAUDOIN Cedric <cedric.baudoin@thalesaleniaspace.com> wrote:
>> Basically, we are not trying to reopen the IW debate :) We came to 
>> this question mostly because
>> 1- for long RTT links, the current IW is clearly suboptimal
>> 2- pacing techniques brings major improvement in term of burst impact 
>> , enabling more aggressive schemes Our need is to  enable to better fill the satellite pipe especially for short TCP flows .
> 
> What sorts of IW and pacing rate values are needed to achieve acceptable performance in the satellite systems you are concerned about?
> 
>>From my perspective, one concern is that using a much bigger IW or default pacing rate value can create scenarios where either the server or client is vulnerable to DoS. Even if the larger IW or pacing rate are only enabled for high RTTs, eg > 500ms. For example, suppose a server sees a 500ms RTT and decides to use a larger IW or pacing rate... what if the 500ms was due not to a large propagation delay from a satellite link, but rather a low-bandwidth, congested, or bufferbloated bottleneck? Then the higher IW and pacing rate are pouring gasoline on a fire.
> 
> BTW, if you are considering writing an Internet Draft on â€œPacing in TCP", make sure you include the Linux TCP pacing work done by Eric Dumazet, which is widely deployed (e.g. on google.com and
> youtube.com):
> 
>   https://lwn.net/Articles/564978/
>   https://www.ietf.org/proceedings/88/slides/slides-88-tcpm-9.pdf
> 
> neal
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
> 


From nobody Mon Nov 23 14:57:34 2015
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From: Yuchung Cheng <ycheng@google.com>
Date: Mon, 23 Nov 2015 14:56:50 -0800
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To: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] How to improve TCP efficiency in a large RTT context?
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On Mon, Nov 23, 2015 at 2:33 PM, Joe Touch <touch@isi.edu> wrote:
>
>
> On 11/23/2015 8:13 AM, SALLANTIN Renaud wrote:
>> I totally agree with the fact that enlarging the IW is dangerous in
>> the situation you described. But I don't see what could be the
>> repercussions of increasing the pacing rate?
>>
>> If the large RTT is due to the propagation delay, and not the
>> congestion, you will indeed add an unnecessary delay (depending on your
>> IW size and pacing rate). But, in the other case, you will probably
>> lower the consequences of the congestion on the segments you sent...
>
> If the large RTT is propagation delay, then increasing the IW is exactly
> the right answer, though you do want to pace the data out as well.
right. the key issue is no easy way to reasonably verify if it's prop
delay or cong or something else.

>
> 2140 TCB parameter reuse ought to fix the RTT issue, as well as pacing
> if pacing is implemented and part of the set of parameters reused across
> connections.
>
> Joe
>
>
>
>
>>
>>
>> PS: don't worry, we know and appreciate Eric's work on the pacing ;-)
>>
>> [@@ THALES ALENIA SPACE INTERNAL @@]
>>
>>
>> -----Message d'origine-----
>> De : Neal Cardwell [mailto:ncardwell@google.com]
>> Envoy=C3=A9 : lundi 23 novembre 2015 17:04
>> =C3=80 : BAUDOIN Cedric
>> Cc : mallman@icir.org; SALLANTIN Renaud; tcpm@ietf.org; Yuchung Cheng; E=
ric Dumazet
>> Objet : Re: [tcpm] How to improve TCP efficiency in a large RTT context?
>>
>> On Mon, Nov 23, 2015 at 5:19 AM, BAUDOIN Cedric <cedric.baudoin@thalesal=
eniaspace.com> wrote:
>>> Basically, we are not trying to reopen the IW debate :) We came to
>>> this question mostly because
>>> 1- for long RTT links, the current IW is clearly suboptimal
>>> 2- pacing techniques brings major improvement in term of burst impact
>>> , enabling more aggressive schemes Our need is to  enable to better fil=
l the satellite pipe especially for short TCP flows .
>>
>> What sorts of IW and pacing rate values are needed to achieve acceptable=
 performance in the satellite systems you are concerned about?
>>
>>>From my perspective, one concern is that using a much bigger IW or defau=
lt pacing rate value can create scenarios where either the server or client=
 is vulnerable to DoS. Even if the larger IW or pacing rate are only enable=
d for high RTTs, eg > 500ms. For example, suppose a server sees a 500ms RTT=
 and decides to use a larger IW or pacing rate... what if the 500ms was due=
 not to a large propagation delay from a satellite link, but rather a low-b=
andwidth, congested, or bufferbloated bottleneck? Then the higher IW and pa=
cing rate are pouring gasoline on a fire.
>>
>> BTW, if you are considering writing an Internet Draft on =E2=80=9CPacing=
 in TCP", make sure you include the Linux TCP pacing work done by Eric Duma=
zet, which is widely deployed (e.g. on google.com and
>> youtube.com):
>>
>>   https://lwn.net/Articles/564978/
>>   https://www.ietf.org/proceedings/88/slides/slides-88-tcpm-9.pdf
>>
>> neal
>> _______________________________________________
>> 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: Joe Touch <touch@isi.edu>
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Subject: Re: [tcpm] How to improve TCP efficiency in a large RTT context?
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On 11/23/2015 2:56 PM, Yuchung Cheng wrote:
> On Mon, Nov 23, 2015 at 2:33 PM, Joe Touch <touch@isi.edu> wrote:
>>
>>
>> On 11/23/2015 8:13 AM, SALLANTIN Renaud wrote:
>>> I totally agree with the fact that enlarging the IW is dangerous in
>>> the situation you described. But I don't see what could be the
>>> repercussions of increasing the pacing rate?
>>>
>>> If the large RTT is due to the propagation delay, and not the
>>> congestion, you will indeed add an unnecessary delay (depending on your
>>> IW size and pacing rate). But, in the other case, you will probably
>>> lower the consequences of the congestion on the segments you sent...
>>
>> If the large RTT is propagation delay, then increasing the IW is exactly
>> the right answer, though you do want to pace the data out as well.
>
> right. the key issue is no easy way to reasonably verify if it's prop
> delay or cong or something else.

We ought to be able to trust that congestion will result in loss. If
it's just queuing delay and it's persistent, then it can be treated
exactly the same as propagation delay.

I.e., RED (or BLUE, or ALTQ, CODEL, etc.) ought to do the right thing
here, but in the absence of other information, TCP ought to treat delay
as delay.

Joe


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I haven't been tracking things very closely...

> Several major players already use IW=32 or higher (with and
> without pacing),

Is this well known?  Could you point me to the results?  (I am just
interested.)

allman




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Subject: Re: [tcpm] How to improve TCP efficiency in a large RTT context?
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On Tue, Nov 24, 2015 at 11:00 AM, Mark Allman <mallman@icir.org> wrote:
>
> I haven't been tracking things very closely...
>
>> Several major players already use IW=32 or higher (with and
>> without pacing),
>
> Is this well known?  Could you point me to the results?  (I am just
> interested.)

Here is one such study:

  http://www.cdnplanet.com/blog/initcwnd-settings-major-cdn-providers/

neal


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> Here is one such study:
>=20
>   http://www.cdnplanet.com/blog/initcwnd-settings-major-cdn-providers/

cool!  thanks!




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From nobody Tue Nov 24 09:01:35 2015
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On 11/24/2015 8:40 AM, SALLANTIN Renaud wrote:
> 
>> If the large RTT is propagation delay, then increasing the IW is exactly the right answer, though you do want to pace the data out as well.
> 
>> 2140 TCB parameter reuse ought to fix the RTT issue, as well as pacing if pacing is implemented and part of the set of parameters reused across connections.
> 
> Do we know the percentage of connection establishments that can benefit from the TCB information? 

It depends entirely on workload and what fraction of time a connection
spends "learning" about a path.

For connections that are very short, where TCP path properties aren't
really used at all, or connections that are very long with persistent
offered load, it won't matter at all.

It's more work than it's worth for those very short connections. It
doesn't hurt long connections.

It's the medium-sized connections and long connections whose offered
load or competing flows vary a lot that are the largest beneficiaries.

Joe


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A proposal for allowing end hosts to set their initial cwnd to
whatever they want ... in the canonical form that we use to propose
such things ...

  draft-allman-tcpm-no-initwin-00.txt

allman


--
http://www.icir.org/mallman/





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From: Theodore V Faber <theodore.v.faber@aero.org>
To: "mallman@icir.org" <mallman@icir.org>, "tcpm@ietf.org" <tcpm@ietf.org>
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From nobody Tue Nov 24 16:54:37 2015
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On 11/24/2015 09:43 AM, Mark Allman wrote:
>
> A proposal for allowing end hosts to set their initial cwnd to
> whatever they want ... in the canonical form that we use to propose
> such things ...
>
>    draft-allman-tcpm-no-initwin-00.txt


Admiral Farragut would be pleased :)

In the vein of giving an inch and they will ask to take a mile, if such 
a paced, large IW is OK for the start of a connection, would it also be 
OK for "slow start after idle?"  Or perhaps the lesser of that IW and 
cwnd when the connection went idle.

rick jones


From nobody Tue Nov 24 17:01:14 2015
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From: Theodore V Faber <theodore.v.faber@aero.org>
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> Admiral Farragut would be pleased :)

Ha!

> In the vein of giving an inch and they will ask to take a mile, if
> such a paced, large IW is OK for the start of a connection, would it
> also be OK for "slow start after idle?"  Or perhaps the lesser of
> that IW and cwnd when the connection went idle.

Perhaps.  The difference is that in this case the host has *some*
notion of the path it is traversing.  But, at some point that notion
is so dated that indeed treating it like a brand new connection from
a CC standpoint seems OK to me.  But, there are probably more
considerations here than at the start of a connection.

allman




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From nobody Tue Nov 24 22:13:42 2015
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Hmm. Is there any guidelines to apply this? We don't need TCB sharing for this?
Also, why it's BCP?

Thanks,
--
Yoshi

On Tue, Nov 24, 2015 at 9:43 AM, Mark Allman <mallman@icir.org> wrote:
>
> A proposal for allowing end hosts to set their initial cwnd to
> whatever they want ... in the canonical form that we use to propose
> such things ...
>
>   draft-allman-tcpm-no-initwin-00.txt
>
> allman
>
>
> --
> http://www.icir.org/mallman/
>
>
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>


From nobody Wed Nov 25 01:46:20 2015
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From: SALLANTIN Renaud <renaud.sallantin@thalesaleniaspace.com>
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In 1(C), I think that "pacing" needs to be defined.

I am in favor of a lower and an upper limit for the pacing time between 2 s=
egments transmission.

If the number of segments you have to send across the RTT makes the pacing =
time lower than a minimal value (TBD), then we'd better not pace.=20
Indeed, pacing adds an unnecessary complexity.=20

If the number of segments you have to send across the RTT makes the pacing =
time higher than a maximal value (TBD), then we'd better use this upper lim=
it as pacing time.
Indeed, after a certain value, increasing again the pacing time between 2 s=
egments transmission is useless and harmful:
*it does not reduce the congestion impact on your segments transmission;=20
*but it adds an unnecessary (and long) delay in case of long RTT. For examp=
le, pacing without upper limit an IW of 10 segments in satcom adds around 5=
00ms...


I know that many points have to be discussed, and that's why we would be in=
terested in a draft on the Pacing.


[@@ THALES ALENIA SPACE INTERNAL @@]


-----Message d'origine-----
De=A0: tcpm [mailto:tcpm-bounces@ietf.org] De la part de Yoshifumi Nishida
Envoy=E9=A0: mercredi 25 novembre 2015 07:14
=C0=A0: Mark Allman
Cc=A0: tcpm@ietf.org
Objet=A0: Re: [tcpm] removing TCP's initial window

Hmm. Is there any guidelines to apply this? We don't need TCB sharing for t=
his?
Also, why it's BCP?

Thanks,
--
Yoshi

On Tue, Nov 24, 2015 at 9:43 AM, Mark Allman <mallman@icir.org> wrote:
>
> A proposal for allowing end hosts to set their initial cwnd to=20
> whatever they want ... in the canonical form that we use to propose=20
> such things ...
>
>   draft-allman-tcpm-no-initwin-00.txt
>
> allman
>
>
> --
> http://www.icir.org/mallman/
>
>
>
>
>
> _______________________________________________
> 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


From nobody Wed Nov 25 02:19:47 2015
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I very much agree about TCB sharing.

On the other hand, I also very much agree with the first sentiment =
expressed in the "reasoning" list:

***
    (a) The author thinks that talking about the initial window for the
        better part of two decades is probably enough.  And, definitely
        boring.
***

So all in all, I appreciate this proposal a lot  (and admittedly, I'm =
also becoming a fan of Mark's combined sense of humor and system design  =
- I still think that tcpx2 was the best solution ever proposed to solve =
header space issues).

Cheers,
Michael


> On 25 Nov 2015, at 07:13, Yoshifumi Nishida <nishida@sfc.wide.ad.jp> =
wrote:
>=20
> Hmm. Is there any guidelines to apply this? We don't need TCB sharing =
for this?
> Also, why it's BCP?
>=20
> Thanks,
> --
> Yoshi
>=20
> On Tue, Nov 24, 2015 at 9:43 AM, Mark Allman <mallman@icir.org> wrote:
>>=20
>> A proposal for allowing end hosts to set their initial cwnd to
>> whatever they want ... in the canonical form that we use to propose
>> such things ...
>>=20
>>  draft-allman-tcpm-no-initwin-00.txt
>>=20
>> allman
>>=20
>>=20
>> --
>> http://www.icir.org/mallman/
>>=20
>>=20
>>=20
>>=20
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Wed Nov 25 04:47:49 2015
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> Hmm. Is there any guidelines to apply this?

Not sure I follow.

(I mean, clearly, it isn't fleshed out very well.  It was just a
sketch to see if there are any actual legs to it.)

> We don't need TCB sharing for this?

Nope.

> Also, why it's BCP?

Oh, sorry.  That is because I quickly re-used the boilerplate from
something else.

allman




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Few thoughts -



1. Following statement is probably not true in many cases,

   e.g., cellular and satellite access networks.

   " (c) An overly aggressive IW is likely to congestion (sic) local networ=
ks

        before burdening remote portions of the path"

   An overly aggressive IW will cause congestion at cellular/satellite head=
-end.



2. Following rationale is a bit weak -

   " (g) Ultimately, being egregiously overly aggressive will not be in

    the sender's best interest---e.g., there will be a fight for

    local resources among the sender's own connections---and

    therefore there is an incentive to be reasonable."



   If being overly aggressive does not affect the sender's own local resour=
ces,

   but affects remote resources instead, a sender might feel incentivized t=
o be

   more aggressive than others, perhaps resulting in an IW-arms-race.



3. Rule (3) in section 1 describes actions after packet drops. We presume i=
t applies to

   ECN marks as well.



4. It would seem logical that IW simply not be a one-size-fits-all value fo=
r a given server.

   A server serves many clients with different time-varying path characteri=
stics.

   Making IW a function of the client path characteristics, perhaps based o=
n TCB sharing

   and/or cached historical information and/or configured parameters would =
seem to be prudent.



Anil



-----Original Message-----
From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Mark Allman
Sent: Wednesday, November 25, 2015 7:48 AM
To: Yoshifumi Nishida
Cc: tcpm@ietf.org
Subject: Re: [tcpm] removing TCP's initial window





> Hmm. Is there any guidelines to apply this?



Not sure I follow.



(I mean, clearly, it isn't fleshed out very well.  It was just a sketch to =
see if there are any actual legs to it.)



> We don't need TCB sharing for this?



Nope.



> Also, why it's BCP?



Oh, sorry.  That is because I quickly re-used the boilerplate from somethin=
g else.



allman







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<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;">Few thoughts -<o:p></o:p></spa=
n></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;">1. Following statement is prob=
ably not true in many cases,
<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><span style=3D"mso-spacerun:ye=
s">&nbsp;&nbsp;&nbsp;</span>e.g., cellular and satellite access networks.<o=
:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><span style=3D"mso-spacerun:ye=
s">&nbsp;&nbsp;
</span>&quot; (c) An overly aggressive IW is likely to congestion (sic) loc=
al networks<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><span style=3D"mso-spacerun:ye=
s">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span>before burdening remote portions of the path&quot;<o:p></o:p></span>=
</p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><span style=3D"mso-spacerun:ye=
s">&nbsp;&nbsp;
</span>An overly aggressive IW will cause congestion at cellular/satellite =
head-end.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;">2. Following rationale is a bi=
t weak -<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><span style=3D"mso-spacerun:ye=
s">&nbsp;&nbsp;
</span>&quot; (g) Ultimately, being egregiously overly aggressive will not =
be in <o:p>
</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><span style=3D"mso-spacerun:ye=
s">&nbsp;&nbsp;&nbsp;&nbsp;</span>the sender's best interest---e.g., there =
will be a fight for<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><span style=3D"mso-spacerun:ye=
s">&nbsp;&nbsp;&nbsp;
</span>local resources among the sender's own connections---and<o:p></o:p><=
/span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><span style=3D"mso-spacerun:ye=
s">&nbsp;&nbsp;&nbsp;
</span>therefore there is an incentive to be reasonable.&quot;<o:p></o:p></=
span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><span style=3D"mso-spacerun:ye=
s">&nbsp;&nbsp;
</span>If being overly aggressive does not affect the sender's own local re=
sources,<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><span style=3D"mso-spacerun:ye=
s">&nbsp;&nbsp;
</span>but affects remote resources instead, a sender might feel incentiviz=
ed to be<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><span style=3D"mso-spacerun:ye=
s">&nbsp;&nbsp;
</span>more aggressive than others, perhaps resulting in an IW-arms-race.<o=
:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;">3. Rule (3) in section 1 descr=
ibes actions after packet drops. We presume it applies to<o:p></o:p></span>=
</p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><span style=3D"mso-spacerun:ye=
s">&nbsp;
</span><span style=3D"mso-spacerun:yes">&nbsp;</span>ECN marks as well.<o:p=
></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;">4. It would seem logical that =
IW simply not be a one-size-fits-all value for a given server.<o:p></o:p></=
span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><span style=3D"mso-spacerun:ye=
s">&nbsp;&nbsp;
</span>A server serves many clients with different time-varying path charac=
teristics.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><span style=3D"mso-spacerun:ye=
s">&nbsp;&nbsp;
</span>Making IW a function of the client path characteristics, perhaps bas=
ed on TCB sharing<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><span style=3D"mso-spacerun:ye=
s">&nbsp;&nbsp;
</span>and/or cached historical information and/or configured parameters wo=
uld seem to be prudent.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:9.0pt;mso-bidi-font-size=
:10.5pt;font-family:&quot;Courier New&quot;">Anil<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">-----Original Message-----<br>
From: tcpm [mailto:tcpm-bounces@ietf.org] On Behalf Of Mark Allman<br>
Sent: Wednesday, November 25, 2015 7:48 AM<br>
To: Yoshifumi Nishida<br>
Cc: tcpm@ietf.org<br>
Subject: Re: [tcpm] removing TCP's initial window</p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt; Hmm. Is there any guidelines to apply this?<=
o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Not sure I follow.<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">(I mean, clearly, it isn't fleshed out very well.=
<span style=3D"mso-spacerun:yes">&nbsp;
</span>It was just a sketch to see if there are any actual legs to it.)<o:p=
></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt; We don't need TCB sharing for this?<o:p></o:=
p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Nope.<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt; Also, why it's BCP?<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Oh, sorry.<span style=3D"mso-spacerun:yes">&nbsp;=
 </span>That is because I quickly re-used the boilerplate from something el=
se.<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">allman<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

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> 2. Following rationale is a bit weak -
>=20
> " (g) Ultimately, being egregiously overly aggressive will not be in=20
> the sender's best interest---e.g., there will be a fight for
> local resources among the sender's own connections---and
> therefore there is an incentive to be reasonable."
>=20
> If being overly aggressive does not affect the sender's own local
> resources, but affects remote resources instead, a sender might
> feel incentivized to be more aggressive than others, perhaps
> resulting in an IW-arms-race.

Yeah- but the thing is that IW isn't the overall driver here.  That
is, I could try to dump as much as the advertised window would
allow.  But, if I see loss then I have to revert back.  And, I have
to repair that loss.  And, so ultimately there is a check here.

> 3. Rule (3) in section 1 describes actions after packet drops. We
> presume it applies to
>=20
> ECN marks as well.

(Sure.)

> 4. It would seem logical that IW simply not be a one-size-fits-all
> value for a given server.=20=20

Right.  There is no one-size-fits-all.  And, so we---and I am
clearly as much to blame as anyone---have chosen a
one-size-fits-nobody approach.

> A server serves many clients with different time-varying path
> characteristics.  Making IW a function of the client path
> characteristics,

Right.  The ideal IW depends on the time-varying conditions.  The
problem is that when starting up we don't have much of an idea about
this.

> perhaps based on TCB sharing and/or cached historical information
> and/or configured parameters would seem to be prudent.

I don't disagree.  Nothing in the draft precludes any of this.  That
is, the draft frees implementations to use whatever they deem as
reasonable for the initial window.  That could be a constant based
on their own upstream.  Or, some function based on the RTT during
the 3WHS.  Or, based on historical characteristics of the given
remote IP / routed block.  Or, whatever.  There are lots of ideas in
the air for arriving at something sensible.  And, there is not one
"sensible" for all situations.  So, let's stop pretending there is
some magic number that solves this and just let implementations /
instantiations puzzle through it for themselves.

allman


=2D-
http://www.icir.org/mallman/
@mallman_icsi




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Why not pace the IW of less than 10 segments?=20


[@@ THALES ALENIA SPACE INTERNAL @@]


-----Message d'origine-----
De=A0: tcpm [mailto:tcpm-bounces@ietf.org] De la part de SALLANTIN Renaud
Envoy=E9=A0: mercredi 25 novembre 2015 10:46
=C0=A0: 'Yoshifumi Nishida'; 'Mark Allman'
Cc=A0: 'tcpm@ietf.org'
Objet=A0: Re: [tcpm] removing TCP's initial window

In 1(C), I think that "pacing" needs to be defined.

I am in favor of a lower and an upper limit for the pacing time between 2 s=
egments transmission.

If the number of segments you have to send across the RTT makes the pacing =
time lower than a minimal value (TBD), then we'd better not pace.=20
Indeed, pacing adds an unnecessary complexity.=20

If the number of segments you have to send across the RTT makes the pacing =
time higher than a maximal value (TBD), then we'd better use this upper lim=
it as pacing time.
Indeed, after a certain value, increasing again the pacing time between 2 s=
egments transmission is useless and harmful:
*it does not reduce the congestion impact on your segments transmission; *b=
ut it adds an unnecessary (and long) delay in case of long RTT. For example=
, pacing without upper limit an IW of 10 segments in satcom adds around 500=
ms...


I know that many points have to be discussed, and that's why we would be in=
terested in a draft on the Pacing.


[@@ THALES ALENIA SPACE INTERNAL @@]


-----Message d'origine-----
De=A0: tcpm [mailto:tcpm-bounces@ietf.org] De la part de Yoshifumi Nishida
Envoy=E9=A0: mercredi 25 novembre 2015 07:14
=C0=A0: Mark Allman
Cc=A0: tcpm@ietf.org
Objet=A0: Re: [tcpm] removing TCP's initial window

Hmm. Is there any guidelines to apply this? We don't need TCB sharing for t=
his?
Also, why it's BCP?

Thanks,
--
Yoshi

On Tue, Nov 24, 2015 at 9:43 AM, Mark Allman <mallman@icir.org> wrote:
>
> A proposal for allowing end hosts to set their initial cwnd to=20
> whatever they want ... in the canonical form that we use to propose=20
> such things ...
>
>   draft-allman-tcpm-no-initwin-00.txt
>
> allman
>
>
> --
> http://www.icir.org/mallman/
>
>
>
>
>
> _______________________________________________
> 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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> Why not pace the IW of less than 10 segments?=20

I didn't say not to.  I said for IW > 10 you MUST.

If you want to, great.  But, we have already agreed on IW10 without
pacing and so it seems like we don't need to add a requirement for
pacing there.

allman




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From nobody Wed Nov 25 10:44:37 2015
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I like the assumptions, esp. that AQM is widely deployed, but they're
just that - assumptions.

I'd like to see some evidence this approach is safe first, because
pacing traffic out seems more likely to mask a "tragedy of the commons"
issue than bursting would have, esp. when AQM isn't available.

Joe

On 11/24/2015 9:43 AM, Mark Allman wrote:
> 
> A proposal for allowing end hosts to set their initial cwnd to
> whatever they want ... in the canonical form that we use to propose
> such things ...
> 
>   draft-allman-tcpm-no-initwin-00.txt
> 
> allman
> 
> 
> --
> http://www.icir.org/mallman/
> 
> 
> 
> 
> 
> 
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
> 


From nobody Wed Nov 25 10:56:27 2015
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On 20/11/15 14:40, SALLANTIN Renaud wrote:
> ·         Is there room to increase even more the Initial Window? Several major players already use IW=32 or higher (with and without pacing), that would really help for satellite communications.  But the question here is how to mitigate the burst impact?

In combination with larger IW, would there be benefit in packets
being marked for "no roundtrip yet"?  During startup of a flow,
a bottleneck router could then fire a source-quench.

This would at least help the case where the bottleneck
is close to the source.
-- 
Cheers,
  Jeremy


From nobody Wed Nov 25 11:24:17 2015
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On 11/25/2015 10:43 AM, Joe Touch wrote:
> I like the assumptions, esp. that AQM is widely deployed, but they're
> just that - assumptions.
>
> I'd like to see some evidence this approach is safe first, because
> pacing traffic out seems more likely to mask a "tragedy of the commons"
> issue than bursting would have, esp. when AQM isn't available.

Much as I'd like to see bursting the IW (Damn the torpedoes and all 
that) because it would enable using TSO/GSO, why would pacing mask a 
tragedy of the AQM-less commons?  If there is no AQM available at the 
bottleneck, it will either trigger a drop at the tail of the queue or 
not, and if it does not, all we have is a slight (one RTT?) delay before 
the sender "really" dumps a load on the bottleneck.

Heck, if we are lucky (well, are we, ever?) that paced IW if it does get 
through without drops may complete the sending to be done anyway, and if 
it doesn't, it suggests there was going to be a queue fill/overflow if 
we started with a classic, smaller IW anyway.  If not during the 
classic, smaller IW itself then during the subsequent RTTs.

rick jones


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On 11/25/2015 11:24 AM, Rick Jones wrote:
> On 11/25/2015 10:43 AM, Joe Touch wrote:
>> I like the assumptions, esp. that AQM is widely deployed, but they're
>> just that - assumptions.
>>
>> I'd like to see some evidence this approach is safe first, because
>> pacing traffic out seems more likely to mask a "tragedy of the commons"
>> issue than bursting would have, esp. when AQM isn't available.
> 
> Much as I'd like to see bursting the IW (Damn the torpedoes and all
> that) because it would enable using TSO/GSO, why would pacing mask a
> tragedy of the AQM-less commons?  If there is no AQM available at the
> bottleneck, it will either trigger a drop at the tail of the queue or
> not, and if it does not, all we have is a slight (one RTT?) delay before
> the sender "really" dumps a load on the bottleneck.

Losses can occur all along a path, they don't have to occur only at one
bottleneck. If you burst, you're more likely to see a drop in one of
those locations than if you pace things out.

In a sense, bursting is like a heavy-handed version of packet-pair in
terms of measuring the drop behavior along the path - you're more likely
to create a situation where you cause a drop of your own packets,
because you're the dominant occupant of the queues at a given router.

> Heck, if we are lucky (well, are we, ever?) that paced IW if it does get
> through without drops may complete the sending to be done anyway, and if
> it doesn't, it suggests there was going to be a queue fill/overflow if
> we started with a classic, smaller IW anyway.  If not during the
> classic, smaller IW itself then during the subsequent RTTs.

Except that if you run over several RTTs, you run a greater chance of
experiencing a drop during one of those RTTs on the way to a larger window.

Joe


From nobody Wed Nov 25 11:59:15 2015
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> On 24 Nov 2015, at 18:43, Mark Allman <mallman@icir.org> wrote:
> 
> 
> A proposal for allowing end hosts to set their initial cwnd to
> whatever they want ... in the canonical form that we use to propose
> such things ...
Hi Marc,

I really like and support this document. Wouldn't is make sense
not to restrict this document to TCP, but also cover SCTP?

Best regards
Michael
> 
>  draft-allman-tcpm-no-initwin-00.txt
> 
> allman
> 
> 
> --
> http://www.icir.org/mallman/
> 
> 
> 
> 
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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> I really like and support this document. Wouldn't is make sense
> not to restrict this document to TCP, but also cover SCTP?

Absolutely.  While I am kicking one beehive, I might as well kick
two! :-)

It is fun to know that I can still make enough noise to make people
think.

allman




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From nobody Wed Nov 25 13:30:34 2015
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Subject: Re: [tcpm] removing TCP's initial window
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One other point to consider:

	If you're seeing loss, it also means you might be causing
	loss for others too.

That's why we increase the window gently until we know better, and why
getting one packet through might be permission to send another, but
should not be interpreted as being safe to send a lot.

Joe

On 11/25/2015 10:43 AM, Joe Touch wrote:
> I like the assumptions, esp. that AQM is widely deployed, but they're
> just that - assumptions.
> 
> I'd like to see some evidence this approach is safe first, because
> pacing traffic out seems more likely to mask a "tragedy of the commons"
> issue than bursting would have, esp. when AQM isn't available.
> 
> Joe
> 
> On 11/24/2015 9:43 AM, Mark Allman wrote:
>>
>> A proposal for allowing end hosts to set their initial cwnd to
>> whatever they want ... in the canonical form that we use to propose
>> such things ...
>>
>>   draft-allman-tcpm-no-initwin-00.txt
>>
>> allman
>>
>>
>> --
>> http://www.icir.org/mallman/
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>


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Subject: Re: [tcpm] How to improve TCP efficiency in a large RTT context?
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On 11/25/2015 10:56 AM, Jeremy Harris wrote:
> On 20/11/15 14:40, SALLANTIN Renaud wrote:
>> ·         Is there room to increase even more the Initial Window? Several major players already use IW=32 or higher (with and without pacing), that would really help for satellite communications.  But the question here is how to mitigate the burst impact?
> 
> In combination with larger IW, would there be benefit in packets
> being marked for "no roundtrip yet"?  During startup of a flow,
> a bottleneck router could then fire a source-quench.

See RF6633.

Those messages are deprecated.

> This would at least help the case where the bottleneck
> is close to the source.

The best we have is ECN marking, but that helps only for the return path
ACKs, i.e., that'll take a round trip to react to.

Joe


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Subject: Re: [tcpm] How to improve TCP efficiency in a large RTT context?
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On 25/11/15 21:33, Joe Touch wrote:
> 
> 
> On 11/25/2015 10:56 AM, Jeremy Harris wrote:
>> On 20/11/15 14:40, SALLANTIN Renaud wrote:
>>> ·         Is there room to increase even more the Initial Window? Several major players already use IW=32 or higher (with and without pacing), that would really help for satellite communications.  But the question here is how to mitigate the burst impact?
>>
>> In combination with larger IW, would there be benefit in packets
>> being marked for "no roundtrip yet"?  During startup of a flow,
>> a bottleneck router could then fire a source-quench.
> 
> See RF6633.
> 
> Those messages are deprecated.

Large IW isn't permitted, either.  We're discussing changes.
-- 
Cheers,
  Jeremy



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On 11/25/2015 1:47 PM, Jeremy Harris wrote:
> On 25/11/15 21:33, Joe Touch wrote:
>>
>>
>> On 11/25/2015 10:56 AM, Jeremy Harris wrote:
>>> On 20/11/15 14:40, SALLANTIN Renaud wrote:
>>>> ·         Is there room to increase even more the Initial Window? Several major players already use IW=32 or higher (with and without pacing), that would really help for satellite communications.  But the question here is how to mitigate the burst impact?
>>>
>>> In combination with larger IW, would there be benefit in packets
>>> being marked for "no roundtrip yet"?  During startup of a flow,
>>> a bottleneck router could then fire a source-quench.
>>
>> See RF6633.
>>
>> Those messages are deprecated.
> 
> Large IW isn't permitted, either.  We're discussing changes.

We don't typically "un-deprecate" things, though.

Source quench is deprecated for a number of reasons, summarized in RFC
6633. It'd be useful to address those before assuming it can simply just
start existing.

Joe


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From: SALLANTIN Renaud <renaud.sallantin@thalesaleniaspace.com>
To: "mallman@icir.org" <mallman@icir.org>, "tcpm@ietf.org" <tcpm@ietf.org>
Date: Thu, 26 Nov 2015 11:30:07 +0100
Thread-Topic: [tcpm] removing TCP's initial window
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Subject: Re: [tcpm] removing TCP's initial window
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Unlike for Jumpstart, you do not modify the TCP behavior after the IW trans=
mission.=20
Don't you think that, in some cases, double the IW may trigger some congest=
ion?=20
Would it be possible to refine the mechanism, and for example, determine wh=
at should be used in the following RTTs  (slow start or CC) in function of =
the IW size?=20

[@@ THALES ALENIA SPACE INTERNAL @@]


-----Message d'origine-----
De=A0: tcpm [mailto:tcpm-bounces@ietf.org] De la part de Mark Allman
Envoy=E9=A0: mardi 24 novembre 2015 18:43
=C0=A0: tcpm@ietf.org
Objet=A0: [tcpm] removing TCP's initial window


A proposal for allowing end hosts to set their initial cwnd to whatever the=
y want ... in the canonical form that we use to propose such things ...

  draft-allman-tcpm-no-initwin-00.txt

allman


--
http://www.icir.org/mallman/





From nobody Thu Nov 26 11:28:00 2015
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From: Pasi Sarolahti <pasi.sarolahti@iki.fi>
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Subject: Re: [tcpm] TCPM notes from IETF-94
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Hi Michael,

Thanks for clarification. I will correct this part in the minutes, along =
with the nit and couple of other updates, and will upload a new version =
soon.

- Pasi


On 18 Nov 2015, at 10:48, Michael Welzl <michawe@ifi.uio.no> wrote:

> Hi,
>=20
> nit: s/what is write/what is right
>=20
> Else: the transcription of what David Black said to Naeem Khademi on =
the mic is misleading - it misses key bits that make it seem like a =
different statement than it was:
>=20
>=20
> The minutes say:
> ***
> David Black: RFC3168 says: don't do anything that breaks the Internet. =
If there is a ECE, react to it the same way as with packet-loss.
> ***
>=20
> ... which makes it sound like this would be his advice and he would =
opposed to experimental alternatives. However, what he really said is =
(transcribed from Meetecho:
> =
http://recs.conf.meetecho.com/Playout/watch.jsp?recording=3DIETF94_TCPM&ch=
apter=3Dchapter_1   at approx. 0:31):
>=20
> ***
> As one of the authors of RFC 3168: I agree that encouraging =
Experimental for congestion control is a really good idea.
> 3168 was written for the broad internet, hence contains some very =
strict language that basically said don't do anything that would break =
things. We had people react to congestion if you CE-mark as if you're =
dropping a packet, we said: the only thing we know for certain is safe =
right now - this was a long time ago - was: react to it as you would =
react to a packet drop. I'm not opposed to interesting innovative things =
being done under the guise of experimentation, I wanna provide some =
history for why 3168 says what it does.
> ***
>=20
> The last sentence matters - it's really quite a different thing.
>=20
> Cheers,
> Michael
>=20
>=20
>=20
>=20
>=20
>> On 16 Nov 2015, at 21:23, Pasi Sarolahti <pasi.sarolahti@iki.fi> =
wrote:
>>=20
>> Hi,
>>=20
>> The minutes of the Yokohama TCPM meeting are available at =
https://www.ietf.org/proceedings/94/minutes/minutes-94-tcpm. Many thanks =
to Christoph and David for compiling them!
>>=20
>> If you have corrections to make, please send them to list, or =
directly to chairs.
>>=20
>> - Pasi
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From nobody Mon Nov 30 06:30:39 2015
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From: Mark Allman <mallman@icir.org>
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> Unlike for Jumpstart, you do not modify the TCP behavior after the
> IW transmission.

I am not sure if I follow.

Regardless of what the IW is or how that is arrived at I think it is
perfectly fine to send 2 * IW in the second RTT of data transmission
if there was no loss in the first window of data.  That is just
traditional slow start.

And, if an implementation wanted to somehow decide to transmit less
than 2 * IW in the second RTT then that would also be fine with me.
(I.e., you are always free to be more conservative than the spec
allows.)=20

These two bits follow directly from the spec and would seem
non-controversial to me.

Does that answer the question?

allman




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From nobody Mon Nov 30 08:02:40 2015
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> One other point to consider:
>=20
> 	If you're seeing loss, it also means you might be causing
> 	loss for others too.

The dynamics are not that simple.

E.g., if the other folks had started faster they might actually be
out of the way when I start and not still sending.

E.g., if I am seeing loss then that is bad for me.  And, I don't
have incentive to do bad things to myself.  If the proposal would
cause pain for others and not me then perhaps I'd buy this point
more.  But, I don't see that it will.  So, in some sense my own self
interest also serves to protect others.

allman




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On 11/30/2015 8:02 AM, Mark Allman wrote:
> 
>> One other point to consider:
>>
>> 	If you're seeing loss, it also means you might be causing
>> 	loss for others too.
> 
> The dynamics are not that simple.
> 
> E.g., if the other folks had started faster they might actually be
> out of the way when I start and not still sending.

They might. They might not.

I'll wager that my "might" is a "probably", though, because the chance
that paced transmission is exactly the only thing dropped at a buffer is
vanishingly small, given AQM.

However, again this is a place where experiments would help, and the
status-quo is the only viable position until we have evidence that
cross-drops aren't the likely result.

Joe

