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From: Martin Stiemerling <mls.ietf@gmail.com>
To: tsvwg@ietf.org
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Date: Wed, 1 Jun 2016 15:19:27 +0200
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Subject: [tsvwg] Review of draft-ietf-tsvwg-gre-in-udp-encap-11
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Dear all,

Please find here my review of draft-ietf-tsvwg-gre-in-udp-encap-11.

General: The draft is ready to with a few smaller issues.

Issues:
- The text says in multiple places "avoid congestion".
However, I do not see that, for instance, rate limiters, will avoid 
congestion. A rate limiter will cause congestion for affected flows but 
potentially keep congestion away from other parts of the network.
(A bit more) correct would be something along the line of "to detect 
congestion and to mitigate congestion".

- This document uses RFC5405bis as reference in many places, though RFC 
5405 is the only stable anchor at this point of time. I am not sure how 
far RFC 5405bis is right now or asked differently, is there a need to 
wait for RFC 5405 to complete?

- I do disagree with "IP-traffic can be assumed to be 
congestion-controlled.", even though it is stated in RFC 5405, as there 
is a lot of IP traffic that is not congestion controlled, such as 
multicast traffic...
Section 4, even states that multicast and broadcast traffic can be 
carried over GRE-UDP! None of both is in general congestion controlled!

- Section 2.1.1 on page 5, point 7 "it MUST set the same DSCP value to 
all fragments." From what value the DSCP must be set? This isn't stated 
here.

- Section 4.1, first sentence:
"   Regarding packet fragmentation, an encapsulator/decapsulator SHOULD
    be compliant with [RFC7588] and perform fragmentation before the
    encapsulation."
RFC 7588 is mentioned here and the SHOULD implies that a 
reader/implementer has to read RFC 7588. However, RFC 7588 is not listed 
under normative references and it is anyhow an informational document, 
which would require a down-ref in the last call.


Editorials:
- There are multiple occurences in the text "IANA Note: Replace TBD1 and 
TBD2 with the IANA-assigned numbers". I suggest to remove these notes 
and just to say in the IANA considerations section that IANA is 
requested to replace these labels with the appropriate port numbers. And 
tell in the IANA considerations section where to find these labels.

- TBD1/TBD2 in figures: replace with a different labels and say that 
these labels do correspondent to the IANA assigned port numbers. The 
figures do not need to state the actual port numbers, as they will do, 
if the text and the IANA text stays as it is right now.

- Section 1.1: Move TMCE up in the list, as it used in the terminology 
even before it is defined.

Thanks,

   Martin


From nobody Wed Jun  1 06:24:52 2016
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Folks-

A long time ago we started a document in TCPM that sought to
synthesize what we have figured out over the years about the RTO
estimator.  Basically, we have learned that an RTO estimator needs a
few aspects for network safety, but does not need to be nailed down
as concretely as we have done in the past (e.g., in RFC 6298).

As this effort evolved it seems like the high-level principles and
requirements are actually not tied to TCP, but are in fact protocol
agnostic.  E.g., before we have some notion of an RTT across a
specific path, an RTO scheme needs to use some initial length of
time to trigger a retransmit.  This equally applies to TCP, SCTP,
RTP, UDP-based apps, etc.

The draft name includes "ietf-tcpm", but is actually broader.  My
own assessment is that the draft enjoys pretty wide consensus in
TCPM.  But, I am looking for technical comments to ensure that the
document doesn't suffer from TCP blinders as it has evolved.

The draft is:

  https://tools.ietf.org/html/draft-ietf-tcpm-rto-consider-03

Happy to hear comments on the list or off the list.

Many thanks!

allman


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




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From: "Black, David" <david.black@emc.com>
To: Martin Stiemerling <mls.ietf@gmail.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: [tsvwg] Review of draft-ietf-tsvwg-gre-in-udp-encap-11
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From nobody Thu Jun  2 02:06:35 2016
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From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: Re: [tsvwg] soliciting comments on draft-ietf-tcpm-rto-consider
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Mark,

As I previously said, I think this is a useful document. However,  I do 
think we need to be careful here when we use terms like "retransmission 
timer algorithms" and "RTO". See below.


On 01/06/2016 14:24, Mark Allman wrote:
>
> Folks-
>
> A long time ago we started a document in TCPM that sought to
> synthesize what we have figured out over the years about the RTO
> estimator.  Basically, we have learned that an RTO estimator needs a
> few aspects for network safety, but does not need to be nailed down
> as concretely as we have done in the past (e.g., in RFC 6298).
>
> As this effort evolved it seems like the high-level principles and
> requirements are actually not tied to TCP, but are in fact protocol
> agnostic.  E.g., before we have some notion of an RTT across a
> specific path, an RTO scheme needs to use some initial length of
> time to trigger a retransmit.  This equally applies to TCP, SCTP,
> RTP, UDP-based apps, etc.
>
> The draft name includes "ietf-tcpm", but is actually broader.  My
> own assessment is that the draft enjoys pretty wide consensus in
> TCPM.  But, I am looking for technical comments to ensure that the
> document doesn't suffer from TCP blinders as it has evolved.
>
> The draft is:
>
>   https://tools.ietf.org/html/draft-ietf-tcpm-rto-consider-03
>
> Happy to hear comments on the list or off the list.
>
> Many thanks!
>
> allman
>
>
> --
> http://www.icir.org/mallman/
>
For TCP/SCTP these terms are fairly well-specified for TCP/SCTP and I 
also agree on what TCP/SCTP should do. The draft uses the term "RTO" 
... As I understand, an RTO expires in TCP  when there is a lack of 
confirmed delivery across a network path. To me, this detects times when 
the sender can no longer be confident that data is being received -- 
i.e., there is risk that congestion has been detected. To be "safe", the 
network needs to reduce the volume of traffic it sends in response to 
detecting this potential congestion. Exponential backoff of the RTO is 
an important part of the TCP response.
However, to make these recommendations applicable to other protocols, I 
think we need to agree about what we mean by timers, retransmission and 
the term "RTO" in the context of all protocols, and make sure the draft 
says what it understands by these terms.

I'll try to explain why I think this is a new problem for UDP - my hope 
is that this will paint a picture against which we can agree some simple 
definitions that apply beyond TCP, it's long and tutorial (and I'm 
typing over breakfast coffee, so maybe has some oversights, but I'm 
hoping the resulting introductory text is concise, once we understand 
each other's starting points).

---

There are very important differences in the way TCP and UDP handle 
retransmission and timers:

* TCP organises data as a reliable byte stream, UDP does not. This means 
that a modern TCP flow typically can (and does) use other methods to 
detect loss and hence retransmit, and the timer-based RTO mechanism is 
not usually the primary retransmission mechanism. Current thinking is 
that this should not be triggered often in normal operation.

* UDP applications can use timers for retransmission: They need to 
respond "similarly" to TCP when they detect loss, in that they must 
reduce their volume of traffic sent in response when they detect 
congestion. They also can use timers for various reasons. Timers can 
also provide retransmission (in UDP, this is not always of the original 
data). Transmission (and retransmission) are often also controlled by 
timers. The term RTO is not used widely in UDP (and note defined in 
DCCP). this sort of retransmission timer expiry is not necessarily 
equivalent to a TCP RTO.

* Applications can attach different priorities to different UDP 
datagrams they send: All data does not need to be equal, some data may 
be considered more important or more time-critical (e.g., think of data 
from a video codec using UDP). It's quite OK for such an application to 
set a short delivery deadline for some data (more responsive), and to 
care much less about loss of other data. If a timer triggers, the app 
can decide which data to retransmit (or send some sort of repair request 
to compensate for the loss), and when to send this. This includes the 
possibility to use timers for critical datagrams, or to use other 
methods. The thing I think that is important is that an app must reduce 
the total volume of data sent went it experiences loss, it is nothing to 
do with whether timers are used. A re-ransmission timeout can be 
different to a TCP RTO.

I disagree for UDP with: (draft-ietf-tcpm-rto-consider-03.txt says: 
"Ideally, if we know a segment will be lost before reaching the 
destination, a second copy of it would be sent immediately after the 
first transmission." The draft says this is a tradeoff, I think this 
becomes one of many design decisions for a partially reliable datagram 
service. The second copy part is purely dependent on the app, etc)

* Frequency of RTT measurement: Many UDP apps do not make per RTT 
measurements of the path. These have some functions that resemble RTO 
expiry in TCP, but they are not the same. They mayuse  less frequent 
feedback messages to confirm the path they use (e.g., RTCP often used 
with reporting intervals greater than 1 RTT). This is usually a 
conscious decision based on application processing or network capacity 
tradeoff. A lack of reception feedback still needs to be regarded as a 
trigger for a congestion response (aka rate reduction or flow 
termination). These methods work on longer timescales than a TCP RTO 
would operate (and often react differently to a TCP CC , e.g., TFRC). 
This use seems to slightly contradict the current draft, (e.g., I doubt 
if there would be RAI consensus that RTP is wrong.)

* Low volume: Some UDP apps do not implement feedback indicating the RTT 
and volume of successful data transmitted across the path needs special 
consideration (RFC5405 talks about this, SIP is an example - this usage 
resembles TCP SYN transmission, and could probably be handled 
similarly). If the app does not know that data is successfully reaching 
the remote endpoint it needs to take other action (such an app can not 
calculate a safe reduction, e.g., halving the rate) - 
exponentially-backed off request messages from a starting value are 
commonly used in this way. This has similarities with the use by TCP of 
the RTO during Open, where a lack of feedback results in a RTO expiry. 
Methods are described in the UDP Guidelines (RFC5405)to cope with 
senders that send very few packets per RTT, or ones do not measure RTT, etc.

* Low volume: Some UDP applications use a keep alive message to confirm 
their path (STUN is an example, there are others). The IETF circuit 
breaker documents also define a set of last-resort methods for detecting 
what i will call persistent congestion for UDP. These methods work on 
longer timescales than a TCP RTO would operate (and react more 
vigourously than CC), but have similaeity in provide a method of last 
resort.  However, I think this use of timers I think does not fall 
within the scope of the document, but the Guidelines need to note these 
timers need to backoff.

* Backoff - and timer retransmission not harmful: UDP needs to avoid 
congestion collapse. For UDP, the best trade-off (whether it is worth 
risking to send a particular retransmission early or not) is best known 
by the application (e.g., for transaction-based apps - DNS?, exponential 
back-off for each failed retransmission would seem appropriate; in 
contrast for multimedia streaming,  fast retransmission of key data is 
often vital, and some unnecessary retransmissions may be desirable - 
even if that then means less lower priority data can be sent; there are 
likely other cases). If an app sends early, then it MUST be considered 
as a segment that is congestion-controlled (i.e., I think equivalent to 
consuming one MSS of cwnd, in TCP language -  the acknowledgment 
ambiguity that would happen in TCP should not occur in a protocol 
designed for this, that's a result of TCP choosing to use culmulative 
ACKs). To me, the first important thing is not whether timers are used 
to trigger retransmissions, but that data transmission avoids congestion 
collapse. The second important thing, is that this use does not confirm 
the path capacity - hence this does not replace the need for  a method 
of detecting congestion in the last resort. This sort of use seems 
outside the scope of your draft.

* efficiency and timers: I agree there can be an efficiency trade-off 
when using timers (noted as you say for TCP - true also of keep-alives, 
etc), if the RTT isn't measured accurately  and then not used correctly 
to calculate a suitable timeout, then early retransmissions will add 
volume to the data sent by an app. That absolutely needs to be taken 
into consideration. If all data were equal, it would be really wise to 
avoid sending data in early retransmissions (e.g., >RTT but <RTO) - 
because it may not be needed, and TCP may not account for such a 
retransmission using cwnd. Setting the RTO timeout then becomes a 
trade-off for TCP. UDP applications that do not have specific needs, 
should by default design ensure they use a SRTT with variance to set the 
limit on expected feedback time. UDP applications that do understand how 
to use timers near the RTT can do so (but then need to do CC as noted 
above).

---

I'm hopefully learning also as we discuss this, and will also try to 
clarify things with my co-editors in the RFC5405.bis document after I 
have worked through this topic a little further.

For the RTO draft, I think it is well worth trying to sort out the the 
scope for the document, and it is really important to define what is 
actually meant by a RTO, especially in the context of a UDP application.

Gorry










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From: Brian Trammell <ietf@trammell.ch>
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Subject: Re: [tsvwg] Last Call: <draft-ietf-tsvwg-rfc5405bis-13.txt> (UDP Usage Guidelines) to Best Current Practice
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Greetings, all,

Apologies for the late last call comment; I have only one, relatively =
minor. I hope it's still useful.

I understand that Section 3 was written to encourage application =
developers not to roll their own transports ("trust us when we say this =
is hard, this document is a list of reasons why") but as written it =
would seem to discourage transport innovation atop UDP (e.g. QUIC, the =
RTCWEB data channel, anything-over-PLUS), which I very much hope was not =
the intent. The problematic recommendation is in the second paragraph:

   These mechanisms are difficult to implement correctly.  For most
   applications, the use of one of the existing IETF transport protocols
   is the simplest method of acquiring the required mechanisms.  Doing
   so also avoids issues that protocols using a new IP protocol number
   face when being deployed over the Internet, where middleboxes that
   only support TCP and UDP are not rare.  Consequently, the RECOMMENDED
   alternative to the UDP usage described in the remainder of this
   section is the use of an IETF transport protocol such as TCP
   [RFC0793], Stream Control Transmission Protocol (SCTP) [RFC4960], and
   SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram
   Congestion Control Protocol (DCCP) [RFC4340] with its different
   congestion control types [RFC4341][RFC4342][RFC5622].

First, this paragraph ignores potential deployment issues with any of =
these other than TCP, which risks seeming out of touch, but this is a =
minor point and probably not worth a late edit. Second, I'm concerned =
this recommendation could be taken as broader than intended, against the =
definition of any new transport protocol encapsulated within UDP that =
performs substantially the same function as the listed protocols.

I think this can be made clearer by simply adding to the list of =
examples:

NEW:

   These mechanisms are difficult to implement correctly.  For most
   applications, the use of one of the existing IETF transport protocols
   is the simplest method of acquiring the required mechanisms.  Doing
   so also avoids issues that protocols using a new IP protocol number
   face when being deployed over the Internet, where middleboxes that
   only support TCP and UDP are not rare.  Consequently, the RECOMMENDED
   alternative to the UDP usage described in the remainder of this
   section is the use of an IETF transport protocol such as TCP
   [RFC0793], Stream Control Transmission Protocol (SCTP) [RFC4960], and
   SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram
   Congestion Control Protocol (DCCP) [RFC4340] with its different
   congestion control types [RFC4341][RFC4342][RFC5622], or transport
   protocols specified by the IETF in the future.

and removing the examples from the summary in section 7:

OLD:

   | SHOULD use a full-featured transport (TCP, SCTP, DCCP)  |         |

NEW:

   | SHOULD use a full-featured transport                    |         |

Thanks, cheers,

Brian


> On 18 May 2016, at 02:17, The IESG <iesg-secretary@ietf.org> wrote:
>=20
>=20
> The IESG has received a request from the Transport Area Working Group =
WG
> (tsvwg) to consider the following document:
> - 'UDP Usage Guidelines'
>  <draft-ietf-tsvwg-rfc5405bis-13.txt> as Best Current Practice
>=20
> The IESG plans to make a decision in the next few weeks, and solicits
> final comments on this action. Please send substantive comments to the
> ietf@ietf.org mailing lists by 2016-05-31. Exceptionally, comments may =
be
> sent to iesg@ietf.org instead. In either case, please retain the
> beginning of the Subject line to allow automated sorting.
>=20
> Abstract
>=20
>=20
>   The User Datagram Protocol (UDP) provides a minimal message-passing
>   transport that has no inherent congestion control mechanisms.  This
>   document provides guidelines on the use of UDP for the designers of
>   applications, tunnels and other protocols that use UDP.  Congestion
>   control guidelines are a primary focus, but the document also
>   provides guidance on other topics, including message sizes,
>   reliability, checksums, middlebox traversal, the use of ECN, DSCPs,
>   and ports.
>=20
>   Because congestion control is critical to the stable operation of =
the
>   Internet, applications and other protocols that choose to use UDP as
>   an Internet transport must employ mechanisms to prevent congestion
>   collapse and to establish some degree of fairness with concurrent
>   traffic.  They may also need to implement additional mechanisms,
>   depending on how they use UDP.
>=20
>   Some guidance is also applicable to the design of other protocols
>   (e.g., protocols layered directly on IP or via IP-based tunnels),
>   especially when these protocols do not themselves provide congestion
>   control.
>=20
>   This document obsoletes RFC5405 and adds guidelines for multicast =
UDP
>   usage.
>=20
>=20
>=20
>=20
> The file can be obtained via
> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/
>=20
> IESG discussion can be tracked via
> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/ballot/
>=20
>=20
> No IPR declarations have been submitted directly on this I-D.
>=20
>=20


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From nobody Thu Jun  2 11:09:24 2016
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To: otroan@employees.org, Xuxiaohu <xuxiaohu@huawei.com>
References: <E83B905A-FF6D-4996-B71A-7921DE4B133B@ericsson.com> <BFC09F5C-D6DF-4B6B-AA95-03919B9F09FB@cisco.com> <573E2A0E.1060609@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D54EB60@NKGEML515-MBX.china.huawei.com> <573F453C.5060908@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D554B73@NKGEML515-MBS.china.huawei.com> <5743303C.5040109@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D55514C@NKGEML515-MBS.china.huawei.com> <5743DD16.3050506@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D555482@NKGEML515-MBS.china.huawei.com> <57448C14.2060203@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5557DE@NKGEML515-MBS.china.huawei.com> <9c462520-eb8e-fcd0-0a08-228f80fbc779@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5596E0@NKGEML515-MBS.china.huawei.com> <8790AF6F-CCD6-43AC-A50E-957B037643F1@employees.org>
From: Joe Touch <touch@isi.edu>
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Date: Thu, 2 Jun 2016 11:08:17 -0700
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Subject: Re: [tsvwg] [nvo3] [Softwires] Is it feasible to perform fragmentation on UDP encapsulated packets.
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On 5/27/2016 3:50 AM, otroan@employees.org wrote:
> It is not possible to implement reassembly complying with IETF RFCs.

a) ATM does this at ridiculously high fragment rates. Granted IP frags
can come out of order, but the fragments are generally much larger.

b) What is the alternative, given we have minimum MTU requirements?

If you're limiting yourself to IPv4 payloads where DF=0, sure, there
there is an alternative. But you've just disabled IPv6 and IPv4 with DF=1.

I.e., it's not possible to NOT implement this and comply with IETF RFCs.

Joe




From nobody Thu Jun  2 11:21:25 2016
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Subject: Re: [tsvwg] Review of draft-ietf-tsvwg-gre-in-udp-encap-11
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On 6/2/2016 11:14 AM, Joe Touch wrote:
>
> On 6/1/2016 6:19 AM, Martin Stiemerling wrote:
>> - Section 4.1, first sentence:
>> "   Regarding packet fragmentation, an encapsulator/decapsulator SHOULD
>>    be compliant with [RFC7588] and perform fragmentation before the
>>    encapsulation."
>> RFC 7588 is mentioned here and the SHOULD implies that a
>> reader/implementer has to read RFC 7588. However, RFC 7588 is not
>> listed under normative references and it is anyhow an informational
>> document, which would require a down-ref in the last call. 
> RFC7588 *describes* how existing deployments operate. It does not
> specify requirements.
>
> It doesn't seem appropriate to refer to being "compliant" with a
> description.

FWIW - correction to the following:
> Further, RFC7588 indicates problems with that configuration and
> describes how its use is thus limited. At best, you would need to copy
> those restrictions as well, but that further begs the question of
> needing a copy of GRE without the benefits of the GRE header.
Too many drafts ;-)

In this case, you do need to copy the caveats in RFC7588, but the doc
already elsewhere justifies the GRE in UDP approach sufficiently.

Joe


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Subject: Re: [tsvwg] Review of draft-ietf-tsvwg-gre-in-udp-encap-11
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On 6/1/2016 6:19 AM, Martin Stiemerling wrote:
> - Section 4.1, first sentence:
> "   Regarding packet fragmentation, an encapsulator/decapsulator SHOULD
>    be compliant with [RFC7588] and perform fragmentation before the
>    encapsulation."
> RFC 7588 is mentioned here and the SHOULD implies that a
> reader/implementer has to read RFC 7588. However, RFC 7588 is not
> listed under normative references and it is anyhow an informational
> document, which would require a down-ref in the last call. 

RFC7588 *describes* how existing deployments operate. It does not
specify requirements.

It doesn't seem appropriate to refer to being "compliant" with a
description.

Further, RFC7588 indicates problems with that configuration and
describes how its use is thus limited. At best, you would need to copy
those restrictions as well, but that further begs the question of
needing a copy of GRE without the benefits of the GRE header.

Joe


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From: "Templin, Fred L" <Fred.L.Templin@boeing.com>
To: Joe Touch <touch@isi.edu>, "otroan@employees.org" <otroan@employees.org>,  Xuxiaohu <xuxiaohu@huawei.com>
Thread-Topic: [Int-area] [nvo3] [Softwires] Is it feasible to perform fragmentation on UDP encapsulated packets.
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Date: Thu, 2 Jun 2016 21:16:53 +0000
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References: <E83B905A-FF6D-4996-B71A-7921DE4B133B@ericsson.com> <BFC09F5C-D6DF-4B6B-AA95-03919B9F09FB@cisco.com> <573E2A0E.1060609@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D54EB60@NKGEML515-MBX.china.huawei.com> <573F453C.5060908@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D554B73@NKGEML515-MBS.china.huawei.com> <5743303C.5040109@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D55514C@NKGEML515-MBS.china.huawei.com> <5743DD16.3050506@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D555482@NKGEML515-MBS.china.huawei.com> <57448C14.2060203@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5557DE@NKGEML515-MBS.china.huawei.com> <9c462520-eb8e-fcd0-0a08-228f80fbc779@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5596E0@NKGEML515-MBS.china.huawei.com> <8790AF6F-CCD6-43AC-A50E-957B037643F1@employees.org> <57507611.5010801@isi.edu>
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Subject: Re: [tsvwg] [Int-area] [nvo3] [Softwires] Is it feasible to perform fragmentation on UDP encapsulated packets.
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Agree with Joe, and also just to mention that RFC4459 discusses tunnel MTU
and fragmentation considerations.

Thanks - Fred

> -----Original Message-----
> From: Int-area [mailto:int-area-bounces@ietf.org] On Behalf Of Joe Touch
> Sent: Thursday, June 02, 2016 11:08 AM
> To: otroan@employees.org; Xuxiaohu <xuxiaohu@huawei.com>
> Cc: Softwires WG <softwires@ietf.org>; nvo3@ietf.org; int-area@ietf.org; =
lisp@ietf.org; tsvwg@ietf.org
> Subject: Re: [Int-area] [nvo3] [Softwires] Is it feasible to perform frag=
mentation on UDP encapsulated packets.
>=20
>=20
>=20
> On 5/27/2016 3:50 AM, otroan@employees.org wrote:
> > It is not possible to implement reassembly complying with IETF RFCs.
>=20
> a) ATM does this at ridiculously high fragment rates. Granted IP frags
> can come out of order, but the fragments are generally much larger.
>=20
> b) What is the alternative, given we have minimum MTU requirements?
>=20
> If you're limiting yourself to IPv4 payloads where DF=3D0, sure, there
> there is an alternative. But you've just disabled IPv6 and IPv4 with DF=
=3D1.
>=20
> I.e., it's not possible to NOT implement this and comply with IETF RFCs.
>=20
> Joe
>=20
>=20
>=20
> _______________________________________________
> Int-area mailing list
> Int-area@ietf.org
> https://www.ietf.org/mailman/listinfo/int-area



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From: Xuxiaohu <xuxiaohu@huawei.com>
To: Joe Touch <touch@isi.edu>, "otroan@employees.org" <otroan@employees.org>
Thread-Topic: [nvo3] [Softwires] Is it feasible to perform fragmentation on UDP encapsulated packets.
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Date: Fri, 3 Jun 2016 02:53:37 +0000
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References: <E83B905A-FF6D-4996-B71A-7921DE4B133B@ericsson.com> <BFC09F5C-D6DF-4B6B-AA95-03919B9F09FB@cisco.com> <573E2A0E.1060609@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D54EB60@NKGEML515-MBX.china.huawei.com> <573F453C.5060908@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D554B73@NKGEML515-MBS.china.huawei.com> <5743303C.5040109@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D55514C@NKGEML515-MBS.china.huawei.com> <5743DD16.3050506@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D555482@NKGEML515-MBS.china.huawei.com> <57448C14.2060203@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5557DE@NKGEML515-MBS.china.huawei.com> <9c462520-eb8e-fcd0-0a08-228f80fbc779@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5596E0@NKGEML515-MBS.china.huawei.com> <8790AF6F-CCD6-43AC-A50E-957B037643F1@employees.org> <57507611.5010801@isi.edu>
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Subject: Re: [tsvwg] [nvo3] [Softwires] Is it feasible to perform fragmentation on UDP encapsulated packets.
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> -----Original Message-----
> From: Joe Touch [mailto:touch@isi.edu]
> Sent: Friday, June 03, 2016 2:08 AM
> To: otroan@employees.org; Xuxiaohu
> Cc: Softwires WG; nvo3@ietf.org; int-area@ietf.org; lisp@ietf.org;
> tsvwg@ietf.org
> Subject: Re: [nvo3] [Softwires] Is it feasible to perform fragmentation o=
n UDP
> encapsulated packets.
>=20
>=20
>=20
> On 5/27/2016 3:50 AM, otroan@employees.org wrote:
> > It is not possible to implement reassembly complying with IETF RFCs.
>=20
> a) ATM does this at ridiculously high fragment rates. Granted IP frags ca=
n come
> out of order, but the fragments are generally much larger.

As pointed in RFC4459,

     "At the time of reassembly, all the information (i.e., all the
      fragments) is normally not available; when the first fragment
      arrives to be reassembled, a buffer of the maximum possible size
      may have to be allocated because the total length of the
      reassembled datagram is not known at that time. Furthermore, as
      fragments might get lost, or be reordered or delayed, the
      reassembly engine has to wait with the partial packet for some
      time (e.g., 60 seconds [9]).  When this would have to be done at
      the line rate, with, for example 10 Gbit/s speed, the length of
      the buffers that reassembly might require would be prohibitive. "

Have you heard the wide adoption of 622M (STM-1) beyond ATM interfaces betw=
een ATM switches in the previous ATM networks? If not, the highest non-ATM =
interface in the past ATM networks was the FE interface which is 100M bps (=
around the year of 2000), If I remembered it correctly. Furthermore, have y=
ou heard the reordering issue with ATM cells? If no, that means once an ATM=
 cell of a given ALL PDU gets lost, all the received ATM cells of that AAL =
PDU would be dropped and therefore the reassembly buffer for that AAL datag=
ram would be released. In other words, there is no need to wait for the los=
t or reordered fragment for a certain period of time before releasing the r=
eassembly buffer. Such behavior is not possible for IP fragmentation and re=
assembly. Last but not least, there is no need to assign a reassembly buffe=
r per AAL PDU (as opposed to per IP datagram in the IP fragmentation and re=
assembly case), instead, only one reassembly buffer per VC since all cells =
within a given VC would be received in order. Since the SAR is only needed =
on the edge of the ATM networks, the number of VCs is very limited. In a wo=
rd, there are significant differences between IP fragmentation and ATM SAR.=
=20

Best regards,
Xiaohu

> b) What is the alternative, given we have minimum MTU requirements?
>=20
> If you're limiting yourself to IPv4 payloads where DF=3D0, sure, there th=
ere is an
> alternative. But you've just disabled IPv6 and IPv4 with DF=3D1.
>=20
> I.e., it's not possible to NOT implement this and comply with IETF RFCs.
>=20
> Joe
>=20
>=20


From nobody Thu Jun  2 21:35:05 2016
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To: Xuxiaohu <xuxiaohu@huawei.com>, "otroan@employees.org" <otroan@employees.org>
References: <E83B905A-FF6D-4996-B71A-7921DE4B133B@ericsson.com> <BFC09F5C-D6DF-4B6B-AA95-03919B9F09FB@cisco.com> <573E2A0E.1060609@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D54EB60@NKGEML515-MBX.china.huawei.com> <573F453C.5060908@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D554B73@NKGEML515-MBS.china.huawei.com> <5743303C.5040109@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D55514C@NKGEML515-MBS.china.huawei.com> <5743DD16.3050506@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D555482@NKGEML515-MBS.china.huawei.com> <57448C14.2060203@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5557DE@NKGEML515-MBS.china.huawei.com> <9c462520-eb8e-fcd0-0a08-228f80fbc779@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5596E0@NKGEML515-MBS.china.huawei.com> <8790AF6F-CCD6-43AC-A50E-957B037643F1@employees.org> <57507611.5010801@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D55B2A4@NKGEML515-MBS.china.huawei.com>
From: Joe Touch <touch@isi.edu>
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Subject: Re: [tsvwg] [nvo3] [Softwires] Is it feasible to perform fragmentation on UDP encapsulated packets.
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On 6/2/2016 7:53 PM, Xuxiaohu wrote:
>
>> -----Original Message-----
>> From: Joe Touch [mailto:touch@isi.edu]
>> Sent: Friday, June 03, 2016 2:08 AM
>> To: otroan@employees.org; Xuxiaohu
>> Cc: Softwires WG; nvo3@ietf.org; int-area@ietf.org; lisp@ietf.org;
>> tsvwg@ietf.org
>> Subject: Re: [nvo3] [Softwires] Is it feasible to perform fragmentation on UDP
>> encapsulated packets.
>>
>>
>>
>> On 5/27/2016 3:50 AM, otroan@employees.org wrote:
>>> It is not possible to implement reassembly complying with IETF RFCs.
>> a) ATM does this at ridiculously high fragment rates. Granted IP frags can come
>> out of order, but the fragments are generally much larger.
> As pointed in RFC4459,
>
>      "At the time of reassembly, all the information (i.e., all the
>       fragments) is normally not available; when the first fragment
>       arrives to be reassembled, a buffer of the maximum possible size
>       may have to be allocated because the total length of the
>       reassembled datagram is not known at that time. Furthermore, as
>       fragments might get lost, or be reordered or delayed, the
>       reassembly engine has to wait with the partial packet for some
>       time (e.g., 60 seconds [9]).  When this would have to be done at
>       the line rate, with, for example 10 Gbit/s speed, the length of
>       the buffers that reassembly might require would be prohibitive. "

Yes, but the alternative of declaring that you don't reassemble has a
cost in terms of dropped segments too.

Taking that drop into account, buffering for a smaller amount of
potential reordering and accounting for reasonable reassembly sizes (for
IPv6, the smallest "max" that's required is 1500) need not be prohibitive.

> Have you heard the wide adoption of 622M (STM-1) beyond ATM interfaces between ATM switches in the previous ATM networks? If not, the highest non-ATM interface in the past ATM networks was the FE interface which is 100M bps (around the year of 2000), If I remembered it correctly. 

OC12 ATM NICs were commonly available in 1998. That was back when
ethernet was pushing 100M and the only other near-gigabit tech was
Myricom (a spin-off of USC/ISI and Caltech). I.e., with the tech
available at that time,  600Mbps SAR was possible.

> Furthermore, have you heard the reordering issue with ATM cells? If no, that means once an ATM cell of a given ALL PDU gets lost, all the received ATM cells of that AAL PDU would be dropped and therefore the reassembly buffer for that AAL datagram would be released. In other words, there is no need to wait for the lost or reordered fragment for a certain period of time before releasing the reassembly buffer.
Reordered no, but just because they arrive in order does not mean they
are required to arrive *adjacent*. The same requirement applies in terms
of needing reassembly buffers for a number of interleaved segments.

>  Such behavior is not possible for IP fragmentation and reassembly. Last but not least, there is no need to assign a reassembly buffer per AAL PDU (as opposed to per IP datagram in the IP fragmentation and reassembly case), instead, only one reassembly buffer per VC since all cells within a given VC would be received in order. Since the SAR is only needed on the edge of the ATM networks, the number of VCs is very limited. 
You could easily decide to allocate a fixed number of reassembly buffers
per IP flow to conserve resources too. The amount of reordering,
together with the number of flows supported would determine your ability
to avoid packet drops -- but recall that drops at high load are not
unusual for IP.

Joe


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From: Xuxiaohu <xuxiaohu@huawei.com>
To: Joe Touch <touch@isi.edu>, "otroan@employees.org" <otroan@employees.org>
Thread-Topic: [nvo3] [Softwires] Is it feasible to perform fragmentation on UDP encapsulated packets.
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Date: Fri, 3 Jun 2016 07:38:40 +0000
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References: <E83B905A-FF6D-4996-B71A-7921DE4B133B@ericsson.com> <BFC09F5C-D6DF-4B6B-AA95-03919B9F09FB@cisco.com> <573E2A0E.1060609@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D54EB60@NKGEML515-MBX.china.huawei.com> <573F453C.5060908@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D554B73@NKGEML515-MBS.china.huawei.com> <5743303C.5040109@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D55514C@NKGEML515-MBS.china.huawei.com> <5743DD16.3050506@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D555482@NKGEML515-MBS.china.huawei.com> <57448C14.2060203@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5557DE@NKGEML515-MBS.china.huawei.com> <9c462520-eb8e-fcd0-0a08-228f80fbc779@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5596E0@NKGEML515-MBS.china.huawei.com> <8790AF6F-CCD6-43AC-A50E-957B037643F1@employees.org> <57507611.5010801@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D55B2A4@NKGEML515-MBS.china.huawei.com> <575108D3.3010506@isi.edu>
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Subject: Re: [tsvwg] [nvo3] [Softwires] Is it feasible to perform fragmentation on UDP encapsulated packets.
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> -----Original Message-----
> From: Joe Touch [mailto:touch@isi.edu]
> Sent: Friday, June 03, 2016 12:34 PM
> To: Xuxiaohu; otroan@employees.org
> Cc: Softwires WG; nvo3@ietf.org; int-area@ietf.org; lisp@ietf.org;
> tsvwg@ietf.org
> Subject: Re: [nvo3] [Softwires] Is it feasible to perform fragmentation o=
n UDP
> encapsulated packets.
>=20
>=20
>=20
> On 6/2/2016 7:53 PM, Xuxiaohu wrote:
> >
> >> -----Original Message-----
> >> From: Joe Touch [mailto:touch@isi.edu]
> >> Sent: Friday, June 03, 2016 2:08 AM
> >> To: otroan@employees.org; Xuxiaohu
> >> Cc: Softwires WG; nvo3@ietf.org; int-area@ietf.org; lisp@ietf.org;
> >> tsvwg@ietf.org
> >> Subject: Re: [nvo3] [Softwires] Is it feasible to perform
> >> fragmentation on UDP encapsulated packets.
> >>
> >>
> >>
> >> On 5/27/2016 3:50 AM, otroan@employees.org wrote:
> >>> It is not possible to implement reassembly complying with IETF RFCs.
> >> a) ATM does this at ridiculously high fragment rates. Granted IP
> >> frags can come out of order, but the fragments are generally much larg=
er.
> > As pointed in RFC4459,
> >
> >      "At the time of reassembly, all the information (i.e., all the
> >       fragments) is normally not available; when the first fragment
> >       arrives to be reassembled, a buffer of the maximum possible size
> >       may have to be allocated because the total length of the
> >       reassembled datagram is not known at that time. Furthermore, as
> >       fragments might get lost, or be reordered or delayed, the
> >       reassembly engine has to wait with the partial packet for some
> >       time (e.g., 60 seconds [9]).  When this would have to be done at
> >       the line rate, with, for example 10 Gbit/s speed, the length of
> >       the buffers that reassembly might require would be prohibitive. "
>=20
> Yes, but the alternative of declaring that you don't reassemble has a cos=
t in
> terms of dropped segments too.

The alternative is to configure the MTU of the core large enough to accommo=
date the added encapsulation header. This is a feasible and proven approach=
 in well-managed SP networks. No packet loss and no forwarding performance =
degradation.

> Taking that drop into account, buffering for a smaller amount of potentia=
l
> reordering and accounting for reasonable reassembly sizes (for IPv6, the
> smallest "max" that's required is 1500) need not be prohibitive.

In the IPv6-in-IPv4 Software network case, assume the MTU of the core is no=
t large enough than 1280+20, all the IPv6 traffic across the tunnels would =
have to be fragmented and then reassembled. Not a small amount.

> > Have you heard the wide adoption of 622M (STM-1) beyond ATM interfaces
> between ATM switches in the previous ATM networks? If not, the highest
> non-ATM interface in the past ATM networks was the FE interface which is =
100M
> bps (around the year of 2000), If I remembered it correctly.
>=20
> OC12 ATM NICs were commonly available in 1998. That was back when

Sorry, there was a spelling mistake ( s/STM-1/STM-4).

OC12/STM-4=3D622M. My question is have you heard the wide adoption of 622M =
(STM-4) BEYOND ATM interfaces (e.g., STM-16/OC48) at that time?

> ethernet was pushing 100M and the only other near-gigabit tech was Myrico=
m
> (a spin-off of USC/ISI and Caltech). I.e., with the tech available at tha=
t time,
> 600Mbps SAR was possible.

Was that 600Mbps SAR capability proved in real networks? And for what servi=
ce?

> > Furthermore, have you heard the reordering issue with ATM cells? If no,=
 that
> means once an ATM cell of a given ALL PDU gets lost, all the received ATM=
 cells
> of that AAL PDU would be dropped and therefore the reassembly buffer for =
that
> AAL datagram would be released. In other words, there is no need to wait =
for
> the lost or reordered fragment for a certain period of time before releas=
ing the
> reassembly buffer.
> Reordered no, but just because they arrive in order does not mean they ar=
e
> required to arrive *adjacent*. The same requirement applies in terms of
> needing reassembly buffers for a number of interleaved segments.

However, the reassembly buffer requirement is limited due to the fact that =
only one buffer is needed per VC and the total number of VCs is not large o=
n the edge of ATM networks. In contrast, the number of IP flows is uncontro=
llable.

> >  Such behavior is not possible for IP fragmentation and reassembly. Las=
t but
> not least, there is no need to assign a reassembly buffer per AAL PDU (as
> opposed to per IP datagram in the IP fragmentation and reassembly case),
> instead, only one reassembly buffer per VC since all cells within a given=
 VC would
> be received in order. Since the SAR is only needed on the edge of the ATM
> networks, the number of VCs is very limited.
> You could easily decide to allocate a fixed number of reassembly buffers =
per IP
> flow to conserve resources too. The amount of reordering, together with t=
he

What about in the DDoS attack case? Do you believe that SPs would allow the=
 existence of that obvious DDoS attack risk in their networks and afford th=
e significant forwarding performance degradation? especially when there has=
 been a feasible and proven solution to the MTU issue (i.e., configure the =
MTU of the core large enough)?

Best regards,
Xiaohu

> number of flows supported would determine your ability to avoid packet dr=
ops
> -- but recall that drops at high load are not unusual for IP.
>=20
> Joe


From nobody Fri Jun  3 05:48:32 2016
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References: <E83B905A-FF6D-4996-B71A-7921DE4B133B@ericsson.com> <BFC09F5C-D6DF-4B6B-AA95-03919B9F09FB@cisco.com> <573E2A0E.1060609@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D54EB60@NKGEML515-MBX.china.huawei.com> <573F453C.5060908@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D554B73@NKGEML515-MBS.china.huawei.com> <5743303C.5040109@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D55514C@NKGEML515-MBS.china.huawei.com> <5743DD16.3050506@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D555482@NKGEML515-MBS.china.huawei.com> <57448C14.2060203@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5557DE@NKGEML515-MBS.china.huawei.com> <9c462520-eb8e-fcd0-0a08-228f80fbc779@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5596E0@NKGEML515-MBS.china.huawei.com> <8790AF6F-CCD6-43AC-A50E-957B037643F1@employees.org> <57507611.5010801@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D55B2A4@NKGEML515-MBS.china.huawei.com> <575108D3.3010506@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D55B37B@NKGEML515-MBS.china.huawei.com>
To: Softwires WG <softwires@ietf.org>
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Subject: Re: [tsvwg] [Softwires] [nvo3] Is it feasible to perform fragmentation on UDP encapsulated packets.
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> 2016-06-03 09:38, Xuxiaohu <xuxiaohu@huawei.com> :
>=20
>> -----Original Message-----
>> From: Joe Touch [mailto:touch@isi.edu]
>> Sent: Friday, June 03, 2016 12:34 PM
...
>> Yes, but the alternative of declaring that you don't reassemble has a cos=
t in
>> terms of dropped segments too.
>=20
> The alternative is to configure the MTU of the core large enough to accomm=
odate the added encapsulation header. This is a feasible and proven approach=
 in well-managed SP networks. No packet loss and no forwarding performance d=
egradation.

Just for information on this subject, an earlier analysis is a available in h=
ttps://tools.ietf.org/html/rfc6751#section-6.4.
(No plan to participate again in IETF technical debates.)

Regards to all,
RD

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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div><br></div><div>2016-06-03 09:38, Xuxia=
ohu &lt;<a href=3D"mailto:xuxiaohu@huawei.com">xuxiaohu@huawei.com</a>&gt; :=
<br></div><blockquote type=3D"cite"><span></span><br><blockquote type=3D"cit=
e"><span>-----Original Message-----</span><br></blockquote><blockquote type=3D=
"cite"><span>From: Joe Touch [<a href=3D"mailto:touch@isi.edu">mailto:touch@=
isi.edu</a>]</span><br></blockquote><blockquote type=3D"cite"><span>Sent: Fri=
day, June 03, 2016 12:34 PM</span></blockquote></blockquote>...<br><blockquo=
te type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><span><=
/span></blockquote></blockquote><blockquote type=3D"cite"><span>Yes, but the=
 alternative of declaring that you don't reassemble has a cost in</span><br>=
</blockquote><blockquote type=3D"cite"><span>terms of dropped segments too.<=
/span><br></blockquote><span></span><br><span>The alternative is to configur=
e the MTU of the core large enough to accommodate the added encapsulation he=
ader. This is a feasible and proven approach in well-managed SP networks. No=
 packet loss and no forwarding performance degradation.</span><br></blockquo=
te><br><div>Just <b>for information</b> on this subject, an earlier analysis=
 is a available in&nbsp;<b><a href=3D"https://tools.ietf.org/html/rfc6751#se=
ction-6">https://tools.ietf.org/html/rfc6751#section-6</a>.</b>4.</div><div>=
(No plan to participate again in IETF technical debates.)</div><div><br></di=
v><div>Regards to all,</div><div>RD<br><blockquote type=3D"cite"><div><span>=
</span></div></blockquote></div></body></html>=

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To: Brian Trammell <ietf@trammell.ch>
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Subject: Re: [tsvwg] Last Call: <draft-ietf-tsvwg-rfc5405bis-13.txt> (UDP Usage Guidelines) to Best Current Practice
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--001a114d3ceea9a1010534613197
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On Thursday, June 2, 2016, Brian Trammell <ietf@trammell.ch> wrote:

> Greetings, all,
>
> Apologies for the late last call comment; I have only one, relatively
> minor. I hope it's still useful.
>
> I understand that Section 3 was written to encourage application
> developers not to roll their own transports ("trust us when we say this is
> hard, this document is a list of reasons why") but as written it would seem
> to discourage transport innovation atop UDP (e.g. QUIC, the RTCWEB data
> channel, anything-over-PLUS), which I very much hope was not the intent.
> The problematic recommendation is in the second paragraph:
>
>    These mechanisms are difficult to implement correctly.  For most
>    applications, the use of one of the existing IETF transport protocols
>    is the simplest method of acquiring the required mechanisms.  Doing
>    so also avoids issues that protocols using a new IP protocol number
>    face when being deployed over the Internet, where middleboxes that
>    only support TCP and UDP are not rare.  Consequently, the RECOMMENDED
>    alternative to the UDP usage described in the remainder of this
>    section is the use of an IETF transport protocol such as TCP
>    [RFC0793], Stream Control Transmission Protocol (SCTP) [RFC4960], and
>    SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram
>    Congestion Control Protocol (DCCP) [RFC4340] with its different
>    congestion control types [RFC4341][RFC4342][RFC5622].
>
> First, this paragraph ignores potential deployment issues with any of
> these other than TCP, which risks seeming out of touch, but this is a minor
> point and probably not worth a late edit. Second, I'm concerned this
> recommendation could be taken as broader than intended, against the
> definition of any new transport protocol encapsulated within UDP that
> performs substantially the same function as the listed protocols.
>
> I think this can be made clearer by simply adding to the list of examples:
>
> NEW:
>
>    These mechanisms are difficult to implement correctly.  For most
>    applications, the use of one of the existing IETF transport protocols
>    is the simplest method of acquiring the required mechanisms.  Doing
>    so also avoids issues that protocols using a new IP protocol number
>    face when being deployed over the Internet, where middleboxes that
>    only support TCP and UDP are not rare.  Consequently, the RECOMMENDED
>    alternative to the UDP usage described in the remainder of this
>    section is the use of an IETF transport protocol such as TCP
>    [RFC0793], Stream Control Transmission Protocol (SCTP) [RFC4960], and
>    SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram
>    Congestion Control Protocol (DCCP) [RFC4340] with its different
>    congestion control types [RFC4341][RFC4342][RFC5622], or transport
>    protocols specified by the IETF in the future.
>
> and removing the examples from the summary in section 7:
>
> OLD:
>
>    | SHOULD use a full-featured transport (TCP, SCTP, DCCP)  |         |
>
> NEW:
>
>    | SHOULD use a full-featured transport                    |         |
>
> Thanks, cheers,
>
> Brian
>
>
> > On 18 May 2016, at 02:17, The IESG <iesg-secretary@ietf.org
> <javascript:;>> wrote:
> >
> >
> > The IESG has received a request from the Transport Area Working Group WG
> > (tsvwg) to consider the following document:
> > - 'UDP Usage Guidelines'
> >  <draft-ietf-tsvwg-rfc5405bis-13.txt> as Best Current Practice
> >
> > The IESG plans to make a decision in the next few weeks, and solicits
> > final comments on this action. Please send substantive comments to the
> > ietf@ietf.org <javascript:;> mailing lists by 2016-05-31.
> Exceptionally, comments may be
> > sent to iesg@ietf.org <javascript:;> instead. In either case, please
> retain the
> > beginning of the Subject line to allow automated sorting.
> >
> > Abstract
> >
> >
> >   The User Datagram Protocol (UDP) provides a minimal message-passing
> >   transport that has no inherent congestion control mechanisms.  This
> >   document provides guidelines on the use of UDP for the designers of
> >   applications, tunnels and other protocols that use UDP.  Congestion
> >   control guidelines are a primary focus, but the document also
> >   provides guidance on other topics, including message sizes,
> >   reliability, checksums, middlebox traversal, the use of ECN, DSCPs,
> >   and ports.
> >
> >   Because congestion control is critical to the stable operation of the
> >   Internet, applications and other protocols that choose to use UDP as
> >   an Internet transport must employ mechanisms to prevent congestion
> >   collapse and to establish some degree of fairness with concurrent
> >   traffic.  They may also need to implement additional mechanisms,
> >   depending on how they use UDP.
> >
> >   Some guidance is also applicable to the design of other protocols
> >   (e.g., protocols layered directly on IP or via IP-based tunnels),
> >   especially when these protocols do not themselves provide congestion
> >   control.
> >
> >   This document obsoletes RFC5405 and adds guidelines for multicast UDP
> >   usage.
> >
> >
> >
> >
> > The file can be obtained via
> > https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/
> >
> > IESG discussion can be tracked via
> > https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/ballot/
> >
> >
> > No IPR declarations have been submitted directly on this I-D.
> >
> >


I disagree.  The text as-is is best and represents a prudent period of
review in the WG.  As suggested many times to both spud and quic, extending
udp is not recommended. We have multiple L4 transport protocols for a
reason, you should innovative in that space without UDP.

CB

--001a114d3ceea9a1010534613197
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<br><br>On Thursday, June 2, 2016, Brian Trammell &lt;<a href=3D"mailto:iet=
f@trammell.ch">ietf@trammell.ch</a>&gt; wrote:<br><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex">Greetings, all,<br>
<br>
Apologies for the late last call comment; I have only one, relatively minor=
. I hope it&#39;s still useful.<br>
<br>
I understand that Section 3 was written to encourage application developers=
 not to roll their own transports (&quot;trust us when we say this is hard,=
 this document is a list of reasons why&quot;) but as written it would seem=
 to discourage transport innovation atop UDP (e.g. QUIC, the RTCWEB data ch=
annel, anything-over-PLUS), which I very much hope was not the intent. The =
problematic recommendation is in the second paragraph:<br>
<br>
=C2=A0 =C2=A0These mechanisms are difficult to implement correctly.=C2=A0 F=
or most<br>
=C2=A0 =C2=A0applications, the use of one of the existing IETF transport pr=
otocols<br>
=C2=A0 =C2=A0is the simplest method of acquiring the required mechanisms.=
=C2=A0 Doing<br>
=C2=A0 =C2=A0so also avoids issues that protocols using a new IP protocol n=
umber<br>
=C2=A0 =C2=A0face when being deployed over the Internet, where middleboxes =
that<br>
=C2=A0 =C2=A0only support TCP and UDP are not rare.=C2=A0 Consequently, the=
 RECOMMENDED<br>
=C2=A0 =C2=A0alternative to the UDP usage described in the remainder of thi=
s<br>
=C2=A0 =C2=A0section is the use of an IETF transport protocol such as TCP<b=
r>
=C2=A0 =C2=A0[RFC0793], Stream Control Transmission Protocol (SCTP) [RFC496=
0], and<br>
=C2=A0 =C2=A0SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Dat=
agram<br>
=C2=A0 =C2=A0Congestion Control Protocol (DCCP) [RFC4340] with its differen=
t<br>
=C2=A0 =C2=A0congestion control types [RFC4341][RFC4342][RFC5622].<br>
<br>
First, this paragraph ignores potential deployment issues with any of these=
 other than TCP, which risks seeming out of touch, but this is a minor poin=
t and probably not worth a late edit. Second, I&#39;m concerned this recomm=
endation could be taken as broader than intended, against the definition of=
 any new transport protocol encapsulated within UDP that performs substanti=
ally the same function as the listed protocols.<br>
<br>
I think this can be made clearer by simply adding to the list of examples:<=
br>
<br>
NEW:<br>
<br>
=C2=A0 =C2=A0These mechanisms are difficult to implement correctly.=C2=A0 F=
or most<br>
=C2=A0 =C2=A0applications, the use of one of the existing IETF transport pr=
otocols<br>
=C2=A0 =C2=A0is the simplest method of acquiring the required mechanisms.=
=C2=A0 Doing<br>
=C2=A0 =C2=A0so also avoids issues that protocols using a new IP protocol n=
umber<br>
=C2=A0 =C2=A0face when being deployed over the Internet, where middleboxes =
that<br>
=C2=A0 =C2=A0only support TCP and UDP are not rare.=C2=A0 Consequently, the=
 RECOMMENDED<br>
=C2=A0 =C2=A0alternative to the UDP usage described in the remainder of thi=
s<br>
=C2=A0 =C2=A0section is the use of an IETF transport protocol such as TCP<b=
r>
=C2=A0 =C2=A0[RFC0793], Stream Control Transmission Protocol (SCTP) [RFC496=
0], and<br>
=C2=A0 =C2=A0SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Dat=
agram<br>
=C2=A0 =C2=A0Congestion Control Protocol (DCCP) [RFC4340] with its differen=
t<br>
=C2=A0 =C2=A0congestion control types [RFC4341][RFC4342][RFC5622], or trans=
port<br>
=C2=A0 =C2=A0protocols specified by the IETF in the future.<br>
<br>
and removing the examples from the summary in section 7:<br>
<br>
OLD:<br>
<br>
=C2=A0 =C2=A0| SHOULD use a full-featured transport (TCP, SCTP, DCCP)=C2=A0=
 |=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0|<br>
<br>
NEW:<br>
<br>
=C2=A0 =C2=A0| SHOULD use a full-featured transport=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|<br>
<br>
Thanks, cheers,<br>
<br>
Brian<br>
<br>
<br>
&gt; On 18 May 2016, at 02:17, The IESG &lt;<a href=3D"javascript:;" onclic=
k=3D"_e(event, &#39;cvml&#39;, &#39;iesg-secretary@ietf.org&#39;)">iesg-sec=
retary@ietf.org</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; The IESG has received a request from the Transport Area Working Group =
WG<br>
&gt; (tsvwg) to consider the following document:<br>
&gt; - &#39;UDP Usage Guidelines&#39;<br>
&gt;=C2=A0 &lt;draft-ietf-tsvwg-rfc5405bis-13.txt&gt; as Best Current Pract=
ice<br>
&gt;<br>
&gt; The IESG plans to make a decision in the next few weeks, and solicits<=
br>
&gt; final comments on this action. Please send substantive comments to the=
<br>
&gt; <a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;iet=
f@ietf.org&#39;)">ietf@ietf.org</a> mailing lists by 2016-05-31. Exceptiona=
lly, comments may be<br>
&gt; sent to <a href=3D"javascript:;" onclick=3D"_e(event, &#39;cvml&#39;, =
&#39;iesg@ietf.org&#39;)">iesg@ietf.org</a> instead. In either case, please=
 retain the<br>
&gt; beginning of the Subject line to allow automated sorting.<br>
&gt;<br>
&gt; Abstract<br>
&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0The User Datagram Protocol (UDP) provides a minimal messag=
e-passing<br>
&gt;=C2=A0 =C2=A0transport that has no inherent congestion control mechanis=
ms.=C2=A0 This<br>
&gt;=C2=A0 =C2=A0document provides guidelines on the use of UDP for the des=
igners of<br>
&gt;=C2=A0 =C2=A0applications, tunnels and other protocols that use UDP.=C2=
=A0 Congestion<br>
&gt;=C2=A0 =C2=A0control guidelines are a primary focus, but the document a=
lso<br>
&gt;=C2=A0 =C2=A0provides guidance on other topics, including message sizes=
,<br>
&gt;=C2=A0 =C2=A0reliability, checksums, middlebox traversal, the use of EC=
N, DSCPs,<br>
&gt;=C2=A0 =C2=A0and ports.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0Because congestion control is critical to the stable opera=
tion of the<br>
&gt;=C2=A0 =C2=A0Internet, applications and other protocols that choose to =
use UDP as<br>
&gt;=C2=A0 =C2=A0an Internet transport must employ mechanisms to prevent co=
ngestion<br>
&gt;=C2=A0 =C2=A0collapse and to establish some degree of fairness with con=
current<br>
&gt;=C2=A0 =C2=A0traffic.=C2=A0 They may also need to implement additional =
mechanisms,<br>
&gt;=C2=A0 =C2=A0depending on how they use UDP.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0Some guidance is also applicable to the design of other pr=
otocols<br>
&gt;=C2=A0 =C2=A0(e.g., protocols layered directly on IP or via IP-based tu=
nnels),<br>
&gt;=C2=A0 =C2=A0especially when these protocols do not themselves provide =
congestion<br>
&gt;=C2=A0 =C2=A0control.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0This document obsoletes RFC5405 and adds guidelines for mu=
lticast UDP<br>
&gt;=C2=A0 =C2=A0usage.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; The file can be obtained via<br>
&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bi=
s/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc=
5405bis/</a><br>
&gt;<br>
&gt; IESG discussion can be tracked via<br>
&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bi=
s/ballot/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-ts=
vwg-rfc5405bis/ballot/</a><br>
&gt;<br>
&gt;<br>
&gt; No IPR declarations have been submitted directly on this I-D.<br>
&gt;<br>
&gt;</blockquote><div><br></div><div>I disagree.=C2=A0 The text as-is is=C2=
=A0best and represents a prudent period of review in the WG.=C2=A0=C2=A0As =
suggested many times to both spud and quic, extending udp is not recommende=
d. We have multiple=C2=A0L4 transport protocols for a reason, you should in=
novative in that space without UDP.=C2=A0</div><div><br></div><div>CB</div>

--001a114d3ceea9a1010534613197--


From nobody Fri Jun  3 08:28:29 2016
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Date: Fri, 03 Jun 2016 16:28:09 +0100
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Subject: Re: [tsvwg] Last Call: <draft-ietf-tsvwg-rfc5405bis-13.txt> (UDP Usage Guidelines) to Best Current Practice
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It is certainly not too late, many thanks - to both Brian and you!

We have various pending comments to look through, and will make sure all 
comments are also taken into consideration, -  next week we'll be 
resolving the conflicts between comments and seeing how best to 
accommodate these in updated text, comments then may be late!

Gorry

On 03/06/2016 16:15, Ca By wrote:
>
>
> On Thursday, June 2, 2016, Brian Trammell <ietf@trammell.ch 
> <mailto:ietf@trammell.ch>> wrote:
>
>     Greetings, all,
>
>     Apologies for the late last call comment; I have only one,
>     relatively minor. I hope it's still useful.
>
>     I understand that Section 3 was written to encourage application
>     developers not to roll their own transports ("trust us when we say
>     this is hard, this document is a list of reasons why") but as
>     written it would seem to discourage transport innovation atop UDP
>     (e.g. QUIC, the RTCWEB data channel, anything-over-PLUS), which I
>     very much hope was not the intent. The problematic recommendation
>     is in the second paragraph:
>
>        These mechanisms are difficult to implement correctly.  For most
>        applications, the use of one of the existing IETF transport
>     protocols
>        is the simplest method of acquiring the required mechanisms.  Doing
>        so also avoids issues that protocols using a new IP protocol number
>        face when being deployed over the Internet, where middleboxes that
>        only support TCP and UDP are not rare.  Consequently, the
>     RECOMMENDED
>        alternative to the UDP usage described in the remainder of this
>        section is the use of an IETF transport protocol such as TCP
>        [RFC0793], Stream Control Transmission Protocol (SCTP)
>     [RFC4960], and
>        SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram
>        Congestion Control Protocol (DCCP) [RFC4340] with its different
>        congestion control types [RFC4341][RFC4342][RFC5622].
>
>     First, this paragraph ignores potential deployment issues with any
>     of these other than TCP, which risks seeming out of touch, but
>     this is a minor point and probably not worth a late edit. Second,
>     I'm concerned this recommendation could be taken as broader than
>     intended, against the definition of any new transport protocol
>     encapsulated within UDP that performs substantially the same
>     function as the listed protocols.
>
>     I think this can be made clearer by simply adding to the list of
>     examples:
>
>     NEW:
>
>        These mechanisms are difficult to implement correctly.  For most
>        applications, the use of one of the existing IETF transport
>     protocols
>        is the simplest method of acquiring the required mechanisms.  Doing
>        so also avoids issues that protocols using a new IP protocol number
>        face when being deployed over the Internet, where middleboxes that
>        only support TCP and UDP are not rare.  Consequently, the
>     RECOMMENDED
>        alternative to the UDP usage described in the remainder of this
>        section is the use of an IETF transport protocol such as TCP
>        [RFC0793], Stream Control Transmission Protocol (SCTP)
>     [RFC4960], and
>        SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram
>        Congestion Control Protocol (DCCP) [RFC4340] with its different
>        congestion control types [RFC4341][RFC4342][RFC5622], or transport
>        protocols specified by the IETF in the future.
>
>     and removing the examples from the summary in section 7:
>
>     OLD:
>
>        | SHOULD use a full-featured transport (TCP, SCTP, DCCP)  |   
>          |
>
>     NEW:
>
>        | SHOULD use a full-featured transport                    |   
>          |
>
>     Thanks, cheers,
>
>     Brian
>
>
>     > On 18 May 2016, at 02:17, The IESG <iesg-secretary@ietf.org
>     <javascript:;>> wrote:
>     >
>     >
>     > The IESG has received a request from the Transport Area Working
>     Group WG
>     > (tsvwg) to consider the following document:
>     > - 'UDP Usage Guidelines'
>     > <draft-ietf-tsvwg-rfc5405bis-13.txt> as Best Current Practice
>     >
>     > The IESG plans to make a decision in the next few weeks, and
>     solicits
>     > final comments on this action. Please send substantive comments
>     to the
>     > ietf@ietf.org <javascript:;> mailing lists by 2016-05-31.
>     Exceptionally, comments may be
>     > sent to iesg@ietf.org <javascript:;> instead. In either case,
>     please retain the
>     > beginning of the Subject line to allow automated sorting.
>     >
>     > Abstract
>     >
>     >
>     >   The User Datagram Protocol (UDP) provides a minimal
>     message-passing
>     >   transport that has no inherent congestion control mechanisms. 
>     This
>     >   document provides guidelines on the use of UDP for the
>     designers of
>     >   applications, tunnels and other protocols that use UDP. 
>     Congestion
>     >   control guidelines are a primary focus, but the document also
>     >   provides guidance on other topics, including message sizes,
>     >   reliability, checksums, middlebox traversal, the use of ECN,
>     DSCPs,
>     >   and ports.
>     >
>     >   Because congestion control is critical to the stable operation
>     of the
>     >   Internet, applications and other protocols that choose to use
>     UDP as
>     >   an Internet transport must employ mechanisms to prevent congestion
>     >   collapse and to establish some degree of fairness with concurrent
>     >   traffic.  They may also need to implement additional mechanisms,
>     >   depending on how they use UDP.
>     >
>     >   Some guidance is also applicable to the design of other protocols
>     >   (e.g., protocols layered directly on IP or via IP-based tunnels),
>     >   especially when these protocols do not themselves provide
>     congestion
>     >   control.
>     >
>     >   This document obsoletes RFC5405 and adds guidelines for
>     multicast UDP
>     >   usage.
>     >
>     >
>     >
>     >
>     > The file can be obtained via
>     > https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/
>     >
>     > IESG discussion can be tracked via
>     > https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/ballot/
>     >
>     >
>     > No IPR declarations have been submitted directly on this I-D.
>     >
>     >
>
>
> I disagree.  The text as-is is best and represents a prudent period of 
> review in the WG.  As suggested many times to both spud and quic, 
> extending udp is not recommended. We have multiple L4 transport 
> protocols for a reason, you should innovative in that space without UDP.
>
> CB


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From: "Templin, Fred L" <Fred.L.Templin@boeing.com>
To: Xuxiaohu <xuxiaohu@huawei.com>, Joe Touch <touch@isi.edu>, "otroan@employees.org" <otroan@employees.org>
Thread-Topic: [nvo3] [Softwires] Is it feasible to perform fragmentation on UDP encapsulated packets.
Thread-Index: AQHRvWr/d3+tgL34oUSVk+7sP6lEoZ/X6QAw
Date: Fri, 3 Jun 2016 16:27:06 +0000
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References: <E83B905A-FF6D-4996-B71A-7921DE4B133B@ericsson.com> <BFC09F5C-D6DF-4B6B-AA95-03919B9F09FB@cisco.com> <573E2A0E.1060609@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D54EB60@NKGEML515-MBX.china.huawei.com> <573F453C.5060908@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D554B73@NKGEML515-MBS.china.huawei.com> <5743303C.5040109@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D55514C@NKGEML515-MBS.china.huawei.com> <5743DD16.3050506@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D555482@NKGEML515-MBS.china.huawei.com> <57448C14.2060203@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5557DE@NKGEML515-MBS.china.huawei.com> <9c462520-eb8e-fcd0-0a08-228f80fbc779@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5596E0@NKGEML515-MBS.china.huawei.com> <8790AF6F-CCD6-43AC-A50E-957B037643F1@employees.org> <57507611.5010801@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D55B2A4@NKGEML515-MBS.china.huawei.com> <575108D3.3010506@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D55B37B@NKGEML515-MBS.china.huawei.com>
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Subject: Re: [tsvwg] [nvo3] [Softwires] Is it feasible to perform fragmentation on UDP encapsulated packets.
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> The alternative is to configure the MTU of the core large enough to accom=
modate the added encapsulation header.

But what if you can't do that? All it takes is one link with a small and/or
misconfigured MTU...

Fred
fred.l.templin@boeing.com

> -----Original Message-----
> From: Int-area [mailto:int-area-bounces@ietf.org] On Behalf Of Xuxiaohu
> Sent: Friday, June 03, 2016 12:39 AM
> To: Joe Touch <touch@isi.edu>; otroan@employees.org
> Cc: Softwires WG <softwires@ietf.org>; nvo3@ietf.org; int-area@ietf.org; =
lisp@ietf.org; tsvwg@ietf.org
> Subject: Re: [Int-area] [nvo3] [Softwires] Is it feasible to perform frag=
mentation on UDP encapsulated packets.
>=20
>=20
>=20
> > -----Original Message-----
> > From: Joe Touch [mailto:touch@isi.edu]
> > Sent: Friday, June 03, 2016 12:34 PM
> > To: Xuxiaohu; otroan@employees.org
> > Cc: Softwires WG; nvo3@ietf.org; int-area@ietf.org; lisp@ietf.org;
> > tsvwg@ietf.org
> > Subject: Re: [nvo3] [Softwires] Is it feasible to perform fragmentation=
 on UDP
> > encapsulated packets.
> >
> >
> >
> > On 6/2/2016 7:53 PM, Xuxiaohu wrote:
> > >
> > >> -----Original Message-----
> > >> From: Joe Touch [mailto:touch@isi.edu]
> > >> Sent: Friday, June 03, 2016 2:08 AM
> > >> To: otroan@employees.org; Xuxiaohu
> > >> Cc: Softwires WG; nvo3@ietf.org; int-area@ietf.org; lisp@ietf.org;
> > >> tsvwg@ietf.org
> > >> Subject: Re: [nvo3] [Softwires] Is it feasible to perform
> > >> fragmentation on UDP encapsulated packets.
> > >>
> > >>
> > >>
> > >> On 5/27/2016 3:50 AM, otroan@employees.org wrote:
> > >>> It is not possible to implement reassembly complying with IETF RFCs=
.
> > >> a) ATM does this at ridiculously high fragment rates. Granted IP
> > >> frags can come out of order, but the fragments are generally much la=
rger.
> > > As pointed in RFC4459,
> > >
> > >      "At the time of reassembly, all the information (i.e., all the
> > >       fragments) is normally not available; when the first fragment
> > >       arrives to be reassembled, a buffer of the maximum possible siz=
e
> > >       may have to be allocated because the total length of the
> > >       reassembled datagram is not known at that time. Furthermore, as
> > >       fragments might get lost, or be reordered or delayed, the
> > >       reassembly engine has to wait with the partial packet for some
> > >       time (e.g., 60 seconds [9]).  When this would have to be done a=
t
> > >       the line rate, with, for example 10 Gbit/s speed, the length of
> > >       the buffers that reassembly might require would be prohibitive.=
 "
> >
> > Yes, but the alternative of declaring that you don't reassemble has a c=
ost in
> > terms of dropped segments too.
>=20
> The alternative is to configure the MTU of the core large enough to accom=
modate the added encapsulation header. This is a feasible
> and proven approach in well-managed SP networks. No packet loss and no fo=
rwarding performance degradation.
>=20
> > Taking that drop into account, buffering for a smaller amount of potent=
ial
> > reordering and accounting for reasonable reassembly sizes (for IPv6, th=
e
> > smallest "max" that's required is 1500) need not be prohibitive.
>=20
> In the IPv6-in-IPv4 Software network case, assume the MTU of the core is =
not large enough than 1280+20, all the IPv6 traffic across
> the tunnels would have to be fragmented and then reassembled. Not a small=
 amount.
>=20
> > > Have you heard the wide adoption of 622M (STM-1) beyond ATM interface=
s
> > between ATM switches in the previous ATM networks? If not, the highest
> > non-ATM interface in the past ATM networks was the FE interface which i=
s 100M
> > bps (around the year of 2000), If I remembered it correctly.
> >
> > OC12 ATM NICs were commonly available in 1998. That was back when
>=20
> Sorry, there was a spelling mistake ( s/STM-1/STM-4).
>=20
> OC12/STM-4=3D622M. My question is have you heard the wide adoption of 622=
M (STM-4) BEYOND ATM interfaces (e.g., STM-16/OC48)
> at that time?
>=20
> > ethernet was pushing 100M and the only other near-gigabit tech was Myri=
com
> > (a spin-off of USC/ISI and Caltech). I.e., with the tech available at t=
hat time,
> > 600Mbps SAR was possible.
>=20
> Was that 600Mbps SAR capability proved in real networks? And for what ser=
vice?
>=20
> > > Furthermore, have you heard the reordering issue with ATM cells? If n=
o, that
> > means once an ATM cell of a given ALL PDU gets lost, all the received A=
TM cells
> > of that AAL PDU would be dropped and therefore the reassembly buffer fo=
r that
> > AAL datagram would be released. In other words, there is no need to wai=
t for
> > the lost or reordered fragment for a certain period of time before rele=
asing the
> > reassembly buffer.
> > Reordered no, but just because they arrive in order does not mean they =
are
> > required to arrive *adjacent*. The same requirement applies in terms of
> > needing reassembly buffers for a number of interleaved segments.
>=20
> However, the reassembly buffer requirement is limited due to the fact tha=
t only one buffer is needed per VC and the total number of
> VCs is not large on the edge of ATM networks. In contrast, the number of =
IP flows is uncontrollable.
>=20
> > >  Such behavior is not possible for IP fragmentation and reassembly. L=
ast but
> > not least, there is no need to assign a reassembly buffer per AAL PDU (=
as
> > opposed to per IP datagram in the IP fragmentation and reassembly case)=
,
> > instead, only one reassembly buffer per VC since all cells within a giv=
en VC would
> > be received in order. Since the SAR is only needed on the edge of the A=
TM
> > networks, the number of VCs is very limited.
> > You could easily decide to allocate a fixed number of reassembly buffer=
s per IP
> > flow to conserve resources too. The amount of reordering, together with=
 the
>=20
> What about in the DDoS attack case? Do you believe that SPs would allow t=
he existence of that obvious DDoS attack risk in their
> networks and afford the significant forwarding performance degradation? e=
specially when there has been a feasible and proven
> solution to the MTU issue (i.e., configure the MTU of the core large enou=
gh)?
>=20
> Best regards,
> Xiaohu
>=20
> > number of flows supported would determine your ability to avoid packet =
drops
> > -- but recall that drops at high load are not unusual for IP.
> >
> > Joe
>=20
> _______________________________________________
> Int-area mailing list
> Int-area@ietf.org
> https://www.ietf.org/mailman/listinfo/int-area



From nobody Fri Jun  3 09:44:56 2016
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Date: Fri, 3 Jun 2016 09:44:40 -0700
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From: Tom Herbert <tom@herbertland.com>
To: Xuxiaohu <xuxiaohu@huawei.com>
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Subject: Re: [tsvwg] [Int-area] [nvo3] [Softwires] Is it feasible to perform fragmentation on UDP encapsulated packets.
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On Fri, Jun 3, 2016 at 12:38 AM, Xuxiaohu <xuxiaohu@huawei.com> wrote:
>
>
>> -----Original Message-----
>> From: Joe Touch [mailto:touch@isi.edu]
>> Sent: Friday, June 03, 2016 12:34 PM
>> To: Xuxiaohu; otroan@employees.org
>> Cc: Softwires WG; nvo3@ietf.org; int-area@ietf.org; lisp@ietf.org;
>> tsvwg@ietf.org
>> Subject: Re: [nvo3] [Softwires] Is it feasible to perform fragmentation =
on UDP
>> encapsulated packets.
>>
>>
>>
>> On 6/2/2016 7:53 PM, Xuxiaohu wrote:
>> >
>> >> -----Original Message-----
>> >> From: Joe Touch [mailto:touch@isi.edu]
>> >> Sent: Friday, June 03, 2016 2:08 AM
>> >> To: otroan@employees.org; Xuxiaohu
>> >> Cc: Softwires WG; nvo3@ietf.org; int-area@ietf.org; lisp@ietf.org;
>> >> tsvwg@ietf.org
>> >> Subject: Re: [nvo3] [Softwires] Is it feasible to perform
>> >> fragmentation on UDP encapsulated packets.
>> >>
>> >>
>> >>
>> >> On 5/27/2016 3:50 AM, otroan@employees.org wrote:
>> >>> It is not possible to implement reassembly complying with IETF RFCs.
>> >> a) ATM does this at ridiculously high fragment rates. Granted IP
>> >> frags can come out of order, but the fragments are generally much lar=
ger.
>> > As pointed in RFC4459,
>> >
>> >      "At the time of reassembly, all the information (i.e., all the
>> >       fragments) is normally not available; when the first fragment
>> >       arrives to be reassembled, a buffer of the maximum possible size
>> >       may have to be allocated because the total length of the
>> >       reassembled datagram is not known at that time. Furthermore, as
>> >       fragments might get lost, or be reordered or delayed, the
>> >       reassembly engine has to wait with the partial packet for some
>> >       time (e.g., 60 seconds [9]).  When this would have to be done at
>> >       the line rate, with, for example 10 Gbit/s speed, the length of
>> >       the buffers that reassembly might require would be prohibitive. =
"
>>
>> Yes, but the alternative of declaring that you don't reassemble has a co=
st in
>> terms of dropped segments too.
>
> The alternative is to configure the MTU of the core large enough to accom=
modate the added encapsulation header. This is a feasible and proven approa=
ch in well-managed SP networks. No packet loss and no forwarding performanc=
e degradation.
>
>> Taking that drop into account, buffering for a smaller amount of potenti=
al
>> reordering and accounting for reasonable reassembly sizes (for IPv6, the
>> smallest "max" that's required is 1500) need not be prohibitive.
>
> In the IPv6-in-IPv4 Software network case, assume the MTU of the core is =
not large enough than 1280+20, all the IPv6 traffic across the tunnels woul=
d have to be fragmented and then reassembled. Not a small amount.
>
>> > Have you heard the wide adoption of 622M (STM-1) beyond ATM interfaces
>> between ATM switches in the previous ATM networks? If not, the highest
>> non-ATM interface in the past ATM networks was the FE interface which is=
 100M
>> bps (around the year of 2000), If I remembered it correctly.
>>
>> OC12 ATM NICs were commonly available in 1998. That was back when
>
> Sorry, there was a spelling mistake ( s/STM-1/STM-4).
>
> OC12/STM-4=3D622M. My question is have you heard the wide adoption of 622=
M (STM-4) BEYOND ATM interfaces (e.g., STM-16/OC48) at that time?
>
>> ethernet was pushing 100M and the only other near-gigabit tech was Myric=
om
>> (a spin-off of USC/ISI and Caltech). I.e., with the tech available at th=
at time,
>> 600Mbps SAR was possible.
>
> Was that 600Mbps SAR capability proved in real networks? And for what ser=
vice?
>
>> > Furthermore, have you heard the reordering issue with ATM cells? If no=
, that
>> means once an ATM cell of a given ALL PDU gets lost, all the received AT=
M cells
>> of that AAL PDU would be dropped and therefore the reassembly buffer for=
 that
>> AAL datagram would be released. In other words, there is no need to wait=
 for
>> the lost or reordered fragment for a certain period of time before relea=
sing the
>> reassembly buffer.
>> Reordered no, but just because they arrive in order does not mean they a=
re
>> required to arrive *adjacent*. The same requirement applies in terms of
>> needing reassembly buffers for a number of interleaved segments.
>
> However, the reassembly buffer requirement is limited due to the fact tha=
t only one buffer is needed per VC and the total number of VCs is not large=
 on the edge of ATM networks. In contrast, the number of IP flows is uncont=
rollable.
>
>> >  Such behavior is not possible for IP fragmentation and reassembly. La=
st but
>> not least, there is no need to assign a reassembly buffer per AAL PDU (a=
s
>> opposed to per IP datagram in the IP fragmentation and reassembly case),
>> instead, only one reassembly buffer per VC since all cells within a give=
n VC would
>> be received in order. Since the SAR is only needed on the edge of the AT=
M
>> networks, the number of VCs is very limited.
>> You could easily decide to allocate a fixed number of reassembly buffers=
 per IP
>> flow to conserve resources too. The amount of reordering, together with =
the
>
> What about in the DDoS attack case? Do you believe that SPs would allow t=
he existence of that obvious DDoS attack risk in their networks and afford =
the significant forwarding performance degradation? especially when there h=
as been a feasible and proven solution to the MTU issue (i.e., configure th=
e MTU of the core large enough)?
>
That is precisely solution #3 in RFC4459:

"Ensure that in the specific environment, the encapsulated packets
will fit in all the paths in the network, e.g., by using MTU bigger
than 1500 in the backbone used for encapsulation."

If you're able to implement this in you network that's great, but it
is not feasible in all situations and hence can't be the standard. We
may not control all the underlay networks (definitely not over the
Internet) and hence not be able to set all the MTUs large enough. Also
an MTU greater than 1500 require jumbo frames, not all networks
enabled those.

One other point, when a device sources or sinks IP packets like in
tunneling it is taking on the role of a host and so we expect host
requirements apply. For instance, routers may perform fragmentation in
IPv4, but they do not perform fragmentation for IPv6 or reassembly
(those are hosts functionalities). There has been at least one attempt
to carve out exceptions in applying host requirements to devices that
perform tunneling, that is an allowance to send a zero UDP checksum in
in IPv6. Documenting this took two RFCs and we still needed a whole
lot of description in at least GRE/UDP on requirements that to use the
non-zero checksum. I don't think standardizing exceptions in
fragmentation for tunnels would be any easier.

Tom

> Best regards,
> Xiaohu
>
>> number of flows supported would determine your ability to avoid packet d=
rops
>> -- but recall that drops at high load are not unusual for IP.
>>
>> Joe
>
> _______________________________________________
> Int-area mailing list
> Int-area@ietf.org
> https://www.ietf.org/mailman/listinfo/int-area


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From: "Fred Baker (fred)" <fred@cisco.com>
To: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>
Thread-Topic: [tsvwg] TSVWG working group :: detailed status update on work items
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Subject: Re: [tsvwg] TSVWG working group :: detailed status update on work items
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> On May 19, 2016, at 3:53 AM, Gorry Fairhurst <gorry@erg.abdn.ac.uk> =
wrote:
>=20
> Drafts under consideration by WG for adoption:
> ---
> draft-szigeti-tsvwg-ieee-802-11  - Document Shepherd: TBD
> Mapping to 802.11
> - Initial presentation at Prague IETF (Summer 2014)
> - Presentation at IETF Nov 2015.
> Thought to fall within TSVWG charter
> -  No call issued for adoption at this time.
> Chair Note: Does this need to reference RC4594 or an update to that =
spec.?
> There's some background work that's likely to surface as proposed =
4594bis and 5127bis drafts.
> DUE: Please discuss on the TSVWG list.
> DUE: Working Group call for adoption.

What is the current status? We have not observed an adoption call, or =
received any reviews.

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From: "Black, David" <david.black@emc.com>
To: "Fred Baker (fred)" <fred@cisco.com>, "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>
Thread-Topic: [tsvwg] TSVWG working group :: detailed status update on work items
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Date: Fri, 3 Jun 2016 17:11:48 +0000
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Subject: Re: [tsvwg] TSVWG working group :: detailed status update on work items
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WG adoption call coming soon ...

Thanks, --David

> -----Original Message-----
> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Fred Baker (fred=
)
> Sent: Friday, June 03, 2016 1:01 PM
> To: gorry@erg.abdn.ac.uk
> Cc: tsvwg WG
> Subject: Re: [tsvwg] TSVWG working group :: detailed status update on wor=
k
> items
>=20
>=20
> > On May 19, 2016, at 3:53 AM, Gorry Fairhurst <gorry@erg.abdn.ac.uk> wro=
te:
> >
> > Drafts under consideration by WG for adoption:
> > ---
> > draft-szigeti-tsvwg-ieee-802-11  - Document Shepherd: TBD
> > Mapping to 802.11
> > - Initial presentation at Prague IETF (Summer 2014)
> > - Presentation at IETF Nov 2015.
> > Thought to fall within TSVWG charter
> > -  No call issued for adoption at this time.
> > Chair Note: Does this need to reference RC4594 or an update to that spe=
c.?
> > There's some background work that's likely to surface as proposed 4594b=
is and
> 5127bis drafts.
> > DUE: Please discuss on the TSVWG list.
> > DUE: Working Group call for adoption.
>=20
> What is the current status? We have not observed an adoption call, or rec=
eived
> any reviews.


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From: Tom Herbert <tom@herbertland.com>
To: Brian Trammell <ietf@trammell.ch>
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Subject: Re: [tsvwg] [Spud] Last Call: <draft-ietf-tsvwg-rfc5405bis-13.txt> (UDP Usage Guidelines) to Best Current Practice
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On Thu, Jun 2, 2016 at 3:11 AM, Brian Trammell <ietf@trammell.ch> wrote:
> Greetings, all,
>
> Apologies for the late last call comment; I have only one, relatively min=
or. I hope it's still useful.
>
> I understand that Section 3 was written to encourage application develope=
rs not to roll their own transports ("trust us when we say this is hard, th=
is document is a list of reasons why") but as written it would seem to disc=
ourage transport innovation atop UDP (e.g. QUIC, the RTCWEB data channel, a=
nything-over-PLUS), which I very much hope was not the intent. The problema=
tic recommendation is in the second paragraph:
>
>    These mechanisms are difficult to implement correctly.  For most
>    applications, the use of one of the existing IETF transport protocols
>    is the simplest method of acquiring the required mechanisms.  Doing
>    so also avoids issues that protocols using a new IP protocol number
>    face when being deployed over the Internet, where middleboxes that
>    only support TCP and UDP are not rare.  Consequently, the RECOMMENDED
>    alternative to the UDP usage described in the remainder of this
>    section is the use of an IETF transport protocol such as TCP
>    [RFC0793], Stream Control Transmission Protocol (SCTP) [RFC4960], and
>    SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram
>    Congestion Control Protocol (DCCP) [RFC4340] with its different
>    congestion control types [RFC4341][RFC4342][RFC5622].
>
> First, this paragraph ignores potential deployment issues with any of the=
se other than TCP, which risks seeming out of touch, but this is a minor po=
int and probably not worth a late edit. Second, I'm concerned this recommen=
dation could be taken as broader than intended, against the definition of a=
ny new transport protocol encapsulated within UDP that performs substantial=
ly the same function as the listed protocols.
>
I would agree, this paragraph also seems a little self
contradictory.There is an acknowledgment that "middleboxes that only
support TCP and UDP are not rare", but then the next sentence
recommends the use of several other protocols besides UDP and TCP. If
I put these two together, the only congested controlled protocol that
is recommended and expected to work on the Internet is TCP.

Tom

> I think this can be made clearer by simply adding to the list of examples=
:
>
> NEW:
>
>    These mechanisms are difficult to implement correctly.  For most
>    applications, the use of one of the existing IETF transport protocols
>    is the simplest method of acquiring the required mechanisms.  Doing
>    so also avoids issues that protocols using a new IP protocol number
>    face when being deployed over the Internet, where middleboxes that
>    only support TCP and UDP are not rare.  Consequently, the RECOMMENDED
>    alternative to the UDP usage described in the remainder of this
>    section is the use of an IETF transport protocol such as TCP
>    [RFC0793], Stream Control Transmission Protocol (SCTP) [RFC4960], and
>    SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram
>    Congestion Control Protocol (DCCP) [RFC4340] with its different
>    congestion control types [RFC4341][RFC4342][RFC5622], or transport
>    protocols specified by the IETF in the future.
>
> and removing the examples from the summary in section 7:
>
> OLD:
>
>    | SHOULD use a full-featured transport (TCP, SCTP, DCCP)  |         |
>
> NEW:
>
>    | SHOULD use a full-featured transport                    |         |
>
> Thanks, cheers,
>
> Brian
>
>
>> On 18 May 2016, at 02:17, The IESG <iesg-secretary@ietf.org> wrote:
>>
>>
>> The IESG has received a request from the Transport Area Working Group WG
>> (tsvwg) to consider the following document:
>> - 'UDP Usage Guidelines'
>>  <draft-ietf-tsvwg-rfc5405bis-13.txt> as Best Current Practice
>>
>> The IESG plans to make a decision in the next few weeks, and solicits
>> final comments on this action. Please send substantive comments to the
>> ietf@ietf.org mailing lists by 2016-05-31. Exceptionally, comments may b=
e
>> sent to iesg@ietf.org instead. In either case, please retain the
>> beginning of the Subject line to allow automated sorting.
>>
>> Abstract
>>
>>
>>   The User Datagram Protocol (UDP) provides a minimal message-passing
>>   transport that has no inherent congestion control mechanisms.  This
>>   document provides guidelines on the use of UDP for the designers of
>>   applications, tunnels and other protocols that use UDP.  Congestion
>>   control guidelines are a primary focus, but the document also
>>   provides guidance on other topics, including message sizes,
>>   reliability, checksums, middlebox traversal, the use of ECN, DSCPs,
>>   and ports.
>>
>>   Because congestion control is critical to the stable operation of the
>>   Internet, applications and other protocols that choose to use UDP as
>>   an Internet transport must employ mechanisms to prevent congestion
>>   collapse and to establish some degree of fairness with concurrent
>>   traffic.  They may also need to implement additional mechanisms,
>>   depending on how they use UDP.
>>
>>   Some guidance is also applicable to the design of other protocols
>>   (e.g., protocols layered directly on IP or via IP-based tunnels),
>>   especially when these protocols do not themselves provide congestion
>>   control.
>>
>>   This document obsoletes RFC5405 and adds guidelines for multicast UDP
>>   usage.
>>
>>
>>
>>
>> The file can be obtained via
>> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/
>>
>> IESG discussion can be tracked via
>> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/ballot/
>>
>>
>> No IPR declarations have been submitted directly on this I-D.
>>
>>
>
>
> _______________________________________________
> Spud mailing list
> Spud@ietf.org
> https://www.ietf.org/mailman/listinfo/spud
>


From nobody Sun Jun  5 22:18:26 2016
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Subject: Re: [tsvwg] [QUIC] [Spud] Last Call: <draft-ietf-tsvwg-rfc5405bis-13.txt> (UDP Usage Guidelines) to Best Current Practice
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On Sat, Jun 4, 2016 at 5:25 PM, Tom Herbert <tom@herbertland.com> wrote:

> On Thu, Jun 2, 2016 at 3:11 AM, Brian Trammell <ietf@trammell.ch> wrote:
> > Greetings, all,
> >
> > Apologies for the late last call comment; I have only one, relatively
> minor. I hope it's still useful.
> >
> > I understand that Section 3 was written to encourage application
> developers not to roll their own transports ("trust us when we say this is
> hard, this document is a list of reasons why") but as written it would seem
> to discourage transport innovation atop UDP (e.g. QUIC, the RTCWEB data
> channel, anything-over-PLUS), which I very much hope was not the intent.
> The problematic recommendation is in the second paragraph:
> >
> >    These mechanisms are difficult to implement correctly.  For most
> >    applications, the use of one of the existing IETF transport protocols
> >    is the simplest method of acquiring the required mechanisms.  Doing
> >    so also avoids issues that protocols using a new IP protocol number
> >    face when being deployed over the Internet, where middleboxes that
> >    only support TCP and UDP are not rare.  Consequently, the RECOMMENDED
> >    alternative to the UDP usage described in the remainder of this
> >    section is the use of an IETF transport protocol such as TCP
> >    [RFC0793], Stream Control Transmission Protocol (SCTP) [RFC4960], and
> >    SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram
> >    Congestion Control Protocol (DCCP) [RFC4340] with its different
> >    congestion control types [RFC4341][RFC4342][RFC5622].
> >
> > First, this paragraph ignores potential deployment issues with any of
> these other than TCP, which risks seeming out of touch, but this is a minor
> point and probably not worth a late edit. Second, I'm concerned this
> recommendation could be taken as broader than intended, against the
> definition of any new transport protocol encapsulated within UDP that
> performs substantially the same function as the listed protocols.
> >
> I would agree, this paragraph also seems a little self
> contradictory.There is an acknowledgment that "middleboxes that only
> support TCP and UDP are not rare", but then the next sentence
> recommends the use of several other protocols besides UDP and TCP. If
> I put these two together, the only congested controlled protocol that
> is recommended and expected to work on the Internet is TCP.
>

+1   TCP (implemented in kernel space) can't possibly evolve congestion
avoidance as fast as the Internet has changed, or will change (example:
good handling of middle boxes that use "policers" rather than some flavor
of buffer-size based packet-drop).
The rationale for using UDP for QUIC was indeed that a new IP number would
never make it through the Internet (other protocol deployment attempts have
commonly verified this).  It turns out that even UDP is partially blocked
(as recently as 2011) by paths to 5-7% of all chrome clients, which lead to
the "automated fallback" elements of QUIC,

Hopefully the benefits that QUIC is bringing to the Internet will not be
outlawed (precluded by such commentary).

Jim


>
> Tom
>
> > I think this can be made clearer by simply adding to the list of
> examples:
> >
> > NEW:
> >
> >    These mechanisms are difficult to implement correctly.  For most
> >    applications, the use of one of the existing IETF transport protocols
> >    is the simplest method of acquiring the required mechanisms.  Doing
> >    so also avoids issues that protocols using a new IP protocol number
> >    face when being deployed over the Internet, where middleboxes that
> >    only support TCP and UDP are not rare.  Consequently, the RECOMMENDED
> >    alternative to the UDP usage described in the remainder of this
> >    section is the use of an IETF transport protocol such as TCP
> >    [RFC0793], Stream Control Transmission Protocol (SCTP) [RFC4960], and
> >    SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram
> >    Congestion Control Protocol (DCCP) [RFC4340] with its different
> >    congestion control types [RFC4341][RFC4342][RFC5622], or transport
> >    protocols specified by the IETF in the future.
> >
> > and removing the examples from the summary in section 7:
> >
> > OLD:
> >
> >    | SHOULD use a full-featured transport (TCP, SCTP, DCCP)  |         |
> >
> > NEW:
> >
> >    | SHOULD use a full-featured transport                    |         |
> >
> > Thanks, cheers,
> >
> > Brian
> >
> >
> >> On 18 May 2016, at 02:17, The IESG <iesg-secretary@ietf.org> wrote:
> >>
> >>
> >> The IESG has received a request from the Transport Area Working Group WG
> >> (tsvwg) to consider the following document:
> >> - 'UDP Usage Guidelines'
> >>  <draft-ietf-tsvwg-rfc5405bis-13.txt> as Best Current Practice
> >>
> >> The IESG plans to make a decision in the next few weeks, and solicits
> >> final comments on this action. Please send substantive comments to the
> >> ietf@ietf.org mailing lists by 2016-05-31. Exceptionally, comments may
> be
> >> sent to iesg@ietf.org instead. In either case, please retain the
> >> beginning of the Subject line to allow automated sorting.
> >>
> >> Abstract
> >>
> >>
> >>   The User Datagram Protocol (UDP) provides a minimal message-passing
> >>   transport that has no inherent congestion control mechanisms.  This
> >>   document provides guidelines on the use of UDP for the designers of
> >>   applications, tunnels and other protocols that use UDP.  Congestion
> >>   control guidelines are a primary focus, but the document also
> >>   provides guidance on other topics, including message sizes,
> >>   reliability, checksums, middlebox traversal, the use of ECN, DSCPs,
> >>   and ports.
> >>
> >>   Because congestion control is critical to the stable operation of the
> >>   Internet, applications and other protocols that choose to use UDP as
> >>   an Internet transport must employ mechanisms to prevent congestion
> >>   collapse and to establish some degree of fairness with concurrent
> >>   traffic.  They may also need to implement additional mechanisms,
> >>   depending on how they use UDP.
> >>
> >>   Some guidance is also applicable to the design of other protocols
> >>   (e.g., protocols layered directly on IP or via IP-based tunnels),
> >>   especially when these protocols do not themselves provide congestion
> >>   control.
> >>
> >>   This document obsoletes RFC5405 and adds guidelines for multicast UDP
> >>   usage.
> >>
> >>
> >>
> >>
> >> The file can be obtained via
> >> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/
> >>
> >> IESG discussion can be tracked via
> >> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/ballot/
> >>
> >>
> >> No IPR declarations have been submitted directly on this I-D.
> >>
> >>
> >
> >
> > _______________________________________________
> > Spud mailing list
> > Spud@ietf.org
> > https://www.ietf.org/mailman/listinfo/spud
> >
>
> _______________________________________________
> QUIC mailing list
> QUIC@ietf.org
> https://www.ietf.org/mailman/listinfo/quic
>

--001a114d8a20402fb30534953324
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Sat, Jun 4, 2016 at 5:25 PM, Tom Herbert <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:tom@herbertland.com" target=3D"_blank">tom@herbertland.com</a>=
&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px=
 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-co=
lor:rgb(204,204,204);padding-left:1ex"><span class=3D"">On Thu, Jun 2, 2016=
 at 3:11 AM, Brian Trammell &lt;<a href=3D"mailto:ietf@trammell.ch">ietf@tr=
ammell.ch</a>&gt; wrote:<br>
&gt; Greetings, all,<br>
&gt;<br>
&gt; Apologies for the late last call comment; I have only one, relatively =
minor. I hope it&#39;s still useful.<br>
&gt;<br>
&gt; I understand that Section 3 was written to encourage application devel=
opers not to roll their own transports (&quot;trust us when we say this is =
hard, this document is a list of reasons why&quot;) but as written it would=
 seem to discourage transport innovation atop UDP (e.g. QUIC, the RTCWEB da=
ta channel, anything-over-PLUS), which I very much hope was not the intent.=
 The problematic recommendation is in the second paragraph:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 These mechanisms are difficult to implement correctly.=C2=
=A0 For most<br>
&gt;=C2=A0 =C2=A0 applications, the use of one of the existing IETF transpo=
rt protocols<br>
&gt;=C2=A0 =C2=A0 is the simplest method of acquiring the required mechanis=
ms.=C2=A0 Doing<br>
&gt;=C2=A0 =C2=A0 so also avoids issues that protocols using a new IP proto=
col number<br>
&gt;=C2=A0 =C2=A0 face when being deployed over the Internet, where middleb=
oxes that<br>
&gt;=C2=A0 =C2=A0 only support TCP and UDP are not rare.=C2=A0 Consequently=
, the RECOMMENDED<br>
&gt;=C2=A0 =C2=A0 alternative to the UDP usage described in the remainder o=
f this<br>
&gt;=C2=A0 =C2=A0 section is the use of an IETF transport protocol such as =
TCP<br>
&gt;=C2=A0 =C2=A0 [RFC0793], Stream Control Transmission Protocol (SCTP) [R=
FC4960], and<br>
&gt;=C2=A0 =C2=A0 SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], o=
r Datagram<br>
&gt;=C2=A0 =C2=A0 Congestion Control Protocol (DCCP) [RFC4340] with its dif=
ferent<br>
&gt;=C2=A0 =C2=A0 congestion control types [RFC4341][RFC4342][RFC5622].<br>
&gt;<br>
&gt; First, this paragraph ignores potential deployment issues with any of =
these other than TCP, which risks seeming out of touch, but this is a minor=
 point and probably not worth a late edit. Second, I&#39;m concerned this r=
ecommendation could be taken as broader than intended, against the definiti=
on of any new transport protocol encapsulated within UDP that performs subs=
tantially the same function as the listed protocols.<br>
&gt;<br>
</span>I would agree, this paragraph also seems a little self<br>
contradictory.There is an acknowledgment that &quot;middleboxes that only<b=
r>
support TCP and UDP are not rare&quot;, but then the next sentence<br>
recommends the use of several other protocols besides UDP and TCP. If<br>
I put these two together, the only congested controlled protocol that<br>
is recommended and expected to work on the Internet is TCP.<br></blockquote=
><div><br></div><div>+1 =C2=A0 TCP (implemented in kernel space) can&#39;t =
possibly evolve congestion avoidance as fast as the Internet has changed, o=
r will change (example: good handling of middle boxes that use &quot;police=
rs&quot; rather than some flavor of buffer-size based packet-drop).=C2=A0</=
div><div>The rationale for using UDP for QUIC was indeed that a new IP numb=
er would never make it through the Internet (other protocol deployment atte=
mpts have commonly verified this).=C2=A0 It turns out that even UDP is part=
ially blocked (as recently as 2011) by paths to 5-7% of all chrome clients,=
 which lead to the &quot;automated fallback&quot; elements of QUIC, =C2=A0=
=C2=A0</div><div><br></div><div>Hopefully the benefits that QUIC is bringin=
g to the Internet will not be outlawed (precluded by such commentary).</div=
><div><br></div><div>Jim</div><div>=C2=A0</div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-s=
tyle:solid;border-left-color:rgb(204,204,204);padding-left:1ex">
<br>
Tom<br>
<div><div class=3D"h5"><br>
&gt; I think this can be made clearer by simply adding to the list of examp=
les:<br>
&gt;<br>
&gt; NEW:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 These mechanisms are difficult to implement correctly.=C2=
=A0 For most<br>
&gt;=C2=A0 =C2=A0 applications, the use of one of the existing IETF transpo=
rt protocols<br>
&gt;=C2=A0 =C2=A0 is the simplest method of acquiring the required mechanis=
ms.=C2=A0 Doing<br>
&gt;=C2=A0 =C2=A0 so also avoids issues that protocols using a new IP proto=
col number<br>
&gt;=C2=A0 =C2=A0 face when being deployed over the Internet, where middleb=
oxes that<br>
&gt;=C2=A0 =C2=A0 only support TCP and UDP are not rare.=C2=A0 Consequently=
, the RECOMMENDED<br>
&gt;=C2=A0 =C2=A0 alternative to the UDP usage described in the remainder o=
f this<br>
&gt;=C2=A0 =C2=A0 section is the use of an IETF transport protocol such as =
TCP<br>
&gt;=C2=A0 =C2=A0 [RFC0793], Stream Control Transmission Protocol (SCTP) [R=
FC4960], and<br>
&gt;=C2=A0 =C2=A0 SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], o=
r Datagram<br>
&gt;=C2=A0 =C2=A0 Congestion Control Protocol (DCCP) [RFC4340] with its dif=
ferent<br>
&gt;=C2=A0 =C2=A0 congestion control types [RFC4341][RFC4342][RFC5622], or =
transport<br>
&gt;=C2=A0 =C2=A0 protocols specified by the IETF in the future.<br>
&gt;<br>
&gt; and removing the examples from the summary in section 7:<br>
&gt;<br>
&gt; OLD:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 | SHOULD use a full-featured transport (TCP, SCTP, DCCP)=
=C2=A0 |=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0|<br>
&gt;<br>
&gt; NEW:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 | SHOULD use a full-featured transport=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|<br>
&gt;<br>
&gt; Thanks, cheers,<br>
&gt;<br>
&gt; Brian<br>
&gt;<br>
&gt;<br>
&gt;&gt; On 18 May 2016, at 02:17, The IESG &lt;<a href=3D"mailto:iesg-secr=
etary@ietf.org">iesg-secretary@ietf.org</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; The IESG has received a request from the Transport Area Working Gr=
oup WG<br>
&gt;&gt; (tsvwg) to consider the following document:<br>
&gt;&gt; - &#39;UDP Usage Guidelines&#39;<br>
&gt;&gt;=C2=A0 &lt;draft-ietf-tsvwg-rfc5405bis-13.txt&gt; as Best Current P=
ractice<br>
&gt;&gt;<br>
&gt;&gt; The IESG plans to make a decision in the next few weeks, and solic=
its<br>
&gt;&gt; final comments on this action. Please send substantive comments to=
 the<br>
&gt;&gt; <a href=3D"mailto:ietf@ietf.org">ietf@ietf.org</a> mailing lists b=
y 2016-05-31. Exceptionally, comments may be<br>
&gt;&gt; sent to <a href=3D"mailto:iesg@ietf.org">iesg@ietf.org</a> instead=
. In either case, please retain the<br>
&gt;&gt; beginning of the Subject line to allow automated sorting.<br>
&gt;&gt;<br>
&gt;&gt; Abstract<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0The User Datagram Protocol (UDP) provides a minimal me=
ssage-passing<br>
&gt;&gt;=C2=A0 =C2=A0transport that has no inherent congestion control mech=
anisms.=C2=A0 This<br>
&gt;&gt;=C2=A0 =C2=A0document provides guidelines on the use of UDP for the=
 designers of<br>
&gt;&gt;=C2=A0 =C2=A0applications, tunnels and other protocols that use UDP=
.=C2=A0 Congestion<br>
&gt;&gt;=C2=A0 =C2=A0control guidelines are a primary focus, but the docume=
nt also<br>
&gt;&gt;=C2=A0 =C2=A0provides guidance on other topics, including message s=
izes,<br>
&gt;&gt;=C2=A0 =C2=A0reliability, checksums, middlebox traversal, the use o=
f ECN, DSCPs,<br>
&gt;&gt;=C2=A0 =C2=A0and ports.<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0Because congestion control is critical to the stable o=
peration of the<br>
&gt;&gt;=C2=A0 =C2=A0Internet, applications and other protocols that choose=
 to use UDP as<br>
&gt;&gt;=C2=A0 =C2=A0an Internet transport must employ mechanisms to preven=
t congestion<br>
&gt;&gt;=C2=A0 =C2=A0collapse and to establish some degree of fairness with=
 concurrent<br>
&gt;&gt;=C2=A0 =C2=A0traffic.=C2=A0 They may also need to implement additio=
nal mechanisms,<br>
&gt;&gt;=C2=A0 =C2=A0depending on how they use UDP.<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0Some guidance is also applicable to the design of othe=
r protocols<br>
&gt;&gt;=C2=A0 =C2=A0(e.g., protocols layered directly on IP or via IP-base=
d tunnels),<br>
&gt;&gt;=C2=A0 =C2=A0especially when these protocols do not themselves prov=
ide congestion<br>
&gt;&gt;=C2=A0 =C2=A0control.<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0This document obsoletes RFC5405 and adds guidelines fo=
r multicast UDP<br>
&gt;&gt;=C2=A0 =C2=A0usage.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; The file can be obtained via<br>
&gt;&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc54=
05bis/" rel=3D"noreferrer" target=3D"_blank">https://datatracker.ietf.org/d=
oc/draft-ietf-tsvwg-rfc5405bis/</a><br>
&gt;&gt;<br>
&gt;&gt; IESG discussion can be tracked via<br>
&gt;&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc54=
05bis/ballot/" rel=3D"noreferrer" target=3D"_blank">https://datatracker.iet=
f.org/doc/draft-ietf-tsvwg-rfc5405bis/ballot/</a><br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; No IPR declarations have been submitted directly on this I-D.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;<br>
&gt;<br>
</div></div>&gt; _______________________________________________<br>
&gt; Spud mailing list<br>
&gt; <a href=3D"mailto:Spud@ietf.org">Spud@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/spud" rel=3D"noreferr=
er" target=3D"_blank">https://www.ietf.org/mailman/listinfo/spud</a><br>
<div class=3D""><div class=3D"h5">&gt;<br>
<br>
_______________________________________________<br>
QUIC mailing list<br>
<a href=3D"mailto:QUIC@ietf.org">QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/quic</a><br>
</div></div></blockquote></div><br></div></div>

--001a114d8a20402fb30534953324--


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From: "Templin, Fred L" <Fred.L.Templin@boeing.com>
To: Tom Herbert <tom@herbertland.com>, Xuxiaohu <xuxiaohu@huawei.com>
Thread-Topic: [tsvwg] [Int-area] [nvo3] [Softwires] Is it feasible to perform fragmentation on UDP encapsulated packets.
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Date: Mon, 6 Jun 2016 20:37:58 +0000
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Subject: Re: [tsvwg] [Int-area] [nvo3] [Softwires] Is it feasible to perform fragmentation on UDP encapsulated packets.
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From nobody Mon Jun  6 13:48:15 2016
Return-Path: <lucy.yong@huawei.com>
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From: Lucy yong <lucy.yong@huawei.com>
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Date: Tue, 7 Jun 2016 22:59:19 -0700
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To: Tom Herbert <tom@herbertland.com>
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Cc: tsvwg WG <tsvwg@ietf.org>, draft-ietf-tsvwg-rfc5405bis@ietf.org, tsvwg-chairs@ietf.org, spud <spud@ietf.org>, quic@ietf.org, ietf@ietf.org
Subject: Re: [tsvwg] [Spud] Last Call: <draft-ietf-tsvwg-rfc5405bis-13.txt> (UDP Usage Guidelines) to Best Current Practice
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--001a11c113a0a771ee0534be01e4
Content-Type: text/plain; charset=UTF-8

Just adding my 2c if this still isn't too late: I agree with Brian that the
wording in the draft seems limiting. His suggested wording does make things
better, but there still is the contradiction that Tom pointed out:

I would agree, this paragraph also seems a little self
> contradictory.There is an acknowledgment that "middleboxes that only
> support TCP and UDP are not rare", but then the next sentence
> recommends the use of several other protocols besides UDP and TCP. If
> I put these two together, the only congested controlled protocol that
> is recommended and expected to work on the Internet is TCP.


This caught my attention as well. As it stands, the recommendation in this
paragraph is very unclear.

What exactly is the goal here? If it is to ensure that the recommendations
allow for deployable protocols, then UDP-based ones must be allowed.
Otherwise, the recommendation seems to be to only use TCP. I don't think
that was the intent... but then what is the intent in this paragraph?

- jana

--001a11c113a0a771ee0534be01e4
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr">Just adding my 2c if this still isn&#39;t too late: I agre=
e with Brian that the wording in the draft seems limiting. His suggested wo=
rding does make things better, but there still is the contradiction that To=
m pointed out:<div><br><div class=3D"gmail_extra"><div class=3D"gmail_quote=
"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex">I would agree, this paragraph also seems a=
 little self<br>
contradictory.There is an acknowledgment that &quot;middleboxes that only<b=
r>
support TCP and UDP are not rare&quot;, but then the next sentence<br>
recommends the use of several other protocols besides UDP and TCP. If<br>
I put these two together, the only congested controlled protocol that<br>
is recommended and expected to work on the Internet is TCP.</blockquote><di=
v><br></div><div>This caught my attention as well. As it stands, the recomm=
endation in this paragraph is very unclear.=C2=A0</div><div><br></div><div>=
What exactly is the goal here? If it is to ensure that the recommendations =
allow for deployable protocols, then UDP-based ones must be allowed. Otherw=
ise, the recommendation seems to be to only use TCP. I don&#39;t think that=
 was the intent... but then what is the intent in this paragraph?</div><div=
><br></div><div>- jana</div></div></div></div></div>

--001a11c113a0a771ee0534be01e4--


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Subject: Re: [tsvwg] [tcpm] New Version Notification for draft-khademi-tcpm-alternativebackoff-ecn-00.txt
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Dear John,

Thank you very much for taking the time to read and comment on this, and =
very sorry for my long delay in getting back to you. Busy days...
More below; in my answers, I try to focus on comments that concern the =
current draft:

> On 01 Jun 2016, at 03:51, John Leslie <john@jlc.net> wrote:
>=20
> Michael Welzl <michawe@ifi.uio.no> wrote:
>> On 31. mai 2016, at 22.34, John Leslie <john@jlc.net> wrote:
>>> Michael Welzl <michawe@ifi.uio.no> wrote:
>>>=20
>>>> 1) a document that only makes the necessary update to RFC3168 to =
allow a
>>>> reaction to ECN that differs from the reaction to loss:
>>>> This is draft-khademi-tsvwg-ecn-response, which I just posted.
>>>=20
>>>  Three cheers!!!
>>>=20
>>>> 2) a document that only contains the concrete change to TCP that we
>>>> suggest (the new multiplication factor):
>>>> This is draft-khademi-tcpm-alternativebackoff-ecn, with details =
below.
>>>>=20
>>>> Please read and discuss item 2 here in TCPM - thanks!!   :)
>>>=20
>>>  You don't specify where to discuss item 1).
>>=20
>> Sorry - I did, but only in the corresponding email that went to TSVWG =
only.
>> =3D> TSVWG is the place for that.
>=20
>   So, this email is going to TSVWG...
>>=20
>>> Thus, I would prefer that alternativebackoff-ecn not limit us to the
>>> scheme in ecn-response.
>>=20
>> You mean the other way round, right?
>=20
>   I'll take your word for it. (The interconnection of documents =
between
> two WGs is enough to confuse me when I'm trying to get out the door to =
a
> dinner meeting!)
>=20
>> If so, then that???s how it is - draft-khademi-tsvwg-ecn-response =
just
>> mentions draft-khademi-tcpm-alternativebackoff-ecn as one example.
>=20
>   I was pretty thoroghly confused when reading ecn-response too =
quickly.
>=20
>   There's an inherent problem talking about ECN: it involves router
> action on the internetworking layer (IP) and sender action on the
> transport layer (e.g. TCP). Reading Section 1, it's hard to separate
> what RFCs 3168 and 5681 say from what we will propose.
>=20
>   In particular, I question whether the "congestion window" is an
> appropriate concept for the internetworking layer. Routers (or other
> "forwarders") simply mark congestion experienced within the two bits
> allocated to ECN in the IP header. It can only confuse folks to
> couple this with what the sender will do when it receives notice of
> the congestion marking.
>=20
>   I understand this pretty well (IMHO); but _I_ was confused. "cwnd"
> and "ssthresh" are internal details of the TCP treatment of congestion
> marking, and we _do_ have to specify _that_ we are changing these; but
> mentioning them in Section 1, but not explaining the changes until
> Section 3, is unnecessarily confusing.

>   When I got to Section 3, I was running out of time. :^(
>=20
>   The actual 3168 text in Section 3.2 has _always_ been problemmatic.
> The concept of "ECN-Capable transport" is poorly defined; bordering
> on PFM "pure f-ing magic" of how the sender is supposed to "receive
> a CE-marked packet".

RFC3168 has some text that isn't specific to TCP and just talks about =
"congestion control algorithms followed at the end systems".
This text doesn't specify how the information about a CE-mark is =
reflected from the receiver to the sender, because it can't - e.g. the =
text probably applies to QUIC just as well as TCP.

Then, it has a TCP-specific section, in which it describes how TCP =
reflects the CE-mark, and how a TCP sender responds to it.
We change both these text pieces in this draft.

You say you were confused by the early mention of "cwnd" and "ssthresh". =
I suggest to address this in the introduction by being explicit there =
about what I just wrote here - that there is a generic statement about =
the behavior of congestion controls, and a TCP-specific section in =
RFC3168, and we're changing both. Then, we can continue to talk about =
TCP as an example.



>   I don't think it's a good idea to try to minimize the text changes
> to RFC 3168. So, IMHO, the fourth paragraph of Section 3.2 is not
> helpful.

This seems to be related to your confusion about how the ECE-mark =
"magically" gets back to the sender. I hope that I was able to clarify =
this enough here - this text necessarily leaves this open because it's =
not about TCP alone.


>   I must note that the eighth paragraph is much better -- except for
> having to guess where the end-quote is.

Okay, thanks for pointing this out: the last sentence (after the end =
quote) should be a paragraph on its own.


> However "the semantics defined
> in [RFC3168]" seems less than helpful. It's hard to find exactly what
> 3168 text is being referenced here: most, at least, require exact
> duplication of the reaction to loss.

Ok, we can probably remove this bit about the semantics. The point was =
to say the same as section 2.2 does.


>   BTW, "should be treated just as a congestion loss" isn't even
> possible IMHO; and we really-should simply excise every occurance
> of this text from 3168-as-modified.

My short answer is that this surely isn't the point of the current =
draft.


>   As I read Section 3.2 quickly, I assumed it intended to specify
> that "the TCP source halves the congestion window 'cwnd' and reduces
> the slow start threshhold 'ssthresh' " -- subject only to substituting
> some other fraction for "halves".
>=20
>   (I'm not sure the more-careful reading I'm doing now actually
> changes this!)

IIUC this is about RFC3168, not our draft.


> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
>   I could continue this, but most readers would die of boredom...
>=20
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
>   What I actually wanted to see was a clear statement that =
ecn-response
> was revising 3168 by removing the "MUST" requirement for these =
algorithms
> when the congestion is marked by the ECN CE instead of packet loss; =
and
> that _an_ experiment is being separately proposed by =
alternativebackoff-ecn
> to experiment with a different reaction by the sender.

I thought our draft does exactly that - with the difference that it =
still says MUST react / reduce the rate, but without the "as if a packet =
was lost" thing.


>   There's still a lot of hand-waving (however we phrase ecn-response)
> about how to tell whether an experiment is working and is safe to =
deploy
> on the big-I Internet. I'm happy to work on that; but it doesn't =
belong
> in this email. Section 3.3 pretty much says that; but the description
> of what limits there may be to experiments is missing. IMHO, it's a
> bad idea to let each experiment change anything it pleases -- there
> should be limits in ecn-response to what is permissible to change.

This is not written in section 3.3 but in our RFC3168 text update =
(section 3.2), where we require a rate reduction by at least 15%.


> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
>   There's yet another issue: how observers of these experiments can
> know what they're observing. I _really_don't_ want to discuss that in
> this email...
>=20
> --
> John Leslie <john@jlc.net>


Cheers,
Michael


From nobody Wed Jun  8 05:20:46 2016
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Subject: [tsvwg] Last Call: <draft-ietf-tsvwg-rfc5405bis-13.txt> (UDP Usage Guidelines) to Best Current Practice
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On Sunday, June 5, 2016, Jim Roskind <JimRoskind@gmail.com
<javascript:_e(%7B%7D,'cvml','JimRoskind@gmail.com');>> wrote:

>
>
> On Sat, Jun 4, 2016 at 5:25 PM, Tom Herbert <tom@herbertland.com> wrote:
>
>> On Thu, Jun 2, 2016 at 3:11 AM, Brian Trammell <ietf@trammell.ch> wrote:
>> > Greetings, all,
>> >
>> > Apologies for the late last call comment; I have only one, relatively
>> minor. I hope it's still useful.
>> >
>> > I understand that Section 3 was written to encourage application
>> developers not to roll their own transports ("trust us when we say this is
>> hard, this document is a list of reasons why") but as written it would seem
>> to discourage transport innovation atop UDP (e.g. QUIC, the RTCWEB data
>> channel, anything-over-PLUS), which I very much hope was not the intent.
>> The problematic recommendation is in the second paragraph:
>> >
>> >    These mechanisms are difficult to implement correctly.  For most
>> >    applications, the use of one of the existing IETF transport protocols
>> >    is the simplest method of acquiring the required mechanisms.  Doing
>> >    so also avoids issues that protocols using a new IP protocol number
>> >    face when being deployed over the Internet, where middleboxes that
>> >    only support TCP and UDP are not rare.  Consequently, the RECOMMENDED
>> >    alternative to the UDP usage described in the remainder of this
>> >    section is the use of an IETF transport protocol such as TCP
>> >    [RFC0793], Stream Control Transmission Protocol (SCTP) [RFC4960], and
>> >    SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram
>> >    Congestion Control Protocol (DCCP) [RFC4340] with its different
>> >    congestion control types [RFC4341][RFC4342][RFC5622].
>> >
>> > First, this paragraph ignores potential deployment issues with any of
>> these other than TCP, which risks seeming out of touch, but this is a minor
>> point and probably not worth a late edit. Second, I'm concerned this
>> recommendation could be taken as broader than intended, against the
>> definition of any new transport protocol encapsulated within UDP that
>> performs substantially the same function as the listed protocols.
>> >
>> I would agree, this paragraph also seems a little self
>> contradictory.There is an acknowledgment that "middleboxes that only
>> support TCP and UDP are not rare", but then the next sentence
>> recommends the use of several other protocols besides UDP and TCP. If
>> I put these two together, the only congested controlled protocol that
>> is recommended and expected to work on the Internet is TCP.
>>
>
> +1   TCP (implemented in kernel space) can't possibly evolve congestion
> avoidance as fast as the Internet has changed, or will change (example:
> good handling of middle boxes that use "policers" rather than some flavor
> of buffer-size based packet-drop).
> The rationale for using UDP for QUIC was indeed that a new IP number would
> never make it through the Internet (other protocol deployment attempts have
> commonly verified this).  It turns out that even UDP is partially blocked
> (as recently as 2011) by paths to 5-7% of all chrome clients, which lead to
> the "automated fallback" elements of QUIC,
>
>
I think this changes with ipv6, no?  Google sees over 26% ipv6 in the usa
and growing quickly

http://www.google.com/intl/en/ipv6/statistics.html

And there is evidence that middle boxes go away in ipv6. .... So perhaps
the world is changing and your assuptions do not hold



>
>
Hopefully the benefits that QUIC is bringing to the Internet will not be
> outlawed (precluded by such commentary).
>
> Jim
>
>
>>
>> Tom
>>
>> > I think this can be made clearer by simply adding to the list of
>> examples:
>> >
>> > NEW:
>> >
>> >    These mechanisms are difficult to implement correctly.  For most
>> >    applications, the use of one of the existing IETF transport protocols
>> >    is the simplest method of acquiring the required mechanisms.  Doing
>> >    so also avoids issues that protocols using a new IP protocol number
>> >    face when being deployed over the Internet, where middleboxes that
>> >    only support TCP and UDP are not rare.  Consequently, the RECOMMENDED
>> >    alternative to the UDP usage described in the remainder of this
>> >    section is the use of an IETF transport protocol such as TCP
>> >    [RFC0793], Stream Control Transmission Protocol (SCTP) [RFC4960], and
>> >    SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram
>> >    Congestion Control Protocol (DCCP) [RFC4340] with its different
>> >    congestion control types [RFC4341][RFC4342][RFC5622], or transport
>> >    protocols specified by the IETF in the future.
>> >
>> > and removing the examples from the summary in section 7:
>> >
>> > OLD:
>> >
>> >    | SHOULD use a full-featured transport (TCP, SCTP, DCCP)  |         |
>> >
>> > NEW:
>> >
>> >    | SHOULD use a full-featured transport                    |         |
>> >
>> > Thanks, cheers,
>> >
>> > Brian
>> >
>> >
>> >> On 18 May 2016, at 02:17, The IESG <iesg-secretary@ietf.org> wrote:
>> >>
>> >>
>> >> The IESG has received a request from the Transport Area Working Group
>> WG
>> >> (tsvwg) to consider the following document:
>> >> - 'UDP Usage Guidelines'
>> >>  <draft-ietf-tsvwg-rfc5405bis-13.txt> as Best Current Practice
>> >>
>> >> The IESG plans to make a decision in the next few weeks, and solicits
>> >> final comments on this action. Please send substantive comments to the
>> >> ietf@ietf.org mailing lists by 2016-05-31. Exceptionally, comments
>> may be
>> >> sent to iesg@ietf.org instead. In either case, please retain the
>> >> beginning of the Subject line to allow automated sorting.
>> >>
>> >> Abstract
>> >>
>> >>
>> >>   The User Datagram Protocol (UDP) provides a minimal message-passing
>> >>   transport that has no inherent congestion control mechanisms.  This
>> >>   document provides guidelines on the use of UDP for the designers of
>> >>   applications, tunnels and other protocols that use UDP.  Congestion
>> >>   control guidelines are a primary focus, but the document also
>> >>   provides guidance on other topics, including message sizes,
>> >>   reliability, checksums, middlebox traversal, the use of ECN, DSCPs,
>> >>   and ports.
>> >>
>> >>   Because congestion control is critical to the stable operation of the
>> >>   Internet, applications and other protocols that choose to use UDP as
>> >>   an Internet transport must employ mechanisms to prevent congestion
>> >>   collapse and to establish some degree of fairness with concurrent
>> >>   traffic.  They may also need to implement additional mechanisms,
>> >>   depending on how they use UDP.
>> >>
>> >>   Some guidance is also applicable to the design of other protocols
>> >>   (e.g., protocols layered directly on IP or via IP-based tunnels),
>> >>   especially when these protocols do not themselves provide congestion
>> >>   control.
>> >>
>> >>   This document obsoletes RFC5405 and adds guidelines for multicast UDP
>> >>   usage.
>> >>
>> >>
>> >>
>> >>
>> >> The file can be obtained via
>> >> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/
>> >>
>> >> IESG discussion can be tracked via
>> >> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/ballot/
>> >>
>> >>
>> >> No IPR declarations have been submitted directly on this I-D.
>> >>
>> >>
>> >
>> >
>> > _______________________________________________
>> > Spud mailing list
>> > Spud@ietf.org
>> > https://www.ietf.org/mailman/listinfo/spud
>> >
>>
>> _______________________________________________
>> QUIC mailing list
>> QUIC@ietf.org
>> https://www.ietf.org/mailman/listinfo/quic
>>
>
>

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<br><br>On Sunday, June 5, 2016, Jim Roskind &lt;<a href=3D"javascript:_e(%=
7B%7D,&#39;cvml&#39;,&#39;JimRoskind@gmail.com&#39;);" target=3D"_blank">Ji=
mRoskind@gmail.com</a>&gt; wrote:<br><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div di=
r=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On =
Sat, Jun 4, 2016 at 5:25 PM, Tom Herbert <span dir=3D"ltr">&lt;<a>tom@herbe=
rtland.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;=
border-left-color:rgb(204,204,204);padding-left:1ex"><span>On Thu, Jun 2, 2=
016 at 3:11 AM, Brian Trammell &lt;<a>ietf@trammell.ch</a>&gt; wrote:<br>
&gt; Greetings, all,<br>
&gt;<br>
&gt; Apologies for the late last call comment; I have only one, relatively =
minor. I hope it&#39;s still useful.<br>
&gt;<br>
&gt; I understand that Section 3 was written to encourage application devel=
opers not to roll their own transports (&quot;trust us when we say this is =
hard, this document is a list of reasons why&quot;) but as written it would=
 seem to discourage transport innovation atop UDP (e.g. QUIC, the RTCWEB da=
ta channel, anything-over-PLUS), which I very much hope was not the intent.=
 The problematic recommendation is in the second paragraph:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 These mechanisms are difficult to implement correctly.=C2=
=A0 For most<br>
&gt;=C2=A0 =C2=A0 applications, the use of one of the existing IETF transpo=
rt protocols<br>
&gt;=C2=A0 =C2=A0 is the simplest method of acquiring the required mechanis=
ms.=C2=A0 Doing<br>
&gt;=C2=A0 =C2=A0 so also avoids issues that protocols using a new IP proto=
col number<br>
&gt;=C2=A0 =C2=A0 face when being deployed over the Internet, where middleb=
oxes that<br>
&gt;=C2=A0 =C2=A0 only support TCP and UDP are not rare.=C2=A0 Consequently=
, the RECOMMENDED<br>
&gt;=C2=A0 =C2=A0 alternative to the UDP usage described in the remainder o=
f this<br>
&gt;=C2=A0 =C2=A0 section is the use of an IETF transport protocol such as =
TCP<br>
&gt;=C2=A0 =C2=A0 [RFC0793], Stream Control Transmission Protocol (SCTP) [R=
FC4960], and<br>
&gt;=C2=A0 =C2=A0 SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], o=
r Datagram<br>
&gt;=C2=A0 =C2=A0 Congestion Control Protocol (DCCP) [RFC4340] with its dif=
ferent<br>
&gt;=C2=A0 =C2=A0 congestion control types [RFC4341][RFC4342][RFC5622].<br>
&gt;<br>
&gt; First, this paragraph ignores potential deployment issues with any of =
these other than TCP, which risks seeming out of touch, but this is a minor=
 point and probably not worth a late edit. Second, I&#39;m concerned this r=
ecommendation could be taken as broader than intended, against the definiti=
on of any new transport protocol encapsulated within UDP that performs subs=
tantially the same function as the listed protocols.<br>
&gt;<br>
</span>I would agree, this paragraph also seems a little self<br>
contradictory.There is an acknowledgment that &quot;middleboxes that only<b=
r>
support TCP and UDP are not rare&quot;, but then the next sentence<br>
recommends the use of several other protocols besides UDP and TCP. If<br>
I put these two together, the only congested controlled protocol that<br>
is recommended and expected to work on the Internet is TCP.<br></blockquote=
><div><br></div><div>+1 =C2=A0 TCP (implemented in kernel space) can&#39;t =
possibly evolve congestion avoidance as fast as the Internet has changed, o=
r will change (example: good handling of middle boxes that use &quot;police=
rs&quot; rather than some flavor of buffer-size based packet-drop).=C2=A0</=
div><div>The rationale for using UDP for QUIC was indeed that a new IP numb=
er would never make it through the Internet (other protocol deployment atte=
mpts have commonly verified this).=C2=A0 It turns out that even UDP is part=
ially blocked (as recently as 2011) by paths to 5-7% of all chrome clients,=
 which lead to the &quot;automated fallback&quot; elements of QUIC, =C2=A0=
=C2=A0</div><div><br></div></div></div></div></blockquote><div><br></div><d=
iv>I think this changes with ipv6, no?=C2=A0<span></span>=C2=A0Google sees =
over 26% ipv6 in the usa and growing quickly</div><div><br></div><div><a hr=
ef=3D"http://www.google.com/intl/en/ipv6/statistics.html">http://www.google=
.com/intl/en/ipv6/statistics.html</a><br></div><div><br></div><div>And ther=
e is evidence that middle boxes go away in ipv6. .... So perhaps the world =
is changing and your assuptions do not hold</div><div><br></div><br><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div cl=
ass=3D"gmail_quote"><div><br></div><div>=C2=A0<br></div></div></div></div><=
/blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"g=
mail_extra"><div class=3D"gmail_quote"><div>Hopefully the benefits that QUI=
C is bringing to the Internet will not be outlawed (precluded by such comme=
ntary).</div><div><br></div><div>Jim</div><div>=C2=A0</div><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;b=
order-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex"=
>
<br>
Tom<br>
<div><div><br>
&gt; I think this can be made clearer by simply adding to the list of examp=
les:<br>
&gt;<br>
&gt; NEW:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 These mechanisms are difficult to implement correctly.=C2=
=A0 For most<br>
&gt;=C2=A0 =C2=A0 applications, the use of one of the existing IETF transpo=
rt protocols<br>
&gt;=C2=A0 =C2=A0 is the simplest method of acquiring the required mechanis=
ms.=C2=A0 Doing<br>
&gt;=C2=A0 =C2=A0 so also avoids issues that protocols using a new IP proto=
col number<br>
&gt;=C2=A0 =C2=A0 face when being deployed over the Internet, where middleb=
oxes that<br>
&gt;=C2=A0 =C2=A0 only support TCP and UDP are not rare.=C2=A0 Consequently=
, the RECOMMENDED<br>
&gt;=C2=A0 =C2=A0 alternative to the UDP usage described in the remainder o=
f this<br>
&gt;=C2=A0 =C2=A0 section is the use of an IETF transport protocol such as =
TCP<br>
&gt;=C2=A0 =C2=A0 [RFC0793], Stream Control Transmission Protocol (SCTP) [R=
FC4960], and<br>
&gt;=C2=A0 =C2=A0 SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], o=
r Datagram<br>
&gt;=C2=A0 =C2=A0 Congestion Control Protocol (DCCP) [RFC4340] with its dif=
ferent<br>
&gt;=C2=A0 =C2=A0 congestion control types [RFC4341][RFC4342][RFC5622], or =
transport<br>
&gt;=C2=A0 =C2=A0 protocols specified by the IETF in the future.<br>
&gt;<br>
&gt; and removing the examples from the summary in section 7:<br>
&gt;<br>
&gt; OLD:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 | SHOULD use a full-featured transport (TCP, SCTP, DCCP)=
=C2=A0 |=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0|<br>
&gt;<br>
&gt; NEW:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 | SHOULD use a full-featured transport=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|<br>
&gt;<br>
&gt; Thanks, cheers,<br>
&gt;<br>
&gt; Brian<br>
&gt;<br>
&gt;<br>
&gt;&gt; On 18 May 2016, at 02:17, The IESG &lt;<a>iesg-secretary@ietf.org<=
/a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; The IESG has received a request from the Transport Area Working Gr=
oup WG<br>
&gt;&gt; (tsvwg) to consider the following document:<br>
&gt;&gt; - &#39;UDP Usage Guidelines&#39;<br>
&gt;&gt;=C2=A0 &lt;draft-ietf-tsvwg-rfc5405bis-13.txt&gt; as Best Current P=
ractice<br>
&gt;&gt;<br>
&gt;&gt; The IESG plans to make a decision in the next few weeks, and solic=
its<br>
&gt;&gt; final comments on this action. Please send substantive comments to=
 the<br>
&gt;&gt; <a>ietf@ietf.org</a> mailing lists by 2016-05-31. Exceptionally, c=
omments may be<br>
&gt;&gt; sent to <a>iesg@ietf.org</a> instead. In either case, please retai=
n the<br>
&gt;&gt; beginning of the Subject line to allow automated sorting.<br>
&gt;&gt;<br>
&gt;&gt; Abstract<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0The User Datagram Protocol (UDP) provides a minimal me=
ssage-passing<br>
&gt;&gt;=C2=A0 =C2=A0transport that has no inherent congestion control mech=
anisms.=C2=A0 This<br>
&gt;&gt;=C2=A0 =C2=A0document provides guidelines on the use of UDP for the=
 designers of<br>
&gt;&gt;=C2=A0 =C2=A0applications, tunnels and other protocols that use UDP=
.=C2=A0 Congestion<br>
&gt;&gt;=C2=A0 =C2=A0control guidelines are a primary focus, but the docume=
nt also<br>
&gt;&gt;=C2=A0 =C2=A0provides guidance on other topics, including message s=
izes,<br>
&gt;&gt;=C2=A0 =C2=A0reliability, checksums, middlebox traversal, the use o=
f ECN, DSCPs,<br>
&gt;&gt;=C2=A0 =C2=A0and ports.<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0Because congestion control is critical to the stable o=
peration of the<br>
&gt;&gt;=C2=A0 =C2=A0Internet, applications and other protocols that choose=
 to use UDP as<br>
&gt;&gt;=C2=A0 =C2=A0an Internet transport must employ mechanisms to preven=
t congestion<br>
&gt;&gt;=C2=A0 =C2=A0collapse and to establish some degree of fairness with=
 concurrent<br>
&gt;&gt;=C2=A0 =C2=A0traffic.=C2=A0 They may also need to implement additio=
nal mechanisms,<br>
&gt;&gt;=C2=A0 =C2=A0depending on how they use UDP.<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0Some guidance is also applicable to the design of othe=
r protocols<br>
&gt;&gt;=C2=A0 =C2=A0(e.g., protocols layered directly on IP or via IP-base=
d tunnels),<br>
&gt;&gt;=C2=A0 =C2=A0especially when these protocols do not themselves prov=
ide congestion<br>
&gt;&gt;=C2=A0 =C2=A0control.<br>
&gt;&gt;<br>
&gt;&gt;=C2=A0 =C2=A0This document obsoletes RFC5405 and adds guidelines fo=
r multicast UDP<br>
&gt;&gt;=C2=A0 =C2=A0usage.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; The file can be obtained via<br>
&gt;&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc54=
05bis/" rel=3D"noreferrer" target=3D"_blank">https://datatracker.ietf.org/d=
oc/draft-ietf-tsvwg-rfc5405bis/</a><br>
&gt;&gt;<br>
&gt;&gt; IESG discussion can be tracked via<br>
&gt;&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc54=
05bis/ballot/" rel=3D"noreferrer" target=3D"_blank">https://datatracker.iet=
f.org/doc/draft-ietf-tsvwg-rfc5405bis/ballot/</a><br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; No IPR declarations have been submitted directly on this I-D.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;<br>
&gt;<br>
</div></div>&gt; _______________________________________________<br>
&gt; Spud mailing list<br>
&gt; <a>Spud@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/spud" rel=3D"noreferr=
er" target=3D"_blank">https://www.ietf.org/mailman/listinfo/spud</a><br>
<div><div>&gt;<br>
<br>
_______________________________________________<br>
QUIC mailing list<br>
<a>QUIC@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/quic" rel=3D"noreferrer" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/quic</a><br>
</div></div></blockquote></div><br></div></div>
</blockquote>

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From: Brian Trammell <ietf@trammell.ch>
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Subject: Re: [tsvwg] Last Call: <draft-ietf-tsvwg-rfc5405bis-13.txt> (UDP Usage Guidelines) to Best Current Practice
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>=20
>> On 08 Jun 2016, at 14:20, Ca By <cb.list6@gmail.com> wrote:
>>=20
>>=20
>>=20
>> On Sunday, June 5, 2016, Jim Roskind <JimRoskind@gmail.com> wrote:
>>=20
>>=20
>> On Sat, Jun 4, 2016 at 5:25 PM, Tom Herbert <tom@herbertland.com> =
wrote:
>> On Thu, Jun 2, 2016 at 3:11 AM, Brian Trammell <ietf@trammell.ch> =
wrote:
>> > Greetings, all,
>> >
>> > Apologies for the late last call comment; I have only one, =
relatively minor. I hope it's still useful.
>> >
>> > I understand that Section 3 was written to encourage application =
developers not to roll their own transports ("trust us when we say this =
is hard, this document is a list of reasons why") but as written it =
would seem to discourage transport innovation atop UDP (e.g. QUIC, the =
RTCWEB data channel, anything-over-PLUS), which I very much hope was not =
the intent. The problematic recommendation is in the second paragraph:
>> >
>> >    These mechanisms are difficult to implement correctly.  For most
>> >    applications, the use of one of the existing IETF transport =
protocols
>> >    is the simplest method of acquiring the required mechanisms.  =
Doing
>> >    so also avoids issues that protocols using a new IP protocol =
number
>> >    face when being deployed over the Internet, where middleboxes =
that
>> >    only support TCP and UDP are not rare.  Consequently, the =
RECOMMENDED
>> >    alternative to the UDP usage described in the remainder of this
>> >    section is the use of an IETF transport protocol such as TCP
>> >    [RFC0793], Stream Control Transmission Protocol (SCTP) =
[RFC4960], and
>> >    SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or =
Datagram
>> >    Congestion Control Protocol (DCCP) [RFC4340] with its different
>> >    congestion control types [RFC4341][RFC4342][RFC5622].
>> >
>> > First, this paragraph ignores potential deployment issues with any =
of these other than TCP, which risks seeming out of touch, but this is a =
minor point and probably not worth a late edit. Second, I'm concerned =
this recommendation could be taken as broader than intended, against the =
definition of any new transport protocol encapsulated within UDP that =
performs substantially the same function as the listed protocols.
>> >
>> I would agree, this paragraph also seems a little self
>> contradictory.There is an acknowledgment that "middleboxes that only
>> support TCP and UDP are not rare", but then the next sentence
>> recommends the use of several other protocols besides UDP and TCP. If
>> I put these two together, the only congested controlled protocol that
>> is recommended and expected to work on the Internet is TCP.
>>=20
>> +1   TCP (implemented in kernel space) can't possibly evolve =
congestion avoidance as fast as the Internet has changed, or will change =
(example: good handling of middle boxes that use "policers" rather than =
some flavor of buffer-size based packet-drop).
>> The rationale for using UDP for QUIC was indeed that a new IP number =
would never make it through the Internet (other protocol deployment =
attempts have commonly verified this).  It turns out that even UDP is =
partially blocked (as recently as 2011) by paths to 5-7% of all chrome =
clients, which lead to the "automated fallback" elements of QUIC,
>>=20
>=20
> I think this changes with ipv6, no?  Google sees over 26% ipv6 in the =
usa and growing quickly
>=20
> http://www.google.com/intl/en/ipv6/statistics.html
>=20
> And there is evidence that middle boxes go away in ipv6. .... So =
perhaps the world is changing and your assuptions do not hold

(dropping ietf@, saving my Narten points since we're a bit off topic on =
the draft last call)

I'd like to see evidence, if you can share it.

We've done a few measurements with Atlas that suggest the brokenness on =
v6 is different, but not necessarily better, but don't have enough data =
points to make an informed statement on the fact. Any time you touch a =
NAPT (which lives at the core of any transition technology for v6-only =
access to the v4 Internet), you run into the "only TCP and UDP have =
ports" fallacy. But yes, we need to compare and contrast v4 and v6 =
access network impairments for non-(6,17) protocol numbers.

Cheers,

Brian

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From: "Black, David" <david.black@emc.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: WG adoption call: draft-szigeti-tsvwg-ieee-802-11 - through June 25
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Date: Thu, 9 Jun 2016 05:14:24 +0000
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Subject: [tsvwg] WG adoption call: draft-szigeti-tsvwg-ieee-802-11 - through June 25
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This is a TSVWG call for interest in WG adoption of draft-szigeti-tsvwg-iee=
e-802-11.
If you have read the draft and support WG adoption of it as a work item
and WG draft, please send an email to the list (tsvwg@ietf.org) saying so.

NB: It is important to say that you've read the draft.  If you're willing
to review/comment on future versions of this draft and/or implement this
draft, please say so on the list or in email directly to the chairs
(tsvwg-chairs@ietf.org).

Authors: We (chairs) assume that you support WG adoption of your draft, so
there's no need to say so on the list.  However, we do need each author to=
=20
publicly state on the list that their direct, personal knowledge (if any) o=
f any
IPR related to this document has already been disclosed, in conformance
with BCPs 78 and 79.

This adoption call will run for about 2 weeks, ending at midnight, prevaili=
ng
US Eastern Time on Saturday June 25.  To get things started, I hereby state
as an individual that I have read the draft and support WG adoption of it.

             Guidelines for DiffServ to IEEE 802.11 Mapping
                   draft-szigeti-tsvwg-ieee-802-11-01

Abstract

   As internet traffic is increasingly sourced-from and destined-to
   wireless endpoints, it is crucial that Quality of Service be aligned
   between wired and wireless networks; however, this is not always the
   case by default.  This is due to the fact that two independent
   standards bodies provide QoS guidance on wired and wireless networks:
   specifically, the IETF specifies standards and design recommendations
   for wired IP networks, while a separate and autonomous standards-
   body, the IEEE, administers the standards for wireless 802.11
   networks.  The purpose of this document is to propose a set
   Differentiated Services Code Point (DSCP) to IEEE 802.11 User
   Priority (UP) mappings to reconcile the marking recommendations
   offered by these two standards bodies, and, as such, to optimize
   wired-and-wireless interconnect QoS.

Thanks, --David
----------------------------------------------------
David L. Black, Distinguished Engineer
EMC, 176 South St., Hopkinton, MA=A0 01748
+1 (508) 293-7953=A0=A0=A0  Cell: +1 (978) 394-7754
david.black@emc.com=A0=A0=A0=A0=A0=A0=A0=20
----------------------------------------------------



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From: "Fred Baker (fred)" <fred@cisco.com>
To: "Black, David" <david.black@emc.com>
Thread-Topic: [tsvwg] WG adoption call: draft-szigeti-tsvwg-ieee-802-11 - through	June 25
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> On Jun 8, 2016, at 11:14 PM, Black, David <david.black@emc.com> wrote:
>=20
> Authors: We (chairs) assume that you support WG adoption of your =
draft, so
> there's no need to say so on the list.  However, we do need each =
author to
> publicly state on the list that their direct, personal knowledge (if =
any) of any
> IPR related to this document has already been disclosed, in =
conformance
> with BCPs 78 and 79.

I know of no such IPR.

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--001a114b322ac81c71053508f333
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On Wednesday, June 8, 2016, Brian Trammell <ietf@trammell.ch> wrote:

> >
> >> On 08 Jun 2016, at 14:20, Ca By <cb.list6@gmail.com <javascript:;>>
> wrote:
> >>
> >>
> >>
> >> On Sunday, June 5, 2016, Jim Roskind <JimRoskind@gmail.com
> <javascript:;>> wrote:
> >>
> >>
> >> On Sat, Jun 4, 2016 at 5:25 PM, Tom Herbert <tom@herbertland.com
> <javascript:;>> wrote:
> >> On Thu, Jun 2, 2016 at 3:11 AM, Brian Trammell <ietf@trammell.ch
> <javascript:;>> wrote:
> >> > Greetings, all,
> >> >
> >> > Apologies for the late last call comment; I have only one, relatively
> minor. I hope it's still useful.
> >> >
> >> > I understand that Section 3 was written to encourage application
> developers not to roll their own transports ("trust us when we say this is
> hard, this document is a list of reasons why") but as written it would seem
> to discourage transport innovation atop UDP (e.g. QUIC, the RTCWEB data
> channel, anything-over-PLUS), which I very much hope was not the intent.
> The problematic recommendation is in the second paragraph:
> >> >
> >> >    These mechanisms are difficult to implement correctly.  For most
> >> >    applications, the use of one of the existing IETF transport
> protocols
> >> >    is the simplest method of acquiring the required mechanisms.  Doing
> >> >    so also avoids issues that protocols using a new IP protocol number
> >> >    face when being deployed over the Internet, where middleboxes that
> >> >    only support TCP and UDP are not rare.  Consequently, the
> RECOMMENDED
> >> >    alternative to the UDP usage described in the remainder of this
> >> >    section is the use of an IETF transport protocol such as TCP
> >> >    [RFC0793], Stream Control Transmission Protocol (SCTP) [RFC4960],
> and
> >> >    SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram
> >> >    Congestion Control Protocol (DCCP) [RFC4340] with its different
> >> >    congestion control types [RFC4341][RFC4342][RFC5622].
> >> >
> >> > First, this paragraph ignores potential deployment issues with any of
> these other than TCP, which risks seeming out of touch, but this is a minor
> point and probably not worth a late edit. Second, I'm concerned this
> recommendation could be taken as broader than intended, against the
> definition of any new transport protocol encapsulated within UDP that
> performs substantially the same function as the listed protocols.
> >> >
> >> I would agree, this paragraph also seems a little self
> >> contradictory.There is an acknowledgment that "middleboxes that only
> >> support TCP and UDP are not rare", but then the next sentence
> >> recommends the use of several other protocols besides UDP and TCP. If
> >> I put these two together, the only congested controlled protocol that
> >> is recommended and expected to work on the Internet is TCP.
> >>
> >> +1   TCP (implemented in kernel space) can't possibly evolve congestion
> avoidance as fast as the Internet has changed, or will change (example:
> good handling of middle boxes that use "policers" rather than some flavor
> of buffer-size based packet-drop).
> >> The rationale for using UDP for QUIC was indeed that a new IP number
> would never make it through the Internet (other protocol deployment
> attempts have commonly verified this).  It turns out that even UDP is
> partially blocked (as recently as 2011) by paths to 5-7% of all chrome
> clients, which lead to the "automated fallback" elements of QUIC,
> >>
> >
> > I think this changes with ipv6, no?  Google sees over 26% ipv6 in the
> usa and growing quickly
> >
> > http://www.google.com/intl/en/ipv6/statistics.html
> >
> > And there is evidence that middle boxes go away in ipv6. .... So perhaps
> the world is changing and your assuptions do not hold
>
> (dropping ietf@, saving my Narten points since we're a bit off topic on
> the draft last call)
>
> I'd like to see evidence, if you can share it.
>
>
" Verizon uses transparent proxy servers on IPv4 but not on IPv6"

https://blogs.akamai.com/2016/06/preparing-for-ipv6-only-mobile-networks-why-and-how.html


We've done a few measurements with Atlas that suggest the brokenness on v6
> is different, but not necessarily better, but don't have enough data points
> to make an informed statement on the fact. Any time you touch a NAPT (which
> lives at the core of any transition technology for v6-only access to the v4
> Internet), you run into the "only TCP and UDP have ports" fallacy. But yes,
> we need to compare and contrast v4 and v6 access network impairments for
> non-(6,17) protocol numbers.
>
>
NAPT is an edge case for v6 users, less than 30% and shrinking by bit
volume as the whales have moved to v6 already

https://blogs.akamai.com/2016/06/four-years-since-world-ipv6-launch-entering-the-mainstream.html



> Cheers,
>
> Brian
>

--001a114b322ac81c71053508f333
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<br><br>On Wednesday, June 8, 2016, Brian Trammell &lt;<a href=3D"mailto:ie=
tf@trammell.ch">ietf@trammell.ch</a>&gt; wrote:<br><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex">&gt;<br>
&gt;&gt; On 08 Jun 2016, at 14:20, Ca By &lt;<a href=3D"javascript:;" oncli=
ck=3D"_e(event, &#39;cvml&#39;, &#39;cb.list6@gmail.com&#39;)">cb.list6@gma=
il.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; On Sunday, June 5, 2016, Jim Roskind &lt;<a href=3D"javascript:;" =
onclick=3D"_e(event, &#39;cvml&#39;, &#39;JimRoskind@gmail.com&#39;)">JimRo=
skind@gmail.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; On Sat, Jun 4, 2016 at 5:25 PM, Tom Herbert &lt;<a href=3D"javascr=
ipt:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;tom@herbertland.com&#39;)"=
>tom@herbertland.com</a>&gt; wrote:<br>
&gt;&gt; On Thu, Jun 2, 2016 at 3:11 AM, Brian Trammell &lt;<a href=3D"java=
script:;" onclick=3D"_e(event, &#39;cvml&#39;, &#39;ietf@trammell.ch&#39;)"=
>ietf@trammell.ch</a>&gt; wrote:<br>
&gt;&gt; &gt; Greetings, all,<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; Apologies for the late last call comment; I have only one, re=
latively minor. I hope it&#39;s still useful.<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; I understand that Section 3 was written to encourage applicat=
ion developers not to roll their own transports (&quot;trust us when we say=
 this is hard, this document is a list of reasons why&quot;) but as written=
 it would seem to discourage transport innovation atop UDP (e.g. QUIC, the =
RTCWEB data channel, anything-over-PLUS), which I very much hope was not th=
e intent. The problematic recommendation is in the second paragraph:<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;=C2=A0 =C2=A0 These mechanisms are difficult to implement corr=
ectly.=C2=A0 For most<br>
&gt;&gt; &gt;=C2=A0 =C2=A0 applications, the use of one of the existing IET=
F transport protocols<br>
&gt;&gt; &gt;=C2=A0 =C2=A0 is the simplest method of acquiring the required=
 mechanisms.=C2=A0 Doing<br>
&gt;&gt; &gt;=C2=A0 =C2=A0 so also avoids issues that protocols using a new=
 IP protocol number<br>
&gt;&gt; &gt;=C2=A0 =C2=A0 face when being deployed over the Internet, wher=
e middleboxes that<br>
&gt;&gt; &gt;=C2=A0 =C2=A0 only support TCP and UDP are not rare.=C2=A0 Con=
sequently, the RECOMMENDED<br>
&gt;&gt; &gt;=C2=A0 =C2=A0 alternative to the UDP usage described in the re=
mainder of this<br>
&gt;&gt; &gt;=C2=A0 =C2=A0 section is the use of an IETF transport protocol=
 such as TCP<br>
&gt;&gt; &gt;=C2=A0 =C2=A0 [RFC0793], Stream Control Transmission Protocol =
(SCTP) [RFC4960], and<br>
&gt;&gt; &gt;=C2=A0 =C2=A0 SCTP Partial Reliability Extension (SCTP-PR) [RF=
C3758], or Datagram<br>
&gt;&gt; &gt;=C2=A0 =C2=A0 Congestion Control Protocol (DCCP) [RFC4340] wit=
h its different<br>
&gt;&gt; &gt;=C2=A0 =C2=A0 congestion control types [RFC4341][RFC4342][RFC5=
622].<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; First, this paragraph ignores potential deployment issues wit=
h any of these other than TCP, which risks seeming out of touch, but this i=
s a minor point and probably not worth a late edit. Second, I&#39;m concern=
ed this recommendation could be taken as broader than intended, against the=
 definition of any new transport protocol encapsulated within UDP that perf=
orms substantially the same function as the listed protocols.<br>
&gt;&gt; &gt;<br>
&gt;&gt; I would agree, this paragraph also seems a little self<br>
&gt;&gt; contradictory.There is an acknowledgment that &quot;middleboxes th=
at only<br>
&gt;&gt; support TCP and UDP are not rare&quot;, but then the next sentence=
<br>
&gt;&gt; recommends the use of several other protocols besides UDP and TCP.=
 If<br>
&gt;&gt; I put these two together, the only congested controlled protocol t=
hat<br>
&gt;&gt; is recommended and expected to work on the Internet is TCP.<br>
&gt;&gt;<br>
&gt;&gt; +1=C2=A0 =C2=A0TCP (implemented in kernel space) can&#39;t possibl=
y evolve congestion avoidance as fast as the Internet has changed, or will =
change (example: good handling of middle boxes that use &quot;policers&quot=
; rather than some flavor of buffer-size based packet-drop).<br>
&gt;&gt; The rationale for using UDP for QUIC was indeed that a new IP numb=
er would never make it through the Internet (other protocol deployment atte=
mpts have commonly verified this).=C2=A0 It turns out that even UDP is part=
ially blocked (as recently as 2011) by paths to 5-7% of all chrome clients,=
 which lead to the &quot;automated fallback&quot; elements of QUIC,<br>
&gt;&gt;<br>
&gt;<br>
&gt; I think this changes with ipv6, no?=C2=A0 Google sees over 26% ipv6 in=
 the usa and growing quickly<br>
&gt;<br>
&gt; <a href=3D"http://www.google.com/intl/en/ipv6/statistics.html" target=
=3D"_blank">http://www.google.com/intl/en/ipv6/statistics.html</a><br>
&gt;<br>
&gt; And there is evidence that middle boxes go away in ipv6. .... So perha=
ps the world is changing and your assuptions do not hold<br>
<br>
(dropping ietf@, saving my Narten points since we&#39;re a bit off topic on=
 the draft last call)<br>
<br>
I&#39;d like to see evidence, if you can share it.<br>
<br></blockquote><div><br></div><div>&quot;=C2=A0<font size=3D"2"><span sty=
le=3D"background-color:rgba(255,255,255,0)">Verizon uses transparent proxy =
servers on IPv4 but not on IPv6&quot;</span></font></div><div><font size=3D=
"2"><span style=3D"background-color:rgba(255,255,255,0)"><br></span></font>=
</div><div><a href=3D"https://blogs.akamai.com/2016/06/preparing-for-ipv6-o=
nly-mobile-networks-why-and-how.html">https://blogs.akamai.com/2016/06/prep=
aring-for-ipv6-only-mobile-networks-why-and-how.html</a><font size=3D"2"><s=
pan style=3D"background-color:rgba(255,255,255,0)"><br></span></font></div>=
<div><font size=3D"2"><span style=3D"background-color:rgba(255,255,255,0)">=
<br></span></font></div><div><font size=3D"2"><span style=3D"background-col=
or:rgba(255,255,255,0)"><br></span></font></div><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex">
We&#39;ve done a few measurements with Atlas that suggest the brokenness on=
 v6 is different, but not necessarily better, but don&#39;t have enough dat=
a points to make an informed statement on the fact. Any time you touch a NA=
PT (which lives at the core of any transition technology for v6-only access=
 to the v4 Internet), you run into the &quot;only TCP and UDP have ports&qu=
ot; fallacy. But yes, we need to compare and contrast v4 and v6 access netw=
ork impairments for non-(6,17) protocol numbers.<br>
<br></blockquote><div><br></div><div>NAPT is an edge case for v6 users, les=
s than 30% and shrinking by bit volume as the whales have moved to v6 alrea=
dy</div><div><br></div><div><a href=3D"https://blogs.akamai.com/2016/06/fou=
r-years-since-world-ipv6-launch-entering-the-mainstream.html">https://blogs=
.akamai.com/2016/06/four-years-since-world-ipv6-launch-entering-the-mainstr=
eam.html</a><br></div><div><br></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">
Cheers,<br>
<br>
Brian<br>
</blockquote>

--001a114b322ac81c71053508f333--


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Date: Sun, 12 Jun 2016 05:11:48 +0000 (UTC)
From: Lloyd Wood <lloyd.wood@yahoo.co.uk>
To: Softwires WG <softwires@ietf.org>,  "int-area@ietf.org" <int-area@ietf.org>,  "lisp@ietf.org" <lisp@ietf.org>, "tsvwg@ietf.org" <tsvwg@ietf.org>,  Xuxiaohu <xuxiaohu@huawei.com>, "nvo3@ietf.org" <nvo3@ietf.org>
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Subject: Re: [tsvwg] Is it feasible to perform fragmentation on UDP encapsulated	packets.
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------=_Part_2492240_1136524516.1465708308517
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 blockquote, div.yahoo_quoted { margin-left: 0 !important; border-left:1px =
#715FFA solid !important; padding-left:1ex !important; background-color:whi=
te !important; }  Fragmentation should be strongly discouraged.
if you've designed a tunnelling solution and you have fragmentation happeni=
ng as a matter of course, you've designed it wrong.=C2=A0

Lloyd Woodlloyd.wood@yahoo.co.uk

On Friday, May 27, 2016, 8:10 PM, Xuxiaohu <xuxiaohu@huawei.com> wrote:

<Note that I have changed the subject of the email hence it has nothing to =
do with the WG adoption call now. It's just a discussion on a particular is=
sue which is related to those WGs which are working on UDP tunnels. The rea=
son for containing the old email is to use it as a background which may be =
useful for better understanding of this particular issue>

The possible side-effect of performing fragmentation on UDP encapsulated pa=
ckets is to worsen the reassembly burden on tunnel egress since fragments o=
f UDP encapsulated packets are more likely to be forwarded across different=
 paths towards the tunnel egress than those of IP or GRE encapsulated packe=
ts.

It seems that most X-over-UDP proposals choose to prohibit the tunnel ingre=
ss from performing fragmentation on UDP encapsulated packets. See the follo=
wing quoted text regarding fragmentation from those X-over-UDP drafts:

LISP:

When an ITR receives a packet from a site-facing interface and adds H
=C2=A0 octets worth of encapsulation to yield a packet size greater than L
=C2=A0 octets, it resolves the MTU issue by first splitting the original
=C2=A0 packet into 2 equal-sized fragments.=C2=A0 A LISP header is then pre=
pended
=C2=A0 to each fragment.

VXLAN:

VTEPs MUST NOT fragment VXLAN packets.=C2=A0 Intermediate routers may
=C2=A0 fragment encapsulated VXLAN packets due to the larger frame size.
=C2=A0 The destination VTEP MAY silently discard such VXLAN fragments.

VXLAN-GPE:

VTEPs MUST never fragment an encapsulated VXLAN GPE packet, and when
=C2=A0 the outer IP header is IPv4, VTEPs MUST set the DF bit in the outer
=C2=A0 IPv4 header.

GEVENE:

=C2=A0 To prevent fragmentation and maximize performance, the best practice
=C2=A0 when using Geneve is to ensure that the MTU of the physical network
=C2=A0 is greater than or equal to the MTU of the encapsulated network plus
=C2=A0 tunnel headers.

GUE:

=C2=A0 =C2=A0 If a packet is fragmented before encapsulation in GUE, all th=
e
=C2=A0 =C2=A0 related fragments must be encapsulated using the same source =
port
=C2=A0 =C2=A0 (inner flow identifier). An operator may set MTU to account f=
or
=C2=A0 =C2=A0 encapsulation overhead and reduce the likelihood of fragmenta=
tion.

GRE/UDP

Regarding packet fragmentation, an encapsulator/decapsulator SHOULD
=C2=A0 be compliant with [RFC7588] and perform fragmentation before the
=C2=A0 encapsulation.

However, the above choice seems conflict with the requirements as described=
 in https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02


I wonder whether the IETF should reach a consensus on whether or not the fr=
agmentation on UDP encapsulated packets should be allowed.
=C2=A0=20
Best regards,
Xiaohu

> -----Original Message-----
> From: Xuxiaohu
> Sent: Thursday, May 26, 2016 4:35 PM
> To: 'Joe Touch'; Fred Baker (fred); Wassim Haddad
> Cc: int-area@ietf.org
> Subject: RE: [Int-area] Call for adoption of draft-xu-intarea-ip-in-udp-0=
3
>=20
>=20
>=20
> > -----Original Message-----
> > From: Joe Touch [mailto:touch@isi.edu]
> > Sent: Thursday, May 26, 2016 2:11 AM
> > To: Xuxiaohu; Fred Baker (fred); Wassim Haddad
> > Cc: int-area@ietf.org
> > Subject: Re: [Int-area] Call for adoption of
> > draft-xu-intarea-ip-in-udp-03
> >
> >
> >
> > On 5/24/2016 7:24 PM, Xuxiaohu wrote:
> > > Hi Joe,
> > >
> > > I wonder whether you want to tell me the following truth by the
> > > example that you gave: no matter whatever improvements we had done
> > > with this draft, those persons who dislike it by the light of nature
> > > would dislike it in the end.
> >
> > The only improvements that would make this doc useful would be to add
> > capabilities already in GRE/UDP or GUE/UDP, which we already have.
>=20
> Let's go over the four things you mentioned earlier in GRE/UDP and GUE/UD=
P:
>=20
> =C2=A0=C2=A0=C2=A0 - stronger checksums
>=20
> In GRE/UDP, in order to use UDP-zero-checksum, it gave the following
> restrictions:
> " 6. UDP Checksum Handling
>=20
>=C2=A0 =C2=A0 6.1. UDP Checksum with IPv4
>=20
>=C2=A0 =C2=A0 For UDP in IPv4, the UDP checksum MUST be processed as speci=
fied in
>=C2=A0 =C2=A0 [RFC768] and [RFC1122] for both transmit and receive. The IP=
v4
>=20
>=20
>=20
> Yong, Crabber, Xu, Herbert=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 [Page 12]
> -------------------------------------------------------------------------=
-------
>=20
> Internet-Draft=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 GRE-in-UDP Encapsulation=
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 March 2016
>=20
>=C2=A0 =C2=A0 header includes a checksum which protects against mis-delive=
ry of
>=C2=A0 =C2=A0 the packet due to corruption of IP addresses. The UDP checks=
um
>=C2=A0 =C2=A0 potentially provides protection against corruption of the UD=
P header,
>=C2=A0 =C2=A0 GRE header, and GRE payload. Disabling the use of checksums =
is a
>=C2=A0 =C2=A0 deployment consideration that should take into account the r=
isk and
>=C2=A0 =C2=A0 effects of packet corruption.
>=20
>=C2=A0 =C2=A0 When a decapsulator receives a packet, the UDP checksum fiel=
d MUST
>=C2=A0 =C2=A0 be processed. If the UDP checksum is non-zero, the decapsula=
tor MUST
>=C2=A0 =C2=A0 verify the checksum before accepting the packet. By default =
a
>=C2=A0 =C2=A0 decapsulator SHOULD accept UDP packets with a zero checksum.=
 A node
>=C2=A0 =C2=A0 MAY be configured to disallow zero checksums per [RFC1122]; =
this may
>=C2=A0 =C2=A0 be done selectively, for instance disallowing zero checksums=
 from
>=C2=A0 =C2=A0 certain hosts that are known to be sending over paths subjec=
t to
>=C2=A0 =C2=A0 packet corruption. If verification of a non-zero checksum fa=
ils, a
>=C2=A0 =C2=A0 decapsulator lacks the capability to verify a non-zero check=
sum, or
>=C2=A0 =C2=A0 a packet with a zero-checksum was received and the decapsula=
tor is
>=C2=A0 =C2=A0 configured to disallow, the packet MUST be dropped and an ev=
ent MAY
>=C2=A0 =C2=A0 be logged.
>=20
>=C2=A0 =C2=A0 6.2. UDP Checksum with IPv6
>=20
>=C2=A0 =C2=A0 For UDP in IPv6, the UDP checksum MUST be processed as speci=
fied in
>=C2=A0 =C2=A0 [RFC768] and [RFC2460] for both transmit and receive.
>=20
>=C2=A0 =C2=A0 When UDP is used over IPv6, the UDP checksum is relied upon =
to
>=C2=A0 =C2=A0 protect both the IPv6 and UDP headers from corruption. As su=
ch, A
>=C2=A0 =C2=A0 default GRE-in-UDP Tunnel MUST perform UDP checksum; A TMCE =
GRE-in-
>=C2=A0 =C2=A0 UDP Tunnel MAY be configured with the UDP zero-checksum mode=
 if the
>=C2=A0 =C2=A0 traffic-managed controlled environment or a set of closely
>=C2=A0 =C2=A0 cooperating traffic-managed controlled environments (such as=
 by
>=C2=A0 =C2=A0 network operators who have agreed to work together in order =
to
>=C2=A0 =C2=A0 jointly provide specific services) meet at least one of foll=
owing
>=C2=A0 =C2=A0 conditions:
>=20
>=C2=A0 =C2=A0 a. It is known (perhaps through knowledge of equipment types=
 and
>=C2=A0 =C2=A0 =C2=A0 lower layer checks) that packet corruption is excepti=
onally
>=C2=A0 =C2=A0 =C2=A0 unlikely and where the operator is willing to take th=
e risk of
>=C2=A0 =C2=A0 =C2=A0 undetected packet corruption.
>=20
>=C2=A0 =C2=A0 b. It is judged through observational measurements (perhaps =
of
>=C2=A0 =C2=A0 =C2=A0 historic or current traffic flows that use a non-zero=
 checksum)
>=C2=A0 =C2=A0 =C2=A0 that the level of packet corruption is tolerably low =
and where
>=C2=A0 =C2=A0 =C2=A0 the operator is willing to take the risk of undetecte=
d packet
>=C2=A0 =C2=A0 =C2=A0 corruption.
>=20
>=20
>=20
>=20
>=20
> Yong, Crabber, Xu, Herbert=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 [Page 13]
> -------------------------------------------------------------------------=
-------
>=20
> Internet-Draft=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 GRE-in-UDP Encapsulation=
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 March 2016
>=20
>=C2=A0 =C2=A0 c. Carrying applications that are tolerant of mis-delivered =
or
>=C2=A0 =C2=A0 =C2=A0 corrupted packets (perhaps through higher layer check=
sum,
>=C2=A0 =C2=A0 =C2=A0 validation, and retransmission or transmission redund=
ancy) where
>=C2=A0 =C2=A0 =C2=A0 the operator is willing to rely on the applications u=
sing the
>=C2=A0 =C2=A0 =C2=A0 tunnel to survive any corrupt packets.
>=20
>=C2=A0 =C2=A0 The following requirements apply to a TMCE GRE-in-UDP tunnel=
 that
>=C2=A0 =C2=A0 use UDP zero-checksum mode:
>=20
>=C2=A0 =C2=A0 =C2=A0 a. Use of the UDP checksum with IPv6 MUST be the defa=
ult
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 configuration of all GRE-in-UDP tunnels.
>=20
>=C2=A0 =C2=A0 =C2=A0 b. The GRE-in-UDP tunnel implementation MUST comply w=
ith all
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 requirements specified in Section 4 of [RFC693=
6] and with
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 requirement 1 specified in Section 5 of [RFC69=
36].
>=20
>=C2=A0 =C2=A0 =C2=A0 c. The tunnel decapsulator SHOULD only allow the use =
of UDP zero-
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 checksum mode for IPv6 on a single received UD=
P Destination
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 Port regardless of the encapsulator. The motiv=
ation for this
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 requirement is possible corruption of the UDP =
Destination Port,
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 which may cause packet delivery to the wrong U=
DP port. If that
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 other UDP port requires the UDP checksum, the =
mis-delivered
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 packet will be discarded.
>=20
>=C2=A0 =C2=A0 =C2=A0 d. It is RECOMMENDED that the UDP zero-checksum mode =
for IPv6 is
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 only enabled for certain selected source addre=
sses. The tunnel
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 decapsulator MUST check that the source and de=
stination IPv6
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 addresses are valid for the GRE-in-UDP tunnel =
on which the
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 packet was received if that tunnel uses UDP ze=
ro-checksum mode
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 and discard any packet for which this check fa=
ils.
>=20
>=C2=A0 =C2=A0 =C2=A0 e. The tunnel encapsulator SHOULD use different IPv6 =
addresses for
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 each GRE-in-UDP tunnel that uses UDP zero-chec=
ksum mode
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 regardless of the decapsulator in order to str=
engthen the
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 decapsulator's check of the IPv6 source addres=
s (i.e., the same
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 IPv6 source address SHOULD NOT be used with mo=
re than one IPv6
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 destination address, independent of whether th=
at destination
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 address is a unicast or multicast address). Wh=
en this is not
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 possible, it is RECOMMENDED to use each source=
 IPv6 address for
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 as few UDP zero-checksum mode GRE-in-UDP tunne=
ls as is feasible.
>=20
>=C2=A0 =C2=A0 =C2=A0 f. When any middlebox exists on the path of a GRE-in-=
UDP tunnel,
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 it is RECOMMENDED to use the default mode, i.e=
. use UDP
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 checksum, to reduce the chance that the encaps=
ulated packets to
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 be dropped.
>=20
>=20
>=20
>=20
>=20
> Yong, Crabber, Xu, Herbert=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 [Page 14]
> -------------------------------------------------------------------------=
-------
>=20
> Internet-Draft=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 GRE-in-UDP Encapsulation=
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 March 2016
>=20
>=C2=A0 =C2=A0 =C2=A0 g. Any middlebox that allows the UDP zero-checksum mo=
de for IPv6
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 MUST comply with requirement 1 and 8-10 in Sec=
tion 5 of
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 [RFC6936].
>=20
>=C2=A0 =C2=A0 =C2=A0 h. Measures SHOULD be taken to prevent IPv6 traffic w=
ith zero UDP
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 checksums from "escaping" to the general Inter=
net; see Section
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 8 for examples of such measures.
>=20
>=C2=A0 =C2=A0 =C2=A0 i. IPv6 traffic with zero UDP checksums MUST be activ=
ely monitored
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 for errors by the network operator. For exampl=
e, the operator
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 may monitor Ethernet layer packet error rates.
>=20
>=C2=A0 =C2=A0 =C2=A0 j. If a packet with a non-zero checksum is received, =
the checksum
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 MUST be verified before accepting the packet. =
This is
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 regardless of whether the tunnel encapsulator =
and decapsulator
>=C2=A0 =C2=A0 =C2=A0 =C2=A0 have been configured with UDP zero-checksum mo=
de.
>=20
>=C2=A0 =C2=A0 The above requirements do not change either the requirements
>=C2=A0 =C2=A0 specified in [RFC2460] as modified by [RFC6935] or the requi=
rements
>=C2=A0 =C2=A0 specified in [RFC6936].
>=20
>=C2=A0 =C2=A0 The requirement to check the source IPv6 address in addition=
 to the
>=C2=A0 =C2=A0 destination IPv6 address, plus the strong recommendation aga=
inst
>=C2=A0 =C2=A0 reuse of source IPv6 addresses among GRE-in-UDP tunnels coll=
ectively
>=C2=A0 =C2=A0 provide some mitigation for the absence of UDP checksum cove=
rage of
>=C2=A0 =C2=A0 the IPv6 header. A traffic-managed controlled environment th=
at
>=C2=A0 =C2=A0 satisfies at least one of three conditions listed above in t=
his
>=C2=A0 =C2=A0 section provides additional assurance.
>=20
>=C2=A0 =C2=A0 A GRE-in-UDP tunnel is suitable for transmission over lower =
layers
>=C2=A0 =C2=A0 in the traffic-managed controlled environments that are allo=
wed by
>=C2=A0 =C2=A0 the exceptions stated above and the rate of corruption of th=
e inner
>=C2=A0 =C2=A0 IP packet on such networks is not expected to increase by co=
mparison
>=C2=A0 =C2=A0 to GRE traffic that is not encapsulated in UDP.=C2=A0 For th=
ese reasons,
>=C2=A0 =C2=A0 GRE-in-UDP does not provide an additional integrity check ex=
cept
>=C2=A0 =C2=A0 when GRE checksum is used when UDP zero-checksum mode is use=
d with
>=C2=A0 =C2=A0 IPv6, and this design is in accordance with requirements 2, =
3 and 5
>=C2=A0 =C2=A0 specified in Section 5 of [RFC6936].
>=20
>=C2=A0 =C2=A0 Generic Router Encapsulation (GRE) does not accumulate incor=
rect
>=C2=A0 =C2=A0 state as a consequence of GRE header corruption. A corrupt G=
RE
>=C2=A0 =C2=A0 packet may result in either packet discard or forwarding of =
the
>=C2=A0 =C2=A0 packet without accumulation of GRE state. Active monitoring =
of GRE-
>=C2=A0 =C2=A0 in-UDP traffic for errors is REQUIRED as occurrence of error=
s will
>=C2=A0 =C2=A0 result in some accumulation of error information outside the
>=C2=A0 =C2=A0 protocol for operational and management purposes. This desig=
n is in
>=C2=A0 =C2=A0 accordance with requirement 4 specified in Section 5 of [RFC=
6936].
>=20
>=20
>=20
> Yong, Crabber, Xu, Herbert=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 [Page 15]
> -------------------------------------------------------------------------=
-------
>=20
> Internet-Draft=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 GRE-in-UDP Encapsulation=
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 March 2016
>=20
>=C2=A0 =C2=A0 The remaining requirements specified in Section 5 of [RFC693=
6] are
>=C2=A0 =C2=A0 not applicable to GRE-in-UDP.=C2=A0 Requirements 6 and 7 do =
not apply
>=C2=A0 =C2=A0 because GRE does not include a control feedback mechanism.
>=C2=A0 =C2=A0 Requirements 8-10 are middlebox requirements that do not app=
ly to
>=C2=A0 =C2=A0 GRE-in-UDP tunnel endpoints (see Section 7.1 for further mid=
dlebox
>=C2=A0 =C2=A0 discussion).
>=20
>=C2=A0 =C2=A0 It is worth mentioning that the use of a zero UDP checksum s=
hould
>=C2=A0 =C2=A0 present the equivalent risk of undetected packet corruption =
when
>=C2=A0 =C2=A0 sending similar packet using GRE-in-IPv6 without UDP [RFC767=
6] and
>=C2=A0 =C2=A0 without GRE checksums.
>=20
>=C2=A0 =C2=A0 In summary, a TMCE GRE-in-UDP Tunnel is allowed to use UDP-z=
ero-
>=C2=A0 =C2=A0 checksum mode for IPv6 when the conditions and requirements =
stated
>=C2=A0 =C2=A0 above are met. Otherwise the UDP checksum need to be used fo=
r IPv6
>=C2=A0 =C2=A0 as specified in [RFC768] and [RFC2460]. Use of GRE checksum =
is
>=C2=A0 =C2=A0 RECOMMENED when the UDP checksum is not used.
> "
>=20
> In GUE, to support UDP-checksum-zero, it said
>=20
> " Therefore, when GUE is used over
>=C2=A0 =C2=A0 IPv6, either the UDP checksum must be enabled or the GUE hea=
der
>=C2=A0 =C2=A0 checksum must be used.=C2=A0 An encapsulator MAY set a zero =
UDP checksum
>=C2=A0 =C2=A0 for performance or implementation reasons, in which case the=
 GUE
>=C2=A0 =C2=A0 header checksum MUST be used or applicable requirements for =
using
>=C2=A0 =C2=A0 zero UDP checksums in [GREUDP] MUST be met. If the UDP check=
sum is
>=C2=A0 =C2=A0 enabled, then the GUE header checksum should not be used sin=
ce it is
>=C2=A0 =C2=A0 mostly redundant."
>=20
> It's easy for me to add the similar words to the IP-in-UDP draft like "th=
e
> applicable requirements for using zero UDP checksum in [GREUDP] MUST be
> met when zero UDP checksum is used by the tunnel ingress". However, the
> major goal for disabling the UDP checksum is to improve the performance.
> When GUE header checksum is used and/or the bunch of applicable
> requirements as described in GRE/UDP are verified, is the goal of improvi=
ng
> performance still achievable? If not, why not directly enable the UDP-che=
cksum
> instead?
>=20
> =C2=A0=C2=A0=C2=A0 - fragmentation support
>=20
> In GRE/UDP, it said
>=20
> " 4.1. MTU and Fragmentation
>=20
>=C2=A0 =C2=A0 Regarding packet fragmentation, an encapsulator/decapsulator=
 SHOULD
>=C2=A0 =C2=A0 be compliant with [RFC7588] and perform fragmentation before=
 the
>=C2=A0 =C2=A0 encapsulation. The size of fragments SHOULD be less or equal=
 to the
>=C2=A0 =C2=A0 PMTU associated with the path between the GRE ingress and th=
e GRE
>=C2=A0 =C2=A0 egress tunnel endpoints minus the GRE and UDP overhead ..."
>=20
> in GUE, it said
>=20
> " 4.9. MTU and fragmentation
>=20
>=C2=A0 =C2=A0 Standard conventions for handling of MTU (Maximum Transmissi=
on Unit)
>=C2=A0 =C2=A0 and fragmentation in conjunction with networking tunnels
>=C2=A0 =C2=A0 (encapsulation of layer 2 or layer 3 packets) should be foll=
owed.
>=C2=A0 =C2=A0 Details are described in MTU and Fragmentation Issues with I=
n-the-
>=C2=A0 =C2=A0 Network Tunneling [RFC4459]... "
>=20
> It seems that the only missing thing in the IP-in-UDP draft is to allow t=
he outer
> fragmentation. However, as it said in
> (https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02#page-13), " ..=
.IPsec
> performs only Outer Fragmentation; this distinguishes it from IP-in-IP, w=
hich
> performs only Inner Fragmentation. " Note that IP-in-IP is the dominant
> encapsulation choice within Softwires networks. In other words, performin=
g
> only inner fragmentation works very well in practice. Furthermore, the ou=
ter
> fragmentation issue (e.g., reassembly cost for the egress) would become e=
ven
> worse since the fragments of X-in-UDP packets are more likely to be forwa=
rded
> across different paths than those of X-in-IP and X-in-GRE packets. Hence,=
 I'm
> wondering whether it's worthwhile to support the outer fragmentation on U=
DP
> encapsulated packets which seems useless in practice.
>=20
> =C2=A0=C2=A0=C2=A0 - signalling support (e.g., to test whether a tunnel i=
s up or
> =C2=A0=C2=A0=C2=A0 to measure MTUs)
>=20
> I haven't found any description of this in both GRE/UDP and GUE. Did you?
>=20
> =C2=A0=C2=A0=C2=A0 - support for robust ID fields (related to fragmentati=
on,
> =C2=A0=C2=A0=C2=A0 e.g., to overcome the limits of IPv4 ID as per RFC 686=
4)
>=20
> I haven't found any description of this in both GRE/UDP and GUE. Did you?
>=20
> Xiaohu
>=20
> > It is not our obligation to find a way for your document to proceed -
> > that onus is on you.
> >
> > Joe

=20


------=_Part_2492240_1136524516.1465708308517
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<html xmlns=3D"http://www.w3.org/1999/xhtml" xmlns:v=3D"urn:schemas-microso=
ft-com:vml" xmlns:o=3D"urn:schemas-microsoft-com:office:office"><head><!--[=
if gte mso 9]><xml><o:OfficeDocumentSettings><o:AllowPNG/><o:PixelsPerInch>=
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<style type=3D"text/css" scoped> blockquote, div.yahoo_quoted { margin-left=
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                Fragmentation should be strongly discouraged.<div><br></div=
><div>if you've designed a tunnelling solution and you have fragmentation h=
appening as a matter of course, you've designed it wrong.&nbsp;<br><br>Lloy=
d Wood<div>lloyd.wood@yahoo.co.uk</div><br><p class=3D"yahoo-quoted-begin" =
style=3D"font-size: 15px; color: #715FFA; padding-top: 15px; margin-top: 0"=
>On Friday, May 27, 2016, 8:10 PM, Xuxiaohu &lt;xuxiaohu@huawei.com&gt; wro=
te:</p><blockquote class=3D"iosymail"><div id=3D"msgSandbox_AMhLyAoAAG2BnV0=
gc2BwILcOkyXZc_TEXT" class=3D"msgSandbox" style=3D"padding: 1.5em 0.5em 0.5=
em 1.2em; word-wrap: break-word;">&lt;Note that I have changed the subject =
of the email hence it has nothing to do with the WG adoption call now. It's=
 just a discussion on a particular issue which is related to those WGs whic=
h are working on UDP tunnels. The reason for containing the old email is to=
 use it as a background which may be useful for better understanding of thi=
s particular issue&gt;<br><br>The possible side-effect of performing fragme=
ntation on UDP encapsulated packets is to worsen the reassembly burden on t=
unnel egress since fragments of UDP encapsulated packets are more likely to=
 be forwarded across different paths towards the tunnel egress than those o=
f IP or GRE encapsulated packets.<br><br>It seems that most X-over-UDP prop=
osals choose to prohibit the tunnel ingress from performing fragmentation o=
n UDP encapsulated packets. See the following quoted text regarding fragmen=
tation from those X-over-UDP drafts:<br><br>LISP:<br><br>When an ITR receiv=
es a packet from a site-facing interface and adds H<br>&nbsp;  octets worth=
 of encapsulation to yield a packet size greater than L<br>&nbsp;  octets, =
it resolves the MTU issue by first splitting the original<br>&nbsp;  packet=
 into 2 equal-sized fragments.&nbsp; A LISP header is then prepended<br>&nb=
sp;  to each fragment.<br><br>VXLAN:<br><br>VTEPs MUST NOT fragment VXLAN p=
ackets.&nbsp; Intermediate routers may<br>&nbsp;  fragment encapsulated VXL=
AN packets due to the larger frame size.<br>&nbsp;  The destination VTEP MA=
Y silently discard such VXLAN fragments.<br><br>VXLAN-GPE:<br><br>VTEPs MUS=
T never fragment an encapsulated VXLAN GPE packet, and when<br>&nbsp;  the =
outer IP header is IPv4, VTEPs MUST set the DF bit in the outer<br>&nbsp;  =
IPv4 header.<br><br>GEVENE:<br><br>&nbsp;  To prevent fragmentation and max=
imize performance, the best practice<br>&nbsp;  when using Geneve is to ens=
ure that the MTU of the physical network<br>&nbsp;  is greater than or equa=
l to the MTU of the encapsulated network plus<br>&nbsp;  tunnel headers.<br=
><br>GUE:<br><br>&nbsp; &nbsp; If a packet is fragmented before encapsulati=
on in GUE, all the<br>&nbsp; &nbsp; related fragments must be encapsulated =
using the same source port<br>&nbsp; &nbsp; (inner flow identifier). An ope=
rator may set MTU to account for<br>&nbsp; &nbsp; encapsulation overhead an=
d reduce the likelihood of fragmentation.<br><br>GRE/UDP<br><br>Regarding p=
acket fragmentation, an encapsulator/decapsulator SHOULD<br>&nbsp;  be comp=
liant with [RFC7588] and perform fragmentation before the<br>&nbsp;  encaps=
ulation.<br><br>However, the above choice seems conflict with the requireme=
nts as described in <a href=3D"https://tools.ietf.org/html/draft-ietf-intar=
ea-tunnels-02" target=3D"_blank">https://tools.ietf.org/html/draft-ietf-int=
area-tunnels-02</a><br><br><br>I wonder whether the IETF should reach a con=
sensus on whether or not the fragmentation on UDP encapsulated packets shou=
ld be allowed.<br>&nbsp; <br>Best regards,<br>Xiaohu<br><br>&gt; -----Origi=
nal Message-----<br>&gt; From: Xuxiaohu<br>&gt; Sent: Thursday, May 26, 201=
6 4:35 PM<br>&gt; To: 'Joe Touch'; Fred Baker (fred); Wassim Haddad<br>&gt;=
 Cc: <a ymailto=3D"mailto:int-area@ietf.org" href=3D"javascript:return">int=
-area@ietf.org</a><br>&gt; Subject: RE: [Int-area] Call for adoption of dra=
ft-xu-intarea-ip-in-udp-03<br>&gt; <br>&gt; <br>&gt; <br>&gt; &gt; -----Ori=
ginal Message-----<br>&gt; &gt; From: Joe Touch [mailto:<a ymailto=3D"mailt=
o:touch@isi.edu" href=3D"javascript:return">touch@isi.edu</a>]<br>&gt; &gt;=
 Sent: Thursday, May 26, 2016 2:11 AM<br>&gt; &gt; To: Xuxiaohu; Fred Baker=
 (fred); Wassim Haddad<br>&gt; &gt; Cc: <a ymailto=3D"mailto:int-area@ietf.=
org" href=3D"javascript:return">int-area@ietf.org</a><br>&gt; &gt; Subject:=
 Re: [Int-area] Call for adoption of<br>&gt; &gt; draft-xu-intarea-ip-in-ud=
p-03<br>&gt; &gt;<br>&gt; &gt;<br>&gt; &gt;<br>&gt; &gt; On 5/24/2016 7:24 =
PM, Xuxiaohu wrote:<br>&gt; &gt; &gt; Hi Joe,<br>&gt; &gt; &gt;<br>&gt; &gt=
; &gt; I wonder whether you want to tell me the following truth by the<br>&=
gt; &gt; &gt; example that you gave: no matter whatever improvements we had=
 done<br>&gt; &gt; &gt; with this draft, those persons who dislike it by th=
e light of nature<br>&gt; &gt; &gt; would dislike it in the end.<br>&gt; &g=
t;<br>&gt; &gt; The only improvements that would make this doc useful would=
 be to add<br>&gt; &gt; capabilities already in GRE/UDP or GUE/UDP, which w=
e already have.<br>&gt; <br>&gt; Let's go over the four things you mentione=
d earlier in GRE/UDP and GUE/UDP:<br>&gt; <br>&gt; &nbsp;&nbsp;&nbsp; - str=
onger checksums<br>&gt; <br>&gt; In GRE/UDP, in order to use UDP-zero-check=
sum, it gave the following<br>&gt; restrictions:<br>&gt; " 6. UDP Checksum =
Handling<br>&gt; <br>&gt;&nbsp; &nbsp; 6.1. UDP Checksum with IPv4<br>&gt; =
<br>&gt;&nbsp; &nbsp; For UDP in IPv4, the UDP checksum MUST be processed a=
s specified in<br>&gt;&nbsp; &nbsp; [RFC768] and [RFC1122] for both transmi=
t and receive. The IPv4<br>&gt; <br>&gt; <br>&gt; <br>&gt; Yong, Crabber, X=
u, Herbert&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; [Page 12]<br>&=
gt; -----------------------------------------------------------------------=
---------<br>&gt; <br>&gt; Internet-Draft&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=
 GRE-in-UDP Encapsulation&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;  March 2=
016<br>&gt; <br>&gt;&nbsp; &nbsp; header includes a checksum which protects=
 against mis-delivery of<br>&gt;&nbsp; &nbsp; the packet due to corruption =
of IP addresses. The UDP checksum<br>&gt;&nbsp; &nbsp; potentially provides=
 protection against corruption of the UDP header,<br>&gt;&nbsp; &nbsp; GRE =
header, and GRE payload. Disabling the use of checksums is a<br>&gt;&nbsp; =
&nbsp; deployment consideration that should take into account the risk and<=
br>&gt;&nbsp; &nbsp; effects of packet corruption.<br>&gt; <br>&gt;&nbsp; &=
nbsp; When a decapsulator receives a packet, the UDP checksum field MUST<br=
>&gt;&nbsp; &nbsp; be processed. If the UDP checksum is non-zero, the decap=
sulator MUST<br>&gt;&nbsp; &nbsp; verify the checksum before accepting the =
packet. By default a<br>&gt;&nbsp; &nbsp; decapsulator SHOULD accept UDP pa=
ckets with a zero checksum. A node<br>&gt;&nbsp; &nbsp; MAY be configured t=
o disallow zero checksums per [RFC1122]; this may<br>&gt;&nbsp; &nbsp; be d=
one selectively, for instance disallowing zero checksums from<br>&gt;&nbsp;=
 &nbsp; certain hosts that are known to be sending over paths subject to<br=
>&gt;&nbsp; &nbsp; packet corruption. If verification of a non-zero checksu=
m fails, a<br>&gt;&nbsp; &nbsp; decapsulator lacks the capability to verify=
 a non-zero checksum, or<br>&gt;&nbsp; &nbsp; a packet with a zero-checksum=
 was received and the decapsulator is<br>&gt;&nbsp; &nbsp; configured to di=
sallow, the packet MUST be dropped and an event MAY<br>&gt;&nbsp; &nbsp; be=
 logged.<br>&gt; <br>&gt;&nbsp; &nbsp; 6.2. UDP Checksum with IPv6<br>&gt; =
<br>&gt;&nbsp; &nbsp; For UDP in IPv6, the UDP checksum MUST be processed a=
s specified in<br>&gt;&nbsp; &nbsp; [RFC768] and [RFC2460] for both transmi=
t and receive.<br>&gt; <br>&gt;&nbsp; &nbsp; When UDP is used over IPv6, th=
e UDP checksum is relied upon to<br>&gt;&nbsp; &nbsp; protect both the IPv6=
 and UDP headers from corruption. As such, A<br>&gt;&nbsp; &nbsp; default G=
RE-in-UDP Tunnel MUST perform UDP checksum; A TMCE GRE-in-<br>&gt;&nbsp; &n=
bsp; UDP Tunnel MAY be configured with the UDP zero-checksum mode if the<br=
>&gt;&nbsp; &nbsp; traffic-managed controlled environment or a set of close=
ly<br>&gt;&nbsp; &nbsp; cooperating traffic-managed controlled environments=
 (such as by<br>&gt;&nbsp; &nbsp; network operators who have agreed to work=
 together in order to<br>&gt;&nbsp; &nbsp; jointly provide specific service=
s) meet at least one of following<br>&gt;&nbsp; &nbsp; conditions:<br>&gt; =
<br>&gt;&nbsp; &nbsp; a. It is known (perhaps through knowledge of equipmen=
t types and<br>&gt;&nbsp; &nbsp; &nbsp;  lower layer checks) that packet co=
rruption is exceptionally<br>&gt;&nbsp; &nbsp; &nbsp;  unlikely and where t=
he operator is willing to take the risk of<br>&gt;&nbsp; &nbsp; &nbsp;  und=
etected packet corruption.<br>&gt; <br>&gt;&nbsp; &nbsp; b. It is judged th=
rough observational measurements (perhaps of<br>&gt;&nbsp; &nbsp; &nbsp;  h=
istoric or current traffic flows that use a non-zero checksum)<br>&gt;&nbsp=
; &nbsp; &nbsp;  that the level of packet corruption is tolerably low and w=
here<br>&gt;&nbsp; &nbsp; &nbsp;  the operator is willing to take the risk =
of undetected packet<br>&gt;&nbsp; &nbsp; &nbsp;  corruption.<br>&gt; <br>&=
gt; <br>&gt; <br>&gt; <br>&gt; <br>&gt; Yong, Crabber, Xu, Herbert&nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; [Page 13]<br>&gt; ---------------=
-----------------------------------------------------------------<br>&gt; <=
br>&gt; Internet-Draft&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; GRE-in-UDP Encapsu=
lation&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;  March 2016<br>&gt; <br>&gt=
;&nbsp; &nbsp; c. Carrying applications that are tolerant of mis-delivered =
or<br>&gt;&nbsp; &nbsp; &nbsp;  corrupted packets (perhaps through higher l=
ayer checksum,<br>&gt;&nbsp; &nbsp; &nbsp;  validation, and retransmission =
or transmission redundancy) where<br>&gt;&nbsp; &nbsp; &nbsp;  the operator=
 is willing to rely on the applications using the<br>&gt;&nbsp; &nbsp; &nbs=
p;  tunnel to survive any corrupt packets.<br>&gt; <br>&gt;&nbsp; &nbsp; Th=
e following requirements apply to a TMCE GRE-in-UDP tunnel that<br>&gt;&nbs=
p; &nbsp; use UDP zero-checksum mode:<br>&gt; <br>&gt;&nbsp; &nbsp; &nbsp; =
a. Use of the UDP checksum with IPv6 MUST be the default<br>&gt;&nbsp; &nbs=
p; &nbsp; &nbsp;  configuration of all GRE-in-UDP tunnels.<br>&gt; <br>&gt;=
&nbsp; &nbsp; &nbsp; b. The GRE-in-UDP tunnel implementation MUST comply wi=
th all<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  requirements specified in Sectio=
n 4 of [RFC6936] and with<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  requirement 1=
 specified in Section 5 of [RFC6936].<br>&gt; <br>&gt;&nbsp; &nbsp; &nbsp; =
c. The tunnel decapsulator SHOULD only allow the use of UDP zero-<br>&gt;&n=
bsp; &nbsp; &nbsp; &nbsp;  checksum mode for IPv6 on a single received UDP =
Destination<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  Port regardless of the enca=
psulator. The motivation for this<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  requi=
rement is possible corruption of the UDP Destination Port,<br>&gt;&nbsp; &n=
bsp; &nbsp; &nbsp;  which may cause packet delivery to the wrong UDP port. =
If that<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  other UDP port requires the UDP=
 checksum, the mis-delivered<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  packet wil=
l be discarded.<br>&gt; <br>&gt;&nbsp; &nbsp; &nbsp; d. It is RECOMMENDED t=
hat the UDP zero-checksum mode for IPv6 is<br>&gt;&nbsp; &nbsp; &nbsp; &nbs=
p;  only enabled for certain selected source addresses. The tunnel<br>&gt;&=
nbsp; &nbsp; &nbsp; &nbsp;  decapsulator MUST check that the source and des=
tination IPv6<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  addresses are valid for t=
he GRE-in-UDP tunnel on which the<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  packe=
t was received if that tunnel uses UDP zero-checksum mode<br>&gt;&nbsp; &nb=
sp; &nbsp; &nbsp;  and discard any packet for which this check fails.<br>&g=
t; <br>&gt;&nbsp; &nbsp; &nbsp; e. The tunnel encapsulator SHOULD use diffe=
rent IPv6 addresses for<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  each GRE-in-UDP=
 tunnel that uses UDP zero-checksum mode<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;=
  regardless of the decapsulator in order to strengthen the<br>&gt;&nbsp; &=
nbsp; &nbsp; &nbsp;  decapsulator's check of the IPv6 source address (i.e.,=
 the same<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  IPv6 source address SHOULD NO=
T be used with more than one IPv6<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  desti=
nation address, independent of whether that destination<br>&gt;&nbsp; &nbsp=
; &nbsp; &nbsp;  address is a unicast or multicast address). When this is n=
ot<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  possible, it is RECOMMENDED to use e=
ach source IPv6 address for<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  as few UDP =
zero-checksum mode GRE-in-UDP tunnels as is feasible.<br>&gt; <br>&gt;&nbsp=
; &nbsp; &nbsp; f. When any middlebox exists on the path of a GRE-in-UDP tu=
nnel,<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  it is RECOMMENDED to use the defa=
ult mode, i.e. use UDP<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  checksum, to red=
uce the chance that the encapsulated packets to<br>&gt;&nbsp; &nbsp; &nbsp;=
 &nbsp;  be dropped.<br>&gt; <br>&gt; <br>&gt; <br>&gt; <br>&gt; <br>&gt; Y=
ong, Crabber, Xu, Herbert&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=
 [Page 14]<br>&gt; --------------------------------------------------------=
------------------------<br>&gt; <br>&gt; Internet-Draft&nbsp; &nbsp; &nbsp=
; &nbsp; &nbsp; GRE-in-UDP Encapsulation&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;  March 2016<br>&gt; <br>&gt;&nbsp; &nbsp; &nbsp; g. Any middlebox th=
at allows the UDP zero-checksum mode for IPv6<br>&gt;&nbsp; &nbsp; &nbsp; &=
nbsp;  MUST comply with requirement 1 and 8-10 in Section 5 of<br>&gt;&nbsp=
; &nbsp; &nbsp; &nbsp;  [RFC6936].<br>&gt; <br>&gt;&nbsp; &nbsp; &nbsp; h. =
Measures SHOULD be taken to prevent IPv6 traffic with zero UDP<br>&gt;&nbsp=
; &nbsp; &nbsp; &nbsp;  checksums from "escaping" to the general Internet; =
see Section<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  8 for examples of such meas=
ures.<br>&gt; <br>&gt;&nbsp; &nbsp; &nbsp; i. IPv6 traffic with zero UDP ch=
ecksums MUST be actively monitored<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  for =
errors by the network operator. For example, the operator<br>&gt;&nbsp; &nb=
sp; &nbsp; &nbsp;  may monitor Ethernet layer packet error rates.<br>&gt; <=
br>&gt;&nbsp; &nbsp; &nbsp; j. If a packet with a non-zero checksum is rece=
ived, the checksum<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  MUST be verified bef=
ore accepting the packet. This is<br>&gt;&nbsp; &nbsp; &nbsp; &nbsp;  regar=
dless of whether the tunnel encapsulator and decapsulator<br>&gt;&nbsp; &nb=
sp; &nbsp; &nbsp;  have been configured with UDP zero-checksum mode.<br>&gt=
; <br>&gt;&nbsp; &nbsp; The above requirements do not change either the req=
uirements<br>&gt;&nbsp; &nbsp; specified in [RFC2460] as modified by [RFC69=
35] or the requirements<br>&gt;&nbsp; &nbsp; specified in [RFC6936].<br>&gt=
; <br>&gt;&nbsp; &nbsp; The requirement to check the source IPv6 address in=
 addition to the<br>&gt;&nbsp; &nbsp; destination IPv6 address, plus the st=
rong recommendation against<br>&gt;&nbsp; &nbsp; reuse of source IPv6 addre=
sses among GRE-in-UDP tunnels collectively<br>&gt;&nbsp; &nbsp; provide som=
e mitigation for the absence of UDP checksum coverage of<br>&gt;&nbsp; &nbs=
p; the IPv6 header. A traffic-managed controlled environment that<br>&gt;&n=
bsp; &nbsp; satisfies at least one of three conditions listed above in this=
<br>&gt;&nbsp; &nbsp; section provides additional assurance.<br>&gt; <br>&g=
t;&nbsp; &nbsp; A GRE-in-UDP tunnel is suitable for transmission over lower=
 layers<br>&gt;&nbsp; &nbsp; in the traffic-managed controlled environments=
 that are allowed by<br>&gt;&nbsp; &nbsp; the exceptions stated above and t=
he rate of corruption of the inner<br>&gt;&nbsp; &nbsp; IP packet on such n=
etworks is not expected to increase by comparison<br>&gt;&nbsp; &nbsp; to G=
RE traffic that is not encapsulated in UDP.&nbsp; For these reasons,<br>&gt=
;&nbsp; &nbsp; GRE-in-UDP does not provide an additional integrity check ex=
cept<br>&gt;&nbsp; &nbsp; when GRE checksum is used when UDP zero-checksum =
mode is used with<br>&gt;&nbsp; &nbsp; IPv6, and this design is in accordan=
ce with requirements 2, 3 and 5<br>&gt;&nbsp; &nbsp; specified in Section 5=
 of [RFC6936].<br>&gt; <br>&gt;&nbsp; &nbsp; Generic Router Encapsulation (=
GRE) does not accumulate incorrect<br>&gt;&nbsp; &nbsp; state as a conseque=
nce of GRE header corruption. A corrupt GRE<br>&gt;&nbsp; &nbsp; packet may=
 result in either packet discard or forwarding of the<br>&gt;&nbsp; &nbsp; =
packet without accumulation of GRE state. Active monitoring of GRE-<br>&gt;=
&nbsp; &nbsp; in-UDP traffic for errors is REQUIRED as occurrence of errors=
 will<br>&gt;&nbsp; &nbsp; result in some accumulation of error information=
 outside the<br>&gt;&nbsp; &nbsp; protocol for operational and management p=
urposes. This design is in<br>&gt;&nbsp; &nbsp; accordance with requirement=
 4 specified in Section 5 of [RFC6936].<br>&gt; <br>&gt; <br>&gt; <br>&gt; =
Yong, Crabber, Xu, Herbert&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
; [Page 15]<br>&gt; -------------------------------------------------------=
-------------------------<br>&gt; <br>&gt; Internet-Draft&nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; GRE-in-UDP Encapsulation&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp;  March 2016<br>&gt; <br>&gt;&nbsp; &nbsp; The remaining requirement=
s specified in Section 5 of [RFC6936] are<br>&gt;&nbsp; &nbsp; not applicab=
le to GRE-in-UDP.&nbsp; Requirements 6 and 7 do not apply<br>&gt;&nbsp; &nb=
sp; because GRE does not include a control feedback mechanism.<br>&gt;&nbsp=
; &nbsp; Requirements 8-10 are middlebox requirements that do not apply to<=
br>&gt;&nbsp; &nbsp; GRE-in-UDP tunnel endpoints (see Section 7.1 for furth=
er middlebox<br>&gt;&nbsp; &nbsp; discussion).<br>&gt; <br>&gt;&nbsp; &nbsp=
; It is worth mentioning that the use of a zero UDP checksum should<br>&gt;=
&nbsp; &nbsp; present the equivalent risk of undetected packet corruption w=
hen<br>&gt;&nbsp; &nbsp; sending similar packet using GRE-in-IPv6 without U=
DP [RFC7676] and<br>&gt;&nbsp; &nbsp; without GRE checksums.<br>&gt; <br>&g=
t;&nbsp; &nbsp; In summary, a TMCE GRE-in-UDP Tunnel is allowed to use UDP-=
zero-<br>&gt;&nbsp; &nbsp; checksum mode for IPv6 when the conditions and r=
equirements stated<br>&gt;&nbsp; &nbsp; above are met. Otherwise the UDP ch=
ecksum need to be used for IPv6<br>&gt;&nbsp; &nbsp; as specified in [RFC76=
8] and [RFC2460]. Use of GRE checksum is<br>&gt;&nbsp; &nbsp; RECOMMENED wh=
en the UDP checksum is not used.<br>&gt; "<br>&gt; <br>&gt; In GUE, to supp=
ort UDP-checksum-zero, it said<br>&gt; <br>&gt; " Therefore, when GUE is us=
ed over<br>&gt;&nbsp; &nbsp;  IPv6, either the UDP checksum must be enabled=
 or the GUE header<br>&gt;&nbsp; &nbsp;  checksum must be used.&nbsp; An en=
capsulator MAY set a zero UDP checksum<br>&gt;&nbsp; &nbsp;  for performanc=
e or implementation reasons, in which case the GUE<br>&gt;&nbsp; &nbsp;  he=
ader checksum MUST be used or applicable requirements for using<br>&gt;&nbs=
p; &nbsp;  zero UDP checksums in [GREUDP] MUST be met. If the UDP checksum =
is<br>&gt;&nbsp; &nbsp;  enabled, then the GUE header checksum should not b=
e used since it is<br>&gt;&nbsp; &nbsp;  mostly redundant."<br>&gt; <br>&gt=
; It's easy for me to add the similar words to the IP-in-UDP draft like "th=
e<br>&gt; applicable requirements for using zero UDP checksum in [GREUDP] M=
UST be<br>&gt; met when zero UDP checksum is used by the tunnel ingress". H=
owever, the<br>&gt; major goal for disabling the UDP checksum is to improve=
 the performance.<br>&gt; When GUE header checksum is used and/or the bunch=
 of applicable<br>&gt; requirements as described in GRE/UDP are verified, i=
s the goal of improving<br>&gt; performance still achievable? If not, why n=
ot directly enable the UDP-checksum<br>&gt; instead?<br>&gt; <br>&gt; &nbsp=
;&nbsp;&nbsp; - fragmentation support<br>&gt; <br>&gt; In GRE/UDP, it said<=
br>&gt; <br>&gt; " 4.1. MTU and Fragmentation<br>&gt; <br>&gt;&nbsp; &nbsp;=
 Regarding packet fragmentation, an encapsulator/decapsulator SHOULD<br>&gt=
;&nbsp; &nbsp; be compliant with [RFC7588] and perform fragmentation before=
 the<br>&gt;&nbsp; &nbsp; encapsulation. The size of fragments SHOULD be le=
ss or equal to the<br>&gt;&nbsp; &nbsp; PMTU associated with the path betwe=
en the GRE ingress and the GRE<br>&gt;&nbsp; &nbsp; egress tunnel endpoints=
 minus the GRE and UDP overhead ..."<br>&gt; <br>&gt; in GUE, it said<br>&g=
t; <br>&gt; " 4.9. MTU and fragmentation<br>&gt; <br>&gt;&nbsp; &nbsp;  Sta=
ndard conventions for handling of MTU (Maximum Transmission Unit)<br>&gt;&n=
bsp; &nbsp;  and fragmentation in conjunction with networking tunnels<br>&g=
t;&nbsp; &nbsp;  (encapsulation of layer 2 or layer 3 packets) should be fo=
llowed.<br>&gt;&nbsp; &nbsp;  Details are described in MTU and Fragmentatio=
n Issues with In-the-<br>&gt;&nbsp; &nbsp;  Network Tunneling [RFC4459]... =
"<br>&gt; <br>&gt; It seems that the only missing thing in the IP-in-UDP dr=
aft is to allow the outer<br>&gt; fragmentation. However, as it said in<br>=
&gt; (<a href=3D"https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02#=
page-13" target=3D"_blank">https://tools.ietf.org/html/draft-ietf-intarea-t=
unnels-02#page-13</a>), " ...IPsec<br>&gt; performs only Outer Fragmentatio=
n; this distinguishes it from IP-in-IP, which<br>&gt; performs only Inner F=
ragmentation. " Note that IP-in-IP is the dominant<br>&gt; encapsulation ch=
oice within Softwires networks. In other words, performing<br>&gt; only inn=
er fragmentation works very well in practice. Furthermore, the outer<br>&gt=
; fragmentation issue (e.g., reassembly cost for the egress) would become e=
ven<br>&gt; worse since the fragments of X-in-UDP packets are more likely t=
o be forwarded<br>&gt; across different paths than those of X-in-IP and X-i=
n-GRE packets. Hence, I'm<br>&gt; wondering whether it's worthwhile to supp=
ort the outer fragmentation on UDP<br>&gt; encapsulated packets which seems=
 useless in practice.<br>&gt; <br>&gt; &nbsp;&nbsp;&nbsp; - signalling supp=
ort (e.g., to test whether a tunnel is up or<br>&gt; &nbsp;&nbsp;&nbsp; to =
measure MTUs)<br>&gt; <br>&gt; I haven't found any description of this in b=
oth GRE/UDP and GUE. Did you?<br>&gt; <br>&gt; &nbsp;&nbsp;&nbsp; - support=
 for robust ID fields (related to fragmentation,<br>&gt; &nbsp;&nbsp;&nbsp;=
 e.g., to overcome the limits of IPv4 ID as per RFC 6864)<br>&gt; <br>&gt; =
I haven't found any description of this in both GRE/UDP and GUE. Did you?<b=
r>&gt; <br>&gt; Xiaohu<br>&gt; <br>&gt; &gt; It is not our obligation to fi=
nd a way for your document to proceed -<br>&gt; &gt; that onus is on you.<b=
r>&gt; &gt;<br>&gt; &gt; Joe<br></div><blockquote>
            </blockquote></blockquote></div>
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From: "Saumya Dikshit (sadikshi)" <sadikshi@cisco.com>
To: Lloyd Wood <lloyd.wood@yahoo.co.uk>, Softwires WG <softwires@ietf.org>, "int-area@ietf.org" <int-area@ietf.org>, "lisp@ietf.org" <lisp@ietf.org>, "tsvwg@ietf.org" <tsvwg@ietf.org>, Xuxiaohu <xuxiaohu@huawei.com>, "nvo3@ietf.org" <nvo3@ietf.org>
Thread-Topic: [nvo3] [tsvwg] Is it feasible to perform fragmentation on UDP encapsulated packets.
Thread-Index: AQHRxHNBx0FCT/ugokeOCF2M3CVw6A==
Date: Sun, 12 Jun 2016 06:25:47 +0000
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References: <E83B905A-FF6D-4996-B71A-7921DE4B133B@ericsson.com> <BFC09F5C-D6DF-4B6B-AA95-03919B9F09FB@cisco.com> <573E2A0E.1060609@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D54EB60@NKGEML515-MBX.china.huawei.com> <573F453C.5060908@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D554B73@NKGEML515-MBS.china.huawei.com> <5743303C.5040109@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D55514C@NKGEML515-MBS.china.huawei.com> <5743DD16.3050506@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D555482@NKGEML515-MBS.china.huawei.com> <57448C14.2060203@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5557DE@NKGEML515-MBS.china.huawei.com> <9c462520-eb8e-fcd0-0a08-228f80fbc779@isi.edu> <1FEE3F8F5CCDE64C9A8E8F4AD27C19EE0D5596E0@NKGEML515-MBS.china.huawei.com> <974977917.2492241.1465708308527.JavaMail.yahoo@mail.yahoo.com>
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Subject: Re: [tsvwg] [nvo3] Is it feasible to perform fragmentation on UDP encapsulated packets.
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In case of typical datacenter brownfield deployments,
the underlay and overlay may be completely disconnected operationally,
hence there can be cases where the mtu mismatch in underlay network may not
be notified to overlay endpoints and subsequently to end hosts.

From: Lloyd Wood <lloyd.wood@yahoo.co.uk<mailto:lloyd.wood@yahoo.co.uk>>
Date: Sunday, June 12, 2016 at 10:41 AM
To: Softwires WG <softwires@ietf.org<mailto:softwires@ietf.org>>, "int-area=
@ietf.org<mailto:int-area@ietf.org>" <int-area@ietf.org<mailto:int-area@iet=
f.org>>, "lisp@ietf.org<mailto:lisp@ietf.org>" <lisp@ietf.org<mailto:lisp@i=
etf.org>>, "tsvwg@ietf.org<mailto:tsvwg@ietf.org>" <tsvwg@ietf.org<mailto:t=
svwg@ietf.org>>, Xuxiaohu <xuxiaohu@huawei.com<mailto:xuxiaohu@huawei.com>>=
, "nvo3@ietf.org<mailto:nvo3@ietf.org>" <nvo3@ietf.org<mailto:nvo3@ietf.org=
>>
Subject: Re: [nvo3] [tsvwg] Is it feasible to perform fragmentation on UDP =
encapsulated packets.

Fragmentation should be strongly discouraged.

if you've designed a tunnelling solution and you have fragmentation happeni=
ng as a matter of course, you've designed it wrong.

Lloyd Wood
lloyd.wood@yahoo.co.uk<mailto:lloyd.wood@yahoo.co.uk>


On Friday, May 27, 2016, 8:10 PM, Xuxiaohu <xuxiaohu@huawei.com<mailto:xuxi=
aohu@huawei.com>> wrote:

<Note that I have changed the subject of the email hence it has nothing to =
do with the WG adoption call now. It's just a discussion on a particular is=
sue which is related to those WGs which are working on UDP tunnels. The rea=
son for containing the old email is to use it as a background which may be =
useful for better understanding of this particular issue>

The possible side-effect of performing fragmentation on UDP encapsulated pa=
ckets is to worsen the reassembly burden on tunnel egress since fragments o=
f UDP encapsulated packets are more likely to be forwarded across different=
 paths towards the tunnel egress than those of IP or GRE encapsulated packe=
ts.

It seems that most X-over-UDP proposals choose to prohibit the tunnel ingre=
ss from performing fragmentation on UDP encapsulated packets. See the follo=
wing quoted text regarding fragmentation from those X-over-UDP drafts:

LISP:

When an ITR receives a packet from a site-facing interface and adds H
  octets worth of encapsulation to yield a packet size greater than L
  octets, it resolves the MTU issue by first splitting the original
  packet into 2 equal-sized fragments.  A LISP header is then prepended
  to each fragment.

VXLAN:

VTEPs MUST NOT fragment VXLAN packets.  Intermediate routers may
  fragment encapsulated VXLAN packets due to the larger frame size.
  The destination VTEP MAY silently discard such VXLAN fragments.

VXLAN-GPE:

VTEPs MUST never fragment an encapsulated VXLAN GPE packet, and when
  the outer IP header is IPv4, VTEPs MUST set the DF bit in the outer
  IPv4 header.

GEVENE:

  To prevent fragmentation and maximize performance, the best practice
  when using Geneve is to ensure that the MTU of the physical network
  is greater than or equal to the MTU of the encapsulated network plus
  tunnel headers.

GUE:

    If a packet is fragmented before encapsulation in GUE, all the
    related fragments must be encapsulated using the same source port
    (inner flow identifier). An operator may set MTU to account for
    encapsulation overhead and reduce the likelihood of fragmentation.

GRE/UDP

Regarding packet fragmentation, an encapsulator/decapsulator SHOULD
  be compliant with [RFC7588] and perform fragmentation before the
  encapsulation.

However, the above choice seems conflict with the requirements as described=
 in https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02


I wonder whether the IETF should reach a consensus on whether or not the fr=
agmentation on UDP encapsulated packets should be allowed.

Best regards,
Xiaohu

> -----Original Message-----
> From: Xuxiaohu
> Sent: Thursday, May 26, 2016 4:35 PM
> To: 'Joe Touch'; Fred Baker (fred); Wassim Haddad
> Cc: int-area@ietf.org<javascript:return>
> Subject: RE: [Int-area] Call for adoption of draft-xu-intarea-ip-in-udp-0=
3
>
>
>
> > -----Original Message-----
> > From: Joe Touch [mailto:touch@isi.edu<javascript:return>]
> > Sent: Thursday, May 26, 2016 2:11 AM
> > To: Xuxiaohu; Fred Baker (fred); Wassim Haddad
> > Cc: int-area@ietf.org<javascript:return>
> > Subject: Re: [Int-area] Call for adoption of
> > draft-xu-intarea-ip-in-udp-03
> >
> >
> >
> > On 5/24/2016 7:24 PM, Xuxiaohu wrote:
> > > Hi Joe,
> > >
> > > I wonder whether you want to tell me the following truth by the
> > > example that you gave: no matter whatever improvements we had done
> > > with this draft, those persons who dislike it by the light of nature
> > > would dislike it in the end.
> >
> > The only improvements that would make this doc useful would be to add
> > capabilities already in GRE/UDP or GUE/UDP, which we already have.
>
> Let's go over the four things you mentioned earlier in GRE/UDP and GUE/UD=
P:
>
>     - stronger checksums
>
> In GRE/UDP, in order to use UDP-zero-checksum, it gave the following
> restrictions:
> " 6. UDP Checksum Handling
>
>    6.1. UDP Checksum with IPv4
>
>    For UDP in IPv4, the UDP checksum MUST be processed as specified in
>    [RFC768] and [RFC1122] for both transmit and receive. The IPv4
>
>
>
> Yong, Crabber, Xu, Herbert                                    [Page 12]
> -------------------------------------------------------------------------=
-------
>
> Internet-Draft          GRE-in-UDP Encapsulation            March 2016
>
>    header includes a checksum which protects against mis-delivery of
>    the packet due to corruption of IP addresses. The UDP checksum
>    potentially provides protection against corruption of the UDP header,
>    GRE header, and GRE payload. Disabling the use of checksums is a
>    deployment consideration that should take into account the risk and
>    effects of packet corruption.
>
>    When a decapsulator receives a packet, the UDP checksum field MUST
>    be processed. If the UDP checksum is non-zero, the decapsulator MUST
>    verify the checksum before accepting the packet. By default a
>    decapsulator SHOULD accept UDP packets with a zero checksum. A node
>    MAY be configured to disallow zero checksums per [RFC1122]; this may
>    be done selectively, for instance disallowing zero checksums from
>    certain hosts that are known to be sending over paths subject to
>    packet corruption. If verification of a non-zero checksum fails, a
>    decapsulator lacks the capability to verify a non-zero checksum, or
>    a packet with a zero-checksum was received and the decapsulator is
>    configured to disallow, the packet MUST be dropped and an event MAY
>    be logged.
>
>    6.2. UDP Checksum with IPv6
>
>    For UDP in IPv6, the UDP checksum MUST be processed as specified in
>    [RFC768] and [RFC2460] for both transmit and receive.
>
>    When UDP is used over IPv6, the UDP checksum is relied upon to
>    protect both the IPv6 and UDP headers from corruption. As such, A
>    default GRE-in-UDP Tunnel MUST perform UDP checksum; A TMCE GRE-in-
>    UDP Tunnel MAY be configured with the UDP zero-checksum mode if the
>    traffic-managed controlled environment or a set of closely
>    cooperating traffic-managed controlled environments (such as by
>    network operators who have agreed to work together in order to
>    jointly provide specific services) meet at least one of following
>    conditions:
>
>    a. It is known (perhaps through knowledge of equipment types and
>      lower layer checks) that packet corruption is exceptionally
>      unlikely and where the operator is willing to take the risk of
>      undetected packet corruption.
>
>    b. It is judged through observational measurements (perhaps of
>      historic or current traffic flows that use a non-zero checksum)
>      that the level of packet corruption is tolerably low and where
>      the operator is willing to take the risk of undetected packet
>      corruption.
>
>
>
>
>
> Yong, Crabber, Xu, Herbert                                    [Page 13]
> -------------------------------------------------------------------------=
-------
>
> Internet-Draft          GRE-in-UDP Encapsulation            March 2016
>
>    c. Carrying applications that are tolerant of mis-delivered or
>      corrupted packets (perhaps through higher layer checksum,
>      validation, and retransmission or transmission redundancy) where
>      the operator is willing to rely on the applications using the
>      tunnel to survive any corrupt packets.
>
>    The following requirements apply to a TMCE GRE-in-UDP tunnel that
>    use UDP zero-checksum mode:
>
>      a. Use of the UDP checksum with IPv6 MUST be the default
>        configuration of all GRE-in-UDP tunnels.
>
>      b. The GRE-in-UDP tunnel implementation MUST comply with all
>        requirements specified in Section 4 of [RFC6936] and with
>        requirement 1 specified in Section 5 of [RFC6936].
>
>      c. The tunnel decapsulator SHOULD only allow the use of UDP zero-
>        checksum mode for IPv6 on a single received UDP Destination
>        Port regardless of the encapsulator. The motivation for this
>        requirement is possible corruption of the UDP Destination Port,
>        which may cause packet delivery to the wrong UDP port. If that
>        other UDP port requires the UDP checksum, the mis-delivered
>        packet will be discarded.
>
>      d. It is RECOMMENDED that the UDP zero-checksum mode for IPv6 is
>        only enabled for certain selected source addresses. The tunnel
>        decapsulator MUST check that the source and destination IPv6
>        addresses are valid for the GRE-in-UDP tunnel on which the
>        packet was received if that tunnel uses UDP zero-checksum mode
>        and discard any packet for which this check fails.
>
>      e. The tunnel encapsulator SHOULD use different IPv6 addresses for
>        each GRE-in-UDP tunnel that uses UDP zero-checksum mode
>        regardless of the decapsulator in order to strengthen the
>        decapsulator's check of the IPv6 source address (i.e., the same
>        IPv6 source address SHOULD NOT be used with more than one IPv6
>        destination address, independent of whether that destination
>        address is a unicast or multicast address). When this is not
>        possible, it is RECOMMENDED to use each source IPv6 address for
>        as few UDP zero-checksum mode GRE-in-UDP tunnels as is feasible.
>
>      f. When any middlebox exists on the path of a GRE-in-UDP tunnel,
>        it is RECOMMENDED to use the default mode, i.e. use UDP
>        checksum, to reduce the chance that the encapsulated packets to
>        be dropped.
>
>
>
>
>
> Yong, Crabber, Xu, Herbert                                    [Page 14]
> -------------------------------------------------------------------------=
-------
>
> Internet-Draft          GRE-in-UDP Encapsulation            March 2016
>
>      g. Any middlebox that allows the UDP zero-checksum mode for IPv6
>        MUST comply with requirement 1 and 8-10 in Section 5 of
>        [RFC6936].
>
>      h. Measures SHOULD be taken to prevent IPv6 traffic with zero UDP
>        checksums from "escaping" to the general Internet; see Section
>        8 for examples of such measures.
>
>      i. IPv6 traffic with zero UDP checksums MUST be actively monitored
>        for errors by the network operator. For example, the operator
>        may monitor Ethernet layer packet error rates.
>
>      j. If a packet with a non-zero checksum is received, the checksum
>        MUST be verified before accepting the packet. This is
>        regardless of whether the tunnel encapsulator and decapsulator
>        have been configured with UDP zero-checksum mode.
>
>    The above requirements do not change either the requirements
>    specified in [RFC2460] as modified by [RFC6935] or the requirements
>    specified in [RFC6936].
>
>    The requirement to check the source IPv6 address in addition to the
>    destination IPv6 address, plus the strong recommendation against
>    reuse of source IPv6 addresses among GRE-in-UDP tunnels collectively
>    provide some mitigation for the absence of UDP checksum coverage of
>    the IPv6 header. A traffic-managed controlled environment that
>    satisfies at least one of three conditions listed above in this
>    section provides additional assurance.
>
>    A GRE-in-UDP tunnel is suitable for transmission over lower layers
>    in the traffic-managed controlled environments that are allowed by
>    the exceptions stated above and the rate of corruption of the inner
>    IP packet on such networks is not expected to increase by comparison
>    to GRE traffic that is not encapsulated in UDP.  For these reasons,
>    GRE-in-UDP does not provide an additional integrity check except
>    when GRE checksum is used when UDP zero-checksum mode is used with
>    IPv6, and this design is in accordance with requirements 2, 3 and 5
>    specified in Section 5 of [RFC6936].
>
>    Generic Router Encapsulation (GRE) does not accumulate incorrect
>    state as a consequence of GRE header corruption. A corrupt GRE
>    packet may result in either packet discard or forwarding of the
>    packet without accumulation of GRE state. Active monitoring of GRE-
>    in-UDP traffic for errors is REQUIRED as occurrence of errors will
>    result in some accumulation of error information outside the
>    protocol for operational and management purposes. This design is in
>    accordance with requirement 4 specified in Section 5 of [RFC6936].
>
>
>
> Yong, Crabber, Xu, Herbert                                    [Page 15]
> -------------------------------------------------------------------------=
-------
>
> Internet-Draft          GRE-in-UDP Encapsulation            March 2016
>
>    The remaining requirements specified in Section 5 of [RFC6936] are
>    not applicable to GRE-in-UDP.  Requirements 6 and 7 do not apply
>    because GRE does not include a control feedback mechanism.
>    Requirements 8-10 are middlebox requirements that do not apply to
>    GRE-in-UDP tunnel endpoints (see Section 7.1 for further middlebox
>    discussion).
>
>    It is worth mentioning that the use of a zero UDP checksum should
>    present the equivalent risk of undetected packet corruption when
>    sending similar packet using GRE-in-IPv6 without UDP [RFC7676] and
>    without GRE checksums.
>
>    In summary, a TMCE GRE-in-UDP Tunnel is allowed to use UDP-zero-
>    checksum mode for IPv6 when the conditions and requirements stated
>    above are met. Otherwise the UDP checksum need to be used for IPv6
>    as specified in [RFC768] and [RFC2460]. Use of GRE checksum is
>    RECOMMENED when the UDP checksum is not used.
> "
>
> In GUE, to support UDP-checksum-zero, it said
>
> " Therefore, when GUE is used over
>    IPv6, either the UDP checksum must be enabled or the GUE header
>    checksum must be used.  An encapsulator MAY set a zero UDP checksum
>    for performance or implementation reasons, in which case the GUE
>    header checksum MUST be used or applicable requirements for using
>    zero UDP checksums in [GREUDP] MUST be met. If the UDP checksum is
>    enabled, then the GUE header checksum should not be used since it is
>    mostly redundant."
>
> It's easy for me to add the similar words to the IP-in-UDP draft like "th=
e
> applicable requirements for using zero UDP checksum in [GREUDP] MUST be
> met when zero UDP checksum is used by the tunnel ingress". However, the
> major goal for disabling the UDP checksum is to improve the performance.
> When GUE header checksum is used and/or the bunch of applicable
> requirements as described in GRE/UDP are verified, is the goal of improvi=
ng
> performance still achievable? If not, why not directly enable the UDP-che=
cksum
> instead?
>
>     - fragmentation support
>
> In GRE/UDP, it said
>
> " 4.1. MTU and Fragmentation
>
>    Regarding packet fragmentation, an encapsulator/decapsulator SHOULD
>    be compliant with [RFC7588] and perform fragmentation before the
>    encapsulation. The size of fragments SHOULD be less or equal to the
>    PMTU associated with the path between the GRE ingress and the GRE
>    egress tunnel endpoints minus the GRE and UDP overhead ..."
>
> in GUE, it said
>
> " 4.9. MTU and fragmentation
>
>    Standard conventions for handling of MTU (Maximum Transmission Unit)
>    and fragmentation in conjunction with networking tunnels
>    (encapsulation of layer 2 or layer 3 packets) should be followed.
>    Details are described in MTU and Fragmentation Issues with In-the-
>    Network Tunneling [RFC4459]... "
>
> It seems that the only missing thing in the IP-in-UDP draft is to allow t=
he outer
> fragmentation. However, as it said in
> (https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02#page-13), " ..=
.IPsec
> performs only Outer Fragmentation; this distinguishes it from IP-in-IP, w=
hich
> performs only Inner Fragmentation. " Note that IP-in-IP is the dominant
> encapsulation choice within Softwires networks. In other words, performin=
g
> only inner fragmentation works very well in practice. Furthermore, the ou=
ter
> fragmentation issue (e.g., reassembly cost for the egress) would become e=
ven
> worse since the fragments of X-in-UDP packets are more likely to be forwa=
rded
> across different paths than those of X-in-IP and X-in-GRE packets. Hence,=
 I'm
> wondering whether it's worthwhile to support the outer fragmentation on U=
DP
> encapsulated packets which seems useless in practice.
>
>     - signalling support (e.g., to test whether a tunnel is up or
>     to measure MTUs)
>
> I haven't found any description of this in both GRE/UDP and GUE. Did you?
>
>     - support for robust ID fields (related to fragmentation,
>     e.g., to overcome the limits of IPv4 ID as per RFC 6864)
>
> I haven't found any description of this in both GRE/UDP and GUE. Did you?
>
> Xiaohu
>
> > It is not our obligation to find a way for your document to proceed -
> > that onus is on you.
> >
> > Joe

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<div>In case of typical datacenter brownfield deployments,&nbsp;</div>
<div>the underlay and overlay may be completely disconnected operationally,=
&nbsp;</div>
<div>hence there can be cases where the mtu mismatch in underlay network ma=
y not</div>
<div>be notified to overlay endpoints and subsequently to end hosts.</div>
<div><br>
</div>
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<span style=3D"font-weight:bold">From: </span>Lloyd Wood &lt;<a href=3D"mai=
lto:lloyd.wood@yahoo.co.uk">lloyd.wood@yahoo.co.uk</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Sunday, June 12, 2016 at 10:4=
1 AM<br>
<span style=3D"font-weight:bold">To: </span>Softwires WG &lt;<a href=3D"mai=
lto:softwires@ietf.org">softwires@ietf.org</a>&gt;, &quot;<a href=3D"mailto=
:int-area@ietf.org">int-area@ietf.org</a>&quot; &lt;<a href=3D"mailto:int-a=
rea@ietf.org">int-area@ietf.org</a>&gt;, &quot;<a href=3D"mailto:lisp@ietf.=
org">lisp@ietf.org</a>&quot;
 &lt;<a href=3D"mailto:lisp@ietf.org">lisp@ietf.org</a>&gt;, &quot;<a href=
=3D"mailto:tsvwg@ietf.org">tsvwg@ietf.org</a>&quot; &lt;<a href=3D"mailto:t=
svwg@ietf.org">tsvwg@ietf.org</a>&gt;, Xuxiaohu &lt;<a href=3D"mailto:xuxia=
ohu@huawei.com">xuxiaohu@huawei.com</a>&gt;, &quot;<a href=3D"mailto:nvo3@i=
etf.org">nvo3@ietf.org</a>&quot;
 &lt;<a href=3D"mailto:nvo3@ietf.org">nvo3@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [nvo3] [tsvwg] Is it f=
easible to perform fragmentation on UDP encapsulated packets.<br>
</div>
<div><br>
</div>
<div xmlns=3D"http://www.w3.org/1999/xhtml" xmlns:v=3D"urn:schemas-microsof=
t-com:vml" xmlns:o=3D"urn:schemas-microsoft-com:office:office">
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<div><style type=3D"text/css" scoped=3D""> blockquote, div.yahoo_quoted { m=
argin-left: 0 !important; border-left:1px #715FFA solid !important; padding=
-left:1ex !important; background-color:white !important; } </style>Fragment=
ation should be strongly discouraged.
<div><br>
</div>
<div>if you've designed a tunnelling solution and you have fragmentation ha=
ppening as a matter of course, you've designed it wrong.&nbsp;<br>
<br>
Lloyd Wood
<div><a href=3D"mailto:lloyd.wood@yahoo.co.uk">lloyd.wood@yahoo.co.uk</a></=
div>
<br>
<p class=3D"yahoo-quoted-begin" style=3D"font-size: 15px; color: #715FFA; p=
adding-top: 15px; margin-top: 0">
On Friday, May 27, 2016, 8:10 PM, Xuxiaohu &lt;<a href=3D"mailto:xuxiaohu@h=
uawei.com">xuxiaohu@huawei.com</a>&gt; wrote:</p>
<blockquote class=3D"iosymail">
<div id=3D"msgSandbox_AMhLyAoAAG2BnV0gc2BwILcOkyXZc_TEXT" class=3D"msgSandb=
ox" style=3D"padding: 1.5em 0.5em 0.5em 1.2em; word-wrap: break-word;">
&lt;Note that I have changed the subject of the email hence it has nothing =
to do with the WG adoption call now. It's just a discussion on a particular=
 issue which is related to those WGs which are working on UDP tunnels. The =
reason for containing the old email
 is to use it as a background which may be useful for better understanding =
of this particular issue&gt;<br>
<br>
The possible side-effect of performing fragmentation on UDP encapsulated pa=
ckets is to worsen the reassembly burden on tunnel egress since fragments o=
f UDP encapsulated packets are more likely to be forwarded across different=
 paths towards the tunnel egress
 than those of IP or GRE encapsulated packets.<br>
<br>
It seems that most X-over-UDP proposals choose to prohibit the tunnel ingre=
ss from performing fragmentation on UDP encapsulated packets. See the follo=
wing quoted text regarding fragmentation from those X-over-UDP drafts:<br>
<br>
LISP:<br>
<br>
When an ITR receives a packet from a site-facing interface and adds H<br>
&nbsp; octets worth of encapsulation to yield a packet size greater than L<=
br>
&nbsp; octets, it resolves the MTU issue by first splitting the original<br=
>
&nbsp; packet into 2 equal-sized fragments.&nbsp; A LISP header is then pre=
pended<br>
&nbsp; to each fragment.<br>
<br>
VXLAN:<br>
<br>
VTEPs MUST NOT fragment VXLAN packets.&nbsp; Intermediate routers may<br>
&nbsp; fragment encapsulated VXLAN packets due to the larger frame size.<br=
>
&nbsp; The destination VTEP MAY silently discard such VXLAN fragments.<br>
<br>
VXLAN-GPE:<br>
<br>
VTEPs MUST never fragment an encapsulated VXLAN GPE packet, and when<br>
&nbsp; the outer IP header is IPv4, VTEPs MUST set the DF bit in the outer<=
br>
&nbsp; IPv4 header.<br>
<br>
GEVENE:<br>
<br>
&nbsp; To prevent fragmentation and maximize performance, the best practice=
<br>
&nbsp; when using Geneve is to ensure that the MTU of the physical network<=
br>
&nbsp; is greater than or equal to the MTU of the encapsulated network plus=
<br>
&nbsp; tunnel headers.<br>
<br>
GUE:<br>
<br>
&nbsp; &nbsp; If a packet is fragmented before encapsulation in GUE, all th=
e<br>
&nbsp; &nbsp; related fragments must be encapsulated using the same source =
port<br>
&nbsp; &nbsp; (inner flow identifier). An operator may set MTU to account f=
or<br>
&nbsp; &nbsp; encapsulation overhead and reduce the likelihood of fragmenta=
tion.<br>
<br>
GRE/UDP<br>
<br>
Regarding packet fragmentation, an encapsulator/decapsulator SHOULD<br>
&nbsp; be compliant with [RFC7588] and perform fragmentation before the<br>
&nbsp; encapsulation.<br>
<br>
However, the above choice seems conflict with the requirements as described=
 in <a href=3D"https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02" t=
arget=3D"_blank">
https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02</a><br>
<br>
<br>
I wonder whether the IETF should reach a consensus on whether or not the fr=
agmentation on UDP encapsulated packets should be allowed.<br>
&nbsp; <br>
Best regards,<br>
Xiaohu<br>
<br>
&gt; -----Original Message-----<br>
&gt; From: Xuxiaohu<br>
&gt; Sent: Thursday, May 26, 2016 4:35 PM<br>
&gt; To: 'Joe Touch'; Fred Baker (fred); Wassim Haddad<br>
&gt; Cc: <a ymailto=3D"mailto:int-area@ietf.org" href=3D"javascript:return"=
>int-area@ietf.org</a><br>
&gt; Subject: RE: [Int-area] Call for adoption of draft-xu-intarea-ip-in-ud=
p-03<br>
&gt; <br>
&gt; <br>
&gt; <br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: Joe Touch [mailto:<a ymailto=3D"mailto:touch@isi.edu" href=
=3D"javascript:return">touch@isi.edu</a>]<br>
&gt; &gt; Sent: Thursday, May 26, 2016 2:11 AM<br>
&gt; &gt; To: Xuxiaohu; Fred Baker (fred); Wassim Haddad<br>
&gt; &gt; Cc: <a ymailto=3D"mailto:int-area@ietf.org" href=3D"javascript:re=
turn">int-area@ietf.org</a><br>
&gt; &gt; Subject: Re: [Int-area] Call for adoption of<br>
&gt; &gt; draft-xu-intarea-ip-in-udp-03<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; On 5/24/2016 7:24 PM, Xuxiaohu wrote:<br>
&gt; &gt; &gt; Hi Joe,<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; I wonder whether you want to tell me the following truth by =
the<br>
&gt; &gt; &gt; example that you gave: no matter whatever improvements we ha=
d done<br>
&gt; &gt; &gt; with this draft, those persons who dislike it by the light o=
f nature<br>
&gt; &gt; &gt; would dislike it in the end.<br>
&gt; &gt;<br>
&gt; &gt; The only improvements that would make this doc useful would be to=
 add<br>
&gt; &gt; capabilities already in GRE/UDP or GUE/UDP, which we already have=
.<br>
&gt; <br>
&gt; Let's go over the four things you mentioned earlier in GRE/UDP and GUE=
/UDP:<br>
&gt; <br>
&gt; &nbsp;&nbsp;&nbsp; - stronger checksums<br>
&gt; <br>
&gt; In GRE/UDP, in order to use UDP-zero-checksum, it gave the following<b=
r>
&gt; restrictions:<br>
&gt; &quot; 6. UDP Checksum Handling<br>
&gt; <br>
&gt;&nbsp; &nbsp; 6.1. UDP Checksum with IPv4<br>
&gt; <br>
&gt;&nbsp; &nbsp; For UDP in IPv4, the UDP checksum MUST be processed as sp=
ecified in<br>
&gt;&nbsp; &nbsp; [RFC768] and [RFC1122] for both transmit and receive. The=
 IPv4<br>
&gt; <br>
&gt; <br>
&gt; <br>
&gt; Yong, Crabber, Xu, Herbert&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; [Page 12]<br>
&gt; ----------------------------------------------------------------------=
----------<br>
&gt; <br>
&gt; Internet-Draft&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; GRE-in-UDP Encapsulat=
ion&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; March 2016<br>
&gt; <br>
&gt;&nbsp; &nbsp; header includes a checksum which protects against mis-del=
ivery of<br>
&gt;&nbsp; &nbsp; the packet due to corruption of IP addresses. The UDP che=
cksum<br>
&gt;&nbsp; &nbsp; potentially provides protection against corruption of the=
 UDP header,<br>
&gt;&nbsp; &nbsp; GRE header, and GRE payload. Disabling the use of checksu=
ms is a<br>
&gt;&nbsp; &nbsp; deployment consideration that should take into account th=
e risk and<br>
&gt;&nbsp; &nbsp; effects of packet corruption.<br>
&gt; <br>
&gt;&nbsp; &nbsp; When a decapsulator receives a packet, the UDP checksum f=
ield MUST<br>
&gt;&nbsp; &nbsp; be processed. If the UDP checksum is non-zero, the decaps=
ulator MUST<br>
&gt;&nbsp; &nbsp; verify the checksum before accepting the packet. By defau=
lt a<br>
&gt;&nbsp; &nbsp; decapsulator SHOULD accept UDP packets with a zero checks=
um. A node<br>
&gt;&nbsp; &nbsp; MAY be configured to disallow zero checksums per [RFC1122=
]; this may<br>
&gt;&nbsp; &nbsp; be done selectively, for instance disallowing zero checks=
ums from<br>
&gt;&nbsp; &nbsp; certain hosts that are known to be sending over paths sub=
ject to<br>
&gt;&nbsp; &nbsp; packet corruption. If verification of a non-zero checksum=
 fails, a<br>
&gt;&nbsp; &nbsp; decapsulator lacks the capability to verify a non-zero ch=
ecksum, or<br>
&gt;&nbsp; &nbsp; a packet with a zero-checksum was received and the decaps=
ulator is<br>
&gt;&nbsp; &nbsp; configured to disallow, the packet MUST be dropped and an=
 event MAY<br>
&gt;&nbsp; &nbsp; be logged.<br>
&gt; <br>
&gt;&nbsp; &nbsp; 6.2. UDP Checksum with IPv6<br>
&gt; <br>
&gt;&nbsp; &nbsp; For UDP in IPv6, the UDP checksum MUST be processed as sp=
ecified in<br>
&gt;&nbsp; &nbsp; [RFC768] and [RFC2460] for both transmit and receive.<br>
&gt; <br>
&gt;&nbsp; &nbsp; When UDP is used over IPv6, the UDP checksum is relied up=
on to<br>
&gt;&nbsp; &nbsp; protect both the IPv6 and UDP headers from corruption. As=
 such, A<br>
&gt;&nbsp; &nbsp; default GRE-in-UDP Tunnel MUST perform UDP checksum; A TM=
CE GRE-in-<br>
&gt;&nbsp; &nbsp; UDP Tunnel MAY be configured with the UDP zero-checksum m=
ode if the<br>
&gt;&nbsp; &nbsp; traffic-managed controlled environment or a set of closel=
y<br>
&gt;&nbsp; &nbsp; cooperating traffic-managed controlled environments (such=
 as by<br>
&gt;&nbsp; &nbsp; network operators who have agreed to work together in ord=
er to<br>
&gt;&nbsp; &nbsp; jointly provide specific services) meet at least one of f=
ollowing<br>
&gt;&nbsp; &nbsp; conditions:<br>
&gt; <br>
&gt;&nbsp; &nbsp; a. It is known (perhaps through knowledge of equipment ty=
pes and<br>
&gt;&nbsp; &nbsp; &nbsp; lower layer checks) that packet corruption is exce=
ptionally<br>
&gt;&nbsp; &nbsp; &nbsp; unlikely and where the operator is willing to take=
 the risk of<br>
&gt;&nbsp; &nbsp; &nbsp; undetected packet corruption.<br>
&gt; <br>
&gt;&nbsp; &nbsp; b. It is judged through observational measurements (perha=
ps of<br>
&gt;&nbsp; &nbsp; &nbsp; historic or current traffic flows that use a non-z=
ero checksum)<br>
&gt;&nbsp; &nbsp; &nbsp; that the level of packet corruption is tolerably l=
ow and where<br>
&gt;&nbsp; &nbsp; &nbsp; the operator is willing to take the risk of undete=
cted packet<br>
&gt;&nbsp; &nbsp; &nbsp; corruption.<br>
&gt; <br>
&gt; <br>
&gt; <br>
&gt; <br>
&gt; <br>
&gt; Yong, Crabber, Xu, Herbert&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; [Page 13]<br>
&gt; ----------------------------------------------------------------------=
----------<br>
&gt; <br>
&gt; Internet-Draft&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; GRE-in-UDP Encapsulat=
ion&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; March 2016<br>
&gt; <br>
&gt;&nbsp; &nbsp; c. Carrying applications that are tolerant of mis-deliver=
ed or<br>
&gt;&nbsp; &nbsp; &nbsp; corrupted packets (perhaps through higher layer ch=
ecksum,<br>
&gt;&nbsp; &nbsp; &nbsp; validation, and retransmission or transmission red=
undancy) where<br>
&gt;&nbsp; &nbsp; &nbsp; the operator is willing to rely on the application=
s using the<br>
&gt;&nbsp; &nbsp; &nbsp; tunnel to survive any corrupt packets.<br>
&gt; <br>
&gt;&nbsp; &nbsp; The following requirements apply to a TMCE GRE-in-UDP tun=
nel that<br>
&gt;&nbsp; &nbsp; use UDP zero-checksum mode:<br>
&gt; <br>
&gt;&nbsp; &nbsp; &nbsp; a. Use of the UDP checksum with IPv6 MUST be the d=
efault<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; configuration of all GRE-in-UDP tunnels.<br=
>
&gt; <br>
&gt;&nbsp; &nbsp; &nbsp; b. The GRE-in-UDP tunnel implementation MUST compl=
y with all<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; requirements specified in Section 4 of [RFC=
6936] and with<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; requirement 1 specified in Section 5 of [RF=
C6936].<br>
&gt; <br>
&gt;&nbsp; &nbsp; &nbsp; c. The tunnel decapsulator SHOULD only allow the u=
se of UDP zero-<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; checksum mode for IPv6 on a single received=
 UDP Destination<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; Port regardless of the encapsulator. The mo=
tivation for this<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; requirement is possible corruption of the U=
DP Destination Port,<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; which may cause packet delivery to the wron=
g UDP port. If that<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; other UDP port requires the UDP checksum, t=
he mis-delivered<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; packet will be discarded.<br>
&gt; <br>
&gt;&nbsp; &nbsp; &nbsp; d. It is RECOMMENDED that the UDP zero-checksum mo=
de for IPv6 is<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; only enabled for certain selected source ad=
dresses. The tunnel<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; decapsulator MUST check that the source and=
 destination IPv6<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; addresses are valid for the GRE-in-UDP tunn=
el on which the<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; packet was received if that tunnel uses UDP=
 zero-checksum mode<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; and discard any packet for which this check=
 fails.<br>
&gt; <br>
&gt;&nbsp; &nbsp; &nbsp; e. The tunnel encapsulator SHOULD use different IP=
v6 addresses for<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; each GRE-in-UDP tunnel that uses UDP zero-c=
hecksum mode<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; regardless of the decapsulator in order to =
strengthen the<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; decapsulator's check of the IPv6 source add=
ress (i.e., the same<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; IPv6 source address SHOULD NOT be used with=
 more than one IPv6<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; destination address, independent of whether=
 that destination<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; address is a unicast or multicast address).=
 When this is not<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; possible, it is RECOMMENDED to use each sou=
rce IPv6 address for<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; as few UDP zero-checksum mode GRE-in-UDP tu=
nnels as is feasible.<br>
&gt; <br>
&gt;&nbsp; &nbsp; &nbsp; f. When any middlebox exists on the path of a GRE-=
in-UDP tunnel,<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; it is RECOMMENDED to use the default mode, =
i.e. use UDP<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; checksum, to reduce the chance that the enc=
apsulated packets to<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; be dropped.<br>
&gt; <br>
&gt; <br>
&gt; <br>
&gt; <br>
&gt; <br>
&gt; Yong, Crabber, Xu, Herbert&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; [Page 14]<br>
&gt; ----------------------------------------------------------------------=
----------<br>
&gt; <br>
&gt; Internet-Draft&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; GRE-in-UDP Encapsulat=
ion&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; March 2016<br>
&gt; <br>
&gt;&nbsp; &nbsp; &nbsp; g. Any middlebox that allows the UDP zero-checksum=
 mode for IPv6<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; MUST comply with requirement 1 and 8-10 in =
Section 5 of<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; [RFC6936].<br>
&gt; <br>
&gt;&nbsp; &nbsp; &nbsp; h. Measures SHOULD be taken to prevent IPv6 traffi=
c with zero UDP<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; checksums from &quot;escaping&quot; to the =
general Internet; see Section<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; 8 for examples of such measures.<br>
&gt; <br>
&gt;&nbsp; &nbsp; &nbsp; i. IPv6 traffic with zero UDP checksums MUST be ac=
tively monitored<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; for errors by the network operator. For exa=
mple, the operator<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; may monitor Ethernet layer packet error rat=
es.<br>
&gt; <br>
&gt;&nbsp; &nbsp; &nbsp; j. If a packet with a non-zero checksum is receive=
d, the checksum<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; MUST be verified before accepting the packe=
t. This is<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; regardless of whether the tunnel encapsulat=
or and decapsulator<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp; have been configured with UDP zero-checksum=
 mode.<br>
&gt; <br>
&gt;&nbsp; &nbsp; The above requirements do not change either the requireme=
nts<br>
&gt;&nbsp; &nbsp; specified in [RFC2460] as modified by [RFC6935] or the re=
quirements<br>
&gt;&nbsp; &nbsp; specified in [RFC6936].<br>
&gt; <br>
&gt;&nbsp; &nbsp; The requirement to check the source IPv6 address in addit=
ion to the<br>
&gt;&nbsp; &nbsp; destination IPv6 address, plus the strong recommendation =
against<br>
&gt;&nbsp; &nbsp; reuse of source IPv6 addresses among GRE-in-UDP tunnels c=
ollectively<br>
&gt;&nbsp; &nbsp; provide some mitigation for the absence of UDP checksum c=
overage of<br>
&gt;&nbsp; &nbsp; the IPv6 header. A traffic-managed controlled environment=
 that<br>
&gt;&nbsp; &nbsp; satisfies at least one of three conditions listed above i=
n this<br>
&gt;&nbsp; &nbsp; section provides additional assurance.<br>
&gt; <br>
&gt;&nbsp; &nbsp; A GRE-in-UDP tunnel is suitable for transmission over low=
er layers<br>
&gt;&nbsp; &nbsp; in the traffic-managed controlled environments that are a=
llowed by<br>
&gt;&nbsp; &nbsp; the exceptions stated above and the rate of corruption of=
 the inner<br>
&gt;&nbsp; &nbsp; IP packet on such networks is not expected to increase by=
 comparison<br>
&gt;&nbsp; &nbsp; to GRE traffic that is not encapsulated in UDP.&nbsp; For=
 these reasons,<br>
&gt;&nbsp; &nbsp; GRE-in-UDP does not provide an additional integrity check=
 except<br>
&gt;&nbsp; &nbsp; when GRE checksum is used when UDP zero-checksum mode is =
used with<br>
&gt;&nbsp; &nbsp; IPv6, and this design is in accordance with requirements =
2, 3 and 5<br>
&gt;&nbsp; &nbsp; specified in Section 5 of [RFC6936].<br>
&gt; <br>
&gt;&nbsp; &nbsp; Generic Router Encapsulation (GRE) does not accumulate in=
correct<br>
&gt;&nbsp; &nbsp; state as a consequence of GRE header corruption. A corrup=
t GRE<br>
&gt;&nbsp; &nbsp; packet may result in either packet discard or forwarding =
of the<br>
&gt;&nbsp; &nbsp; packet without accumulation of GRE state. Active monitori=
ng of GRE-<br>
&gt;&nbsp; &nbsp; in-UDP traffic for errors is REQUIRED as occurrence of er=
rors will<br>
&gt;&nbsp; &nbsp; result in some accumulation of error information outside =
the<br>
&gt;&nbsp; &nbsp; protocol for operational and management purposes. This de=
sign is in<br>
&gt;&nbsp; &nbsp; accordance with requirement 4 specified in Section 5 of [=
RFC6936].<br>
&gt; <br>
&gt; <br>
&gt; <br>
&gt; Yong, Crabber, Xu, Herbert&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; [Page 15]<br>
&gt; ----------------------------------------------------------------------=
----------<br>
&gt; <br>
&gt; Internet-Draft&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; GRE-in-UDP Encapsulat=
ion&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; March 2016<br>
&gt; <br>
&gt;&nbsp; &nbsp; The remaining requirements specified in Section 5 of [RFC=
6936] are<br>
&gt;&nbsp; &nbsp; not applicable to GRE-in-UDP.&nbsp; Requirements 6 and 7 =
do not apply<br>
&gt;&nbsp; &nbsp; because GRE does not include a control feedback mechanism=
.<br>
&gt;&nbsp; &nbsp; Requirements 8-10 are middlebox requirements that do not =
apply to<br>
&gt;&nbsp; &nbsp; GRE-in-UDP tunnel endpoints (see Section 7.1 for further =
middlebox<br>
&gt;&nbsp; &nbsp; discussion).<br>
&gt; <br>
&gt;&nbsp; &nbsp; It is worth mentioning that the use of a zero UDP checksu=
m should<br>
&gt;&nbsp; &nbsp; present the equivalent risk of undetected packet corrupti=
on when<br>
&gt;&nbsp; &nbsp; sending similar packet using GRE-in-IPv6 without UDP [RFC=
7676] and<br>
&gt;&nbsp; &nbsp; without GRE checksums.<br>
&gt; <br>
&gt;&nbsp; &nbsp; In summary, a TMCE GRE-in-UDP Tunnel is allowed to use UD=
P-zero-<br>
&gt;&nbsp; &nbsp; checksum mode for IPv6 when the conditions and requiremen=
ts stated<br>
&gt;&nbsp; &nbsp; above are met. Otherwise the UDP checksum need to be used=
 for IPv6<br>
&gt;&nbsp; &nbsp; as specified in [RFC768] and [RFC2460]. Use of GRE checks=
um is<br>
&gt;&nbsp; &nbsp; RECOMMENED when the UDP checksum is not used.<br>
&gt; &quot;<br>
&gt; <br>
&gt; In GUE, to support UDP-checksum-zero, it said<br>
&gt; <br>
&gt; &quot; Therefore, when GUE is used over<br>
&gt;&nbsp; &nbsp; IPv6, either the UDP checksum must be enabled or the GUE =
header<br>
&gt;&nbsp; &nbsp; checksum must be used.&nbsp; An encapsulator MAY set a ze=
ro UDP checksum<br>
&gt;&nbsp; &nbsp; for performance or implementation reasons, in which case =
the GUE<br>
&gt;&nbsp; &nbsp; header checksum MUST be used or applicable requirements f=
or using<br>
&gt;&nbsp; &nbsp; zero UDP checksums in [GREUDP] MUST be met. If the UDP ch=
ecksum is<br>
&gt;&nbsp; &nbsp; enabled, then the GUE header checksum should not be used =
since it is<br>
&gt;&nbsp; &nbsp; mostly redundant.&quot;<br>
&gt; <br>
&gt; It's easy for me to add the similar words to the IP-in-UDP draft like =
&quot;the<br>
&gt; applicable requirements for using zero UDP checksum in [GREUDP] MUST b=
e<br>
&gt; met when zero UDP checksum is used by the tunnel ingress&quot;. Howeve=
r, the<br>
&gt; major goal for disabling the UDP checksum is to improve the performanc=
e.<br>
&gt; When GUE header checksum is used and/or the bunch of applicable<br>
&gt; requirements as described in GRE/UDP are verified, is the goal of impr=
oving<br>
&gt; performance still achievable? If not, why not directly enable the UDP-=
checksum<br>
&gt; instead?<br>
&gt; <br>
&gt; &nbsp;&nbsp;&nbsp; - fragmentation support<br>
&gt; <br>
&gt; In GRE/UDP, it said<br>
&gt; <br>
&gt; &quot; 4.1. MTU and Fragmentation<br>
&gt; <br>
&gt;&nbsp; &nbsp; Regarding packet fragmentation, an encapsulator/decapsula=
tor SHOULD<br>
&gt;&nbsp; &nbsp; be compliant with [RFC7588] and perform fragmentation bef=
ore the<br>
&gt;&nbsp; &nbsp; encapsulation. The size of fragments SHOULD be less or eq=
ual to the<br>
&gt;&nbsp; &nbsp; PMTU associated with the path between the GRE ingress and=
 the GRE<br>
&gt;&nbsp; &nbsp; egress tunnel endpoints minus the GRE and UDP overhead ..=
.&quot;<br>
&gt; <br>
&gt; in GUE, it said<br>
&gt; <br>
&gt; &quot; 4.9. MTU and fragmentation<br>
&gt; <br>
&gt;&nbsp; &nbsp; Standard conventions for handling of MTU (Maximum Transmi=
ssion Unit)<br>
&gt;&nbsp; &nbsp; and fragmentation in conjunction with networking tunnels<=
br>
&gt;&nbsp; &nbsp; (encapsulation of layer 2 or layer 3 packets) should be f=
ollowed.<br>
&gt;&nbsp; &nbsp; Details are described in MTU and Fragmentation Issues wit=
h In-the-<br>
&gt;&nbsp; &nbsp; Network Tunneling [RFC4459]... &quot;<br>
&gt; <br>
&gt; It seems that the only missing thing in the IP-in-UDP draft is to allo=
w the outer<br>
&gt; fragmentation. However, as it said in<br>
&gt; (<a href=3D"https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02#=
page-13" target=3D"_blank">https://tools.ietf.org/html/draft-ietf-intarea-t=
unnels-02#page-13</a>), &quot; ...IPsec<br>
&gt; performs only Outer Fragmentation; this distinguishes it from IP-in-IP=
, which<br>
&gt; performs only Inner Fragmentation. &quot; Note that IP-in-IP is the do=
minant<br>
&gt; encapsulation choice within Softwires networks. In other words, perfor=
ming<br>
&gt; only inner fragmentation works very well in practice. Furthermore, the=
 outer<br>
&gt; fragmentation issue (e.g., reassembly cost for the egress) would becom=
e even<br>
&gt; worse since the fragments of X-in-UDP packets are more likely to be fo=
rwarded<br>
&gt; across different paths than those of X-in-IP and X-in-GRE packets. Hen=
ce, I'm<br>
&gt; wondering whether it's worthwhile to support the outer fragmentation o=
n UDP<br>
&gt; encapsulated packets which seems useless in practice.<br>
&gt; <br>
&gt; &nbsp;&nbsp;&nbsp; - signalling support (e.g., to test whether a tunne=
l is up or<br>
&gt; &nbsp;&nbsp;&nbsp; to measure MTUs)<br>
&gt; <br>
&gt; I haven't found any description of this in both GRE/UDP and GUE. Did y=
ou?<br>
&gt; <br>
&gt; &nbsp;&nbsp;&nbsp; - support for robust ID fields (related to fragment=
ation,<br>
&gt; &nbsp;&nbsp;&nbsp; e.g., to overcome the limits of IPv4 ID as per RFC =
6864)<br>
&gt; <br>
&gt; I haven't found any description of this in both GRE/UDP and GUE. Did y=
ou?<br>
&gt; <br>
&gt; Xiaohu<br>
&gt; <br>
&gt; &gt; It is not our obligation to find a way for your document to proce=
ed -<br>
&gt; &gt; that onus is on you.<br>
&gt; &gt;<br>
&gt; &gt; Joe<br>
</div>
<blockquote></blockquote>
</blockquote>
</div>
</div>
</div>
</span>
</body>
</html>

--_000_D382FCD56783Csadikshiciscocom_--


From nobody Mon Jun 13 03:43:25 2016
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Subject: [tsvwg] RV: New Version Notification for draft-suznjevic-tsvwg-delay-limits-00.txt
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Hi all,

We have just submitted this draft about Delay Limits for Real-Time =
Services. See =
https://datatracker.ietf.org/doc/draft-suznjevic-tsvwg-delay-limits/=20

It surveys a set of recommendations about the maximum latency tolerated =
by the users of delay-constrained services.  Some recommendations =
already exist for VoIP, but emerging services as e.g. online gaming, =
have different requirements.  Different papers in the literature =
reporting these constraints are surveyed, and a summary of the latency =
limits for each service is finally provided.

It replaces draft-suznjevic-dispatch-delay-limits and =
draft-suznjevic-tsvwg-mtd-tcmtf

Best regards,

Jose and Mirko

> -----Mensaje original-----
> De: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
> Enviado el: lunes, 13 de junio de 2016 9:22
> Para: Mirko Suznjevic <mirko.suznjevic@fer.hr>; Jose Saldana
> <jsaldana@unizar.es>
> Asunto: New Version Notification for =
draft-suznjevic-tsvwg-delay-limits-00.txt
>=20
>=20
> A new version of I-D, draft-suznjevic-tsvwg-delay-limits-00.txt
> has been successfully submitted by Jose Saldana and posted to the IETF
> repository.
>=20
> Name:		draft-suznjevic-tsvwg-delay-limits
> Revision:	00
> Title:		Delay Limits for Real-Time Services
> Document date:	2016-06-13
> Group:		Individual Submission
> Pages:		12
> URL:            =
https://www.ietf.org/internet-drafts/draft-suznjevic-tsvwg-delay-limits-
> 00.txt
> Status:         =
https://datatracker.ietf.org/doc/draft-suznjevic-tsvwg-delay-limits/
> Htmlized:       =
https://tools.ietf.org/html/draft-suznjevic-tsvwg-delay-limits-00
>=20
>=20
> Abstract:
>    Network delay is one of the main factors which can degrade the
>    Quality of Experience (QoE) of the users of network services.  This
>    document surveys a set of recommendations about the maximum latency
>    tolerated by the users of delay-constrained services.  Some
>    recommendations already exist for VoIP, but emerging services as =
e.g.
>    online gaming, have different requirements.  Different papers in =
the
>    literature reporting these constraints are surveyed, and a summary =
of
>    the latency limits for each service is finally provided.
>=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



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Subject: Re: [tsvwg] Is it feasible to perform fragmentation on UDP encapsulated packets.
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On Sat, Jun 11, 2016 at 10:11 PM, Lloyd Wood <lloyd.wood@yahoo.co.uk> wrote:
> Fragmentation should be strongly discouraged.
>
> if you've designed a tunnelling solution and you have fragmentation
> happening as a matter of course, you've designed it wrong.
>

As mentioned before we do not always have control of the underlay MTUs
to be able to set Jumbo Frames to make MTU larger than 1500 to account
for tunnel fragmentation. At Facebook for instance, we run IP tunnels
between L4 load balancers and backend servers. In certain POPs we
cannot get more than 1500 MTU in the path. The ingress MTU into the
load balancer is 1500 so it is possible to receive packets from the
Internet that are greater than the tunnel MTU. In this case we will
fragment. Since the tunnel is behind a firewall and our devices and
the path from load balancers to backends is low loss there is little
concern of being DOSed.

Tom

> Lloyd Wood
> lloyd.wood@yahoo.co.uk
>
> On Friday, May 27, 2016, 8:10 PM, Xuxiaohu <xuxiaohu@huawei.com> wrote:
>
> <Note that I have changed the subject of the email hence it has nothing to
> do with the WG adoption call now. It's just a discussion on a particular
> issue which is related to those WGs which are working on UDP tunnels. The
> reason for containing the old email is to use it as a background which may
> be useful for better understanding of this particular issue>
>
> The possible side-effect of performing fragmentation on UDP encapsulated
> packets is to worsen the reassembly burden on tunnel egress since fragments
> of UDP encapsulated packets are more likely to be forwarded across different
> paths towards the tunnel egress than those of IP or GRE encapsulated
> packets.
>
> It seems that most X-over-UDP proposals choose to prohibit the tunnel
> ingress from performing fragmentation on UDP encapsulated packets. See the
> following quoted text regarding fragmentation from those X-over-UDP drafts:
>
> LISP:
>
> When an ITR receives a packet from a site-facing interface and adds H
>   octets worth of encapsulation to yield a packet size greater than L
>   octets, it resolves the MTU issue by first splitting the original
>   packet into 2 equal-sized fragments.  A LISP header is then prepended
>   to each fragment.
>
> VXLAN:
>
> VTEPs MUST NOT fragment VXLAN packets.  Intermediate routers may
>   fragment encapsulated VXLAN packets due to the larger frame size.
>   The destination VTEP MAY silently discard such VXLAN fragments.
>
> VXLAN-GPE:
>
> VTEPs MUST never fragment an encapsulated VXLAN GPE packet, and when
>   the outer IP header is IPv4, VTEPs MUST set the DF bit in the outer
>   IPv4 header.
>
> GEVENE:
>
>   To prevent fragmentation and maximize performance, the best practice
>   when using Geneve is to ensure that the MTU of the physical network
>   is greater than or equal to the MTU of the encapsulated network plus
>   tunnel headers.
>
> GUE:
>
>     If a packet is fragmented before encapsulation in GUE, all the
>     related fragments must be encapsulated using the same source port
>     (inner flow identifier). An operator may set MTU to account for
>     encapsulation overhead and reduce the likelihood of fragmentation.
>
> GRE/UDP
>
> Regarding packet fragmentation, an encapsulator/decapsulator SHOULD
>   be compliant with [RFC7588] and perform fragmentation before the
>   encapsulation.
>
> However, the above choice seems conflict with the requirements as described
> in https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02
>
>
> I wonder whether the IETF should reach a consensus on whether or not the
> fragmentation on UDP encapsulated packets should be allowed.
>
> Best regards,
> Xiaohu
>
>> -----Original Message-----
>> From: Xuxiaohu
>> Sent: Thursday, May 26, 2016 4:35 PM
>> To: 'Joe Touch'; Fred Baker (fred); Wassim Haddad
>> Cc: int-area@ietf.org
>> Subject: RE: [Int-area] Call for adoption of draft-xu-intarea-ip-in-udp-03
>>
>>
>>
>> > -----Original Message-----
>> > From: Joe Touch [mailto:touch@isi.edu]
>> > Sent: Thursday, May 26, 2016 2:11 AM
>> > To: Xuxiaohu; Fred Baker (fred); Wassim Haddad
>> > Cc: int-area@ietf.org
>> > Subject: Re: [Int-area] Call for adoption of
>> > draft-xu-intarea-ip-in-udp-03
>> >
>> >
>> >
>> > On 5/24/2016 7:24 PM, Xuxiaohu wrote:
>> > > Hi Joe,
>> > >
>> > > I wonder whether you want to tell me the following truth by the
>> > > example that you gave: no matter whatever improvements we had done
>> > > with this draft, those persons who dislike it by the light of nature
>> > > would dislike it in the end.
>> >
>> > The only improvements that would make this doc useful would be to add
>> > capabilities already in GRE/UDP or GUE/UDP, which we already have.
>>
>> Let's go over the four things you mentioned earlier in GRE/UDP and
>> GUE/UDP:
>>
>>     - stronger checksums
>>
>> In GRE/UDP, in order to use UDP-zero-checksum, it gave the following
>> restrictions:
>> " 6. UDP Checksum Handling
>>
>>    6.1. UDP Checksum with IPv4
>>
>>    For UDP in IPv4, the UDP checksum MUST be processed as specified in
>>    [RFC768] and [RFC1122] for both transmit and receive. The IPv4
>>
>>
>>
>> Yong, Crabber, Xu, Herbert                                    [Page 12]
>>
>> --------------------------------------------------------------------------------
>>
>> Internet-Draft          GRE-in-UDP Encapsulation            March 2016
>>
>>    header includes a checksum which protects against mis-delivery of
>>    the packet due to corruption of IP addresses. The UDP checksum
>>    potentially provides protection against corruption of the UDP header,
>>    GRE header, and GRE payload. Disabling the use of checksums is a
>>    deployment consideration that should take into account the risk and
>>    effects of packet corruption.
>>
>>    When a decapsulator receives a packet, the UDP checksum field MUST
>>    be processed. If the UDP checksum is non-zero, the decapsulator MUST
>>    verify the checksum before accepting the packet. By default a
>>    decapsulator SHOULD accept UDP packets with a zero checksum. A node
>>    MAY be configured to disallow zero checksums per [RFC1122]; this may
>>    be done selectively, for instance disallowing zero checksums from
>>    certain hosts that are known to be sending over paths subject to
>>    packet corruption. If verification of a non-zero checksum fails, a
>>    decapsulator lacks the capability to verify a non-zero checksum, or
>>    a packet with a zero-checksum was received and the decapsulator is
>>    configured to disallow, the packet MUST be dropped and an event MAY
>>    be logged.
>>
>>    6.2. UDP Checksum with IPv6
>>
>>    For UDP in IPv6, the UDP checksum MUST be processed as specified in
>>    [RFC768] and [RFC2460] for both transmit and receive.
>>
>>    When UDP is used over IPv6, the UDP checksum is relied upon to
>>    protect both the IPv6 and UDP headers from corruption. As such, A
>>    default GRE-in-UDP Tunnel MUST perform UDP checksum; A TMCE GRE-in-
>>    UDP Tunnel MAY be configured with the UDP zero-checksum mode if the
>>    traffic-managed controlled environment or a set of closely
>>    cooperating traffic-managed controlled environments (such as by
>>    network operators who have agreed to work together in order to
>>    jointly provide specific services) meet at least one of following
>>    conditions:
>>
>>    a. It is known (perhaps through knowledge of equipment types and
>>      lower layer checks) that packet corruption is exceptionally
>>      unlikely and where the operator is willing to take the risk of
>>      undetected packet corruption.
>>
>>    b. It is judged through observational measurements (perhaps of
>>      historic or current traffic flows that use a non-zero checksum)
>>      that the level of packet corruption is tolerably low and where
>>      the operator is willing to take the risk of undetected packet
>>      corruption.
>>
>>
>>
>>
>>
>> Yong, Crabber, Xu, Herbert                                    [Page 13]
>>
>> --------------------------------------------------------------------------------
>>
>> Internet-Draft          GRE-in-UDP Encapsulation            March 2016
>>
>>    c. Carrying applications that are tolerant of mis-delivered or
>>      corrupted packets (perhaps through higher layer checksum,
>>      validation, and retransmission or transmission redundancy) where
>>      the operator is willing to rely on the applications using the
>>      tunnel to survive any corrupt packets.
>>
>>    The following requirements apply to a TMCE GRE-in-UDP tunnel that
>>    use UDP zero-checksum mode:
>>
>>      a. Use of the UDP checksum with IPv6 MUST be the default
>>        configuration of all GRE-in-UDP tunnels.
>>
>>      b. The GRE-in-UDP tunnel implementation MUST comply with all
>>        requirements specified in Section 4 of [RFC6936] and with
>>        requirement 1 specified in Section 5 of [RFC6936].
>>
>>      c. The tunnel decapsulator SHOULD only allow the use of UDP zero-
>>        checksum mode for IPv6 on a single received UDP Destination
>>        Port regardless of the encapsulator. The motivation for this
>>        requirement is possible corruption of the UDP Destination Port,
>>        which may cause packet delivery to the wrong UDP port. If that
>>        other UDP port requires the UDP checksum, the mis-delivered
>>        packet will be discarded.
>>
>>      d. It is RECOMMENDED that the UDP zero-checksum mode for IPv6 is
>>        only enabled for certain selected source addresses. The tunnel
>>        decapsulator MUST check that the source and destination IPv6
>>        addresses are valid for the GRE-in-UDP tunnel on which the
>>        packet was received if that tunnel uses UDP zero-checksum mode
>>        and discard any packet for which this check fails.
>>
>>      e. The tunnel encapsulator SHOULD use different IPv6 addresses for
>>        each GRE-in-UDP tunnel that uses UDP zero-checksum mode
>>        regardless of the decapsulator in order to strengthen the
>>        decapsulator's check of the IPv6 source address (i.e., the same
>>        IPv6 source address SHOULD NOT be used with more than one IPv6
>>        destination address, independent of whether that destination
>>        address is a unicast or multicast address). When this is not
>>        possible, it is RECOMMENDED to use each source IPv6 address for
>>        as few UDP zero-checksum mode GRE-in-UDP tunnels as is feasible.
>>
>>      f. When any middlebox exists on the path of a GRE-in-UDP tunnel,
>>        it is RECOMMENDED to use the default mode, i.e. use UDP
>>        checksum, to reduce the chance that the encapsulated packets to
>>        be dropped.
>>
>>
>>
>>
>>
>> Yong, Crabber, Xu, Herbert                                    [Page 14]
>>
>> --------------------------------------------------------------------------------
>>
>> Internet-Draft          GRE-in-UDP Encapsulation            March 2016
>>
>>      g. Any middlebox that allows the UDP zero-checksum mode for IPv6
>>        MUST comply with requirement 1 and 8-10 in Section 5 of
>>        [RFC6936].
>>
>>      h. Measures SHOULD be taken to prevent IPv6 traffic with zero UDP
>>        checksums from "escaping" to the general Internet; see Section
>>        8 for examples of such measures.
>>
>>      i. IPv6 traffic with zero UDP checksums MUST be actively monitored
>>        for errors by the network operator. For example, the operator
>>        may monitor Ethernet layer packet error rates.
>>
>>      j. If a packet with a non-zero checksum is received, the checksum
>>        MUST be verified before accepting the packet. This is
>>        regardless of whether the tunnel encapsulator and decapsulator
>>        have been configured with UDP zero-checksum mode.
>>
>>    The above requirements do not change either the requirements
>>    specified in [RFC2460] as modified by [RFC6935] or the requirements
>>    specified in [RFC6936].
>>
>>    The requirement to check the source IPv6 address in addition to the
>>    destination IPv6 address, plus the strong recommendation against
>>    reuse of source IPv6 addresses among GRE-in-UDP tunnels collectively
>>    provide some mitigation for the absence of UDP checksum coverage of
>>    the IPv6 header. A traffic-managed controlled environment that
>>    satisfies at least one of three conditions listed above in this
>>    section provides additional assurance.
>>
>>    A GRE-in-UDP tunnel is suitable for transmission over lower layers
>>    in the traffic-managed controlled environments that are allowed by
>>    the exceptions stated above and the rate of corruption of the inner
>>    IP packet on such networks is not expected to increase by comparison
>>    to GRE traffic that is not encapsulated in UDP.  For these reasons,
>>    GRE-in-UDP does not provide an additional integrity check except
>>    when GRE checksum is used when UDP zero-checksum mode is used with
>>    IPv6, and this design is in accordance with requirements 2, 3 and 5
>>    specified in Section 5 of [RFC6936].
>>
>>    Generic Router Encapsulation (GRE) does not accumulate incorrect
>>    state as a consequence of GRE header corruption. A corrupt GRE
>>    packet may result in either packet discard or forwarding of the
>>    packet without accumulation of GRE state. Active monitoring of GRE-
>>    in-UDP traffic for errors is REQUIRED as occurrence of errors will
>>    result in some accumulation of error information outside the
>>    protocol for operational and management purposes. This design is in
>>    accordance with requirement 4 specified in Section 5 of [RFC6936].
>>
>>
>>
>> Yong, Crabber, Xu, Herbert                                    [Page 15]
>>
>> --------------------------------------------------------------------------------
>>
>> Internet-Draft          GRE-in-UDP Encapsulation            March 2016
>>
>>    The remaining requirements specified in Section 5 of [RFC6936] are
>>    not applicable to GRE-in-UDP.  Requirements 6 and 7 do not apply
>>    because GRE does not include a control feedback mechanism.
>>    Requirements 8-10 are middlebox requirements that do not apply to
>>    GRE-in-UDP tunnel endpoints (see Section 7.1 for further middlebox
>>    discussion).
>>
>>    It is worth mentioning that the use of a zero UDP checksum should
>>    present the equivalent risk of undetected packet corruption when
>>    sending similar packet using GRE-in-IPv6 without UDP [RFC7676] and
>>    without GRE checksums.
>>
>>    In summary, a TMCE GRE-in-UDP Tunnel is allowed to use UDP-zero-
>>    checksum mode for IPv6 when the conditions and requirements stated
>>    above are met. Otherwise the UDP checksum need to be used for IPv6
>>    as specified in [RFC768] and [RFC2460]. Use of GRE checksum is
>>    RECOMMENED when the UDP checksum is not used.
>> "
>>
>> In GUE, to support UDP-checksum-zero, it said
>>
>> " Therefore, when GUE is used over
>>    IPv6, either the UDP checksum must be enabled or the GUE header
>>    checksum must be used.  An encapsulator MAY set a zero UDP checksum
>>    for performance or implementation reasons, in which case the GUE
>>    header checksum MUST be used or applicable requirements for using
>>    zero UDP checksums in [GREUDP] MUST be met. If the UDP checksum is
>>    enabled, then the GUE header checksum should not be used since it is
>>    mostly redundant."
>>
>> It's easy for me to add the similar words to the IP-in-UDP draft like "the
>> applicable requirements for using zero UDP checksum in [GREUDP] MUST be
>> met when zero UDP checksum is used by the tunnel ingress". However, the
>> major goal for disabling the UDP checksum is to improve the performance.
>> When GUE header checksum is used and/or the bunch of applicable
>> requirements as described in GRE/UDP are verified, is the goal of
>> improving
>> performance still achievable? If not, why not directly enable the
>> UDP-checksum
>> instead?
>>
>>     - fragmentation support
>>
>> In GRE/UDP, it said
>>
>> " 4.1. MTU and Fragmentation
>>
>>    Regarding packet fragmentation, an encapsulator/decapsulator SHOULD
>>    be compliant with [RFC7588] and perform fragmentation before the
>>    encapsulation. The size of fragments SHOULD be less or equal to the
>>    PMTU associated with the path between the GRE ingress and the GRE
>>    egress tunnel endpoints minus the GRE and UDP overhead ..."
>>
>> in GUE, it said
>>
>> " 4.9. MTU and fragmentation
>>
>>    Standard conventions for handling of MTU (Maximum Transmission Unit)
>>    and fragmentation in conjunction with networking tunnels
>>    (encapsulation of layer 2 or layer 3 packets) should be followed.
>>    Details are described in MTU and Fragmentation Issues with In-the-
>>    Network Tunneling [RFC4459]... "
>>
>> It seems that the only missing thing in the IP-in-UDP draft is to allow
>> the outer
>> fragmentation. However, as it said in
>> (https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02#page-13), "
>> ...IPsec
>> performs only Outer Fragmentation; this distinguishes it from IP-in-IP,
>> which
>> performs only Inner Fragmentation. " Note that IP-in-IP is the dominant
>> encapsulation choice within Softwires networks. In other words, performing
>> only inner fragmentation works very well in practice. Furthermore, the
>> outer
>> fragmentation issue (e.g., reassembly cost for the egress) would become
>> even
>> worse since the fragments of X-in-UDP packets are more likely to be
>> forwarded
>> across different paths than those of X-in-IP and X-in-GRE packets. Hence,
>> I'm
>> wondering whether it's worthwhile to support the outer fragmentation on
>> UDP
>> encapsulated packets which seems useless in practice.
>>
>>     - signalling support (e.g., to test whether a tunnel is up or
>>     to measure MTUs)
>>
>> I haven't found any description of this in both GRE/UDP and GUE. Did you?
>>
>>     - support for robust ID fields (related to fragmentation,
>>     e.g., to overcome the limits of IPv4 ID as per RFC 6864)
>>
>> I haven't found any description of this in both GRE/UDP and GUE. Did you?
>>
>> Xiaohu
>>
>> > It is not our obligation to find a way for your document to proceed -
>> > that onus is on you.
>> >
>> > Joe


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Subject: [tsvwg] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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AVT, RTCWEB and TSVWG

This announces an additional one week WG last call (in AVT) on the 
changes done to "Multimedia Congestion Control: Circuit Breakers for 
Unicast RTP Sessions" that is intended for Proposed Standard. The last 
call concludes on the 21st of June.

So this last call is deemed necessary due to normative changes in the 
behaviour as result of the conclusions of how to resolve the Discuss 
regarding reaction to ECN markings.

If there are no issues in this additional WG last call, the draft will 
be approved for publication as it has already passed IESG review.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-avtcore-rtp-circuit-breakers/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-avtcore-rtp-circuit-breakers-16

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-rtp-circuit-breakers-16

Cheers

Magnus Westerlund
AVTCORE WG chair

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Thread-Topic: [tsvwg] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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In the new ECN text in Section 7, I see:

   Given the above issues, implementations MAY ignore ECN-CE marks when=09
   determining if the congestion circuit breaker triggers, since=09
   excessive persistent congestion will eventually lead to packet loss=09
   that will trigger the circuit breaker.  Doing this will protect the=09
   network from congestion collapse, but might result in sub-optimal=09
   user experience for competing flows that share the bottleneck queue,=09
   since that queue will be driven to overflow, inducing high latency.
   If this is a concern, the only current guidance is for=09
   implementations to treat ECN-CE marked packets as equivalent to lost=09
   packets, whilst being aware that this might trigger the circuit=09
   breaker prematurely in future, depending on how AQM and ECN=09
   deployment evolves.  Developers that implement a circuit breaker=09
   based on ECN-CE marks will need to track future developments in AQM=09
   standards and deployed ECN marking behaviour, and ensure their=09
   implementations are updated to match.

I think "MAY" is inappropriate here - this text reads like a "SHOULD NOT"
requirement with an explanation of what happens when something else
is done.   Proposed rewrite:

   Given the above issues, implementations SHOULD NOT ignore ECN-CE marks
   when	determining whether to trigger the congestion circuit breaker, even=
=09
   though excessive persistent congestion will eventually lead to packet lo=
ss=09
   that triggers the circuit breaker.   Relying on packet loss protects the=
=09
   network from congestion collapse, but might result in sub-optimal=09
   user experience for competing flows that share the bottleneck queue,=09
   since that queue will be driven to overflow, inducing higher latency.

   Current implementation guidance is for
   implementations to treat ECN-CE marked packets as equivalent to lost=09
   packets, whilst being aware that this might trigger the circuit=09
   breaker prematurely in the future, depending on how AQM and ECN=09
   deployment evolves.  Developers that implement a circuit breaker=09
   based on ECN-CE marks will need to track future developments in AQM=09
   standards and deployed ECN marking behaviour, and ensure their=09
   implementations are updated to match.

The original "MAY" language invites implementers to completely ignore
ECN-CE marks, which I think is very bad advice.

Thanks, --David

> -----Original Message-----
> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Magnus Westerlun=
d
> Sent: Tuesday, June 14, 2016 4:15 AM
> To: IETF AVTCore WG
> Cc: rtcweb@ietf.org; tsvwg
> Subject: [tsvwg] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-=
breakers-
> 16
>=20
> AVT, RTCWEB and TSVWG
>=20
> This announces an additional one week WG last call (in AVT) on the
> changes done to "Multimedia Congestion Control: Circuit Breakers for
> Unicast RTP Sessions" that is intended for Proposed Standard. The last
> call concludes on the 21st of June.
>=20
> So this last call is deemed necessary due to normative changes in the
> behaviour as result of the conclusions of how to resolve the Discuss
> regarding reaction to ECN markings.
>=20
> If there are no issues in this additional WG last call, the draft will
> be approved for publication as it has already passed IESG review.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-avtcore-rtp-circuit-breakers/
>=20
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-avtcore-rtp-circuit-breakers-16
>=20
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-rtp-circuit-breake=
rs-16
>=20
> Cheers
>=20
> Magnus Westerlund
> AVTCORE WG chair
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------


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Subject: Re: [tsvwg] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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Hi,

Please see below.

Den 2016-06-14 kl. 15:20, skrev Black, David:
> In the new ECN text in Section 7, I see:
>
>    Given the above issues, implementations MAY ignore ECN-CE marks when	
>    determining if the congestion circuit breaker triggers, since	
>    excessive persistent congestion will eventually lead to packet loss	
>    that will trigger the circuit breaker.  Doing this will protect the	
>    network from congestion collapse, but might result in sub-optimal	
>    user experience for competing flows that share the bottleneck queue,	
>    since that queue will be driven to overflow, inducing high latency.
>    If this is a concern, the only current guidance is for	
>    implementations to treat ECN-CE marked packets as equivalent to lost	
>    packets, whilst being aware that this might trigger the circuit	
>    breaker prematurely in future, depending on how AQM and ECN	
>    deployment evolves.  Developers that implement a circuit breaker	
>    based on ECN-CE marks will need to track future developments in AQM	
>    standards and deployed ECN marking behaviour, and ensure their	
>    implementations are updated to match.
>
> I think "MAY" is inappropriate here - this text reads like a "SHOULD NOT"
> requirement with an explanation of what happens when something else
> is done.   Proposed rewrite:
>
>    Given the above issues, implementations SHOULD NOT ignore ECN-CE marks
>    when	determining whether to trigger the congestion circuit breaker, even	
>    though excessive persistent congestion will eventually lead to packet loss	
>    that triggers the circuit breaker.   Relying on packet loss protects the	
>    network from congestion collapse, but might result in sub-optimal	
>    user experience for competing flows that share the bottleneck queue,	
>    since that queue will be driven to overflow, inducing higher latency.
>
>    Current implementation guidance is for
>    implementations to treat ECN-CE marked packets as equivalent to lost	
>    packets, whilst being aware that this might trigger the circuit	
>    breaker prematurely in the future, depending on how AQM and ECN	
>    deployment evolves.  Developers that implement a circuit breaker	
>    based on ECN-CE marks will need to track future developments in AQM	
>    standards and deployed ECN marking behaviour, and ensure their	
>    implementations are updated to match.
>
> The original "MAY" language invites implementers to completely ignore
> ECN-CE marks, which I think is very bad advice.

So in the discussion between Mirja, the authors and me in attempting to 
resolve the discuss we ended up in MAY for a reason. We on the call 
agreed after a lengthy discussion that treating a ECN-CE mark the same 
way as a loss, is not appropriate. This despite that current in force 
specification indicate that would be the answer. However, as indicated 
in the updated text, there are a number of ongoing work that indicates 
that is not the expected actual marking behavior.

This resulted in another issue, there are no specified marking behavior 
that we define an appropriate response based on. When such a behavior is 
defined one can update circuit breaker with a well defined response to 
ECN-CE marks. So due to the lack of defined behavior we didn't see a 
possibility for using SHOULD NOT, as we can't point to what should be 
implemented. Thus we chose to propose MAY as it then is open to the 
implementation if they can find a well working response or completely 
ignore this.

So far I lack evidence for any significant deployment of ECN for RTP, 
unfortunately. This lessens the impact of this choice.

Secondly, ignoring the ECN-CE marks for a circuit breaker we believe 
will have a limited impact. If the implementation only have circuit 
breaker, i.e. no full fledged congestion controller and uses ECN, they 
can in worst case drive the buffer into the overload regime where it 
starts dropping packets. So if the flow does cause persistent congestion 
it will drive the AQM into packet drops and thus triggering of the 
circuit breaker. Yes, it will be unfair and push away responsive flows. 
But, that will happen also without ECN, is not something that the 
circuit breaker can protect against.

I hope this makes it clearer why we arrived at MAY and don't see the 
possibility to use SHOULD strength wording regarding ECN-CE marks.

Cheers

Magnus Westerlund
AVTCORE Chair.


----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From nobody Tue Jun 14 06:59:36 2016
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Subject: Re: [tsvwg] [rtcweb] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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Black, David <david.black@emc.com> wrote:
> 
> In the new ECN text in Section 7, I see:
> 
>    Given the above issues, implementations MAY ignore ECN-CE marks when	
>    determining if the congestion circuit breaker triggers, since	
>    excessive persistent congestion will eventually lead to packet loss	
>...
> I think "MAY" is inappropriate here - this text reads like a "SHOULD NOT"
> requirement with an explanation of what happens when something else
> is done.

   I'm very sympathetic to your issue: there are implementations of ECN
out there which _will_not_ drop an ECN-capable packet before the sending
queue has overflowed.

   But at the same time, we're headed into changes where ECN-CE marking
will become more frequent than packet-drop.

   We must find a way to avoid ECN-CE triggering the congestion circuit
breaker when the forwarding node has not reached a level of congestion
which could trigger packet loss.

   I'd prefer language (perhaps in an RFC-3168-bis) stating that packet
drop MUST me used, at least occasionally, on packets which would be
ECN-CE marked, when the overall congestion at that node reaches the
drop level for non-ECN-capable packets. (The forwarding node is the only
node with sufficient information to see the issue.)

   (I recommend folks interested in circuit-breaker pay attention to the
L4S BoF being scheduled for Berlin.)

--
John Leslie <john@jlc.net>


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From: "Black, David" <david.black@emc.com>
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Subject: Re: [tsvwg] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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> This resulted in another issue, there are no specified marking behavior
> that we define an appropriate response based on. When such a behavior is
> defined one can update circuit breaker with a well defined response to
> ECN-CE marks. So due to the lack of defined behavior we didn't see a
> possibility for using SHOULD NOT, as we can't point to what should be
> implemented. Thus we chose to propose MAY as it then is open to the
> implementation if they can find a well working response or completely
> ignore this.

My problem is that the new "MAY" sentence is unqualified:

> >    Given the above issues, implementations MAY ignore ECN-CE marks when
> >    determining if the congestion circuit breaker triggers,

That invites implementers to ignore ECN-CE marks without giving this topic =
the
level of consideration anticipated in this discussion and elsewhere in Sect=
ion 7.
If "SHOULD NOT" language is unworkable, please propose alternate text that
qualifies applicability of the "MAY" - e.g., only after careful considerati=
on of
everything that we're now discussing and the rest of the text in Section 7.

I proposed "SHOULD NOT" because it appear that the original RFC 2119 defini=
tion
of that term is a good fit to this situation:

4. SHOULD NOT   This phrase, or the phrase "NOT RECOMMENDED" mean that
   there may exist valid reasons in particular circumstances when the
   particular behavior is acceptable or even useful, but the full
   implications should be understood and the case carefully weighed
   before implementing any behavior described with this label.

Thanks, --David


> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Tuesday, June 14, 2016 9:44 AM
> To: Black, David; IETF AVTCore WG
> Cc: rtcweb@ietf.org; tsvwg
> Subject: Re: [tsvwg] WG Last Call on changes: draft-ietf-avtcore-rtp-circ=
uit-
> breakers-16
>=20
> Hi,
>=20
> Please see below.
>=20
> Den 2016-06-14 kl. 15:20, skrev Black, David:
> > In the new ECN text in Section 7, I see:
> >
> >    Given the above issues, implementations MAY ignore ECN-CE marks when
> >    determining if the congestion circuit breaker triggers, since
> >    excessive persistent congestion will eventually lead to packet loss
> >    that will trigger the circuit breaker.  Doing this will protect the
> >    network from congestion collapse, but might result in sub-optimal
> >    user experience for competing flows that share the bottleneck queue,
> >    since that queue will be driven to overflow, inducing high latency.
> >    If this is a concern, the only current guidance is for
> >    implementations to treat ECN-CE marked packets as equivalent to lost
> >    packets, whilst being aware that this might trigger the circuit
> >    breaker prematurely in future, depending on how AQM and ECN
> >    deployment evolves.  Developers that implement a circuit breaker
> >    based on ECN-CE marks will need to track future developments in AQM
> >    standards and deployed ECN marking behaviour, and ensure their
> >    implementations are updated to match.
> >
> > I think "MAY" is inappropriate here - this text reads like a "SHOULD NO=
T"
> > requirement with an explanation of what happens when something else
> > is done.   Proposed rewrite:
> >
> >    Given the above issues, implementations SHOULD NOT ignore ECN-CE mar=
ks
> >    when	determining whether to trigger the congestion circuit breaker,
> even
> >    though excessive persistent congestion will eventually lead to packe=
t loss
> >    that triggers the circuit breaker.   Relying on packet loss protects=
 the
> >    network from congestion collapse, but might result in sub-optimal
> >    user experience for competing flows that share the bottleneck queue,
> >    since that queue will be driven to overflow, inducing higher latency=
.
> >
> >    Current implementation guidance is for
> >    implementations to treat ECN-CE marked packets as equivalent to lost
> >    packets, whilst being aware that this might trigger the circuit
> >    breaker prematurely in the future, depending on how AQM and ECN
> >    deployment evolves.  Developers that implement a circuit breaker
> >    based on ECN-CE marks will need to track future developments in AQM
> >    standards and deployed ECN marking behaviour, and ensure their
> >    implementations are updated to match.
> >
> > The original "MAY" language invites implementers to completely ignore
> > ECN-CE marks, which I think is very bad advice.
>=20
> So in the discussion between Mirja, the authors and me in attempting to
> resolve the discuss we ended up in MAY for a reason. We on the call
> agreed after a lengthy discussion that treating a ECN-CE mark the same
> way as a loss, is not appropriate. This despite that current in force
> specification indicate that would be the answer. However, as indicated
> in the updated text, there are a number of ongoing work that indicates
> that is not the expected actual marking behavior.
>=20
> This resulted in another issue, there are no specified marking behavior
> that we define an appropriate response based on. When such a behavior is
> defined one can update circuit breaker with a well defined response to
> ECN-CE marks. So due to the lack of defined behavior we didn't see a
> possibility for using SHOULD NOT, as we can't point to what should be
> implemented. Thus we chose to propose MAY as it then is open to the
> implementation if they can find a well working response or completely
> ignore this.
>=20
> So far I lack evidence for any significant deployment of ECN for RTP,
> unfortunately. This lessens the impact of this choice.
>=20
> Secondly, ignoring the ECN-CE marks for a circuit breaker we believe
> will have a limited impact. If the implementation only have circuit
> breaker, i.e. no full fledged congestion controller and uses ECN, they
> can in worst case drive the buffer into the overload regime where it
> starts dropping packets. So if the flow does cause persistent congestion
> it will drive the AQM into packet drops and thus triggering of the
> circuit breaker. Yes, it will be unfair and push away responsive flows.
> But, that will happen also without ECN, is not something that the
> circuit breaker can protect against.
>=20
> I hope this makes it clearer why we arrived at MAY and don't see the
> possibility to use SHOULD strength wording regarding ECN-CE marks.
>=20
> Cheers
>=20
> Magnus Westerlund
> AVTCORE Chair.
>=20
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------


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Date: Wed, 15 Jun 2016 12:11:38 +0100
From: Derek Fawcus <dfawcus+lists-tsvwg@employees.org>
To: tsvwg@ietf.org
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Subject: Re: [tsvwg] I-D Action: draft-ietf-tsvwg-natsupp-09.txt
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On Wed, May 18, 2016 at 01:13:46pm -0700, internet-drafts@ietf.org wrote:
> 
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> This draft is a work item of the Transport Area Working Group of the IETF.
> 
>         Title           : Stream Control Transmission Protocol (SCTP) Network Address Translation Support
>         Authors         : Randall R. Stewart
>                           Michael Tuexen
>                           Irene Ruengeler
> 	Filename        : draft-ietf-tsvwg-natsupp-09.txt
> 	Pages           : 44
> 	Date            : 2016-05-18

So does anyone know if there are any host stacks which support the above extensions yet?
(Even if only process space implementations).

i.e. processing the extended ABORT chunk (with M bit set),  and will use the Port Number
Collision Error Cause to trigger a re-attempt with a new port number?

Basically the situation in section 6.4 plus the handling in section 6.3.
I note that 6.3 mentioned re-attemting with a new V-Tag,  but 6.4 does not mention
the client re-attemping with a new port number (assuming it was an ephemeral port).

As to why:
   Assume a CGN where the map of private addr+port <=> public add+port assignments
   is being shared for TCP/UDP/UDP-Lite/DCCP, it would be useful to be able to use
   the same map for SCTP.

DF


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Subject: Re: [tsvwg] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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> On 14 Jun 2016, at 15:20, Black, David <david.black@emc.com> wrote:
>=20
>> This resulted in another issue, there are no specified marking =
behavior
>> that we define an appropriate response based on. When such a behavior =
is
>> defined one can update circuit breaker with a well defined response =
to
>> ECN-CE marks. So due to the lack of defined behavior we didn't see a
>> possibility for using SHOULD NOT, as we can't point to what should be
>> implemented. Thus we chose to propose MAY as it then is open to the
>> implementation if they can find a well working response or completely
>> ignore this.
>=20
> My problem is that the new "MAY" sentence is unqualified:
>=20
>>>   Given the above issues, implementations MAY ignore ECN-CE marks =
when
>>>   determining if the congestion circuit breaker triggers,
>=20
> That invites implementers to ignore ECN-CE marks without giving this =
topic the
> level of consideration anticipated in this discussion and elsewhere in =
Section 7.
> If "SHOULD NOT" language is unworkable, please propose alternate text =
that
> qualifies applicability of the "MAY" - e.g., only after careful =
consideration of
> everything that we're now discussing and the rest of the text in =
Section 7.
>=20
> I proposed "SHOULD NOT" because it appear that the original RFC 2119 =
definition
> of that term is a good fit to this situation:
>=20
> 4. SHOULD NOT   This phrase, or the phrase "NOT RECOMMENDED" mean that
>   there may exist valid reasons in particular circumstances when the
>   particular behavior is acceptable or even useful, but the full
>   implications should be understood and the case carefully weighed
>   before implementing any behavior described with this label.

The issue with saying =E2=80=9Cimplementations SHOULD NOT ignore ECN-CE =
marks=E2=80=9D, is that it implies that they should respond to those =
marks. However, the outcome of the long discussion we had with Mirja was =
that there=E2=80=99s no consensus on what=E2=80=99s the correct =
response, and concern that implementations picking the wrong choice will =
hinder the deployment of ECN.=20



--=20
Colin Perkins
https://csperkins.org/





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From: "Black, David" <david.black@emc.com>
To: Colin Perkins <csp@csperkins.org>
Thread-Topic: [AVTCORE] [tsvwg] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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Hi Gorry!

Responses to Gorry's lengthy feedback (thanks!) inline.  Also, I
have issued an -04 that takes into account this feedback and offline
conversations with Gorry, David and others.  It is available now:

The whole draft has undergone some editing, but sections 1 & 2 could
use some review.  (Although the entire document is short, so if you
can, please look it over.)

https://datatracker.ietf.org/doc/draft-ietf-tcpm-rto-consider/
http://www.icir.org/mallman/pubs/All16a/

Comments, please!

allman



> * Introduction/Scope.

Re-worked substantially.

> * Comments on the recommendations
>=20
> - Some of these comments relate I think to assumptions about TCP
> specifics, that could be generalised, or may be handed in the
> introduction.
>=20
> (1)
> "the
> RTO MUST be conservatively set to no less than 1 second."
> - If this is TCP or SCTP, then RTO is defined. If more is intended
> RTO needs to have been defined.

New intro defines it.

> " For instance, consider a DNS request from a client to a
> resolver. When the request can be served from the resolver's
> cache the FT likely well approximates the network RTT between
> the client and resolver. However, on a cache miss the resolver
> will have to request the needed information from authoritative
> DNS servers, which will non-trivially increase the FT and
> therefore the FT between the client and resolver does not well
> match the network-based RTT between the two hosts."
>=20
> - If the transport is assumed as DNS/UDP, this needs to be said in
> the example, up until this point the ID only discusses TCP and
> clearly DNS/TCP does not have this concern.

Yes, implicit assumption of UDP.  Now explicit.  Good catch.
Thanks.=20

> " (b) FT observations MUST be taken regularly."

I have re-factored the text a bit in -04.  You might like it better.
I think it reads better.  Although the intent did not change.

> - The language below seems inappropriate in a MUST clause, I think
> this implies it is a SHOULD, and the exception conditions do need to
> be discussed, since the period isn't stated, how is this normative
> requirement to be interpreted?

I guess I don't follow.  I read this as "you MUST take FT samples
regularly" and "regularly SHOULD be once / RTT (or as often as you
exchange data if less frequently than once / RTT".  I.e., the MUST
and the SHOULD are about different things.  I tried to make the
words more clearly delineate this in -04.

> - Maybe this text needs clarification: The current wording
> requirement could be read to contradict RFC5405, that avoids against
> sending UDP probes and traffic when an application is idle. This
> needs more careful explanation. Even for TCP/SCTP, I would not
> advocate requiring keep-alive or heartbeat packets as suggested by
> "the requirement that the FT be sampled continuously throughout the
> lifetime of a connection."

See below (where you return to this issue).

> "For the purpose of this
> requirement, we state that FT samples SHOULD be taken at
> least once per RTT or as frequently as data is exchanged and
> ACKed if that happens less frequently than every RTT."
> and later:
> " Hence, this once-per-RTT sampling requirement is explicitly
> a "SHOULD" and not a "MUST".
>=20
> " However, we also recognize that it may not always be
> practical to take a FT sample this often in all cases."
>=20
> -I agree with the =E2=80=9CSHOULD=E2=80=9D from the perspective of TCP, s=
ome methods
> respond more slowly to congestion that TCP, and sample less
> often. How does this relate to the various repair mechanisms in RTP?
> (they also can have timers and retransmit) - are these counter
> examples where the SHOULD does not apply?

This isn't about how often we respond to congestion.  That is
orthogonal to this document.  We are trying to schedule
retransmissions.  So, what we really want to know---and I think this
is now better defined in the intro text you have seen---is how long
we should await delivery confirmation before tripping a retransmit.
That confirmation could be dictated by the network RTT.  Or, be
longer than that depending on protocol operation.  That is why the
document uses "FT" and not "RTT".  So, I am not sure where you're
going here with the congestion responses.

> =E2=80=9C(c) FT samples used in the computation of the RTO MUST NOT be
> ambiguous.=E2=80=9D
>=20
> - I read this requirement is on the samples. That's not
> correct. Maybe it should be
> more like:
> "An RTO algorithm MUST NOT use FT samples that are ambiguous when
> calculating the RTO."

I changed it to this in -04:

        (d) An RTO mechanism MUST NOT use ambiguous FT samples.

( (c) is now something else---see below---and so this changed to (d) )

> - I wonder if section 3 could also talk about other ways to measure
> RTTs when not sending data, such as keepalive and SCTP heartbeat
> messages.

We don't always care about measuring RTTs.  So, we have to be
careful here, but I have added a 2(c) in section 3 on this topic.
It says:

        (c) FT observations MAY be taken from non-data exchanges.

            Some protocols use keepalives, heartbeats or other messages
            to exchange control information.  To the extent that the
            latency of these transactions mirrors data exchange, they
            can be leveraged to take FT samples within the RTO
            mechanism.  Such samples can help protocols keep their RTO
            accurate during lulls in data transmission.  However, given
            that these messages may not be subject to the same delays as
            data transmission, we do not take a general view on whether
            this is useful or not.

> "The backoff may be removed
> after the successful transmission of non-retransmitted data."
>=20
> - The "may" appears to be protocol permission, can it be "MAY=E2=80=9D?

I think this comes from 6298 wording---which I just checked and is
not in protocol permission language.  I think you're correct here.
I have changed it in -04 to be more precise all the way around:

        The backoff SHOULD be removed after the successful repair of
        the lost data and subsequent transmission of
        non-retransmitted data.

> " A maximum value MAY be placed on the RTO provided it is at least
> 60 seconds (a la [RFC6298])."
>=20
> - To me, there seem to be two requirements here:
>=20
> " A maximum value MAY be placed on the RTO. The maximum MUST NOT exceed
> 60 seconds (as in [RFC6298])."

Sounds reasonable.  And, especially so since you mis-understood it
and translated it wrong! :)

The version in -04 is ...

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


> (4) "An exception is made to this rule if an IETF standardized
> mechanism is used to determine that a particular loss is due to
> a non-congestion event (e.g., packet corruption)."
>=20
> - NiT: I wonder if this can be mis-cited? There are as yet no specs
> in the space published in the RFC series. Could we be more careful
> on endorsing this work, and say perhaps that "exceptions could be
> made to this rule if an IETF standardized ... .

Good call.  I made this change.

> "Additionally,
> RTO-triggered congestion control actions may be reversed when a
> standard mechanism determines that the cause of the loss was not
> congestion after all."
>=20
> - Examples from the RFC series exist, and would be good. E.g. Eifel
> or something else?

I added an "(e.g., [RFC5682])".  Good call.

> - Is this the place to define "spurious", since this is used later.

Hm.  I was trying to figure out how to cleanly wedge that in, but I
couldn't find something I liked.  Instead I tweaked the intro at the
first use of "spurious".

> " We note that research has shown the tension between the
> responsiveness and correctness of retransmission timeouts seems to
> be a fundamental tradeoff [AP99]."
> - I think this paper is about TCP, and I think the tension is clear
> for protocols like TCP.

I added a note.

> "While the implication of these deviations
> from the standard may be more spurious retransmits (per [AP99]), we
> are aware of no large scale problems caused by this change to the
> minimum RTO."
>=20
> - I'd like to argue that a lower RTO can reduce performance for
> paths where this RTO can be set below the RTT. Such paths do exist,
> and middleboxes sometimes split TCP connections to overcome this
> issue. I think at least the ID should note the issue, even if it
> claims that these paths are somehow accommodated within the
> guidance.

I do not quite understand this.  I am not conjuring a case where the
timeout could ever be less than the RTT.=20=20

A middlebox can split the TCP connection so we end up with two
connections A<->M and M<->B.  Are you considering the "RTT" to be
A<->B and are then saying A can actually retransmit based on the
A<->M delay?  Sure.  But, in this case I am not sure how A's TCP
will ever know the A<->B "RTT".  For non-TCP protocols that have a
higher layer delivery confirmation mechanism that comes from B I
could see this.  However, in this case, I am not sure how we'd know
to retransmit on A<->M timescales.

Basically I am confused here.  I am happy to address this if you can
explain to me what you mean.

> * Standards status and RFC 2119 recommendations.
>=20
> "nor limit future standardization of specific RTO mechanisms."
> - I'd need advice, but I thought BCPs did exactly this. Paraphrasing
> slightly, the purpose of the RFC5405 was to impose default
> constraints that future RFCs have to justify their choices
> against. That's limiting. The MUSTs are real requirements and the
> SHOULD strong desirables. The feedback we have to the RFC5405.bis is
> to stengthen more SHOULDs to MUSTs - I'm going to speak to my ADs
> before I agree to any of these. But if the IESG agree, then I think
> that is what BCP status is about.
>=20
> Scope (a) I don=E2=80=99t disagree, unless the community explicity update=
s a
> spec a new BCP doesn't change old RFCs.
>=20
> Scope (v) - To me seems weird. Because it seems to set a "SHOULD"
> over the entire document, and that document contains MUSTs. I can't
> see how an RFC can have a "SHOULD-MUST", but likely the ADs can
> advise during WGLC.

(This has all been removed / changed in a big way.  So, I am not
commenting on it.)

> I worry the following can be cited elsewhere as compliance of a
> protocol or mechanism by the IETF, when really I think this is
> intended to define an acceptable RTO response. Some tightening of
> the language would seem appropriate:
>=20
> "RTO mechanisms that are not standardized but adhere to the
> requirements in the following section are deemed consistent with
> the standards. "
> - sounds dangerously wide when cited as text - I was presuming the
> document intended "are deemed consistent with
> the standards in the RTO behaviour" - or something, I presume you
> don't want people to cite this RFC saying their new protocol foo-bar
> is "consistent with IETF standards" because it does this.
>=20
> "an implementation is considered standards compliant"
> - I quibble again here on language, because I don't think this is
> true generally of the protocol, only true of their RTO response.

(This has all been removed / changed in a big way.  So, I am not
commenting on it.)

> "In other words, the requirements in this document can be viewed as
> specifying the default properties of an RTO mechanism."
> - This seems like an informational statement that could be cited as
> a contradiction to the RFC2119 text unless worded carefully? This
> sort of text needs AD advice if kept.
>=20
> "Specifications can more concretely nail down specifics within these
> defaults or work outside the defaults as necessary."
> - The same comment on the second half of this sentence.

I removed the word "default" from the document.  I think the last
paragraph of the intro better addresses things now.

> "That is, this document's
> stance is to not limit the community's ability to make exceptions
> to the requirements herein for particular cases."
> - If this is BCP, I think this needs to be tighter, I don't think
> any ID can expect to automatically make exceptions to a BCP, by
> simply saying "the WG decided differently" if you want that, then I
> think this document needs to be INFO (I'm not the Chair or AD for
> the document). If it's BCP, I'd expect the consensus to set the best
> current practice and the ADs to block new RFCs with at least a
> DISCUSS to work out why WG XXX was going against what has been IETF
> consensus.

I don't think you can go info and make the same sort of
requirements.

I don't really disagree with what you say.  I.e., I would expect
that if this were a BCP that it would be taken into account when
designing RTOs.  And, if something wanted to be more aggressive or
do something outside the requirements that this be done explicitly
with the BCP text as context.  I.e., I would expect that the WG
would consider this and have an answer for why the deviation is OK
far before it hits the IESG.

> "However,
> implementations that fall within the defaults do not require
> explicit specifications to be considered consistent with the
> standards."
> - I could not parse this sentence, it sounds like it could be
> normative language without a keyword.

Text is gone in -04.

> Section 3:
>=20
> "We now list the requirements that SHOULD apply when designing
> retransmission timeout (RTO) mechanisms."
> - this seems to scope a "MUST" with a preceding "SHOULD". I question
> if this is allowed in a BCP.

I understand what is meant.  But, it is not important enough in this
instance to argue about, IMO, and so I nuked the SHOULD normative
language from the beginning of section 3.




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On 6/11/2016 10:11 PM, Lloyd Wood wrote:
> Fragmentation should be strongly discouraged.

Avoided where possible for sure.
>
> if you've designed a tunnelling solution and you have fragmentation
> happening as a matter of course, you've designed it wrong.

If you have a solution that recurses (X over X, with some intervening
layers or not), and X has a required minimum message size, then you
either have a solution that - in at least some non-trivial situations -
will fragment as a matter of course or a fantasy that cannot exist.

The only reason we avoid this in typical operation is by avoiding X over
X. Once you have a tunnel, that might not be avoidable (or even
detectable, depending on encryption).

Joe

>
> Lloyd Wood
> lloyd.wood@yahoo.co.uk
>
> On Friday, May 27, 2016, 8:10 PM, Xuxiaohu <xuxiaohu@huawei.com> wrote:
>
>     <Note that I have changed the subject of the email hence it has
>     nothing to do with the WG adoption call now. It's just a
>     discussion on a particular issue which is related to those WGs
>     which are working on UDP tunnels. The reason for containing the
>     old email is to use it as a background which may be useful for
>     better understanding of this particular issue>
>
>     The possible side-effect of performing fragmentation on UDP
>     encapsulated packets is to worsen the reassembly burden on tunnel
>     egress since fragments of UDP encapsulated packets are more likely
>     to be forwarded across different paths towards the tunnel egress
>     than those of IP or GRE encapsulated packets.
>
>     It seems that most X-over-UDP proposals choose to prohibit the
>     tunnel ingress from performing fragmentation on UDP encapsulated
>     packets. See the following quoted text regarding fragmentation
>     from those X-over-UDP drafts:
>
>     LISP:
>
>     When an ITR receives a packet from a site-facing interface and adds H
>       octets worth of encapsulation to yield a packet size greater than L
>       octets, it resolves the MTU issue by first splitting the original
>       packet into 2 equal-sized fragments.  A LISP header is then
>     prepended
>       to each fragment.
>
>     VXLAN:
>
>     VTEPs MUST NOT fragment VXLAN packets.  Intermediate routers may
>       fragment encapsulated VXLAN packets due to the larger frame size.
>       The destination VTEP MAY silently discard such VXLAN fragments.
>
>     VXLAN-GPE:
>
>     VTEPs MUST never fragment an encapsulated VXLAN GPE packet, and when
>       the outer IP header is IPv4, VTEPs MUST set the DF bit in the outer
>       IPv4 header.
>
>     GEVENE:
>
>       To prevent fragmentation and maximize performance, the best practice
>       when using Geneve is to ensure that the MTU of the physical network
>       is greater than or equal to the MTU of the encapsulated network plus
>       tunnel headers.
>
>     GUE:
>
>         If a packet is fragmented before encapsulation in GUE, all the
>         related fragments must be encapsulated using the same source port
>         (inner flow identifier). An operator may set MTU to account for
>         encapsulation overhead and reduce the likelihood of fragmentation.
>
>     GRE/UDP
>
>     Regarding packet fragmentation, an encapsulator/decapsulator SHOULD
>       be compliant with [RFC7588] and perform fragmentation before the
>       encapsulation.
>
>     However, the above choice seems conflict with the requirements as
>     described in https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02
>
>
>     I wonder whether the IETF should reach a consensus on whether or
>     not the fragmentation on UDP encapsulated packets should be allowed.
>      
>     Best regards,
>     Xiaohu
>
>     > -----Original Message-----
>     > From: Xuxiaohu
>     > Sent: Thursday, May 26, 2016 4:35 PM
>     > To: 'Joe Touch'; Fred Baker (fred); Wassim Haddad
>     > Cc: int-area@ietf.org <javascript:return>
>     > Subject: RE: [Int-area] Call for adoption of
>     draft-xu-intarea-ip-in-udp-03
>     >
>     >
>     >
>     > > -----Original Message-----
>     > > From: Joe Touch [mailto:touch@isi.edu <javascript:return>]
>     > > Sent: Thursday, May 26, 2016 2:11 AM
>     > > To: Xuxiaohu; Fred Baker (fred); Wassim Haddad
>     > > Cc: int-area@ietf.org <javascript:return>
>     > > Subject: Re: [Int-area] Call for adoption of
>     > > draft-xu-intarea-ip-in-udp-03
>     > >
>     > >
>     > >
>     > > On 5/24/2016 7:24 PM, Xuxiaohu wrote:
>     > > > Hi Joe,
>     > > >
>     > > > I wonder whether you want to tell me the following truth by the
>     > > > example that you gave: no matter whatever improvements we
>     had done
>     > > > with this draft, those persons who dislike it by the light
>     of nature
>     > > > would dislike it in the end.
>     > >
>     > > The only improvements that would make this doc useful would be
>     to add
>     > > capabilities already in GRE/UDP or GUE/UDP, which we already have.
>     >
>     > Let's go over the four things you mentioned earlier in GRE/UDP
>     and GUE/UDP:
>     >
>     >     - stronger checksums
>     >
>     > In GRE/UDP, in order to use UDP-zero-checksum, it gave the following
>     > restrictions:
>     > " 6. UDP Checksum Handling
>     >
>     >    6.1. UDP Checksum with IPv4
>     >
>     >    For UDP in IPv4, the UDP checksum MUST be processed as
>     specified in
>     >    [RFC768] and [RFC1122] for both transmit and receive. The IPv4
>     >
>     >
>     >
>     > Yong, Crabber, Xu, Herbert                                   
>     [Page 12]
>     >
>     --------------------------------------------------------------------------------
>     >
>     > Internet-Draft          GRE-in-UDP Encapsulation           
>     March 2016
>     >
>     >    header includes a checksum which protects against mis-delivery of
>     >    the packet due to corruption of IP addresses. The UDP checksum
>     >    potentially provides protection against corruption of the UDP
>     header,
>     >    GRE header, and GRE payload. Disabling the use of checksums is a
>     >    deployment consideration that should take into account the
>     risk and
>     >    effects of packet corruption.
>     >
>     >    When a decapsulator receives a packet, the UDP checksum field
>     MUST
>     >    be processed. If the UDP checksum is non-zero, the
>     decapsulator MUST
>     >    verify the checksum before accepting the packet. By default a
>     >    decapsulator SHOULD accept UDP packets with a zero checksum.
>     A node
>     >    MAY be configured to disallow zero checksums per [RFC1122];
>     this may
>     >    be done selectively, for instance disallowing zero checksums from
>     >    certain hosts that are known to be sending over paths subject to
>     >    packet corruption. If verification of a non-zero checksum
>     fails, a
>     >    decapsulator lacks the capability to verify a non-zero
>     checksum, or
>     >    a packet with a zero-checksum was received and the
>     decapsulator is
>     >    configured to disallow, the packet MUST be dropped and an
>     event MAY
>     >    be logged.
>     >
>     >    6.2. UDP Checksum with IPv6
>     >
>     >    For UDP in IPv6, the UDP checksum MUST be processed as
>     specified in
>     >    [RFC768] and [RFC2460] for both transmit and receive.
>     >
>     >    When UDP is used over IPv6, the UDP checksum is relied upon to
>     >    protect both the IPv6 and UDP headers from corruption. As such, A
>     >    default GRE-in-UDP Tunnel MUST perform UDP checksum; A TMCE
>     GRE-in-
>     >    UDP Tunnel MAY be configured with the UDP zero-checksum mode
>     if the
>     >    traffic-managed controlled environment or a set of closely
>     >    cooperating traffic-managed controlled environments (such as by
>     >    network operators who have agreed to work together in order to
>     >    jointly provide specific services) meet at least one of following
>     >    conditions:
>     >
>     >    a. It is known (perhaps through knowledge of equipment types and
>     >      lower layer checks) that packet corruption is exceptionally
>     >      unlikely and where the operator is willing to take the risk of
>     >      undetected packet corruption.
>     >
>     >    b. It is judged through observational measurements (perhaps of
>     >      historic or current traffic flows that use a non-zero checksum)
>     >      that the level of packet corruption is tolerably low and where
>     >      the operator is willing to take the risk of undetected packet
>     >      corruption.
>     >
>     >
>     >
>     >
>     >
>     > Yong, Crabber, Xu, Herbert                                   
>     [Page 13]
>     >
>     --------------------------------------------------------------------------------
>     >
>     > Internet-Draft          GRE-in-UDP Encapsulation           
>     March 2016
>     >
>     >    c. Carrying applications that are tolerant of mis-delivered or
>     >      corrupted packets (perhaps through higher layer checksum,
>     >      validation, and retransmission or transmission redundancy)
>     where
>     >      the operator is willing to rely on the applications using the
>     >      tunnel to survive any corrupt packets.
>     >
>     >    The following requirements apply to a TMCE GRE-in-UDP tunnel that
>     >    use UDP zero-checksum mode:
>     >
>     >      a. Use of the UDP checksum with IPv6 MUST be the default
>     >        configuration of all GRE-in-UDP tunnels.
>     >
>     >      b. The GRE-in-UDP tunnel implementation MUST comply with all
>     >        requirements specified in Section 4 of [RFC6936] and with
>     >        requirement 1 specified in Section 5 of [RFC6936].
>     >
>     >      c. The tunnel decapsulator SHOULD only allow the use of UDP
>     zero-
>     >        checksum mode for IPv6 on a single received UDP Destination
>     >        Port regardless of the encapsulator. The motivation for this
>     >        requirement is possible corruption of the UDP Destination
>     Port,
>     >        which may cause packet delivery to the wrong UDP port. If
>     that
>     >        other UDP port requires the UDP checksum, the mis-delivered
>     >        packet will be discarded.
>     >
>     >      d. It is RECOMMENDED that the UDP zero-checksum mode for
>     IPv6 is
>     >        only enabled for certain selected source addresses. The
>     tunnel
>     >        decapsulator MUST check that the source and destination IPv6
>     >        addresses are valid for the GRE-in-UDP tunnel on which the
>     >        packet was received if that tunnel uses UDP zero-checksum
>     mode
>     >        and discard any packet for which this check fails.
>     >
>     >      e. The tunnel encapsulator SHOULD use different IPv6
>     addresses for
>     >        each GRE-in-UDP tunnel that uses UDP zero-checksum mode
>     >        regardless of the decapsulator in order to strengthen the
>     >        decapsulator's check of the IPv6 source address (i.e.,
>     the same
>     >        IPv6 source address SHOULD NOT be used with more than one
>     IPv6
>     >        destination address, independent of whether that destination
>     >        address is a unicast or multicast address). When this is not
>     >        possible, it is RECOMMENDED to use each source IPv6
>     address for
>     >        as few UDP zero-checksum mode GRE-in-UDP tunnels as is
>     feasible.
>     >
>     >      f. When any middlebox exists on the path of a GRE-in-UDP
>     tunnel,
>     >        it is RECOMMENDED to use the default mode, i.e. use UDP
>     >        checksum, to reduce the chance that the encapsulated
>     packets to
>     >        be dropped.
>     >
>     >
>     >
>     >
>     >
>     > Yong, Crabber, Xu, Herbert                                   
>     [Page 14]
>     >
>     --------------------------------------------------------------------------------
>     >
>     > Internet-Draft          GRE-in-UDP Encapsulation           
>     March 2016
>     >
>     >      g. Any middlebox that allows the UDP zero-checksum mode for
>     IPv6
>     >        MUST comply with requirement 1 and 8-10 in Section 5 of
>     >        [RFC6936].
>     >
>     >      h. Measures SHOULD be taken to prevent IPv6 traffic with
>     zero UDP
>     >        checksums from "escaping" to the general Internet; see
>     Section
>     >        8 for examples of such measures.
>     >
>     >      i. IPv6 traffic with zero UDP checksums MUST be actively
>     monitored
>     >        for errors by the network operator. For example, the operator
>     >        may monitor Ethernet layer packet error rates.
>     >
>     >      j. If a packet with a non-zero checksum is received, the
>     checksum
>     >        MUST be verified before accepting the packet. This is
>     >        regardless of whether the tunnel encapsulator and
>     decapsulator
>     >        have been configured with UDP zero-checksum mode.
>     >
>     >    The above requirements do not change either the requirements
>     >    specified in [RFC2460] as modified by [RFC6935] or the
>     requirements
>     >    specified in [RFC6936].
>     >
>     >    The requirement to check the source IPv6 address in addition
>     to the
>     >    destination IPv6 address, plus the strong recommendation against
>     >    reuse of source IPv6 addresses among GRE-in-UDP tunnels
>     collectively
>     >    provide some mitigation for the absence of UDP checksum
>     coverage of
>     >    the IPv6 header. A traffic-managed controlled environment that
>     >    satisfies at least one of three conditions listed above in this
>     >    section provides additional assurance.
>     >
>     >    A GRE-in-UDP tunnel is suitable for transmission over lower
>     layers
>     >    in the traffic-managed controlled environments that are
>     allowed by
>     >    the exceptions stated above and the rate of corruption of the
>     inner
>     >    IP packet on such networks is not expected to increase by
>     comparison
>     >    to GRE traffic that is not encapsulated in UDP.  For these
>     reasons,
>     >    GRE-in-UDP does not provide an additional integrity check except
>     >    when GRE checksum is used when UDP zero-checksum mode is used
>     with
>     >    IPv6, and this design is in accordance with requirements 2, 3
>     and 5
>     >    specified in Section 5 of [RFC6936].
>     >
>     >    Generic Router Encapsulation (GRE) does not accumulate incorrect
>     >    state as a consequence of GRE header corruption. A corrupt GRE
>     >    packet may result in either packet discard or forwarding of the
>     >    packet without accumulation of GRE state. Active monitoring
>     of GRE-
>     >    in-UDP traffic for errors is REQUIRED as occurrence of errors
>     will
>     >    result in some accumulation of error information outside the
>     >    protocol for operational and management purposes. This design
>     is in
>     >    accordance with requirement 4 specified in Section 5 of
>     [RFC6936].
>     >
>     >
>     >
>     > Yong, Crabber, Xu, Herbert                                   
>     [Page 15]
>     >
>     --------------------------------------------------------------------------------
>     >
>     > Internet-Draft          GRE-in-UDP Encapsulation           
>     March 2016
>     >
>     >    The remaining requirements specified in Section 5 of
>     [RFC6936] are
>     >    not applicable to GRE-in-UDP.  Requirements 6 and 7 do not apply
>     >    because GRE does not include a control feedback mechanism.
>     >    Requirements 8-10 are middlebox requirements that do not apply to
>     >    GRE-in-UDP tunnel endpoints (see Section 7.1 for further
>     middlebox
>     >    discussion).
>     >
>     >    It is worth mentioning that the use of a zero UDP checksum should
>     >    present the equivalent risk of undetected packet corruption when
>     >    sending similar packet using GRE-in-IPv6 without UDP
>     [RFC7676] and
>     >    without GRE checksums.
>     >
>     >    In summary, a TMCE GRE-in-UDP Tunnel is allowed to use UDP-zero-
>     >    checksum mode for IPv6 when the conditions and requirements
>     stated
>     >    above are met. Otherwise the UDP checksum need to be used for
>     IPv6
>     >    as specified in [RFC768] and [RFC2460]. Use of GRE checksum is
>     >    RECOMMENED when the UDP checksum is not used.
>     > "
>     >
>     > In GUE, to support UDP-checksum-zero, it said
>     >
>     > " Therefore, when GUE is used over
>     >    IPv6, either the UDP checksum must be enabled or the GUE header
>     >    checksum must be used.  An encapsulator MAY set a zero UDP
>     checksum
>     >    for performance or implementation reasons, in which case the GUE
>     >    header checksum MUST be used or applicable requirements for using
>     >    zero UDP checksums in [GREUDP] MUST be met. If the UDP
>     checksum is
>     >    enabled, then the GUE header checksum should not be used
>     since it is
>     >    mostly redundant."
>     >
>     > It's easy for me to add the similar words to the IP-in-UDP draft
>     like "the
>     > applicable requirements for using zero UDP checksum in [GREUDP]
>     MUST be
>     > met when zero UDP checksum is used by the tunnel ingress".
>     However, the
>     > major goal for disabling the UDP checksum is to improve the
>     performance.
>     > When GUE header checksum is used and/or the bunch of applicable
>     > requirements as described in GRE/UDP are verified, is the goal
>     of improving
>     > performance still achievable? If not, why not directly enable
>     the UDP-checksum
>     > instead?
>     >
>     >     - fragmentation support
>     >
>     > In GRE/UDP, it said
>     >
>     > " 4.1. MTU and Fragmentation
>     >
>     >    Regarding packet fragmentation, an encapsulator/decapsulator
>     SHOULD
>     >    be compliant with [RFC7588] and perform fragmentation before the
>     >    encapsulation. The size of fragments SHOULD be less or equal
>     to the
>     >    PMTU associated with the path between the GRE ingress and the GRE
>     >    egress tunnel endpoints minus the GRE and UDP overhead ..."
>     >
>     > in GUE, it said
>     >
>     > " 4.9. MTU and fragmentation
>     >
>     >    Standard conventions for handling of MTU (Maximum
>     Transmission Unit)
>     >    and fragmentation in conjunction with networking tunnels
>     >    (encapsulation of layer 2 or layer 3 packets) should be followed.
>     >    Details are described in MTU and Fragmentation Issues with
>     In-the-
>     >    Network Tunneling [RFC4459]... "
>     >
>     > It seems that the only missing thing in the IP-in-UDP draft is
>     to allow the outer
>     > fragmentation. However, as it said in
>     >
>     (https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02#page-13),
>     " ...IPsec
>     > performs only Outer Fragmentation; this distinguishes it from
>     IP-in-IP, which
>     > performs only Inner Fragmentation. " Note that IP-in-IP is the
>     dominant
>     > encapsulation choice within Softwires networks. In other words,
>     performing
>     > only inner fragmentation works very well in practice.
>     Furthermore, the outer
>     > fragmentation issue (e.g., reassembly cost for the egress) would
>     become even
>     > worse since the fragments of X-in-UDP packets are more likely to
>     be forwarded
>     > across different paths than those of X-in-IP and X-in-GRE
>     packets. Hence, I'm
>     > wondering whether it's worthwhile to support the outer
>     fragmentation on UDP
>     > encapsulated packets which seems useless in practice.
>     >
>     >     - signalling support (e.g., to test whether a tunnel is up or
>     >     to measure MTUs)
>     >
>     > I haven't found any description of this in both GRE/UDP and GUE.
>     Did you?
>     >
>     >     - support for robust ID fields (related to fragmentation,
>     >     e.g., to overcome the limits of IPv4 ID as per RFC 6864)
>     >
>     > I haven't found any description of this in both GRE/UDP and GUE.
>     Did you?
>     >
>     > Xiaohu
>     >
>     > > It is not our obligation to find a way for your document to
>     proceed -
>     > > that onus is on you.
>     > >
>     > > Joe
>


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    <div class="moz-cite-prefix">On 6/11/2016 10:11 PM, Lloyd Wood
      wrote:<br>
    </div>
    <blockquote
      cite="mid:974977917.2492241.1465708308527.JavaMail.yahoo@mail.yahoo.com"
      type="cite"><!--[if gte mso 9]><xml><o:OfficeDocumentSettings><o:AllowPNG/><o:PixelsPerInch>96</o:PixelsPerInch></o:OfficeDocumentSettings></xml><![endif]-->
      <style type="text/css" scoped=""> blockquote, div.yahoo_quoted { margin-left: 0 !important; border-left:1px #715FFA solid !important; padding-left:1ex !important; background-color:white !important; } </style>
      Fragmentation should be strongly discouraged.</blockquote>
    <br>
    Avoided where possible for sure.<br>
    <blockquote
      cite="mid:974977917.2492241.1465708308527.JavaMail.yahoo@mail.yahoo.com"
      type="cite">
      <div><br>
      </div>
      <div>if you've designed a tunnelling solution and you have
        fragmentation happening as a matter of course, you've designed
        it wrong. <br>
      </div>
    </blockquote>
    <br>
    If you have a solution that recurses (X over X, with some
    intervening layers or not), and X has a required minimum message
    size, then you either have a solution that - in at least some
    non-trivial situations - will fragment as a matter of course or a
    fantasy that cannot exist.<br>
    <br>
    The only reason we avoid this in typical operation is by avoiding X
    over X. Once you have a tunnel, that might not be avoidable (or even
    detectable, depending on encryption).<br>
    <br>
    Joe<br>
    <br>
    <blockquote
      cite="mid:974977917.2492241.1465708308527.JavaMail.yahoo@mail.yahoo.com"
      type="cite">
      <div><br>
        Lloyd Wood
        <div><a class="moz-txt-link-abbreviated" href="mailto:lloyd.wood@yahoo.co.uk">lloyd.wood@yahoo.co.uk</a></div>
        <br>
        <p class="yahoo-quoted-begin" style="font-size: 15px; color:
          #715FFA; padding-top: 15px; margin-top: 0">On Friday, May 27,
          2016, 8:10 PM, Xuxiaohu <a class="moz-txt-link-rfc2396E" href="mailto:xuxiaohu@huawei.com">&lt;xuxiaohu@huawei.com&gt;</a> wrote:</p>
        <blockquote class="iosymail">
          <div id="msgSandbox_AMhLyAoAAG2BnV0gc2BwILcOkyXZc_TEXT"
            class="msgSandbox" style="padding: 1.5em 0.5em 0.5em 1.2em;
            word-wrap: break-word;">&lt;Note that I have changed the
            subject of the email hence it has nothing to do with the WG
            adoption call now. It's just a discussion on a particular
            issue which is related to those WGs which are working on UDP
            tunnels. The reason for containing the old email is to use
            it as a background which may be useful for better
            understanding of this particular issue&gt;<br>
            <br>
            The possible side-effect of performing fragmentation on UDP
            encapsulated packets is to worsen the reassembly burden on
            tunnel egress since fragments of UDP encapsulated packets
            are more likely to be forwarded across different paths
            towards the tunnel egress than those of IP or GRE
            encapsulated packets.<br>
            <br>
            It seems that most X-over-UDP proposals choose to prohibit
            the tunnel ingress from performing fragmentation on UDP
            encapsulated packets. See the following quoted text
            regarding fragmentation from those X-over-UDP drafts:<br>
            <br>
            LISP:<br>
            <br>
            When an ITR receives a packet from a site-facing interface
            and adds H<br>
              octets worth of encapsulation to yield a packet size
            greater than L<br>
              octets, it resolves the MTU issue by first splitting the
            original<br>
              packet into 2 equal-sized fragments.  A LISP header is
            then prepended<br>
              to each fragment.<br>
            <br>
            VXLAN:<br>
            <br>
            VTEPs MUST NOT fragment VXLAN packets.  Intermediate routers
            may<br>
              fragment encapsulated VXLAN packets due to the larger
            frame size.<br>
              The destination VTEP MAY silently discard such VXLAN
            fragments.<br>
            <br>
            VXLAN-GPE:<br>
            <br>
            VTEPs MUST never fragment an encapsulated VXLAN GPE packet,
            and when<br>
              the outer IP header is IPv4, VTEPs MUST set the DF bit in
            the outer<br>
              IPv4 header.<br>
            <br>
            GEVENE:<br>
            <br>
              To prevent fragmentation and maximize performance, the
            best practice<br>
              when using Geneve is to ensure that the MTU of the
            physical network<br>
              is greater than or equal to the MTU of the encapsulated
            network plus<br>
              tunnel headers.<br>
            <br>
            GUE:<br>
            <br>
                If a packet is fragmented before encapsulation in GUE,
            all the<br>
                related fragments must be encapsulated using the same
            source port<br>
                (inner flow identifier). An operator may set MTU to
            account for<br>
                encapsulation overhead and reduce the likelihood of
            fragmentation.<br>
            <br>
            GRE/UDP<br>
            <br>
            Regarding packet fragmentation, an encapsulator/decapsulator
            SHOULD<br>
              be compliant with [RFC7588] and perform fragmentation
            before the<br>
              encapsulation.<br>
            <br>
            However, the above choice seems conflict with the
            requirements as described in <a moz-do-not-send="true"
              href="https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02"
              target="_blank">https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02</a><br>
            <br>
            <br>
            I wonder whether the IETF should reach a consensus on
            whether or not the fragmentation on UDP encapsulated packets
            should be allowed.<br>
              <br>
            Best regards,<br>
            Xiaohu<br>
            <br>
            &gt; -----Original Message-----<br>
            &gt; From: Xuxiaohu<br>
            &gt; Sent: Thursday, May 26, 2016 4:35 PM<br>
            &gt; To: 'Joe Touch'; Fred Baker (fred); Wassim Haddad<br>
            &gt; Cc: <a moz-do-not-send="true"
              ymailto="mailto:int-area@ietf.org"
              href="javascript:return">int-area@ietf.org</a><br>
            &gt; Subject: RE: [Int-area] Call for adoption of
            draft-xu-intarea-ip-in-udp-03<br>
            &gt; <br>
            &gt; <br>
            &gt; <br>
            &gt; &gt; -----Original Message-----<br>
            &gt; &gt; From: Joe Touch [mailto:<a moz-do-not-send="true"
              ymailto="mailto:touch@isi.edu" href="javascript:return">touch@isi.edu</a>]<br>
            &gt; &gt; Sent: Thursday, May 26, 2016 2:11 AM<br>
            &gt; &gt; To: Xuxiaohu; Fred Baker (fred); Wassim Haddad<br>
            &gt; &gt; Cc: <a moz-do-not-send="true"
              ymailto="mailto:int-area@ietf.org"
              href="javascript:return">int-area@ietf.org</a><br>
            &gt; &gt; Subject: Re: [Int-area] Call for adoption of<br>
            &gt; &gt; draft-xu-intarea-ip-in-udp-03<br>
            &gt; &gt;<br>
            &gt; &gt;<br>
            &gt; &gt;<br>
            &gt; &gt; On 5/24/2016 7:24 PM, Xuxiaohu wrote:<br>
            &gt; &gt; &gt; Hi Joe,<br>
            &gt; &gt; &gt;<br>
            &gt; &gt; &gt; I wonder whether you want to tell me the
            following truth by the<br>
            &gt; &gt; &gt; example that you gave: no matter whatever
            improvements we had done<br>
            &gt; &gt; &gt; with this draft, those persons who dislike it
            by the light of nature<br>
            &gt; &gt; &gt; would dislike it in the end.<br>
            &gt; &gt;<br>
            &gt; &gt; The only improvements that would make this doc
            useful would be to add<br>
            &gt; &gt; capabilities already in GRE/UDP or GUE/UDP, which
            we already have.<br>
            &gt; <br>
            &gt; Let's go over the four things you mentioned earlier in
            GRE/UDP and GUE/UDP:<br>
            &gt; <br>
            &gt;     - stronger checksums<br>
            &gt; <br>
            &gt; In GRE/UDP, in order to use UDP-zero-checksum, it gave
            the following<br>
            &gt; restrictions:<br>
            &gt; " 6. UDP Checksum Handling<br>
            &gt; <br>
            &gt;    6.1. UDP Checksum with IPv4<br>
            &gt; <br>
            &gt;    For UDP in IPv4, the UDP checksum MUST be processed
            as specified in<br>
            &gt;    [RFC768] and [RFC1122] for both transmit and
            receive. The IPv4<br>
            &gt; <br>
            &gt; <br>
            &gt; <br>
            &gt; Yong, Crabber, Xu, Herbert                             
                  [Page 12]<br>
            &gt;
--------------------------------------------------------------------------------<br>
            &gt; <br>
            &gt; Internet-Draft          GRE-in-UDP Encapsulation       
                March 2016<br>
            &gt; <br>
            &gt;    header includes a checksum which protects against
            mis-delivery of<br>
            &gt;    the packet due to corruption of IP addresses. The
            UDP checksum<br>
            &gt;    potentially provides protection against corruption
            of the UDP header,<br>
            &gt;    GRE header, and GRE payload. Disabling the use of
            checksums is a<br>
            &gt;    deployment consideration that should take into
            account the risk and<br>
            &gt;    effects of packet corruption.<br>
            &gt; <br>
            &gt;    When a decapsulator receives a packet, the UDP
            checksum field MUST<br>
            &gt;    be processed. If the UDP checksum is non-zero, the
            decapsulator MUST<br>
            &gt;    verify the checksum before accepting the packet. By
            default a<br>
            &gt;    decapsulator SHOULD accept UDP packets with a zero
            checksum. A node<br>
            &gt;    MAY be configured to disallow zero checksums per
            [RFC1122]; this may<br>
            &gt;    be done selectively, for instance disallowing zero
            checksums from<br>
            &gt;    certain hosts that are known to be sending over
            paths subject to<br>
            &gt;    packet corruption. If verification of a non-zero
            checksum fails, a<br>
            &gt;    decapsulator lacks the capability to verify a
            non-zero checksum, or<br>
            &gt;    a packet with a zero-checksum was received and the
            decapsulator is<br>
            &gt;    configured to disallow, the packet MUST be dropped
            and an event MAY<br>
            &gt;    be logged.<br>
            &gt; <br>
            &gt;    6.2. UDP Checksum with IPv6<br>
            &gt; <br>
            &gt;    For UDP in IPv6, the UDP checksum MUST be processed
            as specified in<br>
            &gt;    [RFC768] and [RFC2460] for both transmit and
            receive.<br>
            &gt; <br>
            &gt;    When UDP is used over IPv6, the UDP checksum is
            relied upon to<br>
            &gt;    protect both the IPv6 and UDP headers from
            corruption. As such, A<br>
            &gt;    default GRE-in-UDP Tunnel MUST perform UDP checksum;
            A TMCE GRE-in-<br>
            &gt;    UDP Tunnel MAY be configured with the UDP
            zero-checksum mode if the<br>
            &gt;    traffic-managed controlled environment or a set of
            closely<br>
            &gt;    cooperating traffic-managed controlled environments
            (such as by<br>
            &gt;    network operators who have agreed to work together
            in order to<br>
            &gt;    jointly provide specific services) meet at least one
            of following<br>
            &gt;    conditions:<br>
            &gt; <br>
            &gt;    a. It is known (perhaps through knowledge of
            equipment types and<br>
            &gt;      lower layer checks) that packet corruption is
            exceptionally<br>
            &gt;      unlikely and where the operator is willing to take
            the risk of<br>
            &gt;      undetected packet corruption.<br>
            &gt; <br>
            &gt;    b. It is judged through observational measurements
            (perhaps of<br>
            &gt;      historic or current traffic flows that use a
            non-zero checksum)<br>
            &gt;      that the level of packet corruption is tolerably
            low and where<br>
            &gt;      the operator is willing to take the risk of
            undetected packet<br>
            &gt;      corruption.<br>
            &gt; <br>
            &gt; <br>
            &gt; <br>
            &gt; <br>
            &gt; <br>
            &gt; Yong, Crabber, Xu, Herbert                             
                  [Page 13]<br>
            &gt;
--------------------------------------------------------------------------------<br>
            &gt; <br>
            &gt; Internet-Draft          GRE-in-UDP Encapsulation       
                March 2016<br>
            &gt; <br>
            &gt;    c. Carrying applications that are tolerant of
            mis-delivered or<br>
            &gt;      corrupted packets (perhaps through higher layer
            checksum,<br>
            &gt;      validation, and retransmission or transmission
            redundancy) where<br>
            &gt;      the operator is willing to rely on the
            applications using the<br>
            &gt;      tunnel to survive any corrupt packets.<br>
            &gt; <br>
            &gt;    The following requirements apply to a TMCE
            GRE-in-UDP tunnel that<br>
            &gt;    use UDP zero-checksum mode:<br>
            &gt; <br>
            &gt;      a. Use of the UDP checksum with IPv6 MUST be the
            default<br>
            &gt;        configuration of all GRE-in-UDP tunnels.<br>
            &gt; <br>
            &gt;      b. The GRE-in-UDP tunnel implementation MUST
            comply with all<br>
            &gt;        requirements specified in Section 4 of [RFC6936]
            and with<br>
            &gt;        requirement 1 specified in Section 5 of
            [RFC6936].<br>
            &gt; <br>
            &gt;      c. The tunnel decapsulator SHOULD only allow the
            use of UDP zero-<br>
            &gt;        checksum mode for IPv6 on a single received UDP
            Destination<br>
            &gt;        Port regardless of the encapsulator. The
            motivation for this<br>
            &gt;        requirement is possible corruption of the UDP
            Destination Port,<br>
            &gt;        which may cause packet delivery to the wrong UDP
            port. If that<br>
            &gt;        other UDP port requires the UDP checksum, the
            mis-delivered<br>
            &gt;        packet will be discarded.<br>
            &gt; <br>
            &gt;      d. It is RECOMMENDED that the UDP zero-checksum
            mode for IPv6 is<br>
            &gt;        only enabled for certain selected source
            addresses. The tunnel<br>
            &gt;        decapsulator MUST check that the source and
            destination IPv6<br>
            &gt;        addresses are valid for the GRE-in-UDP tunnel on
            which the<br>
            &gt;        packet was received if that tunnel uses UDP
            zero-checksum mode<br>
            &gt;        and discard any packet for which this check
            fails.<br>
            &gt; <br>
            &gt;      e. The tunnel encapsulator SHOULD use different
            IPv6 addresses for<br>
            &gt;        each GRE-in-UDP tunnel that uses UDP
            zero-checksum mode<br>
            &gt;        regardless of the decapsulator in order to
            strengthen the<br>
            &gt;        decapsulator's check of the IPv6 source address
            (i.e., the same<br>
            &gt;        IPv6 source address SHOULD NOT be used with more
            than one IPv6<br>
            &gt;        destination address, independent of whether that
            destination<br>
            &gt;        address is a unicast or multicast address). When
            this is not<br>
            &gt;        possible, it is RECOMMENDED to use each source
            IPv6 address for<br>
            &gt;        as few UDP zero-checksum mode GRE-in-UDP tunnels
            as is feasible.<br>
            &gt; <br>
            &gt;      f. When any middlebox exists on the path of a
            GRE-in-UDP tunnel,<br>
            &gt;        it is RECOMMENDED to use the default mode, i.e.
            use UDP<br>
            &gt;        checksum, to reduce the chance that the
            encapsulated packets to<br>
            &gt;        be dropped.<br>
            &gt; <br>
            &gt; <br>
            &gt; <br>
            &gt; <br>
            &gt; <br>
            &gt; Yong, Crabber, Xu, Herbert                             
                  [Page 14]<br>
            &gt;
--------------------------------------------------------------------------------<br>
            &gt; <br>
            &gt; Internet-Draft          GRE-in-UDP Encapsulation       
                March 2016<br>
            &gt; <br>
            &gt;      g. Any middlebox that allows the UDP zero-checksum
            mode for IPv6<br>
            &gt;        MUST comply with requirement 1 and 8-10 in
            Section 5 of<br>
            &gt;        [RFC6936].<br>
            &gt; <br>
            &gt;      h. Measures SHOULD be taken to prevent IPv6
            traffic with zero UDP<br>
            &gt;        checksums from "escaping" to the general
            Internet; see Section<br>
            &gt;        8 for examples of such measures.<br>
            &gt; <br>
            &gt;      i. IPv6 traffic with zero UDP checksums MUST be
            actively monitored<br>
            &gt;        for errors by the network operator. For example,
            the operator<br>
            &gt;        may monitor Ethernet layer packet error rates.<br>
            &gt; <br>
            &gt;      j. If a packet with a non-zero checksum is
            received, the checksum<br>
            &gt;        MUST be verified before accepting the packet.
            This is<br>
            &gt;        regardless of whether the tunnel encapsulator
            and decapsulator<br>
            &gt;        have been configured with UDP zero-checksum
            mode.<br>
            &gt; <br>
            &gt;    The above requirements do not change either the
            requirements<br>
            &gt;    specified in [RFC2460] as modified by [RFC6935] or
            the requirements<br>
            &gt;    specified in [RFC6936].<br>
            &gt; <br>
            &gt;    The requirement to check the source IPv6 address in
            addition to the<br>
            &gt;    destination IPv6 address, plus the strong
            recommendation against<br>
            &gt;    reuse of source IPv6 addresses among GRE-in-UDP
            tunnels collectively<br>
            &gt;    provide some mitigation for the absence of UDP
            checksum coverage of<br>
            &gt;    the IPv6 header. A traffic-managed controlled
            environment that<br>
            &gt;    satisfies at least one of three conditions listed
            above in this<br>
            &gt;    section provides additional assurance.<br>
            &gt; <br>
            &gt;    A GRE-in-UDP tunnel is suitable for transmission
            over lower layers<br>
            &gt;    in the traffic-managed controlled environments that
            are allowed by<br>
            &gt;    the exceptions stated above and the rate of
            corruption of the inner<br>
            &gt;    IP packet on such networks is not expected to
            increase by comparison<br>
            &gt;    to GRE traffic that is not encapsulated in UDP.  For
            these reasons,<br>
            &gt;    GRE-in-UDP does not provide an additional integrity
            check except<br>
            &gt;    when GRE checksum is used when UDP zero-checksum
            mode is used with<br>
            &gt;    IPv6, and this design is in accordance with
            requirements 2, 3 and 5<br>
            &gt;    specified in Section 5 of [RFC6936].<br>
            &gt; <br>
            &gt;    Generic Router Encapsulation (GRE) does not
            accumulate incorrect<br>
            &gt;    state as a consequence of GRE header corruption. A
            corrupt GRE<br>
            &gt;    packet may result in either packet discard or
            forwarding of the<br>
            &gt;    packet without accumulation of GRE state. Active
            monitoring of GRE-<br>
            &gt;    in-UDP traffic for errors is REQUIRED as occurrence
            of errors will<br>
            &gt;    result in some accumulation of error information
            outside the<br>
            &gt;    protocol for operational and management purposes.
            This design is in<br>
            &gt;    accordance with requirement 4 specified in Section 5
            of [RFC6936].<br>
            &gt; <br>
            &gt; <br>
            &gt; <br>
            &gt; Yong, Crabber, Xu, Herbert                             
                  [Page 15]<br>
            &gt;
--------------------------------------------------------------------------------<br>
            &gt; <br>
            &gt; Internet-Draft          GRE-in-UDP Encapsulation       
                March 2016<br>
            &gt; <br>
            &gt;    The remaining requirements specified in Section 5 of
            [RFC6936] are<br>
            &gt;    not applicable to GRE-in-UDP.  Requirements 6 and 7
            do not apply<br>
            &gt;    because GRE does not include a control feedback
            mechanism.<br>
            &gt;    Requirements 8-10 are middlebox requirements that do
            not apply to<br>
            &gt;    GRE-in-UDP tunnel endpoints (see Section 7.1 for
            further middlebox<br>
            &gt;    discussion).<br>
            &gt; <br>
            &gt;    It is worth mentioning that the use of a zero UDP
            checksum should<br>
            &gt;    present the equivalent risk of undetected packet
            corruption when<br>
            &gt;    sending similar packet using GRE-in-IPv6 without UDP
            [RFC7676] and<br>
            &gt;    without GRE checksums.<br>
            &gt; <br>
            &gt;    In summary, a TMCE GRE-in-UDP Tunnel is allowed to
            use UDP-zero-<br>
            &gt;    checksum mode for IPv6 when the conditions and
            requirements stated<br>
            &gt;    above are met. Otherwise the UDP checksum need to be
            used for IPv6<br>
            &gt;    as specified in [RFC768] and [RFC2460]. Use of GRE
            checksum is<br>
            &gt;    RECOMMENED when the UDP checksum is not used.<br>
            &gt; "<br>
            &gt; <br>
            &gt; In GUE, to support UDP-checksum-zero, it said<br>
            &gt; <br>
            &gt; " Therefore, when GUE is used over<br>
            &gt;    IPv6, either the UDP checksum must be enabled or the
            GUE header<br>
            &gt;    checksum must be used.  An encapsulator MAY set a
            zero UDP checksum<br>
            &gt;    for performance or implementation reasons, in which
            case the GUE<br>
            &gt;    header checksum MUST be used or applicable
            requirements for using<br>
            &gt;    zero UDP checksums in [GREUDP] MUST be met. If the
            UDP checksum is<br>
            &gt;    enabled, then the GUE header checksum should not be
            used since it is<br>
            &gt;    mostly redundant."<br>
            &gt; <br>
            &gt; It's easy for me to add the similar words to the
            IP-in-UDP draft like "the<br>
            &gt; applicable requirements for using zero UDP checksum in
            [GREUDP] MUST be<br>
            &gt; met when zero UDP checksum is used by the tunnel
            ingress". However, the<br>
            &gt; major goal for disabling the UDP checksum is to improve
            the performance.<br>
            &gt; When GUE header checksum is used and/or the bunch of
            applicable<br>
            &gt; requirements as described in GRE/UDP are verified, is
            the goal of improving<br>
            &gt; performance still achievable? If not, why not directly
            enable the UDP-checksum<br>
            &gt; instead?<br>
            &gt; <br>
            &gt;     - fragmentation support<br>
            &gt; <br>
            &gt; In GRE/UDP, it said<br>
            &gt; <br>
            &gt; " 4.1. MTU and Fragmentation<br>
            &gt; <br>
            &gt;    Regarding packet fragmentation, an
            encapsulator/decapsulator SHOULD<br>
            &gt;    be compliant with [RFC7588] and perform
            fragmentation before the<br>
            &gt;    encapsulation. The size of fragments SHOULD be less
            or equal to the<br>
            &gt;    PMTU associated with the path between the GRE
            ingress and the GRE<br>
            &gt;    egress tunnel endpoints minus the GRE and UDP
            overhead ..."<br>
            &gt; <br>
            &gt; in GUE, it said<br>
            &gt; <br>
            &gt; " 4.9. MTU and fragmentation<br>
            &gt; <br>
            &gt;    Standard conventions for handling of MTU (Maximum
            Transmission Unit)<br>
            &gt;    and fragmentation in conjunction with networking
            tunnels<br>
            &gt;    (encapsulation of layer 2 or layer 3 packets) should
            be followed.<br>
            &gt;    Details are described in MTU and Fragmentation
            Issues with In-the-<br>
            &gt;    Network Tunneling [RFC4459]... "<br>
            &gt; <br>
            &gt; It seems that the only missing thing in the IP-in-UDP
            draft is to allow the outer<br>
            &gt; fragmentation. However, as it said in<br>
            &gt; (<a moz-do-not-send="true"
              href="https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02#page-13"
              target="_blank">https://tools.ietf.org/html/draft-ietf-intarea-tunnels-02#page-13</a>),
            " ...IPsec<br>
            &gt; performs only Outer Fragmentation; this distinguishes
            it from IP-in-IP, which<br>
            &gt; performs only Inner Fragmentation. " Note that IP-in-IP
            is the dominant<br>
            &gt; encapsulation choice within Softwires networks. In
            other words, performing<br>
            &gt; only inner fragmentation works very well in practice.
            Furthermore, the outer<br>
            &gt; fragmentation issue (e.g., reassembly cost for the
            egress) would become even<br>
            &gt; worse since the fragments of X-in-UDP packets are more
            likely to be forwarded<br>
            &gt; across different paths than those of X-in-IP and
            X-in-GRE packets. Hence, I'm<br>
            &gt; wondering whether it's worthwhile to support the outer
            fragmentation on UDP<br>
            &gt; encapsulated packets which seems useless in practice.<br>
            &gt; <br>
            &gt;     - signalling support (e.g., to test whether a
            tunnel is up or<br>
            &gt;     to measure MTUs)<br>
            &gt; <br>
            &gt; I haven't found any description of this in both GRE/UDP
            and GUE. Did you?<br>
            &gt; <br>
            &gt;     - support for robust ID fields (related to
            fragmentation,<br>
            &gt;     e.g., to overcome the limits of IPv4 ID as per RFC
            6864)<br>
            &gt; <br>
            &gt; I haven't found any description of this in both GRE/UDP
            and GUE. Did you?<br>
            &gt; <br>
            &gt; Xiaohu<br>
            &gt; <br>
            &gt; &gt; It is not our obligation to find a way for your
            document to proceed -<br>
            &gt; &gt; that onus is on you.<br>
            &gt; &gt;<br>
            &gt; &gt; Joe<br>
          </div>
          <blockquote> </blockquote>
        </blockquote>
      </div>
    </blockquote>
    <br>
  </body>
</html>

--------------030502040200010703010909--


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From: "Black, David" <david.black@emc.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>, "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: draft-ietf-tcpm-rto-consider-04: (R.3)
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First off, I want to thank Mark for all the work invested in the -04 versio=
n of
this draft - in particular, the new scope statements (S.1 - S.6) in Section=
 2
are important clarifications of what this draft does and what it does not d=
o,
so I'm pleased to see them, thank you.

As Mark noted, there have been some extensive offline conversations.
The purpose of this email is to surface one focal point of those conversati=
ons
so that we can get to and record some form of agreement on the public lists=
.

This concerns (R.3) in Section 2 of the -04:

    (R.3) Alternatively, future RTO mechanism implementations may be
          made directly against the requirements in Section 3 without
          another protocol-specific specification.

The focal point was ensuring that (R.3) does not conflict with (R.1) earlie=
r in
Section 2:

    (R.1) RTO mechanisms that are currently standardized are not updated=20
          or obsoleted by this document.  Implementations are free to
          use these existing specifications as they do now.

IMHO, all the participants in the offline discussion agree that the first s=
entence
of (R.1) is 100% correct as stated (e.g., as this draft does not propose to
update any RFCs).  In 20/20 hindsight, the offline discussion was mostly
about word choice and order in (R.3).

I think (R.3) is ok as is in -04, but it contains a small ambiguity (does "=
future"
modify "RTO mechanism" or "implementations"?) that would benefit from a
small editorial clarification to make it clear that "future" modifies "RTO
mechanisms":

    (R.3) Alternatively, implementations of future RTO mechanisms may be
          made directly against the requirements in Section 3 without
          another protocol-specific specification.

Thanks, --David
----------------------------------------------------
David L. Black, Distinguished Engineer
EMC, 176 South St., Hopkinton, MA=A0 01748
+1 (508) 293-7953=A0=A0=A0  Cell: +1 (978) 394-7754
david.black@emc.com=A0=A0=A0=A0=A0=A0=A0=20
----------------------------------------------------



From nobody Thu Jun 16 05:06:58 2016
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>     (R.3) Alternatively, implementations of future RTO mechanisms may be
>           made directly against the requirements in Section 3 without
>           another protocol-specific specification.

I am fine with this wording and have made a note to use it in -05
(modulo any further comments that roll in, of course).

Thanks,
allman


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




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From nobody Thu Jun 16 15:25:59 2016
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Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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Black, David <david.black@emc.com> wrote:
> 
> ...  I view the current text as providing implementers with too much
> latitude to ignore ECN-CE marks (e.g., because an implementer doesn't
> want to think about this problem space in the first place).

   Understand, we have at least two proposals to make ECN-CE more frequent
than packet drop would be for non-ECN packets: possibly substantially
more frequent. Unless both are killed off, ECN-CE will show up frequently
enough that closing the flow on ECN-CE would kill too many connections.

   If you want circuit-breaking on such connections, there are two ways:
1. convince the forwarding nodes to drop packets if their queue exceeds
   design capacity; or
2. require the sender to send enough not-ECN-capable packets so that our
   receiver will see enough packet-drops when a circuit-breaker should
   activate.

   (I prefer the first option; but I wouldn't object to the second.)

   There really isn't any way for our circuit-breaker to know _how_much_
more frequent the ECN-CE marks may be. :^( We _will_ be sorry if we
allot the same frequency of CE packets as packet-drops to trigger the
circuit-breaker.

> Could someone propose initial text to qualifies the current "MAY ignore"
> statement?

   Essentially, for the second option, you might propose text to the
effect of:
] 
] If too many ECN-CE packets are received, the sender SHOULD send some
] not-ECN-capable packets to determine whether enough packets along the
] path are being dropped to justify activating our circuit-breaker.

   I'm not enthusiastic about adding that; but it would resolve the issue.

   BTW, I'm 100% convinced that either of the proposals being considered
in TSVWG could bring _substantial_ benefit to RTCWEB traffic.

--
John Leslie <john@jlc.net>


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Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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> On 16 Jun 2016, at 23:25, John Leslie <john@jlc.net> wrote:
>=20
> Black, David <david.black@emc.com> wrote:
>>=20
>> ...  I view the current text as providing implementers with too much
>> latitude to ignore ECN-CE marks (e.g., because an implementer doesn't
>> want to think about this problem space in the first place).

I agree, but the argument is that doing so is less harmful than =
deploying a circuit breaker that triggers too often when ECN is used.=20

I=E2=80=99m not sure I believe this argument, though, since it seems =
that any new AQM that applies ECN marks much more often than at present =
will have to consider backwards compatibility, to work with deployed TCP =
(e.g., draft-briscoe-tsvwg-aqm-tcpm-rmcat-l4s-problem uses ECT(1) as a =
signal to use the new marking, while existing implementations set =
ECT(0)). These compatibility mechanisms would seem to prevent the issues =
with the circuit breaker too.

>   Understand, we have at least two proposals to make ECN-CE more =
frequent
> than packet drop would be for non-ECN packets: possibly substantially
> more frequent. Unless both are killed off, ECN-CE will show up =
frequently
> enough that closing the flow on ECN-CE would kill too many =
connections.
>=20
>   If you want circuit-breaking on such connections, there are two =
ways:
> 1. convince the forwarding nodes to drop packets if their queue =
exceeds
>   design capacity; or
> 2. require the sender to send enough not-ECN-capable packets so that =
our
>   receiver will see enough packet-drops when a circuit-breaker should
>   activate.
>=20
>   (I prefer the first option; but I wouldn't object to the second.)
>=20
>   There really isn't any way for our circuit-breaker to know =
_how_much_
> more frequent the ECN-CE marks may be. :^(

This is a problem, both for the circuit breaker, and for the algorithms =
being defined in RMCAT. We do need some understanding what the expected =
marking rates are likely to be, so congestion control and circuit =
breakers can be defined.=20

> We _will_ be sorry if we
> allot the same frequency of CE packets as packet-drops to trigger the
> circuit-breaker.
>=20
>> Could someone propose initial text to qualifies the current "MAY =
ignore"
>> statement?
>=20
>   Essentially, for the second option, you might propose text to the
> effect of:
> ]=20
> ] If too many ECN-CE packets are received, the sender SHOULD send some
> ] not-ECN-capable packets to determine whether enough packets along =
the
> ] path are being dropped to justify activating our circuit-breaker.
>=20
>   I=E2=80=99m not enthusiastic about adding that; but it would resolve =
the issue.

I=E2=80=99m not convinced this would work. The circuit breaker is =
looking at long term trends, and in order to have enough not-ECT packets =
to determine if it should trigger, you=E2=80=99d essentially have to run =
without ECN for some seconds.=20

--=20
Colin Perkins
https://csperkins.org/





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Subject: Re: [tsvwg] [tcpm] draft-ietf-tcpm-rto-consider-04: (R.3)
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Hi Mark,

witting as an individual contributor.=20

First of all, thanks for the update! I think it reads much better now. =
There are 2-3 MUST and SHOULD which we could further discuss, but I =
don=E2=80=99t think that is anything big and I=E2=80=99ll defer this to =
later.

I have a personal opinion on the point below (R-3). I think this point =
is just not needed. I say this not because I don=E2=80=99t agree with =
the statement but spelling this out in an RFC, to me, contradict the =
point of the RFC. I know this point is important for you but let me =
further explain.=20

To me I don=E2=80=99t have a problem if people do implement something =
that is not standard conform. Especially in transport where there are a =
lot of things that don=E2=80=99t have interoperability problem, to me =
this is fully okay. Sometimes we in the IETF even pick up existing =
implementation and document them as informational, because we believe =
the implementation is good and having a stable reference will help =
others to do the right thing.=20

However, when we make a recommend I think we should be clear. This is =
especially helpful for implementators who do not really have a good =
reason to do something different and just look for a general =
recommendation.=20

Further I have to note that we do use the word SHOULD in cases where we =
know that there are good reason to deviate from the proposed =
recommendation. In such a case you could still claim to be standard =
conform (for what it is worth) even when you don=E2=80=99t follow the =
SHOULD.

However basically saying whatever you do is confirm with this =
specification, to me really contradicts the point of making a =
recommendation.

It=E2=80=99s simple if you follow all MUSTs and SHOULDs if case you =
don=E2=80=99t have a good reason to do something else, you are conform =
to the RFC, and otherwise you are simply not conform. That doesn=E2=80=99t=
 say that you always when you implement something have to follow RFCs. =
That=E2=80=99s non-sense. Implementors are free to do what they want and =
we can=E2=80=99t stop them; we can only give reasonable recommendations.

My 2c...

Mirja


> Am 16.06.2016 um 14:07 schrieb Mark Allman <mallman@icir.org>:
>=20
>=20
>>    (R.3) Alternatively, implementations of future RTO mechanisms may =
be
>>          made directly against the requirements in Section 3 without
>>          another protocol-specific specification.
>=20
> I am fine with this wording and have made a note to use it in -05
> (modulo any further comments that roll in, of course).
>=20
> Thanks,
> allman
>=20
>=20
> --
> http://www.icir.org/mallman/
>=20
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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From: "Eggert, Lars" <lars@netapp.com>
To: Takeshi Takahashi <takeshi_takahashi@nict.go.jp>
Thread-Topic: Secdir review of draft-ietf-tsvwg-rfc5405bis-13
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Hi,

thanks for the review! I'll incorporate respective changes into -14.

On 2016-05-31, at 12:06, Takeshi Takahashi =
<takeshi_takahashi@nict.go.jp> wrote:
> In Table 1 "Summary of recommendations", I wonder if the corresponding
> section numbers are correct.
>=20
> [Now]
> "SHOULD avoid using multiple ports"  corresponds to Section 5.1
> and
> "SHOULD use a randomized source port or equivalent technique" =
corresponds to
> Section 5.2
>=20
> [New]
> "SHOULD avoid using multiple ports"  corresponds to Section 5.1.1
> and
> "SHOULD use a randomized source port or equivalent technique" =
corresponds to
> Section 5.1.2
>=20
> I might be wrong, so please check.

You were right that what we had was wrong (copy/paste bug, I think), but =
they pointers should be to Sections 5.1.2 and 6, respectively.

> Also I have seen several typos (especially, missing parentheses around
> referenced section numbers) on this document, so please revise the =
texts
> before the publication of this document.

No typos come up when using "idnits --spellcheck". I also don't quite =
understand what you mean by "missing parentheses around referenced =
section numbers" - could you please clarify?

Thanks,
Lars

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From: "Eggert, Lars" <lars@netapp.com>
To: Tim Chown <Tim.Chown@jisc.ac.uk>
Thread-Topic: OPS-DIR review of draft-ietf-tsvwg-rfc5405bis-133
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Hi,

thanks for the review! I'll incorporate respective changes into -14.

On 2016-06-12, at 0:00, Tim Chown <Tim.Chown@jisc.ac.uk> wrote:
> It might be nice to include some level of discussion of new examples =
of UDP-based protocols devised since RFC 5405 (i.e. since 2008), such as =
QUIC, or applications such as mosh that enable ssh over UDP rather than =
the traditional TCP.  Do these follow the guidance here, for example?

This would add quite a bit of digging - there is no spec for mosh, and =
while there is one for QUIC now, I'd require more time than I currently =
have to analyze if and how it follows the different guidelines we =
describe here. (I will however argue that if the QUIC WG forms, the =
eventual QUIC RFC should do so :-)

> CGNs have appeared since RFC 5405; are there specific issues they pose =
for UDP-based applications which require state to remain?

Other than being generally worse than normal NATs due to even faster =
timeouts? Not that I'm aware of.

> p.16 final para of point 1 - is most UDP-based communication =
congestion controlled?

I'd argue that most sent at volume, yes. Maybe with the exception of =
multicast, but that is limited in reach.

> p.30 when you say =E2=80=9Ccompetes fairly within an order of =
magnitude=E2=80=9D do you mean on a 10G link that UDP can take over 9G?

Yes, actually, although that's obviously the extreme, non-ideal case. We =
wanted to make it clear that there is significant leeway, because TCP =
isn't "fair" to TCP either.

> p.30 in the next paragraph, this suggestion would seem non-trivial to =
implement well/accurately, and then the =E2=80=9Corder of magnitude=E2=80=9D=
 is thrown in too=E2=80=A6

Yes, there is a lot of wiggle room here (on purpose).

> p.30 section 4.2 - perhaps reword as it implies explicit PMTUD to each =
known destination (receiver)?

I'm not sure it implies PMTUD - you'd normally have the MTU information =
available through other means in a multicast scenario.

> p.32 worth adding an example of why this is so valuable, e.g. DNS =
cache poisoning?

Which paragraph?

> p.34 perhaps note that the IPv6 Flow Label is mutable on path (section =
6 of RFC 6437)

Is this important here?

Lars

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> I have a personal opinion on the point below (R-3). I think this
> point is just not needed. I say this not because I don=E2=80=99t agree
> with the statement but spelling this out in an RFC, to me,
> contradict the point of the RFC.

On the one hand, I can understand that opinion.  Why would we write
the RFC if we weren't going to allow these things and so why do we
have to say this?  And, if people think this is unnecessary I am
happy to nuke it.

However, let me sketch the lay of the land as I see it ....   It
seems the meat in this document can be used for different purposes.
And, we have seen opinions on each of these I think.  The document
could be used:

  (a) as input to future RTO specifications,

  (b) as input to future RTO implementations, or

  (c) as both (a) and (b)

My take is (c).  And, so (R.2) is meant to cover case (a), while
(R.3) is meant to cover case (b).

So, while I can see your point about not needing to say something
obvious, I am worried that eliding (R.3) will leave us in one of two
places:

  - In case (a) above.  I.e., because (R.2) discusses only
    specifications and not implementations and so a natural reading
    may well be that this document is meant to only help future
    standardization efforts.

  - Or, in some ambiguous state where clearly (a) holds (via (R.2)),
    but it is not explicit that this covers implementations, too,
    even if we think that it does since the text does not explicitly
    say that.  And, so we end up with confusion.

Does that make sense?  Am I worrying too much about explicitly
including implementations?

allman




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From: "Eggert, Lars" <lars@netapp.com>
To: Brian Trammell <ietf@trammell.ch>
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Subject: Re: [tsvwg] Last Call: <draft-ietf-tsvwg-rfc5405bis-13.txt> (UDP Usage Guidelines) to Best Current Practice
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Addressed in my working copy - thanks

> On 2016-06-02, at 12:11, Brian Trammell <ietf@trammell.ch> wrote:
>=20
> Greetings, all,
>=20
> Apologies for the late last call comment; I have only one, relatively =
minor. I hope it's still useful.
>=20
> I understand that Section 3 was written to encourage application =
developers not to roll their own transports ("trust us when we say this =
is hard, this document is a list of reasons why") but as written it =
would seem to discourage transport innovation atop UDP (e.g. QUIC, the =
RTCWEB data channel, anything-over-PLUS), which I very much hope was not =
the intent. The problematic recommendation is in the second paragraph:
>=20
>   These mechanisms are difficult to implement correctly.  For most
>   applications, the use of one of the existing IETF transport =
protocols
>   is the simplest method of acquiring the required mechanisms.  Doing
>   so also avoids issues that protocols using a new IP protocol number
>   face when being deployed over the Internet, where middleboxes that
>   only support TCP and UDP are not rare.  Consequently, the =
RECOMMENDED
>   alternative to the UDP usage described in the remainder of this
>   section is the use of an IETF transport protocol such as TCP
>   [RFC0793], Stream Control Transmission Protocol (SCTP) [RFC4960], =
and
>   SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram
>   Congestion Control Protocol (DCCP) [RFC4340] with its different
>   congestion control types [RFC4341][RFC4342][RFC5622].
>=20
> First, this paragraph ignores potential deployment issues with any of =
these other than TCP, which risks seeming out of touch, but this is a =
minor point and probably not worth a late edit. Second, I'm concerned =
this recommendation could be taken as broader than intended, against the =
definition of any new transport protocol encapsulated within UDP that =
performs substantially the same function as the listed protocols.
>=20
> I think this can be made clearer by simply adding to the list of =
examples:
>=20
> NEW:
>=20
>   These mechanisms are difficult to implement correctly.  For most
>   applications, the use of one of the existing IETF transport =
protocols
>   is the simplest method of acquiring the required mechanisms.  Doing
>   so also avoids issues that protocols using a new IP protocol number
>   face when being deployed over the Internet, where middleboxes that
>   only support TCP and UDP are not rare.  Consequently, the =
RECOMMENDED
>   alternative to the UDP usage described in the remainder of this
>   section is the use of an IETF transport protocol such as TCP
>   [RFC0793], Stream Control Transmission Protocol (SCTP) [RFC4960], =
and
>   SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram
>   Congestion Control Protocol (DCCP) [RFC4340] with its different
>   congestion control types [RFC4341][RFC4342][RFC5622], or transport
>   protocols specified by the IETF in the future.
>=20
> and removing the examples from the summary in section 7:
>=20
> OLD:
>=20
>   | SHOULD use a full-featured transport (TCP, SCTP, DCCP)  |         =
|
>=20
> NEW:
>=20
>   | SHOULD use a full-featured transport                    |         =
|
>=20
> Thanks, cheers,
>=20
> Brian
>=20
>=20
>> On 18 May 2016, at 02:17, The IESG <iesg-secretary@ietf.org> wrote:
>>=20
>>=20
>> The IESG has received a request from the Transport Area Working Group =
WG
>> (tsvwg) to consider the following document:
>> - 'UDP Usage Guidelines'
>> <draft-ietf-tsvwg-rfc5405bis-13.txt> as Best Current Practice
>>=20
>> The IESG plans to make a decision in the next few weeks, and solicits
>> final comments on this action. Please send substantive comments to =
the
>> ietf@ietf.org mailing lists by 2016-05-31. Exceptionally, comments =
may be
>> sent to iesg@ietf.org instead. In either case, please retain the
>> beginning of the Subject line to allow automated sorting.
>>=20
>> Abstract
>>=20
>>=20
>>  The User Datagram Protocol (UDP) provides a minimal message-passing
>>  transport that has no inherent congestion control mechanisms.  This
>>  document provides guidelines on the use of UDP for the designers of
>>  applications, tunnels and other protocols that use UDP.  Congestion
>>  control guidelines are a primary focus, but the document also
>>  provides guidance on other topics, including message sizes,
>>  reliability, checksums, middlebox traversal, the use of ECN, DSCPs,
>>  and ports.
>>=20
>>  Because congestion control is critical to the stable operation of =
the
>>  Internet, applications and other protocols that choose to use UDP as
>>  an Internet transport must employ mechanisms to prevent congestion
>>  collapse and to establish some degree of fairness with concurrent
>>  traffic.  They may also need to implement additional mechanisms,
>>  depending on how they use UDP.
>>=20
>>  Some guidance is also applicable to the design of other protocols
>>  (e.g., protocols layered directly on IP or via IP-based tunnels),
>>  especially when these protocols do not themselves provide congestion
>>  control.
>>=20
>>  This document obsoletes RFC5405 and adds guidelines for multicast =
UDP
>>  usage.
>>=20
>>=20
>>=20
>>=20
>> The file can be obtained via
>> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/
>>=20
>> IESG discussion can be tracked via
>> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/ballot/
>>=20
>>=20
>> No IPR declarations have been submitted directly on this I-D.
>>=20
>>=20
>=20


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From: "Eggert, Lars" <lars@netapp.com>
To: "mallman@icir.org" <mallman@icir.org>, "draft-ietf-tsvwg-rfc5405bis.all@ietf.org" <draft-ietf-tsvwg-rfc5405bis.all@ietf.org>
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Hi,

On 2016-05-31, at 20:57, Mark Allman <mallman@icir.org> wrote:
>  - I'd be interested to hear why it is RECOMMENDED to adhere to the
>    RTO algorithm in RFC6298.  In my view, in 2016 this is just
>    unnecessary and wrong (see my broader opinion below).  (That
>    said, if someone wants to use that algorithm, I think it is
>    fine... but, prescribing it is not needed, IMO.)

because it's a reasonable example for which we have a spec. I'm happy to =
recommend something else that is reasonable.

>  - I'd also like to see the normative text better underscore the
>    importance of backoff.

More than we currently do?

   It is important to note that the initial timeout value is not the
   maximum possible timeout -- the RECOMMENDED algorithm in [RFC6298]
   doubles the measured RTT value after each timeout (e.g., [RFC3261]).
   After a series of timeouts, this exponential back-off therefore
   results in values that are much longer than the initial value.

Would you suggest text?

>  - "TCP requires the initial RTT to be set to no less than 1
>    second" ... TCP requires the initial ***RTO*** be no less than 1
>    second.  There is a difference between the RTT and the RTO and
>    this draft is pretty liberal with using them interchangeably
>    (which is both confusing and wrong).

Oops. Will check.

> A broader opinion is that we should just elide these words in favor
> of a set of general RTO guidelines that span protocols (see
> draft-ietf-tcpm-rto-consider ; which goes beyond TCP, but is in TCPM
> for now because TCP is where it started, but comments would be
> appreciated from all corners).

We could do that. At this point, I'm in favor of doing whatever get this =
off my plate faster :-)

Lars

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From nobody Fri Jun 17 06:58:32 2016
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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 Transport Area Working Group of the IETF.

        Title           : UDP Usage Guidelines
        Authors         : Lars Eggert
                          Godred Fairhurst
                          Greg Shepherd
	Filename        : draft-ietf-tsvwg-rfc5405bis-14.txt
	Pages           : 57
	Date            : 2016-06-17

Abstract:
   The User Datagram Protocol (UDP) provides a minimal message-passing
   transport that has no inherent congestion control mechanisms.  This
   document provides guidelines on the use of UDP for the designers of
   applications, tunnels and other protocols that use UDP.  Congestion
   control guidelines are a primary focus, but the document also
   provides guidance on other topics, including message sizes,
   reliability, checksums, middlebox traversal, the use of ECN, DSCPs,
   and ports.

   Because congestion control is critical to the stable operation of the
   Internet, applications and other protocols that choose to use UDP as
   an Internet transport must employ mechanisms to prevent congestion
   collapse and to establish some degree of fairness with concurrent
   traffic.  They may also need to implement additional mechanisms,
   depending on how they use UDP.

   Some guidance is also applicable to the design of other protocols
   (e.g., protocols layered directly on IP or via IP-based tunnels),
   especially when these protocols do not themselves provide congestion
   control.

   This document obsoletes RFC5405 and adds guidelines for multicast UDP
   usage.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-tsvwg-rfc5405bis-14

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-rfc5405bis-14


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

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


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From: "Eggert, Lars" <lars@netapp.com>
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This revision attempts to address the various directorate and other =
review we have gotten.

The main open issue in my book is what to do about Mark Allman's comment =
regarding scrapping the RTT/RTO text in favor of simply to (normatively, =
I guess) referring to draft-ietf-tcpm-rto-consider.

Lars

> On 2016-06-17, at 15:58, 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 Transport Area Working Group of the =
IETF.
>=20
>        Title           : UDP Usage Guidelines
>        Authors         : Lars Eggert
>                          Godred Fairhurst
>                          Greg Shepherd
> 	Filename        : draft-ietf-tsvwg-rfc5405bis-14.txt
> 	Pages           : 57
> 	Date            : 2016-06-17
>=20
> Abstract:
>   The User Datagram Protocol (UDP) provides a minimal message-passing
>   transport that has no inherent congestion control mechanisms.  This
>   document provides guidelines on the use of UDP for the designers of
>   applications, tunnels and other protocols that use UDP.  Congestion
>   control guidelines are a primary focus, but the document also
>   provides guidance on other topics, including message sizes,
>   reliability, checksums, middlebox traversal, the use of ECN, DSCPs,
>   and ports.
>=20
>   Because congestion control is critical to the stable operation of =
the
>   Internet, applications and other protocols that choose to use UDP as
>   an Internet transport must employ mechanisms to prevent congestion
>   collapse and to establish some degree of fairness with concurrent
>   traffic.  They may also need to implement additional mechanisms,
>   depending on how they use UDP.
>=20
>   Some guidance is also applicable to the design of other protocols
>   (e.g., protocols layered directly on IP or via IP-based tunnels),
>   especially when these protocols do not themselves provide congestion
>   control.
>=20
>   This document obsoletes RFC5405 and adds guidelines for multicast =
UDP
>   usage.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/
>=20
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-tsvwg-rfc5405bis-14
>=20
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rfc5405bis-14
>=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


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From nobody Fri Jun 17 07:03:37 2016
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Subject: Re: [tsvwg] [tcpm] draft-ietf-tcpm-rto-consider-04: (R.3)
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Again, my personal view is that the statement is not needed, and I find =
it rather confusing that helpful. Based my (limited) experience in =
(Linux) implementors community I would like to add that even though they =
are often not very active on ietf mailing lists and might not follow the =
RFC in their implementation, they still read the draft. So simply =
focusing on the actual recommendation in the doc (and the needed =
statements how this doc related to other docs in the RFC series) is =
sufficient for me and more clear.

Mirja


> Am 17.06.2016 um 15:22 schrieb Mark Allman <mallman@icir.org>:
>=20
>=20
>> I have a personal opinion on the point below (R-3). I think this
>> point is just not needed. I say this not because I don=E2=80=99t =
agree
>> with the statement but spelling this out in an RFC, to me,
>> contradict the point of the RFC.
>=20
> On the one hand, I can understand that opinion.  Why would we write
> the RFC if we weren't going to allow these things and so why do we
> have to say this?  And, if people think this is unnecessary I am
> happy to nuke it.
>=20
> However, let me sketch the lay of the land as I see it ....   It
> seems the meat in this document can be used for different purposes.
> And, we have seen opinions on each of these I think.  The document
> could be used:
>=20
>  (a) as input to future RTO specifications,
>=20
>  (b) as input to future RTO implementations, or
>=20
>  (c) as both (a) and (b)
>=20
> My take is (c).  And, so (R.2) is meant to cover case (a), while
> (R.3) is meant to cover case (b).
>=20
> So, while I can see your point about not needing to say something
> obvious, I am worried that eliding (R.3) will leave us in one of two
> places:
>=20
>  - In case (a) above.  I.e., because (R.2) discusses only
>    specifications and not implementations and so a natural reading
>    may well be that this document is meant to only help future
>    standardization efforts.
>=20
>  - Or, in some ambiguous state where clearly (a) holds (via (R.2)),
>    but it is not explicit that this covers implementations, too,
>    even if we think that it does since the text does not explicitly
>    say that.  And, so we end up with confusion.
>=20
> Does that make sense?  Am I worrying too much about explicitly
> including implementations?
>=20
> allman
>=20
>=20
>=20


From nobody Fri Jun 17 07:04:18 2016
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From: "Black, David" <david.black@emc.com>
To: Colin Perkins <csp@csperkins.org>, John Leslie <john@jlc.net>
Thread-Topic: [rtcweb] [AVTCORE] [tsvwg] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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Date: Fri, 17 Jun 2016 14:03:48 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, IETF AVTCore WG <avt@ietf.org>, tsvwg <tsvwg@ietf.org>
Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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cmtpbnMNCj4gaHR0cHM6Ly9jc3BlcmtpbnMub3JnLw0KPiANCj4gDQo+IA0KDQo=


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I am mulling this.  But, a question...

> So simply focusing on the actual recommendation in the doc (and
> the needed statements how this doc related to other docs in the
> RFC series) is sufficient for me and more clear.

I guess this is counter-intuitive to me as usually implicitness is a
less direct path to clarity than explicitness.

So, would you also then get rid of (R.2)?  So, we use (R.1) to state
how the doc relates to existing specs and then say nothing about the
future (about either specs or implementations).  To me that is at
least a more consistent path than retaining (R.2) and removing
(R.3).

But, I am still mulling ...

Thanks,
allman




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From nobody Fri Jun 17 07:23:06 2016
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I find R-2 less problematic/confusing but also probably not needed. =
Especially seeing the SHOULD in there seems slightly weird to me. To me =
the better approach would be to simply use SHOULDs instead of (most of =
the) MUSTs in the latter text in the doc; and potentially add more text =
there that the SHOULD is a really strong SHOULD and there must be a good =
reason to deviate.

Mirja

=20
> Am 17.06.2016 um 16:13 schrieb Mark Allman <mallman@icir.org>:
>=20
>=20
> I am mulling this.  But, a question...
>=20
>> So simply focusing on the actual recommendation in the doc (and
>> the needed statements how this doc related to other docs in the
>> RFC series) is sufficient for me and more clear.
>=20
> I guess this is counter-intuitive to me as usually implicitness is a
> less direct path to clarity than explicitness.
>=20
> So, would you also then get rid of (R.2)?  So, we use (R.1) to state
> how the doc relates to existing specs and then say nothing about the
> future (about either specs or implementations).  To me that is at
> least a more consistent path than retaining (R.2) and removing
> (R.3).
>=20
> But, I am still mulling ...
>=20
> Thanks,
> allman
>=20
>=20
>=20


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To: "Black, David" <david.black@emc.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>, Colin Perkins <csp@csperkins.org>
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Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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+1 to not use normative language here.

However, please note that having a high level of ECN-CE marks (without =
any losses) means that all packets were received correctly. This =
situation can even occurs without high delays (depending on the AQM =
used), which would just mean the services works perfectly. Therefore for =
me CE marks are a perfect input signal for a congestion control loop =
(where the AQM tell the sender to take action - whatever that means). =
But I=E2=80=99m less concerned than David about eventually ignoring it =
for circuit breaker.

In addition one point on something Magnus wrote earlier:
"If the implementation only have circuit breaker, i.e. no full fledged =
congestion controller and uses ECN, they can in worst case drive the =
buffer into the overload regime where it starts dropping packets. =E2=80=9E=


I=E2=80=99m not sure about this case. ECN is an input signal for =
congestion control. If you don=E2=80=99t use congestion control but only =
a circuit breaker, you should probably not enable ECN. At least it not =
clear to me why you would enable it, and it's definitely not conform to =
the ECN spec. Probably we should say something about this in the =
draft...?

Mirja


> Am 17.06.2016 um 16:03 schrieb Black, David <david.black@emc.com>:
>=20
> Colin,
>=20
>>>> ...  I view the current text as providing implementers with too =
much
>>>> latitude to ignore ECN-CE marks (e.g., because an implementer =
doesn't
>>>> want to think about this problem space in the first place).
>>=20
>> I agree, but the argument is that doing so is less harmful than =
deploying a circuit
>> breaker that triggers too often when ECN is used.
>>=20
>> I=E2=80=99m not sure I believe this argument, though, since it seems =
that any new AQM
>> that applies ECN marks much more often than at present will have to =
consider
>> backwards compatibility, to work with deployed TCP (e.g., =
draft-briscoe-tsvwg-
>> aqm-tcpm-rmcat-l4s-problem uses ECT(1) as a signal to use the new =
marking,
>> while existing implementations set ECT(0)). These compatibility =
mechanisms
>> would seem to prevent the issues with the circuit breaker too.
>=20
> That roughly matches my line of thinking, and I'll observe that the =
original DCTCP
> protocol design that used more aggressive ECN-CE marking was only safe =
for
> Controlled Environment deployments.   See the TSVWG rfc5405bis draft =
for the
> definition of Controlled Environment, and ignore the fact that the =
rfc5405bis
> draft is a UDP draft - this definition is more broadly applicable.
>=20
> Going back over Section 7 in this avtcore draft, my views are:
>=20
> [A] None of these drafts justify a "MAY ignore" response to ECN-CE =
marks:
> 	- draft-khademi-tcpm-alternativebackoff-ecn
> 	- draft-ietf-rmcat-nada
> 	- draft-ietf-rmcat-scream-cc
>=20
> [B] In line with Colin's comment on the L4S draft, I think it's =
incumbent on
> the authors of draft-briscoe-aqm-dualq-coupled to figure out how that =
will
> coexist (or avoid) deployed TCP, and this avtcore draft ought not to =
be
> trying to prejudge what will be done there.
>=20
> So, I don't think the current text in Section 7 has justified the =
unfettered
> "implementations MAY ignore ECN-CE marks" text, as ignoring those =
marks
> is not consistent with any of the four cited drafts.
>=20
> In more detail, I think making changes to normative requirements here =
based
> on [B] is premature, and I would hope that the rmcat WG could be =
encouraged
> to consider the RTP circuit breaker in its congestion control drafts, =
as those CC
> mechanisms are related to the circuit breaker mechanism, hence likely
> to be in related areas of an RTP implementation.
>=20
> That leaves draft-khademi-tcpm-alternativebackoff-ecn, which TSVWG
> will be looking at in Berlin.  If a normative statement about ECN-CE =
reaction
> is going to rest on that draft, then the reference to that draft =
should be
> normative.  Something about doing that strikes me as premature ...
>=20
> I realize that we're trying to predict and accommodate the future, =
which
> is an imprecise undertaking at best.   As an alternative to the =
current text,
> would it be reasonable to say (without any RFC 2119 keywords) that the
> best current guidance is still to treat ECN-CE marks as indicating =
drops,
> with a warning that there is a good possibility of this changing in =
the
> near future due to all of the work in progress cited in Section 7?
>=20
> Thanks, --David
>=20
>> -----Original Message-----
>> From: Colin Perkins [mailto:csp@csperkins.org]
>> Sent: Friday, June 17, 2016 6:14 AM
>> To: John Leslie; Black, David
>> Cc: rtcweb@ietf.org; IETF AVTCore WG; tsvwg
>> Subject: Re: [rtcweb] [AVTCORE] [tsvwg] WG Last Call on changes: =
draft-ietf-
>> avtcore-rtp-circuit-breakers-16
>>=20
>>=20
>>> On 16 Jun 2016, at 23:25, John Leslie <john@jlc.net> wrote:
>>>=20
>>> Black, David <david.black@emc.com> wrote:
>>>>=20
>>>> ...  I view the current text as providing implementers with too =
much
>>>> latitude to ignore ECN-CE marks (e.g., because an implementer =
doesn't
>>>> want to think about this problem space in the first place).
>>=20
>> I agree, but the argument is that doing so is less harmful than =
deploying a circuit
>> breaker that triggers too often when ECN is used.
>>=20
>> I=E2=80=99m not sure I believe this argument, though, since it seems =
that any new AQM
>> that applies ECN marks much more often than at present will have to =
consider
>> backwards compatibility, to work with deployed TCP (e.g., =
draft-briscoe-tsvwg-
>> aqm-tcpm-rmcat-l4s-problem uses ECT(1) as a signal to use the new =
marking,
>> while existing implementations set ECT(0)). These compatibility =
mechanisms
>> would seem to prevent the issues with the circuit breaker too.
>>=20
>>>  Understand, we have at least two proposals to make ECN-CE more =
frequent
>>> than packet drop would be for non-ECN packets: possibly =
substantially
>>> more frequent. Unless both are killed off, ECN-CE will show up =
frequently
>>> enough that closing the flow on ECN-CE would kill too many =
connections.
>>>=20
>>>  If you want circuit-breaking on such connections, there are two =
ways:
>>> 1. convince the forwarding nodes to drop packets if their queue =
exceeds
>>>  design capacity; or
>>> 2. require the sender to send enough not-ECN-capable packets so that =
our
>>>  receiver will see enough packet-drops when a circuit-breaker should
>>>  activate.
>>>=20
>>>  (I prefer the first option; but I wouldn't object to the second.)
>>>=20
>>>  There really isn't any way for our circuit-breaker to know =
_how_much_
>>> more frequent the ECN-CE marks may be. :^(
>>=20
>> This is a problem, both for the circuit breaker, and for the =
algorithms being
>> defined in RMCAT. We do need some understanding what the expected =
marking
>> rates are likely to be, so congestion control and circuit breakers =
can be defined.
>>=20
>>> We _will_ be sorry if we
>>> allot the same frequency of CE packets as packet-drops to trigger =
the
>>> circuit-breaker.
>>>=20
>>>> Could someone propose initial text to qualifies the current "MAY =
ignore"
>>>> statement?
>>>=20
>>>  Essentially, for the second option, you might propose text to the
>>> effect of:
>>> ]
>>> ] If too many ECN-CE packets are received, the sender SHOULD send =
some
>>> ] not-ECN-capable packets to determine whether enough packets along =
the
>>> ] path are being dropped to justify activating our circuit-breaker.
>>>=20
>>>  I=E2=80=99m not enthusiastic about adding that; but it would =
resolve the issue.
>>=20
>> I=E2=80=99m not convinced this would work. The circuit breaker is =
looking at long term
>> trends, and in order to have enough not-ECT packets to determine if =
it should
>> trigger, you=E2=80=99d essentially have to run without ECN for some =
seconds.
>>=20
>> --
>> Colin Perkins
>> https://csperkins.org/
>>=20
>>=20
>>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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To: "Eggert, Lars" <lars@netapp.com>
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Subject: Re: [tsvwg] Last Call: <draft-ietf-tsvwg-rfc5405bis-13.txt> (UDP Usage Guidelines) to Best Current Practice
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> On 17 Jun 2016, at 14:45, Eggert, Lars <lars@netapp.com> wrote:
>=20
> Hi,
>=20
>> On 2016-05-31, at 20:57, Mark Allman <mallman@icir.org> wrote:
>> - I'd be interested to hear why it is RECOMMENDED to adhere to the
>>   RTO algorithm in RFC6298.  In my view, in 2016 this is just
>>   unnecessary and wrong (see my broader opinion below).  (That
>>   said, if someone wants to use that algorithm, I think it is
>>   fine... but, prescribing it is not needed, IMO.)
>=20
> because it's a reasonable example for which we have a spec. I'm happy to r=
ecommend something else that is reasonable.
>=20
>> - I'd also like to see the normative text better underscore the
>>   importance of backoff.
>=20
> More than we currently do?
>=20
>   It is important to note that the initial timeout value is not the
>   maximum possible timeout -- the RECOMMENDED algorithm in [RFC6298]
>   doubles the measured RTT value
/measured rtt/timeout period/ was what I suggested.

> after each timeout (e.g., [RFC3261]).
>   After a series of timeouts, this exponential back-off therefore
>   results in values that are much longer than the initial value.
>=20
> Would you suggest text?
>=20
>> - "TCP requires the initial RTT to be set to no less than 1
>>   second" ... TCP requires the initial ***RTO*** be no less than 1
>>   second.  There is a difference between the RTT and the RTO and
>>   this draft is pretty liberal with using them interchangeably
>>   (which is both confusing and wrong).
>=20
> Oops. Will check.
>=20
 Basically there is not in general a concept of RTO in UDP protocols, althou=
gh reliable protocols could follow TCP.=20

>> A broader opinion is that we should just elide these words in favor
>> of a set of general RTO guidelines that span protocols (see
>> draft-ietf-tcpm-rto-consider ; which goes beyond TCP, but is in TCPM
>> for now because TCP is where it started, but comments would be
>> appreciated from all corners).
>=20
> We could do that. At this point, I'm in favor of doing whatever get this o=
ff my plate faster :-)

The publication status and scope of Marks draft currently in TCPM still need=
s discussion... Including how and whether it will mention UDP...  Let's hope=
 that is clear soon. This ref to that draft was intended to help progress bo=
th. I fear Removing the mention of retransmission from the guidelines to onl=
y an external reference stalls both drafts to a discussion in Berlin...=20

> Lars

I think suggested text would be helpful.

Gorry=


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From: Jana Iyengar <jri@google.com>
Date: Fri, 17 Jun 2016 09:15:00 -0700
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Subject: Re: [tsvwg] Last Call: <draft-ietf-tsvwg-rfc5405bis-13.txt> (UDP Usage Guidelines) to Best Current Practice
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On Fri, Jun 17, 2016 at 9:05 AM, Gorry (erg) <gorry@erg.abdn.ac.uk> wrote:

>
>
> > On 17 Jun 2016, at 14:45, Eggert, Lars <lars@netapp.com> wrote:
> >
> > Hi,
> >
> >> On 2016-05-31, at 20:57, Mark Allman <mallman@icir.org> wrote:
> >> - I'd be interested to hear why it is RECOMMENDED to adhere to the
> >>   RTO algorithm in RFC6298.  In my view, in 2016 this is just
> >>   unnecessary and wrong (see my broader opinion below).  (That
> >>   said, if someone wants to use that algorithm, I think it is
> >>   fine... but, prescribing it is not needed, IMO.)
> >
> > because it's a reasonable example for which we have a spec. I'm happy to
> recommend something else that is reasonable.
> >
> >> - I'd also like to see the normative text better underscore the
> >>   importance of backoff.
> >
> > More than we currently do?
> >
> >   It is important to note that the initial timeout value is not the
> >   maximum possible timeout -- the RECOMMENDED algorithm in [RFC6298]
> >   doubles the measured RTT value
> /measured rtt/timeout period/ was what I suggested.
>
> > after each timeout (e.g., [RFC3261]).
> >   After a series of timeouts, this exponential back-off therefore
> >   results in values that are much longer than the initial value.
> >
> > Would you suggest text?
> >
> >> - "TCP requires the initial RTT to be set to no less than 1
> >>   second" ... TCP requires the initial ***RTO*** be no less than 1
> >>   second.  There is a difference between the RTT and the RTO and
> >>   this draft is pretty liberal with using them interchangeably
> >>   (which is both confusing and wrong).
> >
> > Oops. Will check.
> >
>  Basically there is not in general a concept of RTO in UDP protocols,
> although reliable protocols could follow TCP.


While that's true, there can be an expectation of something like an RTO. We
did this for LEDBAT, and we called it a Congestion Timeout (CTO). From RFC
6817:

update_CTO()
    # implements an RTT estimation mechanism using data
    # transmission times and ACK reception times,
    # which is used to implement a congestion timeout (CTO).
    # If implementing LEDBAT in TCP, sender SHOULD use
    # mechanisms described in RFC 6298
<https://tools.ietf.org/html/rfc6298> [RFC6298
<https://tools.ietf.org/html/rfc6298>],
    # and the CTO would be the same as the retransmission timeout (RTO).


and this is used as follows:

   if no ACKs are received within a CTO:
       # extreme congestion, or significant RTT change.
       # set cwnd to 1MSS and backoff the congestion timer.
       cwnd = 1 * MSS
       CTO = 2 * CTO


The UDP guidelines draft can a reflect similar concept and language.

>
> >> A broader opinion is that we should just elide these words in favor
> >> of a set of general RTO guidelines that span protocols (see
> >> draft-ietf-tcpm-rto-consider ; which goes beyond TCP, but is in TCPM
> >> for now because TCP is where it started, but comments would be
> >> appreciated from all corners).
> >
> > We could do that. At this point, I'm in favor of doing whatever get this
> off my plate faster :-)
>
> The publication status and scope of Marks draft currently in TCPM still
> needs discussion... Including how and whether it will mention UDP...  Let's
> hope that is clear soon. This ref to that draft was intended to help
> progress both. I fear Removing the mention of retransmission from the
> guidelines to only an external reference stalls both drafts to a discussion
> in Berlin...
>

+1 to using rto-consider as the reference if at all possible.

- jana


> > Lars
>
> I think suggested text would be helpful.
>
> Gorry
>

--001a1148af184256e805357ba85e
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Fri, Jun 17, 2016 at 9:05 AM, Gorry (erg) <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:gorry@erg.abdn.ac.uk">gorry@erg.abdn.ac.uk</a>&gt;</span> wro=
te:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;=
border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204=
,204);padding-left:1ex"><span class=3D"gmail-"><br>
<br>
&gt; On 17 Jun 2016, at 14:45, Eggert, Lars &lt;<a href=3D"mailto:lars@neta=
pp.com">lars@netapp.com</a>&gt; wrote:<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt;&gt; On 2016-05-31, at 20:57, Mark Allman &lt;<a href=3D"mailto:mallman=
@icir.org">mallman@icir.org</a>&gt; wrote:<br>
&gt;&gt; - I&#39;d be interested to hear why it is RECOMMENDED to adhere to=
 the<br>
&gt;&gt;=C2=A0 =C2=A0RTO algorithm in RFC6298.=C2=A0 In my view, in 2016 th=
is is just<br>
&gt;&gt;=C2=A0 =C2=A0unnecessary and wrong (see my broader opinion below).=
=C2=A0 (That<br>
&gt;&gt;=C2=A0 =C2=A0said, if someone wants to use that algorithm, I think =
it is<br>
&gt;&gt;=C2=A0 =C2=A0fine... but, prescribing it is not needed, IMO.)<br>
&gt;<br>
&gt; because it&#39;s a reasonable example for which we have a spec. I&#39;=
m happy to recommend something else that is reasonable.<br>
&gt;<br>
&gt;&gt; - I&#39;d also like to see the normative text better underscore th=
e<br>
&gt;&gt;=C2=A0 =C2=A0importance of backoff.<br>
&gt;<br>
&gt; More than we currently do?<br>
&gt;<br>
&gt;=C2=A0 =C2=A0It is important to note that the initial timeout value is =
not the<br>
&gt;=C2=A0 =C2=A0maximum possible timeout -- the RECOMMENDED algorithm in [=
RFC6298]<br>
&gt;=C2=A0 =C2=A0doubles the measured RTT value<br>
</span>/measured rtt/timeout period/ was what I suggested.<br>
<span class=3D"gmail-"><br>
&gt; after each timeout (e.g., [RFC3261]).<br>
&gt;=C2=A0 =C2=A0After a series of timeouts, this exponential back-off ther=
efore<br>
&gt;=C2=A0 =C2=A0results in values that are much longer than the initial va=
lue.<br>
&gt;<br>
&gt; Would you suggest text?<br>
&gt;<br>
&gt;&gt; - &quot;TCP requires the initial RTT to be set to no less than 1<b=
r>
&gt;&gt;=C2=A0 =C2=A0second&quot; ... TCP requires the initial ***RTO*** be=
 no less than 1<br>
&gt;&gt;=C2=A0 =C2=A0second.=C2=A0 There is a difference between the RTT an=
d the RTO and<br>
&gt;&gt;=C2=A0 =C2=A0this draft is pretty liberal with using them interchan=
geably<br>
&gt;&gt;=C2=A0 =C2=A0(which is both confusing and wrong).<br>
&gt;<br>
&gt; Oops. Will check.<br>
&gt;<br>
</span>=C2=A0Basically there is not in general a concept of RTO in UDP prot=
ocols, although reliable protocols could follow TCP.</blockquote><div><br><=
/div><div>While that&#39;s true, there can be an expectation of something l=
ike an RTO. We did this for LEDBAT, and we called it a Congestion Timeout (=
CTO). From RFC 6817:</div><div><pre class=3D"gmail-newpage" style=3D"font-s=
ize:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">update_CTO=
()
    # implements an RTT estimation mechanism using data
    # transmission times and ACK reception times,
    # which is used to implement a congestion timeout (CTO).
    # If implementing LEDBAT in TCP, sender SHOULD use
    # mechanisms described in <a href=3D"https://tools.ietf.org/html/rfc629=
8">RFC 6298</a> [<a href=3D"https://tools.ietf.org/html/rfc6298" title=3D"&=
quot;Computing TCP&#39;s Retransmission Timer&quot;">RFC6298</a>],
    # and the CTO would be the same as the retransmission timeout (RTO).
</pre></div><div><br></div><div>and this is used as follows:</div><div><pre=
 class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin=
-bottom:0px;color:rgb(0,0,0)">   if no ACKs are received within a CTO:
       # extreme congestion, or significant RTT change.
       # set cwnd to 1MSS and backoff the congestion timer.
       cwnd =3D 1 * MSS
       CTO =3D 2 * CTO
</pre></div><div><br></div><div>The UDP guidelines draft can a reflect simi=
lar concept and language.</div><blockquote class=3D"gmail_quote" style=3D"m=
argin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;borde=
r-left-color:rgb(204,204,204);padding-left:1ex"><span class=3D"gmail-"><br>
&gt;&gt; A broader opinion is that we should just elide these words in favo=
r<br>
&gt;&gt; of a set of general RTO guidelines that span protocols (see<br>
&gt;&gt; draft-ietf-tcpm-rto-consider ; which goes beyond TCP, but is in TC=
PM<br>
&gt;&gt; for now because TCP is where it started, but comments would be<br>
&gt;&gt; appreciated from all corners).<br>
&gt;<br>
&gt; We could do that. At this point, I&#39;m in favor of doing whatever ge=
t this off my plate faster :-)<br>
<br>
</span>The publication status and scope of Marks draft currently in TCPM st=
ill needs discussion... Including how and whether it will mention UDP...=C2=
=A0 Let&#39;s hope that is clear soon. This ref to that draft was intended =
to help progress both. I fear Removing the mention of retransmission from t=
he guidelines to only an external reference stalls both drafts to a discuss=
ion in Berlin...<br></blockquote><div><br></div><div>+1 to using rto-consid=
er as the reference if at all possible.</div><div><br></div><div>- jana</di=
v><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0p=
x 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color=
:rgb(204,204,204);padding-left:1ex">
&gt; Lars<br>
<br>
I think suggested text would be helpful.<br>
<span class=3D"gmail-HOEnZb"><font color=3D"#888888"><br>
Gorry<br>
</font></span></blockquote></div><br></div></div>

--001a1148af184256e805357ba85e--


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Subject: Re: [tsvwg] [tcpm] draft-ietf-tcpm-rto-consider-04: (R.3)
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I like to re-state things to see if I understand what is being
said.  So, let me see if I understand what you're suggesting (in
high level terms).

(This is about the RTO document in specific, but more broadly would
seem applicable to single-host algorithm specifications.)

You are starting from the fact that implementers are not beholden to
the RFCs in any tangible way.  And, also the fact that we don't have
any real recourse when the standards are not followed.  So, since
they don't feel constrained by the RFCs, we should approach RFCs in
the same way and not worry about constraints as much as giving our
best recommendations.  We hope our strong recommendations are
listened to and used.  But, if they are not, they are not and there
isn't anything we can do about them regardless of what we say people
"MUST" do.  I.e., the standards process should meet the implementers
where they are.

Is that roughly correct?

(In the above I mean to have no opinion, just trying to understand
your hit.  I may subsequently have an opinion and if I can somehow
get over my usual shyness I might then say it. :) )

allman




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From nobody Fri Jun 17 11:24:20 2016
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From: "Tim Szigeti (szigeti)" <szigeti@cisco.com>
To: "Fred Baker (fred)" <fred@cisco.com>, "Black, David" <david.black@emc.com>
Thread-Topic: [tsvwg] WG adoption call: draft-szigeti-tsvwg-ieee-802-11 - through	June 25
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Date: Fri, 17 Jun 2016 18:24:15 +0000
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References: <CE03DB3D7B45C245BCA0D243277949362F57A2A4@MX307CL04.corp.emc.com> <F40BE8F6-4D68-46CD-8C97-F5EB11992737@cisco.com>
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Subject: Re: [tsvwg] WG adoption call: draft-szigeti-tsvwg-ieee-802-11 - through	June 25
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Same for me - I know of no such IPR.

Thanks,

-tim

-----Original Message-----
From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Fred Baker (fred)
Sent: Thursday, June 09, 2016 7:29 AM
To: Black, David
Cc: tsvwg@ietf.org
Subject: Re: [tsvwg] WG adoption call: draft-szigeti-tsvwg-ieee-802-11 - th=
rough June 25


> On Jun 8, 2016, at 11:14 PM, Black, David <david.black@emc.com> wrote:
>=20
> Authors: We (chairs) assume that you support WG adoption of your=20
> draft, so there's no need to say so on the list.  However, we do need=20
> each author to publicly state on the list that their direct, personal=20
> knowledge (if any) of any IPR related to this document has already=20
> been disclosed, in conformance with BCPs 78 and 79.

I know of no such IPR.


From nobody Fri Jun 17 11:27:42 2016
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From: "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>
To: =?utf-8?B?TWlyamEgS8O8aGxld2luZA==?= <mirja.kuehlewind@tik.ee.ethz.ch>, "mallman@icir.org" <mallman@icir.org>
Thread-Topic: [tsvwg] [tcpm] draft-ietf-tcpm-rto-consider-04: (R.3)
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Subject: Re: [tsvwg] [tcpm] draft-ietf-tcpm-rto-consider-04: (R.3)
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Just one quick one that is easy to answer ...

> For TCP, I am confused by how this wording relates to RFC
> 6298.

This document does nothing to RFC6298.  I.e.,

    (R.1) RTO mechanisms that are currently standardized are not updated=20
          or obsoleted by this document.  Implementations are free to
          use these existing specifications as they do now.

That is, RFC6298 remains as it is now.  Implementers are free to use
the algorithm as it sits in RFC6298.

> For TCP, I fully agree that allowing more flexibility compared to
> RFC 6298 would be useful, and a new RFC 6298bis could reference a
> BCP draft-ietf-tcpm-rto-consider. However, I believe that guidance
> for TCP implementers belongs into a standards track document RFC
> 6298bis. The document draft-ietf-tcpm-rto-consider should focus on
> a set of requirements only.

I am not sure what you mean here.

So, we'd use rto-consider as a set of requirements.  And, then we'd
generate a 6298bis that is less specific than 6298.  So, e.g.,
instead of providing equations like this:

            RTTVAR <- (1 - beta) * RTTVAR + beta * |SRTT - R'|
            SRTT <- (1 - alpha) * SRTT + alpha * R'
            RTO <- SRTT + max (G, K*RTTVAR)

and their corresponding parameter values, we'd just fall back to
some generic language like "use RTT samples and the variance of RTT
samples per [rto-consider] to compute an RTO"?  So, the document is
then a fuzzy re-statement of the [rto-consider] requirements instead
of a specific algorithm.

Something like that?

Or, do you just want a 6298bis that includes a statement at the top
that says "this algorithm follows the requirements in
[rto-consider]"?

Or, something else?  I can't envision how you view this as working.=20

allman




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From nobody Fri Jun 17 12:12:50 2016
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From: "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>
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Thread-Topic: [tsvwg] [tcpm] draft-ietf-tcpm-rto-consider-04: (R.3) 
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> > For TCP, I fully agree that allowing more flexibility compared to RFC=20
> > 6298 would be useful, and a new RFC 6298bis could reference a BCP=20
> > draft-ietf-tcpm-rto-consider. However, I believe that guidance for TCP=
=20
> > implementers belongs into a standards track document RFC 6298bis. The=20
> > document draft-ietf-tcpm-rto-consider should focus on a set of=20
> > requirements only.

> I am not sure what you mean here.

6298bis would have a new (first) section with a MUST that applies the guide=
lines in [rto-consider] specifically to the TCP protocol, mandating that al=
l TCP implementations MUST use retransmission timeout mechanisms matching t=
his minimum set of requirements. This is relatively straightforward given t=
hat [rto-consider] is rather TCP-ish already. But for a number of requireme=
nts the actual TCP terminology and technology could be used.

And then the document would recommend with a MAY (or SHOULD) the algorithm =
in 6298 as default.

And I believe that such a proposed standard for TCP would be closer to what=
 implementations actually do.=20

(And I believe for SCTP it would also be a no-brainer to come up with somet=
hing in TSVWG, but the SCTP experts would have comment on that.)

Michael


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Hi Mark,

> Am 17.06.2016 um 19:09 schrieb Mark Allman <mallman@icir.org>:
>=20
> You are starting from the fact that implementers are not beholden to
> the RFCs in any tangible way.  And, also the fact that we don't have
> any real recourse when the standards are not followed.  So, since
> they don't feel constrained by the RFCs, we should approach RFCs in
> the same way and not worry about constraints as much as giving our
> best recommendations.  We hope our strong recommendations are
> listened to and used.  But, if they are not, they are not and there
> isn't anything we can do about them regardless of what we say people
> "MUST" do.  I.e., the standards process should meet the implementers
> where they are.
>=20
> Is that roughly correct?

Roughly yes. I wouldn=E2=80=99t use the wording "implementers are not =
beholden to
the RFCs in any tangible way=E2=80=9C but anyway...

Even though your statement was scoped to single-host algos, I don=E2=80=99=
t want to let this stand in the room like this. Let me add a few things =
(that are more general and probably not important for the discussion on =
rto though).=20

First, this is slightly different if interoperability is required. But =
even in this case, we can=E2=80=99t control what implementors are doing. =
However, I would say there is a natural incentive in this case to follow =
the spec because other it just won=E2=80=99t work.=20

Then I would also like to say that the consequence of this thinking is =
not that RFCs are useless as anyway nobody will follow anyway. I know =
that is not was you were saying or thinking I assume. I just want to =
clearly state that there is an important role for RFCs to make =
recommendation; as I said implementors read RFC (even if they don=E2=80=99=
t follow).

And finally, the other important aspect is that this liberal approach in =
the Internet, where we of course depend to a certain extend on other =
people doing things right/not too wrong, is that enables innovation in =
the Internet and make the Internet to this great thing we have today. =
E.g if someone invents a complete new service that uses the Internet =
that might not fit our current recommendation. So this freedom is needed =
and deliberate.

With this more philosophical statement, I guess the only this that is =
left to say is:

Have a good weekend!
Mirja




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> 6298bis would have a new (first) section with a MUST that applies
> the guidelines in [rto-consider] specifically to the TCP protocol,
> mandating that all TCP implementations MUST use retransmission
> timeout mechanisms matching this minimum set of requirements. This
> is relatively straightforward given that [rto-consider] is rather
> TCP-ish already. But for a number of requirements the actual TCP
> terminology and technology could be used.=20
>=20
> And then the document would recommend with a MAY (or SHOULD) the
> algorithm in 6298 as default.=20
>=20
> And I believe that such a proposed standard for TCP would be
> closer to what implementations actually do.=20=20

We have been around this before, but for the public record ...

To me this is a bunch of work for, perhaps, epsilon gain.  We state
some high level requirements.  But, then we have to re-state the
high-level requirements after prepending a protocol name and only
then can an implementation can actually leverage the requirements.
Perhaps that forms a nice clean document progression.  But, man,
what a bunch of busy work that has no value add...

Rather, I think this document can be directly applied to both future
specifications and future implementations.  I don't yet have a
well-formed hit on whether we should directly state that in some
fashion (as the doc---perhaps clumsily---tries to do now) or leave
the requirements as our best thinking to be implicitly applied
however folks see fit (my understanding of Mirja's hit).

allman


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




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> > You are starting from the fact that implementers are not beholden to
> > the RFCs in any tangible way.  And, also the fact that we don't have
> > any real recourse when the standards are not followed.  So, since
> > they don't feel constrained by the RFCs, we should approach RFCs in
> > the same way and not worry about constraints as much as giving our
> > best recommendations.  We hope our strong recommendations are
> > listened to and used.  But, if they are not, they are not and there
> > isn't anything we can do about them regardless of what we say people
> > "MUST" do.  I.e., the standards process should meet the implementers
> > where they are.
> >=20
> > Is that roughly correct?
>=20
> Roughly yes.

OK.  Then, I am going to mull this approach.  I view three things on
the table:

  - Mirja's approach: State the recommendations, but not explicitly
    what they apply to and as SHOULDs rather than MUSTs and then let
    people use them as they find useful.

  - Michael's approach: The requirements exist only to guide future
    specs.  And, implementations are explicitly not to leverage
    these directly.

  - Mark's approach: Somewhere in the middle, I guess.  State the
    recommendations, but also explicitly that they are for both
    future specs and future implementations.  (Clearly closer to
    Mirja and Michael.)

I am still thinking.  Others can chime in.  But, only if your name
starts with "M"... because naming conventions are important.

> First, this is slightly different if interoperability is
> required. But even in this case, we can=E2=80=99t control what
> implementors are doing. However, I would say there is a natural
> incentive in this case to follow the spec because other it just
> won=E2=80=99t work.=20=20

(Yep.  This is why I scoped!)

> Then I would also like to say that the consequence of this
> thinking is not that RFCs are useless as anyway nobody will follow
> anyway. I know that is not was you were saying or thinking I
> assume. I just want to clearly state that there is an important
> role for RFCs to make recommendation; as I said implementors read
> RFC (even if they don=E2=80=99t follow).

Well, OK.  I understand what you're saying.

But, what I am struggling with here is that if we make some MUSTs
then we get a little something in return.  That sets an expectation.
And, if the FooBar implementation doesn't follow that expectation
and it causes problems then there is a lever to say "well, gee, we
said you needed to do something and you didn't".  FooBar does not
have to listen.  We cannot make them do something different.  But,
we get a little lever.  So, the question in my mind is whether we
get the same sort of leverage with your looser notion of "just state
the requirements" and "use SHOULDs".  And, I am thinking not.  I.e.,
we didn't say "do this" we said "do this unless you have a good
reason" and FooBar might think they have a good reason (e.g., "helps
my customers, so sorry about your's!").  We can still try to say
"well, we strongly recommended ...", but we're in a bit lesser
position.

The thing I am trying to form an opinion on is "how much lesser".
I.e., I have sort of convinced myself stronger statements are
better.  But, it might be that they are only epsilon better and so
ultimately it doesn't much matter.

Thanks,
allman




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Applying "the future will be different" stops all work.  Period.

allman



> > However, let me sketch the lay of the land as I see it ....   It
> > seems the meat in this document can be used for different purposes.
> > And, we have seen opinions on each of these I think.  The document coul=
d be used:
> >
> >  (a) as input to future RTO specifications,
> >
> >  (b) as input to future RTO implementations, or
> >
> >  (c) as both (a) and (b)
>=20
> In case there was any ambiguity:=20
>=20
> I do NOT support (b) or (c).
>=20
> Specifically, I do not understand how [rto-consider] can foresee
> the requirements of future protocols not standardized by the IETF
> right now. The Internet is subject to permanent change. I conclude
> that the requirements outlined in this document cannot be
> considered "safe" for implementations of protocols in future,
> unknown environments.=20
>=20
> The analysis what is "safe" has to be done on a per protocol
> basis, within the standardization process of that protocol, taking
> [rto-consider] as one input regarding the requirements in the
> global Internet at the time when the document was published.=20
>=20
> Michael (with chair hat off)




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> > 6298bis would have a new (first) section with a MUST that applies
> > the guidelines in [rto-consider] specifically to the TCP protocol,
> > mandating that all TCP implementations MUST use retransmission
> > timeout mechanisms matching this minimum set of requirements. This
> > is relatively straightforward given that [rto-consider] is rather
> > TCP-ish already. But for a number of requirements the actual TCP
> > terminology and technology could be used.
> >
> > And then the document would recommend with a MAY (or SHOULD) the
> > algorithm in 6298 as default.
> >
> > And I believe that such a proposed standard for TCP would be
> > closer to what implementations actually do.
>=20
> We have been around this before, but for the public record ...
>=20
> To me this is a bunch of work for, perhaps, epsilon gain.  We state
> some high level requirements.  But, then we have to re-state the
> high-level requirements after prepending a protocol name and only
> then can an implementation can actually leverage the requirements.

Implementers can read the requirements in rto-consider right now.

What implementers cannot leverage right now is a statement about IETF conse=
nsus that for any protocol standardized in the IETF, any RTO algorithm, in =
all circumstances is always "safe" when it meets the requirements under the=
 constraints stated in the document. And specifically implementers of proto=
cols other than TCP cannot rely on that because a document first has to pas=
s IETF last call and IESG approval before the IETF can state that the guida=
nce it is indeed "best current practice".

> Perhaps that forms a nice clean document progression.  But, man,
> what a bunch of busy work that has no value add...

If [rto-consider] is IETF consensus, IMHO a 6298bis document would be a no-=
brainer in TCPM. Maybe I miss something, but that would be a candidate for =
a WGLC directly on the -00 version, given that we already review [rto-consi=
der]. One added value of 6298bis is that it applies the generic wording in =
[rto-consider] specifically to the TCP. I believe that this is simpler to r=
ead for a TCP implementer than to reverse-engineer protocol-agnostic wordin=
g in a draft that gets longer and longer to be protocol-agnostic (and likel=
y will have to be further extended when it hits IETF last call)? And, as a =
side note, my formal concerns would be addressed, since TCPM would have a d=
ocument we can e.g. proceed on standards track.

So is your concern that nobody would invest cycles to author a 6298bis in T=
CPM? Or something equivalent e.g. for SCTP?

In summary, can we agree that if somebody volunteers to submit 6298bis, whi=
ch references [rto-consider], my proposed wording for [rto-consider] actual=
ly has some gain ("epsilon" or more). Thus, in summary my proposal just sor=
ts out everything in TCPM? Or what is then missing in TCPM?

Michael


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Subject: Re: [tsvwg] [tcpm] draft-ietf-tcpm-rto-consider-04: (R.3)
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Hi Mark,

just on this one point and then I go to sleep=E2=80=A6

> Am 17.06.2016 um 22:16 schrieb Mark Allman <mallman@icir.org>:
>=20
> But, what I am struggling with here is that if we make some MUSTs
> then we get a little something in return.  That sets an expectation.
> And, if the FooBar implementation doesn't follow that expectation
> and it causes problems then there is a lever to say "well, gee, we
> said you needed to do something and you didn't".  FooBar does not
> have to listen.  We cannot make them do something different.  But,
> we get a little lever.  So, the question in my mind is whether we
> get the same sort of leverage with your looser notion of "just state
> the requirements" and "use SHOULDs".  And, I am thinking not.  I.e.,
> we didn't say "do this" we said "do this unless you have a good
> reason" and FooBar might think they have a good reason (e.g., "helps
> my customers, so sorry about your's!").  We can still try to say
> "well, we strongly recommended ...", but we're in a bit lesser
> position.

So SHOULD is the right usage here, because what you actually want to say =
is, do it expect you have a good reason. Yes, not everybody knows how to =
understand SHOULD correctly and there are even different opinion in the =
IETF. And what a good reason is, is even harder to define. So from my =
point the only practical solution is not using MUST but providing as =
much explanation as possible on when the SHOULD might not need to be =
follow and when it clearly should be followed. That=E2=80=99s often also =
not easy, I know. But that's the only way to make your point, also for =
people who are maybe less familiar with the language used in RFCs. Btw. =
there is no requirement to use this normative language. Actually this is =
only meant to help people understanding a document and its requirements =
more quickly. But effectively the message should still be clear even if =
the normative, upper-letter notation is ignored.

Mirja=


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From: "Black, David" <david.black@emc.com>
To: "Eggert, Lars" <lars@netapp.com>, tsvwg WG <tsvwg@ietf.org>
Thread-Topic: [tsvwg] I-D Action: draft-ietf-tsvwg-rfc5405bis-14.txt
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Cc: Mark Allman <mallman@icir.org>
Subject: Re: [tsvwg] I-D Action: draft-ietf-tsvwg-rfc5405bis-14.txt
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> The main open issue in my book is what to do about Mark Allman's comment
> regarding scrapping the RTT/RTO text in favor of simply to (normatively, =
I guess)
> referring to draft-ietf-tcpm-rto-consider.

As draft shepherd, my strong guidance is not to do that because that TCPM d=
raft
does not and will not provide specific UDP guidance; that's the rfc5405bis =
draft's role.

I believe that the rfc5405bis RTT/RTO text is aligned with the TCPM RTO dra=
ft, but
this should be double-checked against the stable version of that draft that=
 emerges
from the ongoing discussion.

Thanks, --David

> -----Original Message-----
> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Eggert, Lars
> Sent: Friday, June 17, 2016 10:01 AM
> To: tsvwg WG
> Cc: Mark Allman
> Subject: Re: [tsvwg] I-D Action: draft-ietf-tsvwg-rfc5405bis-14.txt
>=20
> This revision attempts to address the various directorate and other revie=
w we
> have gotten.
>=20
> The main open issue in my book is what to do about Mark Allman's comment
> regarding scrapping the RTT/RTO text in favor of simply to (normatively, =
I guess)
> referring to draft-ietf-tcpm-rto-consider.
>=20
> Lars
>=20
> > On 2016-06-17, at 15:58, internet-drafts@ietf.org wrote:
> >
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts dire=
ctories.
> > This draft is a work item of the Transport Area Working Group of the IE=
TF.
> >
> >        Title           : UDP Usage Guidelines
> >        Authors         : Lars Eggert
> >                          Godred Fairhurst
> >                          Greg Shepherd
> > 	Filename        : draft-ietf-tsvwg-rfc5405bis-14.txt
> > 	Pages           : 57
> > 	Date            : 2016-06-17
> >
> > Abstract:
> >   The User Datagram Protocol (UDP) provides a minimal message-passing
> >   transport that has no inherent congestion control mechanisms.  This
> >   document provides guidelines on the use of UDP for the designers of
> >   applications, tunnels and other protocols that use UDP.  Congestion
> >   control guidelines are a primary focus, but the document also
> >   provides guidance on other topics, including message sizes,
> >   reliability, checksums, middlebox traversal, the use of ECN, DSCPs,
> >   and ports.
> >
> >   Because congestion control is critical to the stable operation of the
> >   Internet, applications and other protocols that choose to use UDP as
> >   an Internet transport must employ mechanisms to prevent congestion
> >   collapse and to establish some degree of fairness with concurrent
> >   traffic.  They may also need to implement additional mechanisms,
> >   depending on how they use UDP.
> >
> >   Some guidance is also applicable to the design of other protocols
> >   (e.g., protocols layered directly on IP or via IP-based tunnels),
> >   especially when these protocols do not themselves provide congestion
> >   control.
> >
> >   This document obsoletes RFC5405 and adds guidelines for multicast UDP
> >   usage.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/
> >
> > There's also a htmlized version available at:
> > https://tools.ietf.org/html/draft-ietf-tsvwg-rfc5405bis-14
> >
> > A diff from the previous version is available at:
> > https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rfc5405bis-14
> >
> >
> > Please note that it may take a couple of minutes from the time of submi=
ssion
> > until the htmlized version and diff are available at tools.ietf.org.
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >


From nobody Fri Jun 17 23:22:52 2016
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To: =?utf-8?Q?Mirja_K=C3=BChlewind?= <mirja.kuehlewind@tik.ee.ethz.ch>
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, tsvwg <tsvwg@ietf.org>, IETF AVTCore WG <avt@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Colin Perkins <csp@csperkins.org>
Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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I think we SHOULD NOT recommend to use ECN marks as inputs to a CB. See belo=
w:

> On 17 Jun 2016, at 16:02, Mirja K=C3=BChlewind <mirja.kuehlewind@tik.ee.et=
hz.ch> wrote:
>=20
> +1 to not use normative language here.
>=20
> However, please note that having a high level of ECN-CE marks (without any=
 losses) means that all packets were received correctly. This situation can e=
ven occurs without high delays (depending on the AQM used), which would just=
 mean the services works perfectly. Therefore for me CE marks are a perfect i=
nput signal for a congestion control loop (where the AQM tell the sender to t=
ake action - whatever that means).

We may in future figure out ways to do this to detect significant failure us=
ing a rate adaptive transport and ECN e.g.  Observing 100% CE marks or somet=
hing, for an RTP flow that is trying to send well below its peak rate decide=
d by CC -- but I think this is speculating at an algorithm and adding detail=
s here is not a good idea. Especially as AQM continues to evolve.

> But I=E2=80=99m less concerned than David about eventually ignoring it for=
 circuit breaker.
>=20
Agree. Loss is the measurement that a CB MUST respond to.

> In addition one point on something Magnus wrote earlier:
> "If the implementation only have circuit breaker, i.e. no full fledged con=
gestion controller and uses ECN, they can in worst case drive the buffer int=
o the overload regime where it starts dropping packets. =E2=80=9E
>=20
> I=E2=80=99m not sure about this case. ECN is an input signal for congestio=
n control. If you don=E2=80=99t use congestion control but only a circuit br=
eaker, you should probably not enable ECN. At least it not clear to me why y=
ou would enable it, and it's definitely not conform to the ECN spec. Probabl=
y we should say something about this in the draft...?
>=20
Agree, enabling ECN without a responsive CC is going to lead to trouble.

> Mirja
>=20
Gorry

>> Am 17.06.2016 um 16:03 schrieb Black, David <david.black@emc.com>:
>>=20
>> Colin,
>>=20
>>>>> ...  I view the current text as providing implementers with too much
>>>>> latitude to ignore ECN-CE marks (e.g., because an implementer doesn't
>>>>> want to think about this problem space in the first place).
>>>=20
>>> I agree, but the argument is that doing so is less harmful than deployin=
g a circuit
>>> breaker that triggers too often when ECN is used.
>>>=20
>>> I=E2=80=99m not sure I believe this argument, though, since it seems tha=
t any new AQM
>>> that applies ECN marks much more often than at present will have to cons=
ider
>>> backwards compatibility, to work with deployed TCP (e.g., draft-briscoe-=
tsvwg-
>>> aqm-tcpm-rmcat-l4s-problem uses ECT(1) as a signal to use the new markin=
g,
>>> while existing implementations set ECT(0)). These compatibility mechanis=
ms
>>> would seem to prevent the issues with the circuit breaker too.
>>=20
>> That roughly matches my line of thinking, and I'll observe that the origi=
nal DCTCP
>> protocol design that used more aggressive ECN-CE marking was only safe fo=
r
>> Controlled Environment deployments.   See the TSVWG rfc5405bis draft for t=
he
>> definition of Controlled Environment, and ignore the fact that the rfc540=
5bis
>> draft is a UDP draft - this definition is more broadly applicable.
>>=20
>> Going back over Section 7 in this avtcore draft, my views are:
>>=20
>> [A] None of these drafts justify a "MAY ignore" response to ECN-CE marks:=

>>    - draft-khademi-tcpm-alternativebackoff-ecn
>>    - draft-ietf-rmcat-nada
>>    - draft-ietf-rmcat-scream-cc
>>=20
>> [B] In line with Colin's comment on the L4S draft, I think it's incumbent=
 on
>> the authors of draft-briscoe-aqm-dualq-coupled to figure out how that wil=
l
>> coexist (or avoid) deployed TCP, and this avtcore draft ought not to be
>> trying to prejudge what will be done there.
>>=20
>> So, I don't think the current text in Section 7 has justified the unfette=
red
>> "implementations MAY ignore ECN-CE marks" text, as ignoring those marks
>> is not consistent with any of the four cited drafts.
>>=20
>> In more detail, I think making changes to normative requirements here bas=
ed
>> on [B] is premature, and I would hope that the rmcat WG could be encourag=
ed
>> to consider the RTP circuit breaker in its congestion control drafts, as t=
hose CC
>> mechanisms are related to the circuit breaker mechanism, hence likely
>> to be in related areas of an RTP implementation.
>>=20
>> That leaves draft-khademi-tcpm-alternativebackoff-ecn, which TSVWG
>> will be looking at in Berlin.  If a normative statement about ECN-CE reac=
tion
>> is going to rest on that draft, then the reference to that draft should b=
e
>> normative.  Something about doing that strikes me as premature ...
>>=20
>> I realize that we're trying to predict and accommodate the future, which
>> is an imprecise undertaking at best.   As an alternative to the current t=
ext,
>> would it be reasonable to say (without any RFC 2119 keywords) that the
>> best current guidance is still to treat ECN-CE marks as indicating drops,=

>> with a warning that there is a good possibility of this changing in the
>> near future due to all of the work in progress cited in Section 7?
>>=20
>> Thanks, --David
>>=20
>>> -----Original Message-----
>>> From: Colin Perkins [mailto:csp@csperkins.org]
>>> Sent: Friday, June 17, 2016 6:14 AM
>>> To: John Leslie; Black, David
>>> Cc: rtcweb@ietf.org; IETF AVTCore WG; tsvwg
>>> Subject: Re: [rtcweb] [AVTCORE] [tsvwg] WG Last Call on changes: draft-i=
etf-
>>> avtcore-rtp-circuit-breakers-16
>>>=20
>>>=20
>>>> On 16 Jun 2016, at 23:25, John Leslie <john@jlc.net> wrote:
>>>>=20
>>>> Black, David <david.black@emc.com> wrote:
>>>>>=20
>>>>> ...  I view the current text as providing implementers with too much
>>>>> latitude to ignore ECN-CE marks (e.g., because an implementer doesn't
>>>>> want to think about this problem space in the first place).
>>>=20
>>> I agree, but the argument is that doing so is less harmful than deployin=
g a circuit
>>> breaker that triggers too often when ECN is used.
>>>=20
>>> I=E2=80=99m not sure I believe this argument, though, since it seems tha=
t any new AQM
>>> that applies ECN marks much more often than at present will have to cons=
ider
>>> backwards compatibility, to work with deployed TCP (e.g., draft-briscoe-=
tsvwg-
>>> aqm-tcpm-rmcat-l4s-problem uses ECT(1) as a signal to use the new markin=
g,
>>> while existing implementations set ECT(0)). These compatibility mechanis=
ms
>>> would seem to prevent the issues with the circuit breaker too.
>>>=20
>>>> Understand, we have at least two proposals to make ECN-CE more frequent=

>>>> than packet drop would be for non-ECN packets: possibly substantially
>>>> more frequent. Unless both are killed off, ECN-CE will show up frequent=
ly
>>>> enough that closing the flow on ECN-CE would kill too many connections.=

>>>>=20
>>>> If you want circuit-breaking on such connections, there are two ways:
>>>> 1. convince the forwarding nodes to drop packets if their queue exceeds=

>>>> design capacity; or
>>>> 2. require the sender to send enough not-ECN-capable packets so that ou=
r
>>>> receiver will see enough packet-drops when a circuit-breaker should
>>>> activate.
>>>>=20
>>>> (I prefer the first option; but I wouldn't object to the second.)
>>>>=20
>>>> There really isn't any way for our circuit-breaker to know _how_much_
>>>> more frequent the ECN-CE marks may be. :^(
>>>=20
>>> This is a problem, both for the circuit breaker, and for the algorithms b=
eing
>>> defined in RMCAT. We do need some understanding what the expected markin=
g
>>> rates are likely to be, so congestion control and circuit breakers can b=
e defined.
>>>=20
>>>> We _will_ be sorry if we
>>>> allot the same frequency of CE packets as packet-drops to trigger the
>>>> circuit-breaker.
>>>>=20
>>>>> Could someone propose initial text to qualifies the current "MAY ignor=
e"
>>>>> statement?
>>>>=20
>>>> Essentially, for the second option, you might propose text to the
>>>> effect of:
>>>> ]
>>>> ] If too many ECN-CE packets are received, the sender SHOULD send some
>>>> ] not-ECN-capable packets to determine whether enough packets along the=

>>>> ] path are being dropped to justify activating our circuit-breaker.
>>>>=20
>>>> I=E2=80=99m not enthusiastic about adding that; but it would resolve th=
e issue.
>>>=20
>>> I=E2=80=99m not convinced this would work. The circuit breaker is lookin=
g at long term
>>> trends, and in order to have enough not-ECT packets to determine if it s=
hould
>>> trigger, you=E2=80=99d essentially have to run without ECN for some seco=
nds.
>>>=20
>>> --
>>> Colin Perkins
>>> https://csperkins.org/
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Mon Jun 20 05:50:33 2016
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From: "Takeshi Takahashi" <takeshi_takahashi@nict.go.jp>
To: "'Eggert, Lars'" <lars@netapp.com>
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Date: Mon, 20 Jun 2016 21:50:32 +0900
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Cc: draft-ietf-tsvwg-rfc5405bis.all@ietf.org, tsvwg@ietf.org, iesg@ietf.org, secdir@ietf.org
Subject: Re: [tsvwg] Secdir review of draft-ietf-tsvwg-rfc5405bis-13
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Hi,

> > Also I have seen several typos (especially, missing parentheses around
> > referenced section numbers) on this document, so please revise the
> > texts before the publication of this document.
> 
> No typos come up when using "idnits --spellcheck". I also don't quite
> understand what you mean by "missing parentheses around referenced section
> numbers" - could you please clarify?

These typos are very trivial, but let me list them here.
Since I'm not a native speaker (writer) of English, so I might be wrong, but
I hope you could take a look at them.
I was reading version 13 of the draft, and version 14 (that was published a
couple of days ago) already have fixed most of the typo I found.

1. in Section 3.1.7,

Old) "requires for"
New) "required for" (already fixed in version 14)

2. in Section 3.1.7,

Old) "... by the UDP congestion control method Section 3.1 that is not less
than the reaction of TCP..."

It could be helpful if you could use parentheses, e.g., "...control method
(see Section 3.1) that..." or "...control method described in Section 3.1)
that..."

3. in Section 3.1.8

Old) "across multiple networks the between ..."
New) "across multiple networks between ..." (already fixed in version 14)

4. in Section 3.1.9

Old) "into unprovisioned Internet paths (e.g., [RFC7510]) To protect"

The sentence might need a period in between; i.e., "into unprovisioned
Internet paths (e.g., [RFC7510]). To protect"

5. in Section 7

Old) "for tunnels carrying IP Traffic,"

It could be a non-capital letter; i.e., "for tunnels carrying IP traffic,"

I like the summary table inserted in Section 7.
Thank you for preparing the summary table.

Kind regards,
Take



> -----Original Message-----
> From: Eggert, Lars [mailto:lars@netapp.com]
> Sent: Friday, June 17, 2016 9:39 PM
> To: Takeshi Takahashi
> Cc: iesg@ietf.org; secdir@ietf.org;
> draft-ietf-tsvwg-rfc5405bis.all@ietf.org; tsvwg@ietf.org
> Subject: Re: Secdir review of draft-ietf-tsvwg-rfc5405bis-13
> 
> Hi,
> 
> thanks for the review! I'll incorporate respective changes into -14.
> 
> On 2016-05-31, at 12:06, Takeshi Takahashi <takeshi_takahashi@nict.go.jp>
> wrote:
> > In Table 1 "Summary of recommendations", I wonder if the corresponding
> > section numbers are correct.
> >
> > [Now]
> > "SHOULD avoid using multiple ports"  corresponds to Section 5.1 and
> > "SHOULD use a randomized source port or equivalent technique"
> > corresponds to Section 5.2
> >
> > [New]
> > "SHOULD avoid using multiple ports"  corresponds to Section 5.1.1 and
> > "SHOULD use a randomized source port or equivalent technique"
> > corresponds to Section 5.1.2
> >
> > I might be wrong, so please check.
> 
> You were right that what we had was wrong (copy/paste bug, I think), but
> they pointers should be to Sections 5.1.2 and 6, respectively.
> 
> > Also I have seen several typos (especially, missing parentheses around
> > referenced section numbers) on this document, so please revise the
> > texts before the publication of this document.
> 
> No typos come up when using "idnits --spellcheck". I also don't quite
> understand what you mean by "missing parentheses around referenced section
> numbers" - could you please clarify?
> 
> Thanks,
> Lars


From nobody Mon Jun 20 06:17:43 2016
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From: "Black, David" <david.black@emc.com>
To: "Gorry (erg)" <gorry@erg.abdn.ac.uk>, =?utf-8?B?TWlyamEgS8O8aGxld2luZA==?= <mirja.kuehlewind@tik.ee.ethz.ch>
Thread-Topic: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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References: <ccf9f2d7-2694-4336-0ec9-ccfebfeb0120@ericsson.com> <CE03DB3D7B45C245BCA0D243277949362F585D3E@MX307CL04.corp.emc.com> <d97e30a7-70f5-26d0-c3a4-0497c669f5f6@ericsson.com> <CE03DB3D7B45C245BCA0D243277949362F586054@MX307CL04.corp.emc.com> <D19E595F-7C66-4AE9-92B4-D550A93F634D@csperkins.org> <CE03DB3D7B45C245BCA0D243277949362F589335@MX307CL04.corp.emc.com> <20160616222548.GB77166@verdi> <0643E158-BF26-4692-8167-B7A959CB20CE@csperkins.org> <CE03DB3D7B45C245BCA0D243277949362F596DBC@MX307CL04.corp.emc.com> <E16BEA87-1D0F-48F1-A9AC-2729079D581D@tik.ee.ethz.ch> <8C16F1C6-B4A7-4BB4-B215-D7E7EAF308F8@erg.abdn.ac.uk>
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, tsvwg <tsvwg@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Colin Perkins <csp@csperkins.org>, IETF AVTCore WG <avt@ietf.org>
Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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To: "Scharf\, Michael \(Nokia - DE\)" <michael.scharf@nokia.com>
From: Mark Allman <mallman@icir.org>
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Subject: Re: [tsvwg] [tcpm] draft-ietf-tcpm-rto-consider-04: (R.3)
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> If [rto-consider] is IETF consensus, IMHO a 6298bis document would
> be a no-brainer in TCPM.

But, it isn't necessary.  RFC 6298 is a document about *one*
algorithm.  We could invent parallel algorithms.  And, write an RFC
that specifies each.  RFC 6298 just happens to be the only RTO we
have taken the bother to develop and write up.  So, RFC 6298 (or a
bis) doesn't have to convey all of our TCP RTO understanding.  It is
just one algorithm for doing it.

So, yeah, we can write a preamble to RFC 6298 that says "this agrees
with [rto-consider]", but that doesn't change the algorithm.  It
isn't a technical change at all.  It is more words in service of
what?  A tidy---by some definition---document progression is all I
can come up with.  It doesn't advance things in any meaningful way.

> One added value of 6298bis is that it applies the
> generic wording in [rto-consider] specifically to the TCP. I
> believe that this is simpler to read for a TCP implementer than to
> reverse-engineer protocol-agnostic wording in a draft that gets
> longer and longer to be protocol-agnostic (and likely will have to
> be further extended when it hits IETF last call)?

The whole thing hasn't really gotten much longer.  And, it is still
quite short.  And, I don't think that someone steeped in TCP will
have a difficult time reading the generic form of the requirements
there now.  These are not particularly foreign concepts we're
talking about.

> So is your concern that nobody would invest cycles to author a
> 6298bis in TCPM? Or something equivalent e.g. for SCTP?=20

My concern is for doing work that just doesn't need to be done.  How
many times do you want to re-state the same thing?  TCP, SCTP.  What
about UDP?  I doubt you'll find the notions in 5405bis to be quite
right because they apply the rto-consider requirements to all
UDP-based protocols.  And, so, I think all the same arguments you're
making now would have to be applied there, too, and so we'll have to
go and write N more of these things for N UDP-based apps.  That's
saying the same thing a lot.  And, God forbid if we ever had to
change something...

The point is that except for document series tidiness I have seen no
reason to make these requirements only apply to future specs---and
not implementations---as you advocate.

allman




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From: "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>
To: "mallman@icir.org" <mallman@icir.org>
Thread-Topic: [tsvwg] [tcpm] draft-ietf-tcpm-rto-consider-04: (R.3) 
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> > If [rto-consider] is IETF consensus, IMHO a 6298bis document would be=20
> > a no-brainer in TCPM.

> But, it isn't necessary.  RFC 6298 is a document about *one* algorithm.

This is not how I read RFC 6298. It already has a MUST in the abstract:

   This document defines the standard algorithm that Transmission
   Control Protocol (TCP) senders are required to use to compute and
   manage their retransmission timer.  It expands on the discussion in
   Section 4.2.3.1 of RFC 1122 and upgrades the requirement of
   supporting the algorithm from a SHOULD to a MUST.  This document
   obsoletes RFC 2988.

RFC 6298 does mention that other algorithms could exist, with the following=
 wording:

   In some situations, it may be beneficial for a TCP sender to be more
   conservative than the algorithms detailed in this document allow.
   However, a TCP MUST NOT be more aggressive than the following
   algorithms allow.  This document obsoletes RFC 2988 [PA00].

But RFC 6298 does not have the notion of *one* algorithm.

So, in my reading, [rto-consider] contradicts quite explicitly the standard=
s track specification of TCP. What do I miss here?

In my view RFC 6298 needs to be updated if TCPM believes that a TCP impleme=
ntation fully compliant to IETF standards should be allowed to have other a=
lgorithms.

As mentioned already, I would support publishing a 6298bis for that purpose=
. I believe that it is important that TCPM standards track documents are (1=
) consistent and (2) well aligned with the major TCP implementations.

The following to me is a very straightforward and clean approach:

* [rto-consider] provides the general protocol-agnostic requirements, as sp=
ecified in the wording of my review

* 5405bis applies it to UDP

* 6298bis applies it TCP and cleanly updates 6298 on standards track

* SCTP should be a low-hanging fruit as well

* Other protocols owned by other working groups have to review [rto-conside=
r] and come to an own consensus anyway

And, possibly even more importantly, reading protocol-specific guidance sho=
uld typically also be simpler for implementers since they don't have to gue=
ss how to apply generic wording to the specific protocol semantics.

In short, I am not aware of any relevant downside of that very clean approa=
ch.

Michael


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From: "Black, David" <david.black@emc.com>
To: "mallman@icir.org" <mallman@icir.org>, "Scharf, Michael (Nokia - DE)" <michael.scharf@nokia.com>
Thread-Topic: [tsvwg] [tcpm] draft-ietf-tcpm-rto-consider-04: (R.3)
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Addressing solely this UDP item ...

>  What about UDP?  I doubt you'll find the notions in 5405bis to be quite
> right because they apply the rto-consider requirements to all
> UDP-based protocols.

The desired structure here (IMHO) is that the TCPM rto-consider draft
states the general principles, and the TSVWG rfc5405bis draft applies them
to UDP.  Then individual UDP-based protocol drafts are expected to follow
the guidance in rfc5405bis, or explain why there are good reasons for
doing otherwise.

> And, so, I think all the same arguments you're
> making now would have to be applied there, too, and so we'll have to
> go and write N more of these things for N UDP-based apps. =20

Not exactly - any UDP-based protocol RFC-to-be is going to have to
reference rfc5405bis, as I'd expect the TSV ADs to Discuss drafts that
don't ;-).  The resulting RTO considerations will show up in each main
protocol draft, not a small separate stand-alone draft.

In practice, we'll probably see more cross-area interaction to head off
this class of issue before it hits the IESG.  FWIW, draft-ietf-forces-inter=
felfb
is a recently-worked example in the area of congestion considerations,
and I do want to thank the responsible RTG AD (Alia) for noticing that
the congestion concerns could use a bit more attention, and encouraging
the draft authors to address them in advance of the draft going to the IESG=
.

Thanks, --David

> -----Original Message-----
> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Mark Allman
> Sent: Monday, June 20, 2016 10:45 AM
> To: Scharf, Michael (Nokia - DE)
> Cc: tcpm@ietf.org; tsvwg@ietf.org
> Subject: Re: [tsvwg] [tcpm] draft-ietf-tcpm-rto-consider-04: (R.3)
>=20
>=20
> > If [rto-consider] is IETF consensus, IMHO a 6298bis document would
> > be a no-brainer in TCPM.
>=20
> But, it isn't necessary.  RFC 6298 is a document about *one*
> algorithm.  We could invent parallel algorithms.  And, write an RFC
> that specifies each.  RFC 6298 just happens to be the only RTO we
> have taken the bother to develop and write up.  So, RFC 6298 (or a
> bis) doesn't have to convey all of our TCP RTO understanding.  It is
> just one algorithm for doing it.
>=20
> So, yeah, we can write a preamble to RFC 6298 that says "this agrees
> with [rto-consider]", but that doesn't change the algorithm.  It
> isn't a technical change at all.  It is more words in service of
> what?  A tidy---by some definition---document progression is all I
> can come up with.  It doesn't advance things in any meaningful way.
>=20
> > One added value of 6298bis is that it applies the
> > generic wording in [rto-consider] specifically to the TCP. I
> > believe that this is simpler to read for a TCP implementer than to
> > reverse-engineer protocol-agnostic wording in a draft that gets
> > longer and longer to be protocol-agnostic (and likely will have to
> > be further extended when it hits IETF last call)?
>=20
> The whole thing hasn't really gotten much longer.  And, it is still
> quite short.  And, I don't think that someone steeped in TCP will
> have a difficult time reading the generic form of the requirements
> there now.  These are not particularly foreign concepts we're
> talking about.
>=20
> > So is your concern that nobody would invest cycles to author a
> > 6298bis in TCPM? Or something equivalent e.g. for SCTP?
>=20
> My concern is for doing work that just doesn't need to be done.  How
> many times do you want to re-state the same thing?  TCP, SCTP.  What
> about UDP?  I doubt you'll find the notions in 5405bis to be quite
> right because they apply the rto-consider requirements to all
> UDP-based protocols.  And, so, I think all the same arguments you're
> making now would have to be applied there, too, and so we'll have to
> go and write N more of these things for N UDP-based apps.  That's
> saying the same thing a lot.  And, God forbid if we ever had to
> change something...
>=20
> The point is that except for document series tidiness I have seen no
> reason to make these requirements only apply to future specs---and
> not implementations---as you advocate.
>=20
> allman
>=20
>=20


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Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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> On 20. jun. 2016, at 15.16, Black, David <david.black@emc.com> wrote:
>=20
>>> But I=E2=80=99m less concerned than David about eventually ignoring =
it for circuit
>> breaker.
>>>=20
>> Agree. Loss is the measurement that a CB MUST respond to.
>=20
> Mumble.   I would be ok with a clear discouragement for use of ECN-CE =
marks, accompanied by the sort of design rationale here, or even better, =
a clear statement that lost packets for the purpose of the RTP circuit =
breaker have to be actually lost without getting into whether or not =
ECN-CE marks are involved -i.e., the RTP circuit breaker is specified =
against actual drops as a network protection backstop.
>=20
> A related concern is that ECN marks may overstate equivalent loss =
behavior - a simplistic queue management discipline that marks every =
packet when the queue is over a threshold (NB: this class of marking =
behavior is NOT RECOMMENDED - a real AQM SHOULD be used) could yield a =
run of ECN-CE marks that would not cause a corresponding with a run of =
packet drops.   This is among the reasons that TCP reacts to ECN-CE =
marks only once per RTT, and might be a reason to treat multiple ECN-CE =
marks in an RTT interval as not representing drops of all packets for =
the RTP circuit breaker's TCP-equivalent throughput calculation.

I=E2=80=99m not sure we need such complicated logic to find a case where =
ECN marks are different from packet drops:

Basically, they simply aren=E2=80=99t - even =E2=80=9Creal=E2=80=9D AQMs =
marking isn=E2=80=99t exactly the same as a packet drop: the marks =
themselves inform you that an AQM did its job, and with modern AQMs like =
CoDel / PIE etc., you=E2=80=99re probably getting this from a shallow =
queue. Chances are that this is less than a BDP worth of queuing, which =
is our justification for recommending a different back-off behavior in =
draft-khademi-tsvwg-ecn-response-00 and =
draft-khademi-tcpm-alternativebackoff-ecn-00

So the point is not that AQMs would treat ECN marking and dropping =
differently - it=E2=80=99s that ECN indicates an AQM, and hence probably =
a shallow queue. With a drop, you just don=E2=80=99t know.

Back to the CB, I think an AQM marking at a shallow queue (like e.g. =
CoDel) is indeed quite different from a =E2=80=9Cbroken connection=E2=80=9D=
.

Cheers,
Michael


>=20
> Thanks, --David
>=20
>> -----Original Message-----
>> From: Gorry (erg) [mailto:gorry@erg.abdn.ac.uk]
>> Sent: Saturday, June 18, 2016 2:23 AM
>> To: Mirja K=C3=BChlewind
>> Cc: Black, David; Magnus Westerlund; Colin Perkins; rtcweb@ietf.org; =
IETF
>> AVTCore WG; tsvwg
>> Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: =
draft-ietf-
>> avtcore-rtp-circuit-breakers-16
>>=20
>> I think we SHOULD NOT recommend to use ECN marks as inputs to a CB. =
See
>> below:
>>=20
>>> On 17 Jun 2016, at 16:02, Mirja K=C3=BChlewind =
<mirja.kuehlewind@tik.ee.ethz.ch>
>> wrote:
>>>=20
>>> +1 to not use normative language here.
>>>=20
>>> However, please note that having a high level of ECN-CE marks =
(without any
>> losses) means that all packets were received correctly. This =
situation can even
>> occurs without high delays (depending on the AQM used), which would =
just
>> mean the services works perfectly. Therefore for me CE marks are a =
perfect input
>> signal for a congestion control loop (where the AQM tell the sender =
to take action
>> - whatever that means).
>>=20
>> We may in future figure out ways to do this to detect significant =
failure using a
>> rate adaptive transport and ECN e.g.  Observing 100% CE marks or =
something, for
>> an RTP flow that is trying to send well below its peak rate decided =
by CC -- but I
>> think this is speculating at an algorithm and adding details here is =
not a good idea.
>> Especially as AQM continues to evolve.
>>=20
>>> But I=E2=80=99m less concerned than David about eventually ignoring =
it for circuit
>> breaker.
>>>=20
>> Agree. Loss is the measurement that a CB MUST respond to.
>>=20
>>> In addition one point on something Magnus wrote earlier:
>>> "If the implementation only have circuit breaker, i.e. no full =
fledged congestion
>> controller and uses ECN, they can in worst case drive the buffer into =
the overload
>> regime where it starts dropping packets. =E2=80=9E
>>>=20
>>> I=E2=80=99m not sure about this case. ECN is an input signal for =
congestion control. If you
>> don=E2=80=99t use congestion control but only a circuit breaker, you =
should probably not
>> enable ECN. At least it not clear to me why you would enable it, and =
it's definitely
>> not conform to the ECN spec. Probably we should say something about =
this in the
>> draft...?
>>>=20
>> Agree, enabling ECN without a responsive CC is going to lead to =
trouble.
>>=20
>>> Mirja
>>>=20
>> Gorry
>>=20
>>>> Am 17.06.2016 um 16:03 schrieb Black, David <david.black@emc.com>:
>>>>=20
>>>> Colin,
>>>>=20
>>>>>>> ...  I view the current text as providing implementers with too =
much
>>>>>>> latitude to ignore ECN-CE marks (e.g., because an implementer =
doesn't
>>>>>>> want to think about this problem space in the first place).
>>>>>=20
>>>>> I agree, but the argument is that doing so is less harmful than =
deploying a
>> circuit
>>>>> breaker that triggers too often when ECN is used.
>>>>>=20
>>>>> I=E2=80=99m not sure I believe this argument, though, since it =
seems that any new
>> AQM
>>>>> that applies ECN marks much more often than at present will have =
to
>> consider
>>>>> backwards compatibility, to work with deployed TCP (e.g., =
draft-briscoe-
>> tsvwg-
>>>>> aqm-tcpm-rmcat-l4s-problem uses ECT(1) as a signal to use the new =
marking,
>>>>> while existing implementations set ECT(0)). These compatibility =
mechanisms
>>>>> would seem to prevent the issues with the circuit breaker too.
>>>>=20
>>>> That roughly matches my line of thinking, and I'll observe that the =
original
>> DCTCP
>>>> protocol design that used more aggressive ECN-CE marking was only =
safe for
>>>> Controlled Environment deployments.   See the TSVWG rfc5405bis =
draft for
>> the
>>>> definition of Controlled Environment, and ignore the fact that the =
rfc5405bis
>>>> draft is a UDP draft - this definition is more broadly applicable.
>>>>=20
>>>> Going back over Section 7 in this avtcore draft, my views are:
>>>>=20
>>>> [A] None of these drafts justify a "MAY ignore" response to ECN-CE =
marks:
>>>>   - draft-khademi-tcpm-alternativebackoff-ecn
>>>>   - draft-ietf-rmcat-nada
>>>>   - draft-ietf-rmcat-scream-cc
>>>>=20
>>>> [B] In line with Colin's comment on the L4S draft, I think it's =
incumbent on
>>>> the authors of draft-briscoe-aqm-dualq-coupled to figure out how =
that will
>>>> coexist (or avoid) deployed TCP, and this avtcore draft ought not =
to be
>>>> trying to prejudge what will be done there.
>>>>=20
>>>> So, I don't think the current text in Section 7 has justified the =
unfettered
>>>> "implementations MAY ignore ECN-CE marks" text, as ignoring those =
marks
>>>> is not consistent with any of the four cited drafts.
>>>>=20
>>>> In more detail, I think making changes to normative requirements =
here based
>>>> on [B] is premature, and I would hope that the rmcat WG could be
>> encouraged
>>>> to consider the RTP circuit breaker in its congestion control =
drafts, as those CC
>>>> mechanisms are related to the circuit breaker mechanism, hence =
likely
>>>> to be in related areas of an RTP implementation.
>>>>=20
>>>> That leaves draft-khademi-tcpm-alternativebackoff-ecn, which TSVWG
>>>> will be looking at in Berlin.  If a normative statement about =
ECN-CE reaction
>>>> is going to rest on that draft, then the reference to that draft =
should be
>>>> normative.  Something about doing that strikes me as premature ...
>>>>=20
>>>> I realize that we're trying to predict and accommodate the future, =
which
>>>> is an imprecise undertaking at best.   As an alternative to the =
current text,
>>>> would it be reasonable to say (without any RFC 2119 keywords) that =
the
>>>> best current guidance is still to treat ECN-CE marks as indicating =
drops,
>>>> with a warning that there is a good possibility of this changing in =
the
>>>> near future due to all of the work in progress cited in Section 7?
>>>>=20
>>>> Thanks, --David
>>>>=20
>>>>> -----Original Message-----
>>>>> From: Colin Perkins [mailto:csp@csperkins.org]
>>>>> Sent: Friday, June 17, 2016 6:14 AM
>>>>> To: John Leslie; Black, David
>>>>> Cc: rtcweb@ietf.org; IETF AVTCore WG; tsvwg
>>>>> Subject: Re: [rtcweb] [AVTCORE] [tsvwg] WG Last Call on changes: =
draft-ietf-
>>>>> avtcore-rtp-circuit-breakers-16
>>>>>=20
>>>>>=20
>>>>>> On 16 Jun 2016, at 23:25, John Leslie <john@jlc.net> wrote:
>>>>>>=20
>>>>>> Black, David <david.black@emc.com> wrote:
>>>>>>>=20
>>>>>>> ...  I view the current text as providing implementers with too =
much
>>>>>>> latitude to ignore ECN-CE marks (e.g., because an implementer =
doesn't
>>>>>>> want to think about this problem space in the first place).
>>>>>=20
>>>>> I agree, but the argument is that doing so is less harmful than =
deploying a
>> circuit
>>>>> breaker that triggers too often when ECN is used.
>>>>>=20
>>>>> I=E2=80=99m not sure I believe this argument, though, since it =
seems that any new
>> AQM
>>>>> that applies ECN marks much more often than at present will have =
to
>> consider
>>>>> backwards compatibility, to work with deployed TCP (e.g., =
draft-briscoe-
>> tsvwg-
>>>>> aqm-tcpm-rmcat-l4s-problem uses ECT(1) as a signal to use the new =
marking,
>>>>> while existing implementations set ECT(0)). These compatibility =
mechanisms
>>>>> would seem to prevent the issues with the circuit breaker too.
>>>>>=20
>>>>>> Understand, we have at least two proposals to make ECN-CE more
>> frequent
>>>>>> than packet drop would be for non-ECN packets: possibly =
substantially
>>>>>> more frequent. Unless both are killed off, ECN-CE will show up =
frequently
>>>>>> enough that closing the flow on ECN-CE would kill too many =
connections.
>>>>>>=20
>>>>>> If you want circuit-breaking on such connections, there are two =
ways:
>>>>>> 1. convince the forwarding nodes to drop packets if their queue =
exceeds
>>>>>> design capacity; or
>>>>>> 2. require the sender to send enough not-ECN-capable packets so =
that our
>>>>>> receiver will see enough packet-drops when a circuit-breaker =
should
>>>>>> activate.
>>>>>>=20
>>>>>> (I prefer the first option; but I wouldn't object to the second.)
>>>>>>=20
>>>>>> There really isn't any way for our circuit-breaker to know =
_how_much_
>>>>>> more frequent the ECN-CE marks may be. :^(
>>>>>=20
>>>>> This is a problem, both for the circuit breaker, and for the =
algorithms being
>>>>> defined in RMCAT. We do need some understanding what the expected
>> marking
>>>>> rates are likely to be, so congestion control and circuit breakers =
can be
>> defined.
>>>>>=20
>>>>>> We _will_ be sorry if we
>>>>>> allot the same frequency of CE packets as packet-drops to trigger =
the
>>>>>> circuit-breaker.
>>>>>>=20
>>>>>>> Could someone propose initial text to qualifies the current "MAY =
ignore"
>>>>>>> statement?
>>>>>>=20
>>>>>> Essentially, for the second option, you might propose text to the
>>>>>> effect of:
>>>>>> ]
>>>>>> ] If too many ECN-CE packets are received, the sender SHOULD send =
some
>>>>>> ] not-ECN-capable packets to determine whether enough packets =
along the
>>>>>> ] path are being dropped to justify activating our =
circuit-breaker.
>>>>>>=20
>>>>>> I=E2=80=99m not enthusiastic about adding that; but it would =
resolve the issue.
>>>>>=20
>>>>> I=E2=80=99m not convinced this would work. The circuit breaker is =
looking at long term
>>>>> trends, and in order to have enough not-ECT packets to determine =
if it
>> should
>>>>> trigger, you=E2=80=99d essentially have to run without ECN for =
some seconds.
>>>>>=20
>>>>> --
>>>>> Colin Perkins
>>>>> https://csperkins.org/
>>>>=20
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From nobody Mon Jun 20 09:52:59 2016
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Subject: Re: [tsvwg] [tcpm] draft-ietf-tcpm-rto-consider-04: (R.3)
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> So, in my reading, [rto-consider] contradicts quite explicitly the
> standards track specification of TCP. What do I miss here?=20

"Contradict" ... maybe.  In the strict sense of the word.  What it
says is that we have learned things and figured out that only a
subset of 6298 is necessary for network safety and so we can offer
flexibility in the areas that are not as important for network
safety.  But, it doesn't alter or call into question the algorithm
in 6298 and folks can readily use it as they always have.  But,
deviation from the precise details is also in some cases quite fine,
as well.

The usual thing we do is to learn and update things.  We increase
the initial cwnd recommendation (which obsoletes the previous
recommendation).  Or, we refine SACK-based loss recovery (which
obsoletes the old scheme with new thinking).  Or, whatnot.  But, in
this case we are saying simultaneously that (a) we can relax on the
details and (b) the details of this one existing spec are still
valid and fine.  This is different than our Usual Case, I think.
But, since we are not taking something and making it different or
better this is not the Usual Case.  We are saying is we have
learned:

  - that there are a class of specific RTO mechanisms that are safe
    and fine,

  - that there is no one-size-fits-all in the class and=20

  - that the class includes the old thing so we cannot simply
    replace it

Instead of honing our view or changing our view we are broadening
our view.  And, this is (I guess) difficult.

On the rest, I could reply.  But, we know where each other stands at
this point and you and I don't need to continue to circle.  We
simply disagree.

allman




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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 Transport Area Working Group of the IETF.

        Title           : Diffserv-Interconnection classes and practice
        Authors         : Ruediger Geib
                          David L. Black
	Filename        : draft-ietf-tsvwg-diffserv-intercon-06.txt
	Pages           : 21
	Date            : 2016-06-20

Abstract:
   This document defines a limited common set of Diffserv PHBs and
   codepoints (DSCPs) to be applied at (inter)connections of two
   separately administered and operated networks, and explains how this
   approach can simplify network configuration and operation.  Many
   network providers operate MPLS using Treatment Aggregates for traffic
   marked with different Diffserv PHBs, and use MPLS for interconnection
   with other networks.  This document offers a simple interconnection
   approach that may simplify operation of Diffserv for network
   interconnection among providers that use MPLS and apply the Short-
   Pipe tunnel mode.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-diffserv-intercon/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-tsvwg-diffserv-intercon-06

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-diffserv-intercon-06


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

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


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From: <Ruediger.Geib@telekom.de>
To: <david.black@emc.com>
Thread-Topic: [tsvwg] WG adoption call: draft-szigeti-tsvwg-ieee-802-11 - through	June 25
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David,

I support adaption.

Regards, Ruediger


-----Urspr=FCngliche Nachricht-----
Von: tsvwg [mailto:tsvwg-bounces@ietf.org] Im Auftrag von Black, David
Gesendet: Donnerstag, 9. Juni 2016 07:14
An: tsvwg@ietf.org
Betreff: [tsvwg] WG adoption call: draft-szigeti-tsvwg-ieee-802-11 - throug=
h June 25

This is a TSVWG call for interest in WG adoption of draft-szigeti-tsvwg-iee=
e-802-11.
If you have read the draft and support WG adoption of it as a work item and=
 WG draft, please send an email to the list (tsvwg@ietf.org) saying so.

NB: It is important to say that you've read the draft.  If you're willing t=
o review/comment on future versions of this draft and/or implement this dra=
ft, please say so on the list or in email directly to the chairs (tsvwg-cha=
irs@ietf.org).

Authors: We (chairs) assume that you support WG adoption of your draft, so =
there's no need to say so on the list.  However, we do need each author to =
publicly state on the list that their direct, personal knowledge (if any) o=
f any IPR related to this document has already been disclosed, in conforman=
ce with BCPs 78 and 79.

This adoption call will run for about 2 weeks, ending at midnight, prevaili=
ng US Eastern Time on Saturday June 25.  To get things started, I hereby st=
ate as an individual that I have read the draft and support WG adoption of =
it.

             Guidelines for DiffServ to IEEE 802.11 Mapping
                   draft-szigeti-tsvwg-ieee-802-11-01

Abstract

   As internet traffic is increasingly sourced-from and destined-to
   wireless endpoints, it is crucial that Quality of Service be aligned
   between wired and wireless networks; however, this is not always the
   case by default.  This is due to the fact that two independent
   standards bodies provide QoS guidance on wired and wireless networks:
   specifically, the IETF specifies standards and design recommendations
   for wired IP networks, while a separate and autonomous standards-
   body, the IEEE, administers the standards for wireless 802.11
   networks.  The purpose of this document is to propose a set
   Differentiated Services Code Point (DSCP) to IEEE 802.11 User
   Priority (UP) mappings to reconcile the marking recommendations
   offered by these two standards bodies, and, as such, to optimize
   wired-and-wireless interconnect QoS.

Thanks, --David
----------------------------------------------------
David L. Black, Distinguished Engineer
EMC, 176 South St., Hopkinton, MA=A0 01748
+1 (508) 293-7953=A0=A0=A0  Cell: +1 (978) 394-7754
david.black@emc.com
----------------------------------------------------



From nobody Thu Jun 23 07:14:00 2016
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Subject: [tsvwg] The TSVWG WG has placed draft-szigeti-tsvwg-ieee-802-11 in state "Call For Adoption By WG Issued"
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The TSVWG WG has placed draft-szigeti-tsvwg-ieee-802-11 in state 
Call For Adoption By WG Issued (entered by Gorry Fairhurst)

The document is available at
https://datatracker.ietf.org/doc/draft-szigeti-tsvwg-ieee-802-11/


Comment:
To end June 25th 2016


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Subject: [tsvwg] WGLC for draft-ietf-tsvwg-diffserv-intercon - to end 10th July 2016
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This email announces a TSVWG Working Group Last Call (WGLC) on:

            Diffserv-Interconnection classes and practice
                     Intended status: Informational
https://tools.ietf.org/html/draft-ietf-tsvwg-diffserv-intercon-06

This WGLC will run for 2 weeks, ending on 10th July 2016. The results
of the WGLC will be presented at the Berlin IETF meeting.

Comments should be sent to the tsvwg@ietf.org list.   Please cc: the WG
chairs at tsvwg-chairs@ietf.org to track your comments as part of the WGLC
process.

As part of this WGLC, comments are requested on the usability of this
draft, and the relation to published RFCs.

Thanks, Gorry
(TSVWG Co-Chair)



From nobody Thu Jun 23 07:47:12 2016
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Date: Thu, 23 Jun 2016 15:47:07 +0100
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Subject: [tsvwg] Please tell us if you have new work or would like to present a draft in TSVWG
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 Hi tsvwg'ers,

This is a request for people to offer presentations on drafts they would
like to be presented to the tsvwg working group at the Berlin IETF.

Priority will be assigned to time that progresses WG drafts, to bring
individual drafts to the attention of the WG and (in least order of
priority) to describe new work relevant to the TSVWG Charter.

Please send all requests and any corrections to the chairs
<tsvwg-chairs@ietf.org> asap, and we'll publish a preliminary meeting
agenda.

Gorry and David
(tsvwg co-chairs)


From nobody Thu Jun 23 09:38:42 2016
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Black, David" <david.black@emc.com>
Thread-Topic: [tsvwg] WG adoption call: draft-szigeti-tsvwg-ieee-802-11 - through June 25
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Subject: Re: [tsvwg] WG adoption call: draft-szigeti-tsvwg-ieee-802-11 - through June 25
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The authors and I discussed some changes to address all of my comments on t=
his draft.

I strongly support adoptions of this draft - the industry needs some common=
 default behavior on theses mappings and we need them sooner rather than la=
ter.


On Jun 8, 2016, at 11:14 PM, Black, David <david.black@emc.com> wrote:

This is a TSVWG call for interest in WG adoption of draft-szigeti-tsvwg-iee=
e-802-11.
If you have read the draft and support WG adoption of it as a work item
and WG draft, please send an email to the list (tsvwg@ietf.org) saying so.

NB: It is important to say that you've read the draft.  If you're willing
to review/comment on future versions of this draft and/or implement this
draft, please say so on the list or in email directly to the chairs
(tsvwg-chairs@ietf.org).


--_000_FA73E76812D64FD4967184F9FC2766ADciscocom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body class=3D"ApplePlainTextBody" dir=3D"auto" style=3D"word-wrap: break-w=
ord; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">
<br>
The authors and I discussed some changes to address all of my comments on t=
his draft.
<br>
<br>
I strongly support adoptions of this draft - the industry needs some common=
 default behavior on theses mappings and we need them sooner rather than la=
ter.
<br>
<br>
<br>
<blockquote type=3D"cite">On Jun 8, 2016, at 11:14 PM, Black, David &lt;dav=
id.black@emc.com&gt; wrote:<br>
<br>
This is a TSVWG call for interest in WG adoption of draft-szigeti-tsvwg-iee=
e-802-11.<br>
If you have read the draft and support WG adoption of it as a work item<br>
and WG draft, please send an email to the list (tsvwg@ietf.org) saying so.<=
br>
<br>
NB: It is important to say that you've read the draft. &nbsp;If you're will=
ing<br>
to review/comment on future versions of this draft and/or implement this<br=
>
draft, please say so on the list or in email directly to the chairs<br>
(tsvwg-chairs@ietf.org).<br>
</blockquote>
<br>
</body>
</html>

--_000_FA73E76812D64FD4967184F9FC2766ADciscocom_--


From nobody Thu Jun 23 20:34:06 2016
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To: tsvwg@ietf.org
References: <d72081e2edc7bca0275854c24a567715.squirrel@erg.abdn.ac.uk>
From: Brian E Carpenter <brian.e.carpenter@gmail.com>
Organization: University of Auckland
Message-ID: <124e9c95-2cb4-d9ee-5555-5de6e5d94874@gmail.com>
Date: Fri, 24 Jun 2016 15:34:10 +1200
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Cc: tsvwg-chairs@ietf.org
Subject: Re: [tsvwg] WGLC for draft-ietf-tsvwg-diffserv-intercon - to end 10th July 2016
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I've been tracking this document since its inception. IMHO it is ready for
publication and should be of considerable value to operators, especially
if RTCWEB gets off the ground. I think it correctly fits in with the other
diffserv-related RFCs.

Regards
   Brian Carpenter


On 24/06/2016 02:39, gorry@erg.abdn.ac.uk wrote:
> 
> This email announces a TSVWG Working Group Last Call (WGLC) on:
> 
>             Diffserv-Interconnection classes and practice
>                      Intended status: Informational
> https://tools.ietf.org/html/draft-ietf-tsvwg-diffserv-intercon-06
> 
> This WGLC will run for 2 weeks, ending on 10th July 2016. The results
> of the WGLC will be presented at the Berlin IETF meeting.
> 
> Comments should be sent to the tsvwg@ietf.org list.   Please cc: the WG
> chairs at tsvwg-chairs@ietf.org to track your comments as part of the WGLC
> process.
> 
> As part of this WGLC, comments are requested on the usability of this
> draft, and the relation to published RFCs.
> 
> Thanks, Gorry
> (TSVWG Co-Chair)
> 
> 
> 


From nobody Fri Jun 24 05:17:15 2016
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From: Michael Welzl <michawe@ifi.uio.no>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/tsvwg/ghqmMDhezxQq1tfaRiZKxWFBjPM>
Subject: [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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Dear all,

We recently did DSCP related measurements that lead us to propose a =
change to draft-ietf-tsvwg-rtcweb-qos.
These measurements are reported in this paper:
=
http://heim.ifi.uio.no/~michawe/research/publications/anrw16-measurement.p=
df
... which will be presented at the ANRW workshop, the Saturday before =
the IETF meeting.

The paper reports on tests between people's homes and 3 servers that we =
ran - one in Oregon (amazon cloud) and two in Norway. This gave us 185 =
distinct paths (IP address pairs). The test was a TCP handshake, done =
with raw sockets, and we played around with values in the IP header. =
Meanwhile, Runa (main author) did some more tests from various public =
places during a trip from Norway to India, essentially confirming the =
finding we discuss here; we'll ask for a slot to report about both the =
data in the ANRW paper and the newer data in the MAPRG session.

The number of data points isn't exactly huge in both cases, but still =
relevant, we believe: it is about a persistent failure that we saw on =
different paths, and so we think it's probably reasonable to take it =
into account (given that a fix shouldn't be very complicated either).

We used, as input, DSCP values from the table in =
draft-ietf-tsvwg-rtcweb-qos.
Here are some sentences copy + pasted from our paper:

***
"To minimize the chance that congestion-based drops make us believe in a =
failure
to communicate using certain values in the IP header, we re-tried failed =
packet
exchanges up to three times, and we sent an ICMP packet just ahead of =
every
measurement packet. We only assumed a communication failure when the =
test
failed three times and the ICMP packet succeeded."

"Considering table 1 in [6], we assumed the flow type =E2=80=9CInteractive=
 Video with or
without Audio=E2=80=9D with application priorities =E2=80=9CVery low=E2=80=
=9D (DSCP value CS1 (8)), =E2=80=9CLow=E2=80=9D
(DF (0)) or =E2=80=9CMedium=E2=80=9D (AF42 (36)), and flow type =
=E2=80=9CAudio=E2=80=9D with application prioritiy
=E2=80=9CHigh=E2=80=9D (EF PHB (46))."

"We did see consistent packet drops too: on 23 paths for DSCP value 8, =
21 paths
for 36, 19 for 46."
***

We also saw DSCP value changes, but nothing very alarming there - in one =
case, AF42 was changed to AF41, which is even nice  :)     however, this =
consistent drop behavior that appears only when using a DSCP value !=3D =
0 isn't good at all. Now these things could in principle have happened =
close to our 3 servers, meaning there were only few devices that did it, =
but in fact this measurement was a part of a larger measurement campaign =
in which we also checked where such packet drops typically happened. The =
result was: most such drops seem to be close to the client, which may =
indicate that there were in fact multiple such boxes.

We believe that draft-ietf-tsvwg-rtcweb-qos should say: if setting the =
DSCP values as proposed here consistently produces packet drops, senders =
should fall back to DSCP 0. WebRTC shouldn't "fail" because of the DSCP.

Cheers,
Michael


From nobody Fri Jun 24 09:09:30 2016
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Subject: [tsvwg] tsvwg - Requested sessions have been scheduled for IETF 96
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Dear Gorry Fairhurst,

The session(s) that you have requested have been scheduled.
Below is the scheduled session information followed by
the original request. 

tsvwg Session 1 (2:00:00)
    Friday, Morning Session I 1000-1200
    Room Name: Bellevue size: 200
    ---------------------------------------------
    tsvwg Session 2 (1:30:00)
    Wednesday, Afternoon Session II 1550-1720
    Room Name: Charlottenburg II/III size: 175
    ---------------------------------------------
    


Request Information:


---------------------------------------------------------
Working Group Name: Transport Area Working Group
Area Name: Transport Area
Session Requester: Gorry Fairhurst

Number of Sessions: 2
Length of Session(s):  2 Hours, 1.5 Hours
Number of Attendees: 75
Conflicts to Avoid: 
 First Priority: tsvarea iccrg aqm rmcat rtcweb tcpm mptcp nvo3 dclcrg taps ippm opsawg opsarea intarea
 Second Priority: v6ops tram tcpinc nfsv4 dispatch
 Third Priority: ccamp artarea precis


Special Requests:
  [If rtcweb has 2 sessions, 1 must not conflict] 
[Must not conflict with Transport Area BoFs]

---------------------------------------------------------


From nobody Fri Jun 24 13:25:15 2016
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To: Michael Welzl <michawe@ifi.uio.no>, tsvwg@ietf.org
References: <CB087237-108A-44B1-8293-3498F24A2303@ifi.uio.no>
From: Brian E Carpenter <brian.e.carpenter@gmail.com>
Organization: University of Auckland
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Archived-At: <https://mailarchive.ietf.org/arch/msg/tsvwg/THrhKbZb3AZQbMKgQkAkkJeoE50>
Subject: Re: [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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> We believe that draft-ietf-tsvwg-rtcweb-qos should say: if setting the =
DSCP values as proposed here consistently produces packet drops, senders =
should fall back to DSCP 0. WebRTC shouldn't "fail" because of the DSCP.

This is an interesting and worrying result indeed, but it's in no way spe=
cific
to WebRTC. I think a generic RFC ("Happy Eyeballs with Differentiated Ser=
vices"?)
would be the way to go. But (as with a lot of recent discussion about IPv=
6
extension headers causing packet drops) we really need to understand why =
this
happens. Is it traffic policing that fails to reclassify and re-mark pack=
ets,
misguided firewalls, or what?

Regards
   Brian

On 25/06/2016 00:17, Michael Welzl wrote:
> Dear all,
>=20
> We recently did DSCP related measurements that lead us to propose a cha=
nge to draft-ietf-tsvwg-rtcweb-qos.
> These measurements are reported in this paper:
> http://heim.ifi.uio.no/~michawe/research/publications/anrw16-measuremen=
t.pdf
> ... which will be presented at the ANRW workshop, the Saturday before t=
he IETF meeting.
>=20
> The paper reports on tests between people's homes and 3 servers that we=
 ran - one in Oregon (amazon cloud) and two in Norway. This gave us 185 d=
istinct paths (IP address pairs). The test was a TCP handshake, done with=
 raw sockets, and we played around with values in the IP header. Meanwhil=
e, Runa (main author) did some more tests from various public places duri=
ng a trip from Norway to India, essentially confirming the finding we dis=
cuss here; we'll ask for a slot to report about both the data in the ANRW=
 paper and the newer data in the MAPRG session.
>=20
> The number of data points isn't exactly huge in both cases, but still r=
elevant, we believe: it is about a persistent failure that we saw on diff=
erent paths, and so we think it's probably reasonable to take it into acc=
ount (given that a fix shouldn't be very complicated either).
>=20
> We used, as input, DSCP values from the table in draft-ietf-tsvwg-rtcwe=
b-qos.
> Here are some sentences copy + pasted from our paper:
>=20
> ***
> "To minimize the chance that congestion-based drops make us believe in =
a failure
> to communicate using certain values in the IP header, we re-tried faile=
d packet
> exchanges up to three times, and we sent an ICMP packet just ahead of e=
very
> measurement packet. We only assumed a communication failure when the te=
st
> failed three times and the ICMP packet succeeded."
>=20
> "Considering table 1 in [6], we assumed the flow type =E2=80=9CInteract=
ive Video with or
> without Audio=E2=80=9D with application priorities =E2=80=9CVery low=E2=
=80=9D (DSCP value CS1 (8)), =E2=80=9CLow=E2=80=9D
> (DF (0)) or =E2=80=9CMedium=E2=80=9D (AF42 (36)), and flow type =E2=80=9C=
Audio=E2=80=9D with application prioritiy
> =E2=80=9CHigh=E2=80=9D (EF PHB (46))."
>=20
> "We did see consistent packet drops too: on 23 paths for DSCP value 8, =
21 paths
> for 36, 19 for 46."
> ***
>=20
> We also saw DSCP value changes, but nothing very alarming there - in on=
e case, AF42 was changed to AF41, which is even nice  :)     however, thi=
s consistent drop behavior that appears only when using a DSCP value !=3D=
 0 isn't good at all. Now these things could in principle have happened c=
lose to our 3 servers, meaning there were only few devices that did it, b=
ut in fact this measurement was a part of a larger measurement campaign i=
n which we also checked where such packet drops typically happened. The r=
esult was: most such drops seem to be close to the client, which may indi=
cate that there were in fact multiple such boxes.
>=20
> We believe that draft-ietf-tsvwg-rtcweb-qos should say: if setting the =
DSCP values as proposed here consistently produces packet drops, senders =
should fall back to DSCP 0. WebRTC shouldn't "fail" because of the DSCP.
>=20
> Cheers,
> Michael
>=20
>=20


From nobody Fri Jun 24 13:36:50 2016
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Subject: [tsvwg] I-D Action: draft-szigeti-tsvwg-ieee-802-11-02.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 Transport Area Working Group of the IETF.

        Title           : Guidelines for DiffServ to IEEE 802.11 Mapping
        Authors         : Tim Szigeti
                          Fred Baker
	Filename        : draft-szigeti-tsvwg-ieee-802-11-02.txt
	Pages           : 27
	Date            : 2016-06-24

Abstract:
   As internet traffic is increasingly sourced-from and destined-to
   wireless endpoints, it is crucial that Quality of Service be aligned
   between wired and wireless networks; however, this is not always the
   case by default.  This is due to the fact that two independent
   standards bodies provide QoS guidance on wired and wireless networks:
   specifically, the IETF specifies standards and design recommendations
   for wired IP networks, while a separate and autonomous standards-
   body, the IEEE, administers the standards for wireless 802.11
   networks.  The purpose of this document is to propose a set
   Differentiated Services Code Point (DSCP) to IEEE 802.11 User
   Priority (UP) mappings to reconcile the marking recommendations
   offered by these two standards bodies, and, as such, to optimize
   wired-and-wireless interconnect QoS.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-szigeti-tsvwg-ieee-802-11/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-szigeti-tsvwg-ieee-802-11-02

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-szigeti-tsvwg-ieee-802-11-02


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

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


From nobody Fri Jun 24 13:52:25 2016
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To: Brian E Carpenter <brian.e.carpenter@gmail.com>
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Subject: Re: [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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Hi,


> On 24. jun. 2016, at 22.25, Brian E Carpenter =
<brian.e.carpenter@gmail.com> wrote:
>=20
>> We believe that draft-ietf-tsvwg-rtcweb-qos should say: if setting =
the DSCP values as proposed here consistently produces packet drops, =
senders should fall back to DSCP 0. WebRTC shouldn't "fail" because of =
the DSCP.
>=20
> This is an interesting and worrying result indeed, but it's in no way =
specific
> to WebRTC. I think a generic RFC ("Happy Eyeballs with Differentiated =
Services"?)
> would be the way to go.

Well, if you use the DSCP within your own administrative domain for =
DiffServ, you should be safe=E2=80=A6. so I guess this problem mostly =
matters for usage across domains, or even end-to-end.
=3D> It seems to me that it=E2=80=99s broader than WebRTC in exactly the =
same way as RFC7657 is broader than WebRTC.=20

As a side note, I think the term =E2=80=9CHappy Eyeballs=E2=80=9D is =
misleading here. This term, to me, indicates parallel usage of multiple =
packet types to see which one(s) succeed(s) and then try to make the =
best choice.
This isn=E2=80=99t what you would do here - we=E2=80=99re talking about =
a rare failure situation. It=E2=80=99s really only something I=E2=80=99d =
suggest as a fall-back, upon seeing communication failing entirely, over =
a longer period (perhaps in line with a circuit-breaker).


> But (as with a lot of recent discussion about IPv6
> extension headers causing packet drops) we really need to understand =
why this
> happens. Is it traffic policing that fails to reclassify and re-mark =
packets,
> misguided firewalls, or what?

I have no clue=E2=80=A6 but as my previous email said, we have =
indications that this behavior occurred close to the senders (most of =
them being in people=E2=80=99s homes), hence it=E2=80=99s more likely =
that this was done by a middlebox or an edge router than by a core =
router.
That=E2=80=99s as much as we know...

Cheers,
Michael


>=20
> Regards
>   Brian
>=20
> On 25/06/2016 00:17, Michael Welzl wrote:
>> Dear all,
>>=20
>> We recently did DSCP related measurements that lead us to propose a =
change to draft-ietf-tsvwg-rtcweb-qos.
>> These measurements are reported in this paper:
>> =
http://heim.ifi.uio.no/~michawe/research/publications/anrw16-measurement.p=
df
>> ... which will be presented at the ANRW workshop, the Saturday before =
the IETF meeting.
>>=20
>> The paper reports on tests between people's homes and 3 servers that =
we ran - one in Oregon (amazon cloud) and two in Norway. This gave us =
185 distinct paths (IP address pairs). The test was a TCP handshake, =
done with raw sockets, and we played around with values in the IP =
header. Meanwhile, Runa (main author) did some more tests from various =
public places during a trip from Norway to India, essentially confirming =
the finding we discuss here; we'll ask for a slot to report about both =
the data in the ANRW paper and the newer data in the MAPRG session.
>>=20
>> The number of data points isn't exactly huge in both cases, but still =
relevant, we believe: it is about a persistent failure that we saw on =
different paths, and so we think it's probably reasonable to take it =
into account (given that a fix shouldn't be very complicated either).
>>=20
>> We used, as input, DSCP values from the table in =
draft-ietf-tsvwg-rtcweb-qos.
>> Here are some sentences copy + pasted from our paper:
>>=20
>> ***
>> "To minimize the chance that congestion-based drops make us believe =
in a failure
>> to communicate using certain values in the IP header, we re-tried =
failed packet
>> exchanges up to three times, and we sent an ICMP packet just ahead of =
every
>> measurement packet. We only assumed a communication failure when the =
test
>> failed three times and the ICMP packet succeeded."
>>=20
>> "Considering table 1 in [6], we assumed the flow type =E2=80=9CInteract=
ive Video with or
>> without Audio=E2=80=9D with application priorities =E2=80=9CVery =
low=E2=80=9D (DSCP value CS1 (8)), =E2=80=9CLow=E2=80=9D
>> (DF (0)) or =E2=80=9CMedium=E2=80=9D (AF42 (36)), and flow type =
=E2=80=9CAudio=E2=80=9D with application prioritiy
>> =E2=80=9CHigh=E2=80=9D (EF PHB (46))."
>>=20
>> "We did see consistent packet drops too: on 23 paths for DSCP value =
8, 21 paths
>> for 36, 19 for 46."
>> ***
>>=20
>> We also saw DSCP value changes, but nothing very alarming there - in =
one case, AF42 was changed to AF41, which is even nice  :)     however, =
this consistent drop behavior that appears only when using a DSCP value =
!=3D 0 isn't good at all. Now these things could in principle have =
happened close to our 3 servers, meaning there were only few devices =
that did it, but in fact this measurement was a part of a larger =
measurement campaign in which we also checked where such packet drops =
typically happened. The result was: most such drops seem to be close to =
the client, which may indicate that there were in fact multiple such =
boxes.
>>=20
>> We believe that draft-ietf-tsvwg-rtcweb-qos should say: if setting =
the DSCP values as proposed here consistently produces packet drops, =
senders should fall back to DSCP 0. WebRTC shouldn't "fail" because of =
the DSCP.
>>=20
>> Cheers,
>> Michael
>>=20
>>=20
>=20


From nobody Sat Jun 25 07:12:51 2016
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Date: Sat, 25 Jun 2016 15:06:59 +0100
From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: [tsvwg] My comments on  draft-ietf-tcpm-rto-consider-04
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https://tools.ietf.org/html/draft-ietf-tcpm-rto-consider-04

I have previously made many comments on the text in rev-03 of this 
draft, and after a careful read, I'd like to now note that in large 
part, rev -04 addresses my concerns regarding scope and applicability.

There still appears to be a global scope for the document as “ SHOULD” 
in (R.2), and then a nested “MUST” within that “SHOULD”. I see others 
have also noted this, but I'd prefer an AD or someone with more insight 
into RFC2119 to decide on this.

Likewise, I understand TCPM needs to decide upon the set of requirements 
for TCP, and TSVWG for SCTP and UDP.

That said, I'd thank Mark for his heavy work to make this more 
applicable. A small number of comments follow.

Gorry

--- Comments on draft-ietf-tcpm-rto-consider-04

Section 1.

This sentence is rather cumbersome to read:
OLD:
However, despite our best intentions and most robust mechanisms, the 
only thing we can truly depend on is the passage of time and therefore 
our ultimate backstop to ensuring reliability is a timeout and re-try 
mechanism.

Is it worth considering reordering the thoughts, say:
NEW:
However, despite our best intentions and most robust mechanisms, the 
ultimate backstop to ensuring reliability is a timeout and re-try 
mechanism, because the only thing we can truly depend on is the passage 
of time.
—
/a for/
OLD:
protocol-agnostic requirements for RTO mechanisms that provide a for 
network safety.
NEW:
OLD:
protocol-agnostic requirements for RTO mechanisms that provide network 
safety.
——
Section 2:
NiT: Is it worth replacing “must” with needs to, or deciding this 
actually is a requirement and then using RFC2119 MUST. I’d be happy 
either way, but don’t think “MUST” fits in this section either.
OLD:
independently. I.e., if host A communicates with hosts B and C, A must 
use independent RTOs for traffic sent to B and C.
NEW:
independently. I.e., if host A communicates with hosts B and C, A needs 
to use independent RTOs for traffic sent to B and C.
—
OLD:
"like UDP-based DNS []"
- missing reference.
---
OLD:
than the 1 second specified in [RFC6298]. While the implication of
these deviations from the standard may be more spurious retransmits
(per [AP99]), we are aware of no large scale problems caused by this
change to the minimum RTO.”

- I do not see the wisdom in the last clause. As an example there have 
been issues (albeit a time ago with a RTO less than 1 second being the 
default - I think in,one version of Solaris) that resulted in TCP having 
much degraded throughput over paths with a RTT greater than the RTO. A 
small initial RTO may still be a problem (e.g., when traffic after a 
failure is routed over a backup path), and I suggest this is not a case 
we should advocate. Indeed, this ID currently makes a requirement that: 
"RTO MUST be conservatively set to no less than 1 second”, which to me 
seems a correct interpretation of the RFC-series, so why does the 
sentence above appear to suggest otherwise?
---



From nobody Sun Jun 26 07:52:01 2016
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From: Hagen Paul Pfeifer <hagen@jauu.net>
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Subject: [tsvwg] Linux TCP RTO
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For the fellows not constantly following the Linux kernel maillinglist: we are
currently try to adjust Linux's TCP RTO behavior to implement a RFC 6298
compliant calculation[1]:

	https://www.mail-archive.com/netdev@vger.kernel.org/msg114487.html


We wrote an small document where we highlighted the current Linux
implementation, validated our 6298 compliant implementation and presented
measurements comparing both mechanisms:

	https://docs.google.com/document/d/1pKmPfnQb6fDK4qpiNVkN8cQyGE4wYDZukcuZfR-BnnM


Yuchung tested the modification in large scale at Googles web servers (HTTP,
HTTP2) and first results looks promising:

	https://www.mail-archive.com/netdev@vger.kernel.org/msg116052.html


Hagen


[1] some aspects of Linux differs too much to implement a 100% compliant RFC
6298 mechanism - especially the MinRTO definition differs.


From nobody Mon Jun 27 00:34:44 2016
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Thread-Topic: [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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Subject: Re: [tsvwg] [rtcweb] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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AVT, RTCWEB, TSVWG and ADS.

I have to conclude that there is not consensus on the proposed text. It 
does appear that this subject do require additional discussion and most 
likely drafting of new text.

Please continue to provide input into the matter. I would like to 
conclude this subject no later than by the end of the Berlin meeting.

Cheers

Magnus

Den 2016-06-14 kl. 10:14, skrev Magnus Westerlund:
> AVT, RTCWEB and TSVWG
>
> This announces an additional one week WG last call (in AVT) on the
> changes done to "Multimedia Congestion Control: Circuit Breakers for
> Unicast RTP Sessions" that is intended for Proposed Standard. The last
> call concludes on the 21st of June.
>
> So this last call is deemed necessary due to normative changes in the
> behaviour as result of the conclusions of how to resolve the Discuss
> regarding reaction to ECN markings.
>
> If there are no issues in this additional WG last call, the draft will
> be approved for publication as it has already passed IESG review.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-avtcore-rtp-circuit-breakers/
>
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-avtcore-rtp-circuit-breakers-16
>
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-rtp-circuit-breakers-16
>
>
> Cheers
>
> Magnus Westerlund
> AVTCORE WG chair
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> Färögatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From nobody Mon Jun 27 07:42:46 2016
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Subject: Re: [tsvwg] [rtcweb] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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Thanks, Magnus!

On 27 Jun 2016, at 8:46, Magnus Westerlund wrote:

> AVT, RTCWEB, TSVWG and ADS.
>
> I have to conclude that there is not consensus on the proposed text. 
> It does appear that this subject do require additional discussion and 
> most likely drafting of new text.
>
> Please continue to provide input into the matter. I would like to 
> conclude this subject no later than by the end of the Berlin meeting.
>
> Cheers
>
> Magnus
>
> Den 2016-06-14 kl. 10:14, skrev Magnus Westerlund:
>> AVT, RTCWEB and TSVWG
>>
>> This announces an additional one week WG last call (in AVT) on the
>> changes done to "Multimedia Congestion Control: Circuit Breakers for
>> Unicast RTP Sessions" that is intended for Proposed Standard. The 
>> last
>> call concludes on the 21st of June.
>>
>> So this last call is deemed necessary due to normative changes in the
>> behaviour as result of the conclusions of how to resolve the Discuss
>> regarding reaction to ECN markings.
>>
>> If there are no issues in this additional WG last call, the draft 
>> will
>> be approved for publication as it has already passed IESG review.
>>
>>
>> The IETF datatracker status page for this draft is:
>> https://datatracker.ietf.org/doc/draft-ietf-avtcore-rtp-circuit-breakers/
>>
>> There's also a htmlized version available at:
>> https://tools.ietf.org/html/draft-ietf-avtcore-rtp-circuit-breakers-16
>>
>> A diff from the previous version is available at:
>> https://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-rtp-circuit-breakers-16
>>
>>
>> Cheers
>>
>> Magnus Westerlund
>> AVTCORE WG chair
>>
>> ----------------------------------------------------------------------
>> Services, Media and Network features, Ericsson Research EAB/TXM
>> ----------------------------------------------------------------------
>> Ericsson AB                 | Phone  +46 10 7148287
>> Färögatan 6                 | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
> -- 
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> Färögatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------


From nobody Mon Jun 27 07:57:58 2016
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Subject: Re: [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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Dear Ruediger,

Thanks a lot for your interest!
More below:

> On 27 Jun 2016, at 09:34, Ruediger.Geib@telekom.de wrote:
>=20
> Hi Michael,
>=20
> Thanks, an interesting test set up.
>=20
> The first IP layer policy point in a network may be the (fixed =
network) BRAS or another node terminating and authorizing network =
access. I can't tell you exactly how these are configured in the network =
of the company I work for, but I think it is rewrite with CS0 and =
transport. I know that the peering routers are configured that way.
>=20
> Your doc says:=20
> Marking packets as AF42 (36) provoked another undefined value (4) on =
14 paths, .... Value 46 (EF PHB), on the other hand, was turned into 6 =
=E2=80=93 yet another undefined value =E2=80=93 on 17 paths.=20
>=20
> This looks like a so called "IP precedence" rewrite: 100100 is =
rewritten to 000100 and 101110 to 000110. It's possible that 001000 was =
rewritten to 000000 that way too in a similar number of cases.
> I favor standardizing this behavior, as it is a typical and simple =
MPLS edge node configuration. =20

Interesting, thanks!


> In Germany, layer 2 interconnection agreements between national =
providers resulted in a recommendation to drop packets with unexpected =
Ethernet priority markings at carrier interconnection. I dislike this =
idea and commented against it, but I wasn't directly involved in =
negotiations.
>=20
> =46rom experience in rolling out new services, there seem to be German =
network operators dropping packets with unexpected DSCPs at IP =
interconnections.

This is very interesting - and I agree it's the wrong thing to do  ("be =
liberal in what you accept" ...).

Runa Barik (the Ph.D. student whose work this is) is planning to give a =
presentation to MAPRG about this, where he'll also talk about a new set =
of measurements that he did during a trip from Norway to India. There, =
he saw the same problem, and he also carried out traceroute-type =
measurements with the changed packets to see where it happened - with =
the result that it often is *inside* the network, i.e. not just 2-3 hops =
or so away from the client  (which is where many other packet drops =
happened). Even though it's a very small datapoint, it would confirm =
what you say about the location of this problem.


> I couldn't read from your doc whether you differentiated between =
fixed, WiFi and LTE/3G access. Please add this information, as policies =
may differ depending on the type of access. I think to recall that e.g. =
DPI was detected more often in wireless networks when EU regulation =
probed for that (no, I don't think one needs DPI to detect an unexpected =
DSCP - my point is to indicate that there differences in policy in =
general).

Sorry: we didn't - but it's good advice, we will incorporate this in the =
future!


Cheers,
Michael


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From nobody Mon Jun 27 08:41:41 2016
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References: <ccf9f2d7-2694-4336-0ec9-ccfebfeb0120@ericsson.com> <CE03DB3D7B45C245BCA0D243277949362F585D3E@MX307CL04.corp.emc.com> <d97e30a7-70f5-26d0-c3a4-0497c669f5f6@ericsson.com> <CE03DB3D7B45C245BCA0D243277949362F586054@MX307CL04.corp.emc.com> <D19E595F-7C66-4AE9-92B4-D550A93F634D@csperkins.org> <CE03DB3D7B45C245BCA0D243277949362F589335@MX307CL04.corp.emc.com> <20160616222548.GB77166@verdi> <0643E158-BF26-4692-8167-B7A959CB20CE@csperkins.org> <CE03DB3D7B45C245BCA0D243277949362F596DBC@MX307CL04.corp.emc.com> <E16BEA87-1D0F-48F1-A9AC-2729079D581D@tik.ee.ethz.ch> <8C16F1C6-B4A7-4BB4-B215-D7E7EAF308F8@erg.abdn.ac.uk> <CE03DB3D7B45C245BCA0D243277949362F59C41D@MX307CL04.corp.emc.com> <3E053A65-2698-4749-8E3D-E0451DF84011@ifi.uio.no> <BF6B00CC65FD2D45A326E74492B2C19FB76A6433@FR711WXCHMBA05.zeu.alcatel-lucent.com>
Date: Mon, 27 Jun 2016 16:38:16 +0100
From: gorry@erg.abdn.ac.uk
To: "De Schepper, Koen (Nokia - BE)" <koen.de_schepper@nokia-bell-labs.com>
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Cc: gorry@erg.abdn.ac.uk, Magnus Westerlund <magnus.westerlund@ericsson.com>, tsvwg <tsvwg@ietf.org>, Michael Welzl <michawe@ifi.uio.no>, IETF AVTCore WG <avt@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Colin Perkins <csp@csperkins.org>
Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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I think thinking of L4S is maybe off at a tangent. The question really is
about the interpretation of loss and CE-mark as equivalent. I argued that 
each ECN-CE mark should not be counted as equivalent to a lost segment -
in this context we should use ECN to drive a CC algorithm and we should be
cautious to avoid requiring its use within a Circuit Breaker - optional
use, if you understand how to interpret a reaction to many CE-marks as
excessive congestion, are permitted.

Gorry

> As far as I understand, this draft is related to circuit breakers in
> end-systems, right?
>
> It is the end system that determines the use of ECN (currently marking
> non-ect for drop and ect(0) for Classic ECN).
>
> In L4S we don't plan to change the behavior of Classic ECN, and ABE's
> behavior should be close to non-ABE ECN. So I guess there is no problem of
> describing the behavior of how a Classic ECN based sender would respond
> today.
>
> As we only want to significantly change the network behavior of ect(1)
> marking, can we solve this issue by recommending (or even requiring)
> senders to mark only ect(0) and describing the classic ECN circuit
> breaker? When L4S gets defined, also an L4S based circuit breaker
> extension can be defined for senders that want to use the L4S service
> (when senders send ect(1) packets).
>
> Regards,
> Koen.
>
>
>> -----Original Message-----
>> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Michael Welzl
>> Sent: maandag 20 juni 2016 18:36
>> To: Black, David
>> Cc: <gorry@erg.abdn.ac.uk> Fairhurst; Magnus Westerlund; tsvwg; IETF
>> AVTCore WG; rtcweb@ietf.org; Colin Perkins
>> Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-
>> ietf-avtcore-rtp-circuit-breakers-16
>>
>>
>> > On 20. jun. 2016, at 15.16, Black, David <david.black@emc.com> wrote:
>> >
>> >>> But I’m less concerned than David about eventually ignoring it
>> for
>> circuit
>> >> breaker.
>> >>>
>> >> Agree. Loss is the measurement that a CB MUST respond to.
>> >
>> > Mumble.   I would be ok with a clear discouragement for use of ECN-CE
>> marks, accompanied by the sort of design rationale here, or even
>> better,
>> a clear statement that lost packets for the purpose of the RTP circuit
>> breaker have to be actually lost without getting into whether or not
>> ECN-CE marks are involved -i.e., the RTP circuit breaker is specified
>> against actual drops as a network protection backstop.
>> >
>> > A related concern is that ECN marks may overstate equivalent loss
>> behavior - a simplistic queue management discipline that marks every
>> packet when the queue is over a threshold (NB: this class of marking
>> behavior is NOT RECOMMENDED - a real AQM SHOULD be used) could yield a
>> run of ECN-CE marks that would not cause a corresponding with a run of
>> packet drops.   This is among the reasons that TCP reacts to ECN-CE
>> marks only once per RTT, and might be a reason to treat multiple ECN-CE
>> marks in an RTT interval as not representing drops of all packets for
>> the RTP circuit breaker's TCP-equivalent throughput calculation.
>>
>> I’m not sure we need such complicated logic to find a case where ECN
>> marks are different from packet drops:
>>
>> Basically, they simply aren’t - even “real” AQMs marking isn’t
>> exactly
>> the same as a packet drop: the marks themselves inform you that an AQM
>> did its job, and with modern AQMs like CoDel / PIE etc., you’re
>> probably
>> getting this from a shallow queue. Chances are that this is less than a
>> BDP worth of queuing, which is our justification for recommending a
>> different back-off behavior in draft-khademi-tsvwg-ecn-response-00 and
>> draft-khademi-tcpm-alternativebackoff-ecn-00
>>
>> So the point is not that AQMs would treat ECN marking and dropping
>> differently - it’s that ECN indicates an AQM, and hence probably a
>> shallow queue. With a drop, you just don’t know.
>>
>> Back to the CB, I think an AQM marking at a shallow queue (like e.g.
>> CoDel) is indeed quite different from a “broken connection”.
>>
>> Cheers,
>> Michael
>>
>>
>> >
>> > Thanks, --David
>> >
>> >> -----Original Message-----
>> >> From: Gorry (erg) [mailto:gorry@erg.abdn.ac.uk]
>> >> Sent: Saturday, June 18, 2016 2:23 AM
>> >> To: Mirja Kühlewind
>> >> Cc: Black, David; Magnus Westerlund; Colin Perkins; rtcweb@ietf.org;
>> IETF
>> >> AVTCore WG; tsvwg
>> >> Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes:
>> draft-ietf-
>> >> avtcore-rtp-circuit-breakers-16
>> >>
>> >> I think we SHOULD NOT recommend to use ECN marks as inputs to a CB.
>> See
>> >> below:
>> >>
>> >>> On 17 Jun 2016, at 16:02, Mirja Kühlewind
>> <mirja.kuehlewind@tik.ee.ethz.ch>
>> >> wrote:
>> >>>
>> >>> +1 to not use normative language here.
>> >>>
>> >>> However, please note that having a high level of ECN-CE marks
>> (without any
>> >> losses) means that all packets were received correctly. This
>> situation can even
>> >> occurs without high delays (depending on the AQM used), which would
>> just
>> >> mean the services works perfectly. Therefore for me CE marks are a
>> perfect input
>> >> signal for a congestion control loop (where the AQM tell the sender
>> to take action
>> >> - whatever that means).
>> >>
>> >> We may in future figure out ways to do this to detect significant
>> failure using a
>> >> rate adaptive transport and ECN e.g.  Observing 100% CE marks or
>> something, for
>> >> an RTP flow that is trying to send well below its peak rate decided
>> by CC -- but I
>> >> think this is speculating at an algorithm and adding details here is
>> not a good idea.
>> >> Especially as AQM continues to evolve.
>> >>
>> >>> But I’m less concerned than David about eventually ignoring it
>> for
>> circuit
>> >> breaker.
>> >>>
>> >> Agree. Loss is the measurement that a CB MUST respond to.
>> >>
>> >>> In addition one point on something Magnus wrote earlier:
>> >>> "If the implementation only have circuit breaker, i.e. no full
>> fledged congestion
>> >> controller and uses ECN, they can in worst case drive the buffer
>> into
>> the overload
>> >> regime where it starts dropping packets. „
>> >>>
>> >>> I’m not sure about this case. ECN is an input signal for
>> congestion
>> control. If you
>> >> don’t use congestion control but only a circuit breaker, you
>> should
>> probably not
>> >> enable ECN. At least it not clear to me why you would enable it, and
>> it's definitely
>> >> not conform to the ECN spec. Probably we should say something about
>> this in the
>> >> draft...?
>> >>>
>> >> Agree, enabling ECN without a responsive CC is going to lead to
>> trouble.
>> >>
>> >>> Mirja
>> >>>
>> >> Gorry
>> >>
>> >>>> Am 17.06.2016 um 16:03 schrieb Black, David <david.black@emc.com>:
>> >>>>
>> >>>> Colin,
>> >>>>
>> >>>>>>> ...  I view the current text as providing implementers with too
>> much
>> >>>>>>> latitude to ignore ECN-CE marks (e.g., because an implementer
>> doesn't
>> >>>>>>> want to think about this problem space in the first place).
>> >>>>>
>> >>>>> I agree, but the argument is that doing so is less harmful than
>> deploying a
>> >> circuit
>> >>>>> breaker that triggers too often when ECN is used.
>> >>>>>
>> >>>>> I’m not sure I believe this argument, though, since it seems
>> that
>> any new
>> >> AQM
>> >>>>> that applies ECN marks much more often than at present will have
>> to
>> >> consider
>> >>>>> backwards compatibility, to work with deployed TCP (e.g., draft-
>> briscoe-
>> >> tsvwg-
>> >>>>> aqm-tcpm-rmcat-l4s-problem uses ECT(1) as a signal to use the new
>> marking,
>> >>>>> while existing implementations set ECT(0)). These compatibility
>> mechanisms
>> >>>>> would seem to prevent the issues with the circuit breaker too.
>> >>>>
>> >>>> That roughly matches my line of thinking, and I'll observe that
>> the
>> original
>> >> DCTCP
>> >>>> protocol design that used more aggressive ECN-CE marking was only
>> safe for
>> >>>> Controlled Environment deployments.   See the TSVWG rfc5405bis
>> draft for
>> >> the
>> >>>> definition of Controlled Environment, and ignore the fact that the
>> rfc5405bis
>> >>>> draft is a UDP draft - this definition is more broadly applicable.
>> >>>>
>> >>>> Going back over Section 7 in this avtcore draft, my views are:
>> >>>>
>> >>>> [A] None of these drafts justify a "MAY ignore" response to ECN-CE
>> marks:
>> >>>>   - draft-khademi-tcpm-alternativebackoff-ecn
>> >>>>   - draft-ietf-rmcat-nada
>> >>>>   - draft-ietf-rmcat-scream-cc
>> >>>>
>> >>>> [B] In line with Colin's comment on the L4S draft, I think it's
>> incumbent on
>> >>>> the authors of draft-briscoe-aqm-dualq-coupled to figure out how
>> that will
>> >>>> coexist (or avoid) deployed TCP, and this avtcore draft ought not
>> to be
>> >>>> trying to prejudge what will be done there.
>> >>>>
>> >>>> So, I don't think the current text in Section 7 has justified the
>> unfettered
>> >>>> "implementations MAY ignore ECN-CE marks" text, as ignoring those
>> marks
>> >>>> is not consistent with any of the four cited drafts.
>> >>>>
>> >>>> In more detail, I think making changes to normative requirements
>> here based
>> >>>> on [B] is premature, and I would hope that the rmcat WG could be
>> >> encouraged
>> >>>> to consider the RTP circuit breaker in its congestion control
>> drafts, as those CC
>> >>>> mechanisms are related to the circuit breaker mechanism, hence
>> likely
>> >>>> to be in related areas of an RTP implementation.
>> >>>>
>> >>>> That leaves draft-khademi-tcpm-alternativebackoff-ecn, which TSVWG
>> >>>> will be looking at in Berlin.  If a normative statement about ECN-
>> CE reaction
>> >>>> is going to rest on that draft, then the reference to that draft
>> should be
>> >>>> normative.  Something about doing that strikes me as premature ...
>> >>>>
>> >>>> I realize that we're trying to predict and accommodate the future,
>> which
>> >>>> is an imprecise undertaking at best.   As an alternative to the
>> current text,
>> >>>> would it be reasonable to say (without any RFC 2119 keywords) that
>> the
>> >>>> best current guidance is still to treat ECN-CE marks as indicating
>> drops,
>> >>>> with a warning that there is a good possibility of this changing
>> in
>> the
>> >>>> near future due to all of the work in progress cited in Section 7?
>> >>>>
>> >>>> Thanks, --David
>> >>>>
>> >>>>> -----Original Message-----
>> >>>>> From: Colin Perkins [mailto:csp@csperkins.org]
>> >>>>> Sent: Friday, June 17, 2016 6:14 AM
>> >>>>> To: John Leslie; Black, David
>> >>>>> Cc: rtcweb@ietf.org; IETF AVTCore WG; tsvwg
>> >>>>> Subject: Re: [rtcweb] [AVTCORE] [tsvwg] WG Last Call on changes:
>> draft-ietf-
>> >>>>> avtcore-rtp-circuit-breakers-16
>> >>>>>
>> >>>>>
>> >>>>>> On 16 Jun 2016, at 23:25, John Leslie <john@jlc.net> wrote:
>> >>>>>>
>> >>>>>> Black, David <david.black@emc.com> wrote:
>> >>>>>>>
>> >>>>>>> ...  I view the current text as providing implementers with too
>> much
>> >>>>>>> latitude to ignore ECN-CE marks (e.g., because an implementer
>> doesn't
>> >>>>>>> want to think about this problem space in the first place).
>> >>>>>
>> >>>>> I agree, but the argument is that doing so is less harmful than
>> deploying a
>> >> circuit
>> >>>>> breaker that triggers too often when ECN is used.
>> >>>>>
>> >>>>> I’m not sure I believe this argument, though, since it seems
>> that
>> any new
>> >> AQM
>> >>>>> that applies ECN marks much more often than at present will have
>> to
>> >> consider
>> >>>>> backwards compatibility, to work with deployed TCP (e.g., draft-
>> briscoe-
>> >> tsvwg-
>> >>>>> aqm-tcpm-rmcat-l4s-problem uses ECT(1) as a signal to use the new
>> marking,
>> >>>>> while existing implementations set ECT(0)). These compatibility
>> mechanisms
>> >>>>> would seem to prevent the issues with the circuit breaker too.
>> >>>>>
>> >>>>>> Understand, we have at least two proposals to make ECN-CE more
>> >> frequent
>> >>>>>> than packet drop would be for non-ECN packets: possibly
>> substantially
>> >>>>>> more frequent. Unless both are killed off, ECN-CE will show up
>> frequently
>> >>>>>> enough that closing the flow on ECN-CE would kill too many
>> connections.
>> >>>>>>
>> >>>>>> If you want circuit-breaking on such connections, there are two
>> ways:
>> >>>>>> 1. convince the forwarding nodes to drop packets if their queue
>> exceeds
>> >>>>>> design capacity; or
>> >>>>>> 2. require the sender to send enough not-ECN-capable packets so
>> that our
>> >>>>>> receiver will see enough packet-drops when a circuit-breaker
>> should
>> >>>>>> activate.
>> >>>>>>
>> >>>>>> (I prefer the first option; but I wouldn't object to the
>> second.)
>> >>>>>>
>> >>>>>> There really isn't any way for our circuit-breaker to know
>> _how_much_
>> >>>>>> more frequent the ECN-CE marks may be. :^(
>> >>>>>
>> >>>>> This is a problem, both for the circuit breaker, and for the
>> algorithms being
>> >>>>> defined in RMCAT. We do need some understanding what the expected
>> >> marking
>> >>>>> rates are likely to be, so congestion control and circuit
>> breakers
>> can be
>> >> defined.
>> >>>>>
>> >>>>>> We _will_ be sorry if we
>> >>>>>> allot the same frequency of CE packets as packet-drops to
>> trigger
>> the
>> >>>>>> circuit-breaker.
>> >>>>>>
>> >>>>>>> Could someone propose initial text to qualifies the current
>> "MAY
>> ignore"
>> >>>>>>> statement?
>> >>>>>>
>> >>>>>> Essentially, for the second option, you might propose text to
>> the
>> >>>>>> effect of:
>> >>>>>> ]
>> >>>>>> ] If too many ECN-CE packets are received, the sender SHOULD
>> send
>> some
>> >>>>>> ] not-ECN-capable packets to determine whether enough packets
>> along the
>> >>>>>> ] path are being dropped to justify activating our circuit-
>> breaker.
>> >>>>>>
>> >>>>>> I’m not enthusiastic about adding that; but it would resolve
>> the
>> issue.
>> >>>>>
>> >>>>> I’m not convinced this would work. The circuit breaker is
>> looking
>> at long term
>> >>>>> trends, and in order to have enough not-ECT packets to determine
>> if it
>> >> should
>> >>>>> trigger, you’d essentially have to run without ECN for some
>> seconds.
>> >>>>>
>> >>>>> --
>> >>>>> Colin Perkins
>> >>>>> https://csperkins.org/
>> >>>>
>> >>>> _______________________________________________
>> >>>> rtcweb mailing list
>> >>>> rtcweb@ietf.org
>> >>>> https://www.ietf.org/mailman/listinfo/rtcweb
>> >
>
>


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From: "De Schepper, Koen (Nokia - BE)" <koen.de_schepper@nokia-bell-labs.com>
To: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>
Thread-Topic: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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Date: Mon, 27 Jun 2016 19:12:50 +0000
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, tsvwg <tsvwg@ietf.org>, Michael Welzl <michawe@ifi.uio.no>, IETF AVTCore WG <avt@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Colin Perkins <csp@csperkins.org>
Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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From: "Black, David" <david.black@emc.com>
To: "De Schepper, Koen (Nokia - BE)" <koen.de_schepper@nokia-bell-labs.com>
Thread-Topic: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, tsvwg <tsvwg@ietf.org>, IETF AVTCore WG <avt@ietf.org>
Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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References: <ccf9f2d7-2694-4336-0ec9-ccfebfeb0120@ericsson.com> <CE03DB3D7B45C245BCA0D243277949362F585D3E@MX307CL04.corp.emc.com> <d97e30a7-70f5-26d0-c3a4-0497c669f5f6@ericsson.com> <CE03DB3D7B45C245BCA0D243277949362F586054@MX307CL04.corp.emc.com> <D19E595F-7C66-4AE9-92B4-D550A93F634D@csperkins.org> <CE03DB3D7B45C245BCA0D243277949362F589335@MX307CL04.corp.emc.com> <20160616222548.GB77166@verdi> <0643E158-BF26-4692-8167-B7A959CB20CE@csperkins.org> <CE03DB3D7B45C245BCA0D243277949362F596DBC@MX307CL04.corp.emc.com> <E16BEA87-1D0F-48F1-A9AC-2729079D581D@tik.ee.ethz.ch> <8C16F1C6-B4A7-4BB4-B215-D7E7EAF308F8@erg.abdn.ac.uk> <CE03DB3D7B45C245BCA0D243277949362F59C41D@MX307CL04.corp.emc.com> <3E053A65-2698-4749-8E3D-E0451DF84011@ifi.uio.no> <BF6B00CC65FD2D45A326E74492B2C19FB76A6433@FR711WXCHMBA05.zeu.alcatel-lucent.com> <32a23d69d22062669f78df806a4eb6b8.squirrel@erg.abdn.ac.uk> <BF6B00CC65FD2D45A326E74492B2C19FB76A659B@FR711WXCHMBA05.zeu.alcatel-lucent.com>
To: "De Schepper, Koen (Nokia - BE)" <koen.de_schepper@nokia-bell-labs.com>
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Cc: "<gorry@erg.abdn.ac.uk> Fairhurst" <gorry@erg.abdn.ac.uk>, Magnus Westerlund <magnus.westerlund@ericsson.com>, tsvwg <tsvwg@ietf.org>, IETF AVTCore WG <avt@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Colin Perkins <csp@csperkins.org>
Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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> On 27. jun. 2016, at 21.12, De Schepper, Koen (Nokia - BE) =
<koen.de_schepper@nokia-bell-labs.com> wrote:
>=20
>=20
>> -----Original Message-----
>> From: gorry@erg.abdn.ac.uk [mailto:gorry@erg.abdn.ac.uk]
>> Sent: maandag 27 juni 2016 17:38
>> To: De Schepper, Koen (Nokia - BE)
>> Cc: Michael Welzl; Black, David; gorry@erg.abdn.ac.uk; Magnus
>> Westerlund; tsvwg; IETF AVTCore WG; rtcweb@ietf.org; Colin Perkins
>> Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: =
draft-
>> ietf-avtcore-rtp-circuit-breakers-16
>>=20
>> I think thinking of L4S is maybe off at a tangent. The question =
really
>> is
>> about the interpretation of loss and CE-mark as equivalent. I argued
>> that
>> each ECN-CE mark should not be counted as equivalent to a lost =
segment -
>=20
> Why not? As long as an AQM is marking at the same rate as dropping, a=20=

> 100% marking means that non-ecn flows are being dropped at a 100%, =
not?

Yes you can construct an AQM mechanism to act precisely like that.
Do we want rules that make every other behavior =E2=80=9Cillegal=E2=80=9D =
?  Should this really be how end systems should interpret all forms of =
CE-marking?

I think we both would agree that the answer is no: both the L4S and the =
ABE work are based on the premise that a CE-mark does *NOT* necessarily =
mean the same as packet loss would.

So I agree with Gorry here - a circuit breaker is a bit of a special =
beast. It really mustn=E2=80=99t trigger by accident.

Cheers,
Michael



>=20
> Koen.
>=20
>=20
>> in this context we should use ECN to drive a CC algorithm and we =
should
>> be
>> cautious to avoid requiring its use within a Circuit Breaker - =
optional
>> use, if you understand how to interpret a reaction to many CE-marks =
as
>> excessive congestion, are permitted.
>>=20
>> Gorry
>>=20
>>> As far as I understand, this draft is related to circuit breakers in
>>> end-systems, right?
>>>=20
>>> It is the end system that determines the use of ECN (currently =
marking
>>> non-ect for drop and ect(0) for Classic ECN).
>>>=20
>>> In L4S we don't plan to change the behavior of Classic ECN, and =
ABE's
>>> behavior should be close to non-ABE ECN. So I guess there is no
>> problem of
>>> describing the behavior of how a Classic ECN based sender would
>> respond
>>> today.
>>>=20
>>> As we only want to significantly change the network behavior of =
ect(1)
>>> marking, can we solve this issue by recommending (or even requiring)
>>> senders to mark only ect(0) and describing the classic ECN circuit
>>> breaker? When L4S gets defined, also an L4S based circuit breaker
>>> extension can be defined for senders that want to use the L4S =
service
>>> (when senders send ect(1) packets).
>>>=20
>>> Regards,
>>> Koen.
>>>=20
>>>=20
>>>> -----Original Message-----
>>>> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Michael
>> Welzl
>>>> Sent: maandag 20 juni 2016 18:36
>>>> To: Black, David
>>>> Cc: <gorry@erg.abdn.ac.uk> Fairhurst; Magnus Westerlund; tsvwg; =
IETF
>>>> AVTCore WG; rtcweb@ietf.org; Colin Perkins
>>>> Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes:
>> draft-
>>>> ietf-avtcore-rtp-circuit-breakers-16
>>>>=20
>>>>=20
>>>>> On 20. jun. 2016, at 15.16, Black, David <david.black@emc.com>
>> wrote:
>>>>>=20
>>>>>>> But I=C3=A2=E2=82=AC=E2=84=A2m less concerned than David about =
eventually ignoring it
>>>> for
>>>> circuit
>>>>>> breaker.
>>>>>>>=20
>>>>>> Agree. Loss is the measurement that a CB MUST respond to.
>>>>>=20
>>>>> Mumble.   I would be ok with a clear discouragement for use of =
ECN-
>> CE
>>>> marks, accompanied by the sort of design rationale here, or even
>>>> better,
>>>> a clear statement that lost packets for the purpose of the RTP
>> circuit
>>>> breaker have to be actually lost without getting into whether or =
not
>>>> ECN-CE marks are involved -i.e., the RTP circuit breaker is =
specified
>>>> against actual drops as a network protection backstop.
>>>>>=20
>>>>> A related concern is that ECN marks may overstate equivalent loss
>>>> behavior - a simplistic queue management discipline that marks =
every
>>>> packet when the queue is over a threshold (NB: this class of =
marking
>>>> behavior is NOT RECOMMENDED - a real AQM SHOULD be used) could =
yield
>> a
>>>> run of ECN-CE marks that would not cause a corresponding with a run
>> of
>>>> packet drops.   This is among the reasons that TCP reacts to ECN-CE
>>>> marks only once per RTT, and might be a reason to treat multiple =
ECN-
>> CE
>>>> marks in an RTT interval as not representing drops of all packets =
for
>>>> the RTP circuit breaker's TCP-equivalent throughput calculation.
>>>>=20
>>>> I=C3=A2=E2=82=AC=E2=84=A2m not sure we need such complicated logic =
to find a case where
>> ECN
>>>> marks are different from packet drops:
>>>>=20
>>>> Basically, they simply aren=C3=A2=E2=82=AC=E2=84=A2t - even =
=C3=A2=E2=82=AC=C5=93real=C3=A2=E2=82=AC=C2=9D AQMs marking
>> isn=C3=A2=E2=82=AC=E2=84=A2t
>>>> exactly
>>>> the same as a packet drop: the marks themselves inform you that an
>> AQM
>>>> did its job, and with modern AQMs like CoDel / PIE etc., =
you=C3=A2=E2=82=AC=E2=84=A2re
>>>> probably
>>>> getting this from a shallow queue. Chances are that this is less =
than
>> a
>>>> BDP worth of queuing, which is our justification for recommending a
>>>> different back-off behavior in draft-khademi-tsvwg-ecn-response-00
>> and
>>>> draft-khademi-tcpm-alternativebackoff-ecn-00
>>>>=20
>>>> So the point is not that AQMs would treat ECN marking and dropping
>>>> differently - it=C3=A2=E2=82=AC=E2=84=A2s that ECN indicates an =
AQM, and hence probably a
>>>> shallow queue. With a drop, you just don=C3=A2=E2=82=AC=E2=84=A2t =
know.
>>>>=20
>>>> Back to the CB, I think an AQM marking at a shallow queue (like =
e.g.
>>>> CoDel) is indeed quite different from a =C3=A2=E2=82=AC=C5=93broken =
connection=C3=A2=E2=82=AC=C2=9D.
>>>>=20
>>>> Cheers,
>>>> Michael
>>>>=20
>>>>=20
>>>>>=20
>>>>> Thanks, --David
>>>>>=20
>>>>>> -----Original Message-----
>>>>>> From: Gorry (erg) [mailto:gorry@erg.abdn.ac.uk]
>>>>>> Sent: Saturday, June 18, 2016 2:23 AM
>>>>>> To: Mirja K=C3=83=C2=BChlewind
>>>>>> Cc: Black, David; Magnus Westerlund; Colin Perkins;
>> rtcweb@ietf.org;
>>>> IETF
>>>>>> AVTCore WG; tsvwg
>>>>>> Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes:
>>>> draft-ietf-
>>>>>> avtcore-rtp-circuit-breakers-16
>>>>>>=20
>>>>>> I think we SHOULD NOT recommend to use ECN marks as inputs to a
>> CB.
>>>> See
>>>>>> below:
>>>>>>=20
>>>>>>> On 17 Jun 2016, at 16:02, Mirja K=C3=83=C2=BChlewind
>>>> <mirja.kuehlewind@tik.ee.ethz.ch>
>>>>>> wrote:
>>>>>>>=20
>>>>>>> +1 to not use normative language here.
>>>>>>>=20
>>>>>>> However, please note that having a high level of ECN-CE marks
>>>> (without any
>>>>>> losses) means that all packets were received correctly. This
>>>> situation can even
>>>>>> occurs without high delays (depending on the AQM used), which
>> would
>>>> just
>>>>>> mean the services works perfectly. Therefore for me CE marks are =
a
>>>> perfect input
>>>>>> signal for a congestion control loop (where the AQM tell the
>> sender
>>>> to take action
>>>>>> - whatever that means).
>>>>>>=20
>>>>>> We may in future figure out ways to do this to detect significant
>>>> failure using a
>>>>>> rate adaptive transport and ECN e.g.  Observing 100% CE marks or
>>>> something, for
>>>>>> an RTP flow that is trying to send well below its peak rate
>> decided
>>>> by CC -- but I
>>>>>> think this is speculating at an algorithm and adding details here
>> is
>>>> not a good idea.
>>>>>> Especially as AQM continues to evolve.
>>>>>>=20
>>>>>>> But I=C3=A2=E2=82=AC=E2=84=A2m less concerned than David about =
eventually ignoring it
>>>> for
>>>> circuit
>>>>>> breaker.
>>>>>>>=20
>>>>>> Agree. Loss is the measurement that a CB MUST respond to.
>>>>>>=20
>>>>>>> In addition one point on something Magnus wrote earlier:
>>>>>>> "If the implementation only have circuit breaker, i.e. no full
>>>> fledged congestion
>>>>>> controller and uses ECN, they can in worst case drive the buffer
>>>> into
>>>> the overload
>>>>>> regime where it starts dropping packets. =C3=A2=E2=82=AC=C5=BE
>>>>>>>=20
>>>>>>> I=C3=A2=E2=82=AC=E2=84=A2m not sure about this case. ECN is an =
input signal for
>>>> congestion
>>>> control. If you
>>>>>> don=C3=A2=E2=82=AC=E2=84=A2t use congestion control but only a =
circuit breaker, you
>>>> should
>>>> probably not
>>>>>> enable ECN. At least it not clear to me why you would enable it,
>> and
>>>> it's definitely
>>>>>> not conform to the ECN spec. Probably we should say something
>> about
>>>> this in the
>>>>>> draft...?
>>>>>>>=20
>>>>>> Agree, enabling ECN without a responsive CC is going to lead to
>>>> trouble.
>>>>>>=20
>>>>>>> Mirja
>>>>>>>=20
>>>>>> Gorry
>>>>>>=20
>>>>>>>> Am 17.06.2016 um 16:03 schrieb Black, David
>> <david.black@emc.com>:
>>>>>>>>=20
>>>>>>>> Colin,
>>>>>>>>=20
>>>>>>>>>>> ...  I view the current text as providing implementers with
>> too
>>>> much
>>>>>>>>>>> latitude to ignore ECN-CE marks (e.g., because an =
implementer
>>>> doesn't
>>>>>>>>>>> want to think about this problem space in the first place).
>>>>>>>>>=20
>>>>>>>>> I agree, but the argument is that doing so is less harmful =
than
>>>> deploying a
>>>>>> circuit
>>>>>>>>> breaker that triggers too often when ECN is used.
>>>>>>>>>=20
>>>>>>>>> I=C3=A2=E2=82=AC=E2=84=A2m not sure I believe this argument, =
though, since it seems
>>>> that
>>>> any new
>>>>>> AQM
>>>>>>>>> that applies ECN marks much more often than at present will
>> have
>>>> to
>>>>>> consider
>>>>>>>>> backwards compatibility, to work with deployed TCP (e.g.,
>> draft-
>>>> briscoe-
>>>>>> tsvwg-
>>>>>>>>> aqm-tcpm-rmcat-l4s-problem uses ECT(1) as a signal to use the
>> new
>>>> marking,
>>>>>>>>> while existing implementations set ECT(0)). These =
compatibility
>>>> mechanisms
>>>>>>>>> would seem to prevent the issues with the circuit breaker too.
>>>>>>>>=20
>>>>>>>> That roughly matches my line of thinking, and I'll observe that
>>>> the
>>>> original
>>>>>> DCTCP
>>>>>>>> protocol design that used more aggressive ECN-CE marking was
>> only
>>>> safe for
>>>>>>>> Controlled Environment deployments.   See the TSVWG rfc5405bis
>>>> draft for
>>>>>> the
>>>>>>>> definition of Controlled Environment, and ignore the fact that
>> the
>>>> rfc5405bis
>>>>>>>> draft is a UDP draft - this definition is more broadly
>> applicable.
>>>>>>>>=20
>>>>>>>> Going back over Section 7 in this avtcore draft, my views are:
>>>>>>>>=20
>>>>>>>> [A] None of these drafts justify a "MAY ignore" response to =
ECN-
>> CE
>>>> marks:
>>>>>>>>  - draft-khademi-tcpm-alternativebackoff-ecn
>>>>>>>>  - draft-ietf-rmcat-nada
>>>>>>>>  - draft-ietf-rmcat-scream-cc
>>>>>>>>=20
>>>>>>>> [B] In line with Colin's comment on the L4S draft, I think it's
>>>> incumbent on
>>>>>>>> the authors of draft-briscoe-aqm-dualq-coupled to figure out =
how
>>>> that will
>>>>>>>> coexist (or avoid) deployed TCP, and this avtcore draft ought
>> not
>>>> to be
>>>>>>>> trying to prejudge what will be done there.
>>>>>>>>=20
>>>>>>>> So, I don't think the current text in Section 7 has justified
>> the
>>>> unfettered
>>>>>>>> "implementations MAY ignore ECN-CE marks" text, as ignoring
>> those
>>>> marks
>>>>>>>> is not consistent with any of the four cited drafts.
>>>>>>>>=20
>>>>>>>> In more detail, I think making changes to normative =
requirements
>>>> here based
>>>>>>>> on [B] is premature, and I would hope that the rmcat WG could =
be
>>>>>> encouraged
>>>>>>>> to consider the RTP circuit breaker in its congestion control
>>>> drafts, as those CC
>>>>>>>> mechanisms are related to the circuit breaker mechanism, hence
>>>> likely
>>>>>>>> to be in related areas of an RTP implementation.
>>>>>>>>=20
>>>>>>>> That leaves draft-khademi-tcpm-alternativebackoff-ecn, which
>> TSVWG
>>>>>>>> will be looking at in Berlin.  If a normative statement about
>> ECN-
>>>> CE reaction
>>>>>>>> is going to rest on that draft, then the reference to that =
draft
>>>> should be
>>>>>>>> normative.  Something about doing that strikes me as premature
>> ...
>>>>>>>>=20
>>>>>>>> I realize that we're trying to predict and accommodate the
>> future,
>>>> which
>>>>>>>> is an imprecise undertaking at best.   As an alternative to the
>>>> current text,
>>>>>>>> would it be reasonable to say (without any RFC 2119 keywords)
>> that
>>>> the
>>>>>>>> best current guidance is still to treat ECN-CE marks as
>> indicating
>>>> drops,
>>>>>>>> with a warning that there is a good possibility of this =
changing
>>>> in
>>>> the
>>>>>>>> near future due to all of the work in progress cited in Section
>> 7?
>>>>>>>>=20
>>>>>>>> Thanks, --David
>>>>>>>>=20
>>>>>>>>> -----Original Message-----
>>>>>>>>> From: Colin Perkins [mailto:csp@csperkins.org]
>>>>>>>>> Sent: Friday, June 17, 2016 6:14 AM
>>>>>>>>> To: John Leslie; Black, David
>>>>>>>>> Cc: rtcweb@ietf.org; IETF AVTCore WG; tsvwg
>>>>>>>>> Subject: Re: [rtcweb] [AVTCORE] [tsvwg] WG Last Call on
>> changes:
>>>> draft-ietf-
>>>>>>>>> avtcore-rtp-circuit-breakers-16
>>>>>>>>>=20
>>>>>>>>>=20
>>>>>>>>>> On 16 Jun 2016, at 23:25, John Leslie <john@jlc.net> wrote:
>>>>>>>>>>=20
>>>>>>>>>> Black, David <david.black@emc.com> wrote:
>>>>>>>>>>>=20
>>>>>>>>>>> ...  I view the current text as providing implementers with
>> too
>>>> much
>>>>>>>>>>> latitude to ignore ECN-CE marks (e.g., because an =
implementer
>>>> doesn't
>>>>>>>>>>> want to think about this problem space in the first place).
>>>>>>>>>=20
>>>>>>>>> I agree, but the argument is that doing so is less harmful =
than
>>>> deploying a
>>>>>> circuit
>>>>>>>>> breaker that triggers too often when ECN is used.
>>>>>>>>>=20
>>>>>>>>> I=C3=A2=E2=82=AC=E2=84=A2m not sure I believe this argument, =
though, since it seems
>>>> that
>>>> any new
>>>>>> AQM
>>>>>>>>> that applies ECN marks much more often than at present will
>> have
>>>> to
>>>>>> consider
>>>>>>>>> backwards compatibility, to work with deployed TCP (e.g.,
>> draft-
>>>> briscoe-
>>>>>> tsvwg-
>>>>>>>>> aqm-tcpm-rmcat-l4s-problem uses ECT(1) as a signal to use the
>> new
>>>> marking,
>>>>>>>>> while existing implementations set ECT(0)). These =
compatibility
>>>> mechanisms
>>>>>>>>> would seem to prevent the issues with the circuit breaker too.
>>>>>>>>>=20
>>>>>>>>>> Understand, we have at least two proposals to make ECN-CE =
more
>>>>>> frequent
>>>>>>>>>> than packet drop would be for non-ECN packets: possibly
>>>> substantially
>>>>>>>>>> more frequent. Unless both are killed off, ECN-CE will show =
up
>>>> frequently
>>>>>>>>>> enough that closing the flow on ECN-CE would kill too many
>>>> connections.
>>>>>>>>>>=20
>>>>>>>>>> If you want circuit-breaking on such connections, there are
>> two
>>>> ways:
>>>>>>>>>> 1. convince the forwarding nodes to drop packets if their
>> queue
>>>> exceeds
>>>>>>>>>> design capacity; or
>>>>>>>>>> 2. require the sender to send enough not-ECN-capable packets
>> so
>>>> that our
>>>>>>>>>> receiver will see enough packet-drops when a circuit-breaker
>>>> should
>>>>>>>>>> activate.
>>>>>>>>>>=20
>>>>>>>>>> (I prefer the first option; but I wouldn't object to the
>>>> second.)
>>>>>>>>>>=20
>>>>>>>>>> There really isn't any way for our circuit-breaker to know
>>>> _how_much_
>>>>>>>>>> more frequent the ECN-CE marks may be. :^(
>>>>>>>>>=20
>>>>>>>>> This is a problem, both for the circuit breaker, and for the
>>>> algorithms being
>>>>>>>>> defined in RMCAT. We do need some understanding what the
>> expected
>>>>>> marking
>>>>>>>>> rates are likely to be, so congestion control and circuit
>>>> breakers
>>>> can be
>>>>>> defined.
>>>>>>>>>=20
>>>>>>>>>> We _will_ be sorry if we
>>>>>>>>>> allot the same frequency of CE packets as packet-drops to
>>>> trigger
>>>> the
>>>>>>>>>> circuit-breaker.
>>>>>>>>>>=20
>>>>>>>>>>> Could someone propose initial text to qualifies the current
>>>> "MAY
>>>> ignore"
>>>>>>>>>>> statement?
>>>>>>>>>>=20
>>>>>>>>>> Essentially, for the second option, you might propose text to
>>>> the
>>>>>>>>>> effect of:
>>>>>>>>>> ]
>>>>>>>>>> ] If too many ECN-CE packets are received, the sender SHOULD
>>>> send
>>>> some
>>>>>>>>>> ] not-ECN-capable packets to determine whether enough packets
>>>> along the
>>>>>>>>>> ] path are being dropped to justify activating our circuit-
>>>> breaker.
>>>>>>>>>>=20
>>>>>>>>>> I=C3=A2=E2=82=AC=E2=84=A2m not enthusiastic about adding =
that; but it would resolve
>>>> the
>>>> issue.
>>>>>>>>>=20
>>>>>>>>> I=C3=A2=E2=82=AC=E2=84=A2m not convinced this would work. The =
circuit breaker is
>>>> looking
>>>> at long term
>>>>>>>>> trends, and in order to have enough not-ECT packets to
>> determine
>>>> if it
>>>>>> should
>>>>>>>>> trigger, you=C3=A2=E2=82=AC=E2=84=A2d essentially have to run =
without ECN for some
>>>> seconds.
>>>>>>>>>=20
>>>>>>>>> --
>>>>>>>>> Colin Perkins
>>>>>>>>> https://csperkins.org/
>>>>>>>>=20
>>>>>>>> _______________________________________________
>>>>>>>> rtcweb mailing list
>>>>>>>> rtcweb@ietf.org
>>>>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>=20
>>>=20
>>>=20
>=20


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Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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David,


> On 27. jun. 2016, at 22.09, Black, David <david.black@emc.com> wrote:
>=20
>> As long as an AQM is marking at the same rate as dropping
>=20
> That's an interesting assumption - it should be true for AQMs vetted
> here in the past, but there are easy ways for it not to hold (e.g., if =
dropping
> or marking is based on queue occupancy, it is possible that dropping
> reduces queue occupancy in a fashion that marking does not).
>=20
> For ECN "classic" (i.e., see RFC 3168) where ECN-CE markings are =
treated
> as drop-equivalent, that is for congestion control purposes, which is =
similar
> to, (but not the same as) the throughput estimation usage for the RTP =
circuit
> breaker.    I'll note that ECN "classic" was designed congestion =
control
> algorithms for react to ECN-CE marks once per RTT, independent of how
> many ECN-CE marks are observed in an RTT.
>=20
> Gorry wrote:
>=20
>>> in this context we should use ECN to drive a CC algorithm and we =
should be
>>> cautious to avoid requiring its use within a Circuit Breaker - =
optional
>>> use, if you understand how to interpret a reaction to many CE-marks =
as
>>> excessive congestion, are permitted.
>=20
> Something like that may be workable, starting with a clear distinction =
between
> the use of ECN by CC (routine, active at all times) and ECN by a =
circuit
> breaker (monitors for evidence that things have gotten bad, only =
activated
> when things get bad).   This would baseline the RTP circuit breaker on =
drops
> and allow use of ECN as additional evidence of problems, in contrast =
to
> congestion control where ECN-CE is effectively treated as =
drop-equivalent.
>=20
> I'm not quite sure how to specify "use of ECN as additional evidence" =
of
> "excessive congestion" as drop-equivalence is about the best we have
> for current guidance.

I fail to parse that sentence, so maybe I=E2=80=99m getting you wrong, =
but anyway I wonder: what=E2=80=99s even the point of this?
Why even bother considering CE-marks as information for a circuit =
breaker?

CE-marks may *not* indicate *excessive* congestion - and since you say =
=E2=80=9Cadditional evidence=E2=80=9D: I don=E2=80=99t think that a =
combination of loss and CE-marks makes this any better? CE-marks may be =
produced by a shallow queue, which can be rather =E2=80=9Cmild=E2=80=9D =
congestion, at least in the light of what a circuit breaker should =
consider=E2=80=A6

Cheers,
Michael


From nobody Mon Jun 27 15:02:43 2016
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> On 27 Jun 2016, at 21:52, Michael Welzl <michawe@ifi.uio.no> wrote:
>=20
> David,
>=20
>=20
>> On 27. jun. 2016, at 22.09, Black, David <david.black@emc.com> wrote:
>>=20
>>> As long as an AQM is marking at the same rate as dropping
>>=20
>> That's an interesting assumption - it should be true for AQMs vetted
>> here in the past, but there are easy ways for it not to hold (e.g., =
if dropping
>> or marking is based on queue occupancy, it is possible that dropping
>> reduces queue occupancy in a fashion that marking does not).
>>=20
>> For ECN "classic" (i.e., see RFC 3168) where ECN-CE markings are =
treated
>> as drop-equivalent, that is for congestion control purposes, which is =
similar
>> to, (but not the same as) the throughput estimation usage for the RTP =
circuit
>> breaker.    I'll note that ECN "classic" was designed congestion =
control
>> algorithms for react to ECN-CE marks once per RTT, independent of how
>> many ECN-CE marks are observed in an RTT.
>>=20
>> Gorry wrote:
>>=20
>>>> in this context we should use ECN to drive a CC algorithm and we =
should be
>>>> cautious to avoid requiring its use within a Circuit Breaker - =
optional
>>>> use, if you understand how to interpret a reaction to many CE-marks =
as
>>>> excessive congestion, are permitted.
>>=20
>> Something like that may be workable, starting with a clear =
distinction between
>> the use of ECN by CC (routine, active at all times) and ECN by a =
circuit
>> breaker (monitors for evidence that things have gotten bad, only =
activated
>> when things get bad).   This would baseline the RTP circuit breaker =
on drops
>> and allow use of ECN as additional evidence of problems, in contrast =
to
>> congestion control where ECN-CE is effectively treated as =
drop-equivalent.
>>=20
>> I'm not quite sure how to specify "use of ECN as additional evidence" =
of
>> "excessive congestion" as drop-equivalence is about the best we have
>> for current guidance.
>=20
> I fail to parse that sentence, so maybe I=E2=80=99m getting you wrong, =
but anyway I wonder: what=E2=80=99s even the point of this?
> Why even bother considering CE-marks as information for a circuit =
breaker?

Because the alternative is that we only break the circuit once the queue =
has been driven into overflow, and packets have been lost. We want to =
avoid that, since it causes latency, and too much latency is very bad =
for the user experience.=20

> CE-marks may *not* indicate *excessive* congestion - and since you say =
=E2=80=9Cadditional evidence=E2=80=9D: I don=E2=80=99t think that a =
combination of loss and CE-marks makes this any better? CE-marks may be =
produced by a shallow queue, which can be rather =E2=80=9Cmild=E2=80=9D =
congestion, at least in the light of what a circuit breaker should =
consider=E2=80=A6

Surely this is just arguing for a different threshold for a circuit =
breaker triggered by ECN-CE marks (using a modern, small queue, AQM) =
than for one triggered by loss (or ECN marks considered equivalent to =
loss)?=20

If I understand the L4S proposal correctly, that would be treat ECN-CE =
marks on ECT(0) marked flows as equivalent to loss, but treat ECN-CE =
marks on ECT(1) marked flows with a (much) higher threshold.=20

Assuming, in all cases, that there=E2=80=99s a parallel congestion =
control algorithm running (and RMCAT has figured out the right =
congestion response for that; the proposals now treat ECN-CE and loss =
very similarly).

--=20
Colin Perkins
https://csperkins.org/





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> On 28. jun. 2016, at 00.02, Colin Perkins <csp@csperkins.org> wrote:
>=20
>=20
>> On 27 Jun 2016, at 21:52, Michael Welzl <michawe@ifi.uio.no> wrote:
>>=20
>> David,
>>=20
>>=20
>>> On 27. jun. 2016, at 22.09, Black, David <david.black@emc.com> =
wrote:
>>>=20
>>>> As long as an AQM is marking at the same rate as dropping
>>>=20
>>> That's an interesting assumption - it should be true for AQMs vetted
>>> here in the past, but there are easy ways for it not to hold (e.g., =
if dropping
>>> or marking is based on queue occupancy, it is possible that dropping
>>> reduces queue occupancy in a fashion that marking does not).
>>>=20
>>> For ECN "classic" (i.e., see RFC 3168) where ECN-CE markings are =
treated
>>> as drop-equivalent, that is for congestion control purposes, which =
is similar
>>> to, (but not the same as) the throughput estimation usage for the =
RTP circuit
>>> breaker.    I'll note that ECN "classic" was designed congestion =
control
>>> algorithms for react to ECN-CE marks once per RTT, independent of =
how
>>> many ECN-CE marks are observed in an RTT.
>>>=20
>>> Gorry wrote:
>>>=20
>>>>> in this context we should use ECN to drive a CC algorithm and we =
should be
>>>>> cautious to avoid requiring its use within a Circuit Breaker - =
optional
>>>>> use, if you understand how to interpret a reaction to many =
CE-marks as
>>>>> excessive congestion, are permitted.
>>>=20
>>> Something like that may be workable, starting with a clear =
distinction between
>>> the use of ECN by CC (routine, active at all times) and ECN by a =
circuit
>>> breaker (monitors for evidence that things have gotten bad, only =
activated
>>> when things get bad).   This would baseline the RTP circuit breaker =
on drops
>>> and allow use of ECN as additional evidence of problems, in contrast =
to
>>> congestion control where ECN-CE is effectively treated as =
drop-equivalent.
>>>=20
>>> I'm not quite sure how to specify "use of ECN as additional =
evidence" of
>>> "excessive congestion" as drop-equivalence is about the best we have
>>> for current guidance.
>>=20
>> I fail to parse that sentence, so maybe I=E2=80=99m getting you =
wrong, but anyway I wonder: what=E2=80=99s even the point of this?
>> Why even bother considering CE-marks as information for a circuit =
breaker?
>=20
> Because the alternative is that we only break the circuit once the =
queue has been driven into overflow, and packets have been lost. We want =
to avoid that, since it causes latency, and too much latency is very bad =
for the user experience.=20

Well - the better way out would be for the application to react. Maybe =
this is me misunderstanding the circuit breaker, but I did think it=E2=80=99=
s more like a last resort=E2=80=A6 you just don=E2=80=99t want to be =
trigger-happy with such a thing?


>> CE-marks may *not* indicate *excessive* congestion - and since you =
say =E2=80=9Cadditional evidence=E2=80=9D: I don=E2=80=99t think that a =
combination of loss and CE-marks makes this any better? CE-marks may be =
produced by a shallow queue, which can be rather =E2=80=9Cmild=E2=80=9D =
congestion, at least in the light of what a circuit breaker should =
consider=E2=80=A6
>=20
> Surely this is just arguing for a different threshold for a circuit =
breaker triggered by ECN-CE marks (using a modern, small queue, AQM) =
than for one triggered by loss (or ECN marks considered equivalent to =
loss)?=20

If you have room for yet another code point, for the circuit breaker =
only?  :)    Or maybe I just misunderstand you here?


> If I understand the L4S proposal correctly, that would be treat ECN-CE =
marks on ECT(0) marked flows as equivalent to loss, but treat ECN-CE =
marks on ECT(1) marked flows with a (much) higher threshold.=20

L4S would not change anything about how ECT(0) marked flows are treated, =
and would CE-mark packets carrying ECT(1) with an instantaneous queue - =
i.e. a much *lower* threshold. But that=E2=80=99s not the issue - I =
agree there=E2=80=99s no problem with L4S.

The compatibility problem does exist with the ABE proposal, which works =
off ECT(0).

The ABE proposal exploits a very simple fact: that CE-marks are, by =
definition, *not* the same as loss (see David Black=E2=80=99s previous =
email where he says "if dropping or marking is based on queue occupancy, =
it is possible that dropping reduces queue occupancy in a fashion that =
marking does not=E2=80=9D). Indeed, queue dynamics play out differently =
when packets are dropped or marked  ( see Section 7 with Figures 13/14 =
in =
https://www.duo.uio.no/bitstream/handle/10852/37381/khademi-AQM_Kids_TR434=
.pdf ) .

Losses may stem from a DropTail (FIFO) queue somewhere along the path - =
CE-marks are, however, very likely to only be caused by an AQM =
algorithm. TCP=E2=80=99s built-in reaction to loss yields full link =
utilization only when there=E2=80=99s at least a BDP worth of queuing. =
This is a lot of latency - when the queue is full this doubles the RTT. =
Modern AQM mechanisms strive to maintain a much smaller average queue =
size, and this is where they mark packets.

So: if we react to CE-marks the same way as to loss, CoDel and PIE let =
us underutilize the link.

Thus, it makes more sense to interpret the signal for what it is: an =
indication that there was congestion, but from a queue that might be =
much smaller than a BDP.


> Assuming, in all cases, that there=E2=80=99s a parallel congestion =
control algorithm running

If you assume that there=E2=80=99s a parallel congestion control =
algorithm running, I understand even less why you want to feed ECN =
CE-marks into the circuit breaker. The congestion control algorithm =
should already deal with them.
=20

> (and RMCAT has figured out the right congestion response for that; the =
proposals now treat ECN-CE and loss very similarly).

I disagree that this is the =E2=80=9Cright=E2=80=9D congestion response. =
It=E2=80=99s a workable one, sure. Nothing extremely terrible will =
happen if congestion controllers treat ECN-CE and loss similarly - it =
just yields unnecessarily poor utilization with ECN, with modern AQMs  =
(unless one backs off by less than TCP would in response to loss too, =
which is good if there=E2=80=99s an AQM in place but may be quite bad =
otherwise).

Bottom line: it really does mean something different, and it seems wrong =
to me to act as if that wasn=E2=80=99t the case - just because we=E2=80=99=
ve always done so.

Cheers,
Michael


From nobody Mon Jun 27 16:03:25 2016
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From: "Fred Baker (fred)" <fred@cisco.com>
To: John Leslie <john@jlc.net>
Thread-Topic: [tsvwg] [rtcweb] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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Date: Mon, 27 Jun 2016 23:03:03 +0000
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References: <ccf9f2d7-2694-4336-0ec9-ccfebfeb0120@ericsson.com> <CE03DB3D7B45C245BCA0D243277949362F585D3E@MX307CL04.corp.emc.com> <20160614135927.GE39331@verdi>
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On Jun 14, 2016, at 6:59 AM, John Leslie <john@jlc.net> wrote:
>   We must find a way to avoid ECN-CE triggering the congestion circuit
> breaker when the forwarding node has not reached a level of congestion
> which could trigger packet loss.

I'm not sure I agree.

I tend to think that ECN is relatively cheap - it asks the sender to =
reduce its effective window but doesn't make him retransmit anything and =
causes no delivery delay - while packet loss is relatively expensive in =
that it makes the sender retransmit and introduces at least a one RTT =
hiccup ( to detect and retransmit the lost packet) in the transmission =
stream. Neither actually changes the rate at which anyone sends (we talk =
about the signal asking the sender to slow down, but the rate =
demonstrably doesn't actually change - we're asking the sender to not =
pack the pipe so full).

When ECN does ask someone to reduce their effective window, if they are =
typically increasing it by one per RTT, ECN could very logically reduce =
it by a small amount, such as two or three. If it reduced it by two, for =
example, that would cause it to skip the next two transmission =
opportunities (the next two received acknowledgements, which might =
actually arrive in the same ack), and would be back to the same level =
two RTTs later. If ECN is typically keeping the delay relatively =
shallow, it is oscillating between "a little shallower" and " a little =
less shallow". So it gets CE every other RTT? So what? If I'm dropping a =
packet, or a couple of packets, every other RTT, we're going to see =
notes on NANOG and other lists. "Your network's broken again. It's =
dropping packets."

I could be argued into setting the CE threshold a little lower than the =
discard threshold. There are probably some mechanisms we'd do well to =
explore, but...

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From nobody Mon Jun 27 16:59:36 2016
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From: "Black, David" <david.black@emc.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: draft-szigeti-tsvwg-ieee-802-11 - Adopted by TSVWG
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Subject: [tsvwg] draft-szigeti-tsvwg-ieee-802-11 - Adopted by TSVWG
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The WG chairs conclusion is that the rough consensus of the TSVWG WG is to =
adopt
this draft as a WG draft.   The authors should submit a draft-ietf-tsvwg-* =
version, and
this draft will be included in the WG drafts portion of the TSVWG Berlin me=
eting agenda.

Thanks, --David

> -----Original Message-----
> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Black, David
> Sent: Thursday, June 09, 2016 1:14 AM
> To: tsvwg@ietf.org
> Subject: [tsvwg] WG adoption call: draft-szigeti-tsvwg-ieee-802-11 - thro=
ugh June
> 25
>=20
> This is a TSVWG call for interest in WG adoption of draft-szigeti-tsvwg-i=
eee-802-
> 11.
> If you have read the draft and support WG adoption of it as a work item
> and WG draft, please send an email to the list (tsvwg@ietf.org) saying so=
.
>=20
> NB: It is important to say that you've read the draft.  If you're willing
> to review/comment on future versions of this draft and/or implement this
> draft, please say so on the list or in email directly to the chairs
> (tsvwg-chairs@ietf.org).
>=20
> Authors: We (chairs) assume that you support WG adoption of your draft, s=
o
> there's no need to say so on the list.  However, we do need each author t=
o
> publicly state on the list that their direct, personal knowledge (if any)=
 of any
> IPR related to this document has already been disclosed, in conformance
> with BCPs 78 and 79.
>=20
> This adoption call will run for about 2 weeks, ending at midnight, prevai=
ling
> US Eastern Time on Saturday June 25.  To get things started, I hereby sta=
te
> as an individual that I have read the draft and support WG adoption of it=
.
>=20
>              Guidelines for DiffServ to IEEE 802.11 Mapping
>                    draft-szigeti-tsvwg-ieee-802-11-01
>=20
> Abstract
>=20
>    As internet traffic is increasingly sourced-from and destined-to
>    wireless endpoints, it is crucial that Quality of Service be aligned
>    between wired and wireless networks; however, this is not always the
>    case by default.  This is due to the fact that two independent
>    standards bodies provide QoS guidance on wired and wireless networks:
>    specifically, the IETF specifies standards and design recommendations
>    for wired IP networks, while a separate and autonomous standards-
>    body, the IEEE, administers the standards for wireless 802.11
>    networks.  The purpose of this document is to propose a set
>    Differentiated Services Code Point (DSCP) to IEEE 802.11 User
>    Priority (UP) mappings to reconcile the marking recommendations
>    offered by these two standards bodies, and, as such, to optimize
>    wired-and-wireless interconnect QoS.
>=20
> Thanks, --David
> ----------------------------------------------------
> David L. Black, Distinguished Engineer
> EMC, 176 South St., Hopkinton, MA=A0 01748
> +1 (508) 293-7953=A0=A0=A0  Cell: +1 (978) 394-7754
> david.black@emc.com
> ----------------------------------------------------
>=20


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From: John Leslie <john@jlc.net>
To: Colin Perkins <csp@csperkins.org>
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References: <E16BEA87-1D0F-48F1-A9AC-2729079D581D@tik.ee.ethz.ch> <8C16F1C6-B4A7-4BB4-B215-D7E7EAF308F8@erg.abdn.ac.uk> <CE03DB3D7B45C245BCA0D243277949362F59C41D@MX307CL04.corp.emc.com> <3E053A65-2698-4749-8E3D-E0451DF84011@ifi.uio.no> <BF6B00CC65FD2D45A326E74492B2C19FB76A6433@FR711WXCHMBA05.zeu.alcatel-lucent.com> <32a23d69d22062669f78df806a4eb6b8.squirrel@erg.abdn.ac.uk> <BF6B00CC65FD2D45A326E74492B2C19FB76A659B@FR711WXCHMBA05.zeu.alcatel-lucent.com> <CE03DB3D7B45C245BCA0D243277949362F5AEE02@MX307CL04.corp.emc.com> <6E35FB6C-CA98-413C-B7AE-75402A968017@ifi.uio.no> <3FD27BBF-8E2D-4A42-86A0-C4C0692FF8C9@csperkins.org>
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Subject: Re: [tsvwg] [AVTCORE] [rtcweb] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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Colin Perkins <csp@csperkins.org> wrote:
>  On 27 Jun 2016, at 21:52, Michael Welzl <michawe@ifi.uio.no> wrote:
>
>> Why even bother considering CE-marks as information for a circuit breaker?
> 
> Because the alternative is that we only break the circuit once the queue
> has been driven into overflow, and packets have been lost.
> We want to avoid that, since it causes latency, and too much latency is
> very bad for the user experience. 

   If you want a circuit-breaker to control latency, why aren't you measuring
latency?

--
John Leslie <john@jlc.net>


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From: "Fred Baker (fred)" <fred@cisco.com>
To: John Leslie <john@jlc.net>
Thread-Topic: [tsvwg] [AVTCORE] [rtcweb] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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> On Jun 27, 2016, at 5:04 PM, John Leslie <john@jlc.net> wrote:
>=20
> Colin Perkins <csp@csperkins.org> wrote:
>> On 27 Jun 2016, at 21:52, Michael Welzl <michawe@ifi.uio.no> wrote:
>>=20
>>> Why even bother considering CE-marks as information for a circuit =
breaker?
>>=20
>> Because the alternative is that we only break the circuit once the =
queue
>> has been driven into overflow, and packets have been lost.
>> We want to avoid that, since it causes latency, and too much latency =
is
>> very bad for the user experience.
>=20
>   If you want a circuit-breaker to control latency, why aren't you =
measuring
> latency?

You mean, like CoDel?

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From nobody Mon Jun 27 18:04:41 2016
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From: "Black, David" <david.black@emc.com>
To: Michael Welzl <michawe@ifi.uio.no>, Colin Perkins <csp@csperkins.org>
Thread-Topic: [AVTCORE] [rtcweb] [tsvwg] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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Subject: Re: [tsvwg] [AVTCORE] [rtcweb] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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From nobody Mon Jun 27 23:36:50 2016
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Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-ieee-802-11-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Transport Area Working Group of the IETF.

        Title           : Guidelines for DiffServ to IEEE 802.11 Mapping
        Authors         : Tim Szigeti
                          Fred Baker
	Filename        : draft-ietf-tsvwg-ieee-802-11-00.txt
	Pages           : 27
	Date            : 2016-06-27

Abstract:
   As internet traffic is increasingly sourced-from and destined-to
   wireless endpoints, it is crucial that Quality of Service be aligned
   between wired and wireless networks; however, this is not always the
   case by default.  This is due to the fact that two independent
   standards bodies provide QoS guidance on wired and wireless networks:
   specifically, the IETF specifies standards and design recommendations
   for wired IP networks, while a separate and autonomous standards-
   body, the IEEE, administers the standards for wireless 802.11
   networks.  The purpose of this document is to propose a set
   Differentiated Services Code Point (DSCP) to IEEE 802.11 User
   Priority (UP) mappings to reconcile the marking recommendations
   offered by these two standards bodies, and, as such, to optimize
   wired-and-wireless interconnect QoS.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-ieee-802-11/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-tsvwg-ieee-802-11-00


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

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


From nobody Tue Jun 28 07:47:21 2016
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Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-rfc5405bis-15.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 Transport Area Working Group of the IETF.

        Title           : UDP Usage Guidelines
        Authors         : Lars Eggert
                          Godred Fairhurst
                          Greg Shepherd
	Filename        : draft-ietf-tsvwg-rfc5405bis-15.txt
	Pages           : 58
	Date            : 2016-06-28

Abstract:
   The User Datagram Protocol (UDP) provides a minimal message-passing
   transport that has no inherent congestion control mechanisms.  This
   document provides guidelines on the use of UDP for the designers of
   applications, tunnels and other protocols that use UDP.  Congestion
   control guidelines are a primary focus, but the document also
   provides guidance on other topics, including message sizes,
   reliability, checksums, middlebox traversal, the use of ECN, DSCPs,
   and ports.

   Because congestion control is critical to the stable operation of the
   Internet, applications and other protocols that choose to use UDP as
   an Internet transport must employ mechanisms to prevent congestion
   collapse and to establish some degree of fairness with concurrent
   traffic.  They may also need to implement additional mechanisms,
   depending on how they use UDP.

   Some guidance is also applicable to the design of other protocols
   (e.g., protocols layered directly on IP or via IP-based tunnels),
   especially when these protocols do not themselves provide congestion
   control.

   This document obsoletes RFC5405 and adds guidelines for multicast UDP
   usage.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-tsvwg-rfc5405bis-15

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-rfc5405bis-15


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.

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ftp://ftp.ietf.org/internet-drafts/


From nobody Tue Jun 28 08:01:55 2016
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This should address all comments raised during IETF LC, apart from =
Mark's unhappiness about the current RTT/RTO text (although I did =
attempt to at least correct the use of the terms).

On 2016-06-28, at 16:47, 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 Transport Area Working Group of the =
IETF.
>=20
>        Title           : UDP Usage Guidelines
>        Authors         : Lars Eggert
>                          Godred Fairhurst
>                          Greg Shepherd
> 	Filename        : draft-ietf-tsvwg-rfc5405bis-15.txt
> 	Pages           : 58
> 	Date            : 2016-06-28
>=20
> Abstract:
>   The User Datagram Protocol (UDP) provides a minimal message-passing
>   transport that has no inherent congestion control mechanisms.  This
>   document provides guidelines on the use of UDP for the designers of
>   applications, tunnels and other protocols that use UDP.  Congestion
>   control guidelines are a primary focus, but the document also
>   provides guidance on other topics, including message sizes,
>   reliability, checksums, middlebox traversal, the use of ECN, DSCPs,
>   and ports.
>=20
>   Because congestion control is critical to the stable operation of =
the
>   Internet, applications and other protocols that choose to use UDP as
>   an Internet transport must employ mechanisms to prevent congestion
>   collapse and to establish some degree of fairness with concurrent
>   traffic.  They may also need to implement additional mechanisms,
>   depending on how they use UDP.
>=20
>   Some guidance is also applicable to the design of other protocols
>   (e.g., protocols layered directly on IP or via IP-based tunnels),
>   especially when these protocols do not themselves provide congestion
>   control.
>=20
>   This document obsoletes RFC5405 and adds guidelines for multicast =
UDP
>   usage.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rfc5405bis/
>=20
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-tsvwg-rfc5405bis-15
>=20
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-rfc5405bis-15
>=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


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From nobody Tue Jun 28 08:08:39 2016
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From: "Black, David" <david.black@emc.com>
To: Tom Herbert <tom@herbertland.com>, "adrian@olddog.co.uk" <adrian@olddog.co.uk>
Thread-Topic: RTG Dir QA review of draft-ietf-nvo3-gue - TSVWG coordination
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Date: Tue, 28 Jun 2016 15:08:02 +0000
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Subject: [tsvwg] RTG Dir QA review of draft-ietf-nvo3-gue - TSVWG coordination
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[+tsvwg - this draft is undergoing a WG adoption call to move it from nvo3 =
to intarea]

Commenting on  just the TSVWG coordination concerns, as a TSVWG co-chair.

Legend:
> > Adrian Farrel (RTG Dir reviewer)
> Tom Herbert (GUE draft author)
David Black (TSVWG co-chair)

> > It seems to me that there is some careful coordination needed with othe=
r
> > work on encapsulation of transport or network protocols in UDP. This
> > idea clearly has value in NVO3, but I should have thought it sat better
> > in the TSVWG. I hope the NVO3 chairs have discussed this with the TSVWG
> > chairs to ensure that there is no friction. This is particularly
> > important because it will be important to recognise that only one of
> > this draft and draft-manner-tsvwg-gut is likely to make it to RFC.
> >
> Much of the content around TSV issues (e.g. zero UDP6 checksum and
> congestion control) is based on that in MPLS/UDP and GRE/UDP.

And those are the two most relevant documents to work from.  GRE/UDP needs
a little more tweaking before RFC publication is requested.  Specifically, =
I need to
supply some improved congestion control text to the authors - doing that is=
 on
my "round tuit" list.
=20
> > You seem to have correctly addressed the three issues that have most
> > worried the TSVWG (checksum, congestion and security), so that is all
> > good, but I would recommend getting the TSVWG involved for a full and
> > detailed review now and for each future revision of the document. In
> > fact, I would have tended towards making this a TSVWG document, but so
> > long as the chairs, the ADs, and the WGs are happy, that should be fine=
.
> >
> > ---
> >
> > Overall, this work is a good idea and needed. When we did MPLS-in-UDP
> > there was a background proposal to generalise and only burn one port
> > number for al UDP encapsulations. This achieves that end.
> >
> > However, I think this proposal may be too general and too extensible.
> > Future-proof is good, but there seem to be a lot of bells and whistles
> > defined here that have no specific use proposed, and no indication that
> > a future use might ever be defined. I think it is one thing that it
> > should be possible to extend a protocol, and another that it defines
> > multiple fields and extension mechanisms that might never be used.
> >
> Yes, two of the authors on this draft are also authors for GRE/UDP.
> GRE/UDP has received excellent review in TSVWG and we would like to
> have similar review of GUE.

That will definitely happen ;-).   My preference is for encapsulation heade=
r
design to be done in WGs that work on encapsulations - TSVWG is not
such a WG, but is more than happy to advise on Transport concerns, such
as those identified above.

Thanks, --David


From nobody Tue Jun 28 09:10:08 2016
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From: "Fred Baker (fred)" <fred@cisco.com>
To: tsvwg <tsvwg@ietf.org>
Thread-Topic: I-D Action: draft-ietf-tsvwg-ieee-802-11-00.txt
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For the record, a diff against draft-szigeti-* may be found at =
https://tools.ietf.org//rfcdiff?url1=3Dhttps://tools.ietf.org/id/draft-szi=
geti-tsvwg-ieee-802-11-02.txt&url2=3Dhttps://tools.ietf.org/id/draft-ietf-=
tsvwg-ieee-802-11-00.txt

> On Jun 27, 2016, at 11:36 PM, 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 Transport Area Working Group of the =
IETF.
>=20
>        Title           : Guidelines for DiffServ to IEEE 802.11 =
Mapping
>        Authors         : Tim Szigeti
>                          Fred Baker
> 	Filename        : draft-ietf-tsvwg-ieee-802-11-00.txt
> 	Pages           : 27
> 	Date            : 2016-06-27
>=20
> Abstract:
>   As internet traffic is increasingly sourced-from and destined-to
>   wireless endpoints, it is crucial that Quality of Service be aligned
>   between wired and wireless networks; however, this is not always the
>   case by default.  This is due to the fact that two independent
>   standards bodies provide QoS guidance on wired and wireless =
networks:
>   specifically, the IETF specifies standards and design =
recommendations
>   for wired IP networks, while a separate and autonomous standards-
>   body, the IEEE, administers the standards for wireless 802.11
>   networks.  The purpose of this document is to propose a set
>   Differentiated Services Code Point (DSCP) to IEEE 802.11 User
>   Priority (UP) mappings to reconcile the marking recommendations
>   offered by these two standards bodies, and, as such, to optimize
>   wired-and-wireless interconnect QoS.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-ieee-802-11/
>=20
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-tsvwg-ieee-802-11-00
>=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
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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Subject: Re: [tsvwg] [AVTCORE] [rtcweb] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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> On 28 Jun 2016, at 01:04, John Leslie <john@jlc.net> wrote:
> Colin Perkins <csp@csperkins.org> wrote:
>> On 27 Jun 2016, at 21:52, Michael Welzl <michawe@ifi.uio.no> wrote:
>>=20
>>> Why even bother considering CE-marks as information for a circuit =
breaker?
>>=20
>> Because the alternative is that we only break the circuit once the =
queue
>> has been driven into overflow, and packets have been lost.
>> We want to avoid that, since it causes latency, and too much latency =
is
>> very bad for the user experience.=20
>=20
>   If you want a circuit-breaker to control latency, why aren't you =
measuring
> latency?

Because the RTP circuit breaker needs to work with unmodified RFC =
3550-style RTCP, which doesn=E2=80=99t return RTT samples sufficiently =
often to allow that, and because we don=E2=80=99t care about the precise =
latency, we just don=E2=80=99t want to fill a large queue.

--=20
Colin Perkins
https://csperkins.org/





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Subject: Re: [tsvwg] [AVTCORE] [rtcweb] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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> On 28 Jun 2016, at 02:04, Black, David <david.black@emc.com> wrote:
>=20
> Trying to shorten up this thread again ...
>=20
>>>>> I'm not quite sure how to specify "use of ECN as additional =
evidence" of
>>>>> "excessive congestion" as drop-equivalence is about the best we =
have
>>>>> for current guidance.
>>>>=20
>>>> I fail to parse that sentence, so maybe I=E2=80=99m getting you =
wrong, but anyway I
>>>> wonder: what=E2=80=99s even the point of this?
>>>> Why even bother considering CE-marks as information for a circuit =
breaker?
>>>=20
>>> Because the alternative is that we only break the circuit once the =
queue has
>> been driven into overflow, and packets have been lost. We want to =
avoid that,
>> since it causes latency, and too much latency is very bad for the =
user experience.
>>=20
>> Well - the better way out would be for the application to react. =
Maybe this is me
>> misunderstanding the circuit breaker, but I did think it=E2=80=99s =
more like a last resort=E2=80=A6
>> you just don=E2=80=99t want to be trigger-happy with such a thing?
>=20
> Well, the RTP circuit breaker draft is not trigger happy - for its =
congestion circuit
> breaker to trip, RTP has to be sending at 10x the rate that TCP would =
send under
> those conditions, based on the TCP throughput equation.  See:
>=20
> =
https://tools.ietf.org/html/draft-ietf-avtcore-rtp-circuit-breakers-16#sec=
tion-4.3
>=20
> The issue here is - when calculating the comparable TCP throughput, =
how are ECN-CE
> marks used to determine the loss rate input to the TCP throughput =
equation?  Do
> ECN-CE marked packets count as having arrived or having been dropped?

Right - or do they count somewhere between the two.

> When things are relatively stable and the ECN-CE marks are being used =
to nudge
> the sender's rate based on what the network can absorb, whether ECN-CE =
marks
> count as losses or not is probably immaterial - the 10x divergence =
from the TCP
> throughput equation's rate is not going to arise, and the circuit =
breaker won't trip.
> The circuit breaker is only supposed to trip when things are seriously =
wrong.

Correct.

> (1) If the RTP congestion circuit breaker trips based on ECN-CE marks =
alone,
> something feels intuitively wrong - how'd we get to RTP running at 10x =
the
> comparable TCP sending rate with no losses?  Perhaps the circuit =
breaker
> shouldn=E2=80=99t trip on ECN-CE marks alone?

Shouldn=E2=80=99t the comparable rate to trigger the circuit breaker be =
10x that given to a TCP flow subject to the same ECN-CE marking rate? If =
the TCP treats ECN-CE as equivalent to loss, for congestion response, =
then the circuit breaker should do so to, etc.

> (2) At the other extreme, the congestion circuit breaker clearly has =
to trip if RTP
> gets to 10x the comparable TCP sending rate based on losses alone.  =
This is the
> baseline for the circuit breaker to provide network protection as =
intended.
>=20
> So, going back to Gorry's suggestion to use ECN-CE marks as =
"additional evidence,"
> here's a straw proposal to shoot at ... factor in ECN-CE marks as =
additional losses
> *only when* losses are already occurring.  =20
>=20
> For example, we could specify that for the RTP congestion circuit =
breaker to trip, the
> RTP sending rate has to be:
> 	- 10x the equivalent TCP sending rate based on counting ECN-CE =
marked
> 		packets as lost; AND
> 	- 3x the equivalent sending rate based on actual drops (i.e., =
counting
> 		ECN-CE marked packets as delivered).
> The "3x" above is an off-the-top-of-my-head factor that attempts to =
roughly
> equally weight the inputs (3 is close to the square root of 10) - pick =
a different
> number if that weighting feels wrong.
>=20
> This would force drops to occur and then consider ECN-CE marks as =
additional evidence
> that something is wrong in the network.
>=20
> Another possible rationale for this mixing is that if drops start =
occurring, then many of
> the new and proposed uses of ECN that treat ECN-CE marks as less than =
loss-equivalent
> are outside their intended operating envelopes/regions.

Clearly if the queue has been driven to overflow, so that packet loss is =
occurring, then the AQM is outside its intended operating regime. I=E2=80=99=
m not sure we need to push it so far, though. Is there not a regime =
where the ECN-CE marking rate indicates excessive congestion, before the =
queue overflows and drops packets?=20

--=20
Colin Perkins
https://csperkins.org/





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Hi, all,

See below.

Updated to correct/clarify text, legacy issues, etc.

Joe


-------- Forwarded Message --------
Subject: 	New Version Notification for draft-touch-tsvwg-udp-options-03.txt
Date: 	Tue, 28 Jun 2016 15:02:53 -0700
From: 	internet-drafts@ietf.org
To: 	Joe Touch <touch@isi.edu>, Dr. Joseph D. Touch <touch@isi.edu>



A new version of I-D, draft-touch-tsvwg-udp-options-03.txt
has been successfully submitted by Joe Touch and posted to the
IETF repository.

Name:		draft-touch-tsvwg-udp-options
Revision:	03
Title:		Transport Options for UDP
Document date:	2016-06-28
Group:		Individual Submission
Pages:		13
URL:            https://www.ietf.org/internet-drafts/draft-touch-tsvwg-udp-options-03.txt
Status:         https://datatracker.ietf.org/doc/draft-touch-tsvwg-udp-options/
Htmlized:       https://tools.ietf.org/html/draft-touch-tsvwg-udp-options-03
Diff:           https://www.ietf.org/rfcdiff?url2=draft-touch-tsvwg-udp-options-03

Abstract:
   Transport protocols are extended through the use of transport header
   options. This document experimentally extends UDP to provide a
   location, syntax, and semantics for transport layer options.

                                                                                  


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The IETF Secretariat




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    Hi, all,<br>
    <br>
    See below.<br>
    <br>
    Updated to correct/clarify text, legacy issues, etc.<br>
    <br>
    Joe<br>
    <div class="moz-forward-container"><br>
      <br>
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        cellspacing="0">
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            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:
            </th>
            <td>New Version Notification for
              draft-touch-tsvwg-udp-options-03.txt</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
            <td>Tue, 28 Jun 2016 15:02:53 -0700</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
            <td>Joe Touch <a class="moz-txt-link-rfc2396E" href="mailto:touch@isi.edu">&lt;touch@isi.edu&gt;</a>, Dr. Joseph D. Touch
              <a class="moz-txt-link-rfc2396E" href="mailto:touch@isi.edu">&lt;touch@isi.edu&gt;</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>A new version of I-D, draft-touch-tsvwg-udp-options-03.txt
has been successfully submitted by Joe Touch and posted to the
IETF repository.

Name:		draft-touch-tsvwg-udp-options
Revision:	03
Title:		Transport Options for UDP
Document date:	2016-06-28
Group:		Individual Submission
Pages:		13
URL:            <a class="moz-txt-link-freetext" href="https://www.ietf.org/internet-drafts/draft-touch-tsvwg-udp-options-03.txt">https://www.ietf.org/internet-drafts/draft-touch-tsvwg-udp-options-03.txt</a>
Status:         <a class="moz-txt-link-freetext" href="https://datatracker.ietf.org/doc/draft-touch-tsvwg-udp-options/">https://datatracker.ietf.org/doc/draft-touch-tsvwg-udp-options/</a>
Htmlized:       <a class="moz-txt-link-freetext" href="https://tools.ietf.org/html/draft-touch-tsvwg-udp-options-03">https://tools.ietf.org/html/draft-touch-tsvwg-udp-options-03</a>
Diff:           <a class="moz-txt-link-freetext" href="https://www.ietf.org/rfcdiff?url2=draft-touch-tsvwg-udp-options-03">https://www.ietf.org/rfcdiff?url2=draft-touch-tsvwg-udp-options-03</a>

Abstract:
   Transport protocols are extended through the use of transport header
   options. This document experimentally extends UDP to provide a
   location, syntax, and semantics for transport layer options.

                                                                                  


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</pre>
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In-Reply-To: <2E09525C-C1AD-41D1-AE22-865518FA0FBE@csperkins.org>
Date: Wed, 29 Jun 2016 10:54:32 +0200
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References: <ccf9f2d7-2694-4336-0ec9-ccfebfeb0120@ericsson.com> <CE03DB3D7B45C245BCA0D243277949362F585D3E@MX307CL04.corp.emc.com> <d97e30a7-70f5-26d0-c3a4-0497c669f5f6@ericsson.com> <CE03DB3D7B45C245BCA0D243277949362F586054@MX307CL04.corp.emc.com> <D19E595F-7C66-4AE9-92B4-D550A93F634D@csperkins.org> <CE03DB3D7B45C245BCA0D243277949362F589335@MX307CL04.corp.emc.com> <20160616222548.GB77166@verdi> <0643E158-BF26-4692-8167-B7A959CB20CE@csperkins.org> <CE03DB3D7B45C245BCA0D243277949362F596DBC@MX307CL04.corp.emc.com> <E16BEA87-1D0F-48F1-A9AC-2729079D581D@tik.ee.ethz.ch> <8C16F1C6-B4A7-4BB4-B215-D7E7EAF308F8@erg.abdn.ac.uk> <CE03DB3D7B45C245BCA0D243277949362F59C41D@MX307CL04.corp.emc.com> <3E053A65-2698-4749-8E3D-E0451DF84011@ifi.uio.no> <BF6B00CC65FD2D45A326E74492B2C19FB76A6433@FR711WXCHMBA05.zeu.alcatel-lucent.com> <32a23d69d22062669f78df806a4eb6b8.squirrel@erg.abdn.ac.uk> <BF6B00CC65FD2D45A326E74492B2C19FB76A659B@FR711WXCHMBA05.zeu.alcatel-lucent.com> <CE03DB3D7B45C245BCA0D24327! 7949362 F5 AEE02@MX307CL04.corp.emc.com> <6E35FB6C-CA98-413C-B7AE-75402A968017@ifi.uio.no> <3FD27BBF-8E2D-4A42-86A0-C4C0692FF8C9@csperkins.org> <A1874131-D163-4740-98B9-61F055230A04@ifi.uio.no> <CE03DB3D7B45C245BCA0D243277949362F5AFAE1@MX307CL04.corp.emc.com> <2E09525C-C1AD-41D1-AE22-865518FA0FBE@csperkins.org>
To: Colin Perkins <csp@csperkins.org>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, tsvwg <tsvwg@ietf.org>, IETF AVTCore WG <avt@ietf.org>
Subject: Re: [tsvwg] [AVTCORE] [rtcweb] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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> On 29. jun. 2016, at 00.02, Colin Perkins <csp@csperkins.org> wrote:
>=20
>=20
>> On 28 Jun 2016, at 02:04, Black, David <david.black@emc.com> wrote:
>>=20
>> Trying to shorten up this thread again ...
>>=20
>>>>>> I'm not quite sure how to specify "use of ECN as additional =
evidence" of
>>>>>> "excessive congestion" as drop-equivalence is about the best we =
have
>>>>>> for current guidance.
>>>>>=20
>>>>> I fail to parse that sentence, so maybe I=E2=80=99m getting you =
wrong, but anyway I
>>>>> wonder: what=E2=80=99s even the point of this?
>>>>> Why even bother considering CE-marks as information for a circuit =
breaker?
>>>>=20
>>>> Because the alternative is that we only break the circuit once the =
queue has
>>> been driven into overflow, and packets have been lost. We want to =
avoid that,
>>> since it causes latency, and too much latency is very bad for the =
user experience.
>>>=20
>>> Well - the better way out would be for the application to react. =
Maybe this is me
>>> misunderstanding the circuit breaker, but I did think it=E2=80=99s =
more like a last resort=E2=80=A6
>>> you just don=E2=80=99t want to be trigger-happy with such a thing?
>>=20
>> Well, the RTP circuit breaker draft is not trigger happy - for its =
congestion circuit
>> breaker to trip, RTP has to be sending at 10x the rate that TCP would =
send under
>> those conditions, based on the TCP throughput equation.  See:
>>=20
>> =
https://tools.ietf.org/html/draft-ietf-avtcore-rtp-circuit-breakers-16#sec=
tion-4.3
>>=20
>> The issue here is - when calculating the comparable TCP throughput, =
how are ECN-CE
>> marks used to determine the loss rate input to the TCP throughput =
equation?  Do
>> ECN-CE marked packets count as having arrived or having been dropped?
>=20
> Right - or do they count somewhere between the two.

Let=E2=80=99s see them clearly for what they are.
They mean: the path is *not* broken (they have arrived!), and a probably =
an AQM mechanism, potentially using a shallow queue, marked them to =
indicate congestion. I think =E2=80=9Csomewhere between the two=E2=80=9D =
really doesn=E2=80=99t capture this well.


>> When things are relatively stable and the ECN-CE marks are being used =
to nudge
>> the sender's rate based on what the network can absorb, whether =
ECN-CE marks
>> count as losses or not is probably immaterial - the 10x divergence =
from the TCP
>> throughput equation's rate is not going to arise, and the circuit =
breaker won't trip.
>> The circuit breaker is only supposed to trip when things are =
seriously wrong.
>=20
> Correct.
>=20
>> (1) If the RTP congestion circuit breaker trips based on ECN-CE marks =
alone,
>> something feels intuitively wrong - how'd we get to RTP running at =
10x the
>> comparable TCP sending rate with no losses?  Perhaps the circuit =
breaker
>> shouldn=E2=80=99t trip on ECN-CE marks alone?
>=20
> Shouldn=E2=80=99t the comparable rate to trigger the circuit breaker =
be 10x that given to a TCP flow subject to the same ECN-CE marking rate? =
If the TCP treats ECN-CE as equivalent to loss, for congestion response, =
then the circuit breaker should do so to, etc.

First, TCP shouldn=E2=80=99t (treat ECN-CE as equivalent to loss), and =
so the circuit breaker shouldn=E2=80=99t.
Second, I guess you=E2=80=99re talking about the equation. Well that =
goes completely wrong anyway (the derivation assumes packets to be lost, =
not marked; then again, you=E2=80=99re using loss, not the loss event =
ratio; then again, you=E2=80=99re close to this with ECN perhaps, using =
=E2=80=9Ctraditional=E2=80=9D ECN receiver behavior).


>> (2) At the other extreme, the congestion circuit breaker clearly has =
to trip if RTP
>> gets to 10x the comparable TCP sending rate based on losses alone.  =
This is the
>> baseline for the circuit breaker to provide network protection as =
intended.
>>=20
>> So, going back to Gorry's suggestion to use ECN-CE marks as =
"additional evidence,"
>> here's a straw proposal to shoot at ... factor in ECN-CE marks as =
additional losses
>> *only when* losses are already occurring.  =20

I think this is very reasonable.


>> For example, we could specify that for the RTP congestion circuit =
breaker to trip, the
>> RTP sending rate has to be:
>> 	- 10x the equivalent TCP sending rate based on counting ECN-CE =
marked
>> 		packets as lost; AND
>> 	- 3x the equivalent sending rate based on actual drops (i.e., =
counting
>> 		ECN-CE marked packets as delivered).
>> The "3x" above is an off-the-top-of-my-head factor that attempts to =
roughly
>> equally weight the inputs (3 is close to the square root of 10) - =
pick a different
>> number if that weighting feels wrong.
>>=20
>> This would force drops to occur and then consider ECN-CE marks as =
additional evidence
>> that something is wrong in the network.
>>=20
>> Another possible rationale for this mixing is that if drops start =
occurring, then many of
>> the new and proposed uses of ECN that treat ECN-CE marks as less than =
loss-equivalent
>> are outside their intended operating envelopes/regions.
>=20
> Clearly if the queue has been driven to overflow, so that packet loss =
is occurring, then the AQM is outside its intended operating regime. =
I=E2=80=99m not sure we need to push it so far, though. Is there not a =
regime where the ECN-CE marking rate indicates excessive congestion, =
before the queue overflows and drops packets?=20

Shouldn=E2=80=99t a congestion control mechanism react well before that?

Cheers,
Michael


From nobody Wed Jun 29 07:02:21 2016
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From: "Black, David" <david.black@emc.com>
To: "Black, David" <david.black@emc.com>, Martin Stiemerling <mls.ietf@gmail.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: RFC 5405bis draft - non-congestion-controlled IP traffic
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Date: Wed, 29 Jun 2016 14:01:39 +0000
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Subject: [tsvwg] RFC 5405bis draft - non-congestion-controlled IP traffic
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From nobody Wed Jun 29 07:44:45 2016
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From: "Black, David" <david.black@emc.com>
To: Michael Welzl <michawe@ifi.uio.no>, Colin Perkins <csp@csperkins.org>
Thread-Topic: [AVTCORE] [rtcweb] [tsvwg] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, tsvwg <tsvwg@ietf.org>, IETF AVTCore WG <avt@ietf.org>
Subject: Re: [tsvwg] [AVTCORE] [rtcweb] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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From nobody Wed Jun 29 07:54:28 2016
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To: "Black, David" <david.black@emc.com>
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Subject: Re: [tsvwg] [AVTCORE] [rtcweb] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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> On 29. jun. 2016, at 16.44, Black, David <david.black@emc.com> wrote:
>=20
> I wrote:=20
>=20
>>> Another possible rationale for this mixing is that if drops start =
occurring, then many of
>>> the new and proposed uses of ECN that treat ECN-CE marks as less =
than loss-equivalent
>>> are outside their intended operating envelopes/regions.
>=20
> Colin responded:
>=20
>> Clearly if the queue has been driven to overflow, so that packet loss =
is
>> occurring, then the AQM is outside its intended operating regime. =
I=E2=80=99m not sure
>> we need to push it so far, though. Is there not a regime where the =
ECN-CE
>> marking rate indicates excessive congestion, before the queue =
overflows and
>> drops packets?
>=20
> Yes, but that may not be relevant to this discussion.   My hypothesis =
for this discussion
> is that actual drops will occur well before RTP  is running at 10x the =
TCP equivalent rate
> (based on drops and ECN-CE marks), and that 10x factor is the trip =
threshold for the
> circuit breaker, which is the focus of this discussion.=20
>=20
> I would think that the "regime where the ECN-CE marking rate indicates =
excessive
> congestion, before the queue overflows and drops packets?" would not =
extend as
> far as RTP running at 10x the TCP equivalent rate, which is where the =
RTP circuit
> breaker trips.

Seems reasonable to me=E2=80=A6.

>=20
> This is all a thought exercise - I'm happy to be shown to be wrong =
based on actual
> data and experience with "running code" ...
>=20
> To which Michael responded:=20
>=20
>> Shouldn=E2=80=99t a congestion control mechanism react well before =
that?
>=20
> Yes, if there is one ;-).  This is RTP =E2=80=A6

Earlier, in the same conversation, so - I assume - the same context, =
Colin wrote:
"Assuming, in all cases, that there=E2=80=99s a parallel congestion =
control algorithm running=E2=80=9D

Cheers,
Michael


From nobody Wed Jun 29 13:01:27 2016
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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 Transport Area Working Group of the IETF.

        Title           : GRE-in-UDP Encapsulation
        Authors         : Lucy Yong
                          Edward Crabbe
                          Xiaohu Xu
                          Tom Herbert
	Filename        : draft-ietf-tsvwg-gre-in-udp-encap-12.txt
	Pages           : 24
	Date            : 2016-06-29

Abstract:
   This document specifies a method of encapsulating network protocol
   packet within GRE and UDP headers. This GRE-in-UDP encapsulation
   allows the UDP source port field to be used as an entropy field.
   This may be used for load balancing of GRE traffic in transit
   networks using existing ECMP mechanisms. This document also
   specifies GRE-in-UDP tunnel requirements for two applicability
   scenarios: (1) general Internet; (2) a traffic-managed controlled
   environment. The controlled environment has less restrictive
   requirements than the general Internet.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-gre-in-udp-encap/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-tsvwg-gre-in-udp-encap-12

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-gre-in-udp-encap-12


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

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


From nobody Wed Jun 29 13:08:06 2016
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From: "Eggert, Lars" <lars@netapp.com>
To: "Black, David" <david.black@emc.com>
Thread-Topic: [tsvwg] RFC 5405bis draft - non-congestion-controlled IP traffic
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Subject: Re: [tsvwg] RFC 5405bis draft - non-congestion-controlled IP traffic
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I made both of these changes in our working copy of -16. Will wait a =
little while before submitting in case there are more change requests.

Thanks,
Lars

On 2016-06-29, at 16:01, Black, David <david.black@emc.com> wrote:
>=20
> A while back, in response to Martin Stiemerling's comments on the =
GRE/UDP draft,
> I wrote:
>=20
>> [2] IP traffic congestion control assumptions.
>>=20
>>> - I do disagree with "IP-traffic can be assumed to be
>>> congestion-controlled.", even though it is stated in RFC 5405, as =
there
>>> is a lot of IP traffic that is not congestion controlled, such as
>>> multicast traffic...
>>> Section 4, even states that multicast and broadcast traffic can be
>>> carried over GRE-UDP! None of both is in general congestion =
controlled!
>>=20
>> Well, this is actually an issue with the rfc5405bis draft, and this =
issue needs
>> to be fixed there, as RFC 5405 is (and the rfc5405bis draft will be) =
the
>> Transport Area's statement to the rest of the IETF on congestion =
control
>> guidelines and requirements for UDP usage.
>=20
> This is indeed an issue with the 5405bis draft.  In particular, Martin =
has
> effectively pointed out that this text in Section 3.1.11 is wrong =
because
> it encompasses multicast and broadcast traffic:
>=20
>   IP-based traffic is generally assumed to be congestion-controlled,
>   i.e., it is assumed that the transport protocols generating IP-based
>   traffic at the sender already employ mechanisms that are sufficient
>   to address congestion on the path.  Consequently, a tunnel carrying
>   IP-based traffic should already interact appropriately with other
>   traffic sharing the path, and specific congestion control mechanisms
>   for the tunnel are not necessary.
>=20
> Changing "IP-based traffic" to "IP-based unicast traffic" in three =
places
> in the above paragraph should correct this problem.
>=20
> In addition, multicast needs to be called out as an exception in the =
next
> paragraph:
>=20
> OLD
>   However, if the IP traffic in the tunnel is known to not be
>   congestion-controlled, additional measures are RECOMMENDED to limit
>   the impact of the tunneled traffic on other traffic sharing the =
path.
>=20
>   The following guidelines define these possible cases in more detail:
> NEW
>   However, if the IP traffic in the tunnel is known to not be
>   congestion-controlled, additional measures are RECOMMENDED to limit
>   the impact of the tunneled traffic on other traffic sharing the =
path.   For
>   the specific case of a tunnel that carries IP multicast traffic, see =
section 4.1.
>=20
>   The following guidelines define these possible cases in more detail:
>=20
> As a WG chair and draft shepherd, I think these changes are important,
> as the original text has already been misread/misinterpreted by more
> than one UDP tunnel designer.
>=20
> Thanks, --David


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From nobody Thu Jun 30 07:51:39 2016
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To: Michael Welzl <michawe@ifi.uio.no>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, tsvwg <tsvwg@ietf.org>, IETF AVTCore WG <avt@ietf.org>
Subject: Re: [tsvwg] [AVTCORE] [rtcweb] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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> On 29 Jun 2016, at 15:54, Michael Welzl <michawe@ifi.uio.no> wrote:
>> On 29. jun. 2016, at 16.44, Black, David <david.black@emc.com> wrote:
>> I wrote:=20
>>=20
>>>> Another possible rationale for this mixing is that if drops start =
occurring, then many of
>>>> the new and proposed uses of ECN that treat ECN-CE marks as less =
than loss-equivalent
>>>> are outside their intended operating envelopes/regions.
>>=20
>> Colin responded:
>>=20
>>> Clearly if the queue has been driven to overflow, so that packet =
loss is
>>> occurring, then the AQM is outside its intended operating regime. =
I=E2=80=99m not sure
>>> we need to push it so far, though. Is there not a regime where the =
ECN-CE
>>> marking rate indicates excessive congestion, before the queue =
overflows and
>>> drops packets?
>>=20
>> Yes, but that may not be relevant to this discussion.   My hypothesis =
for this discussion
>> is that actual drops will occur well before RTP  is running at 10x =
the TCP equivalent rate
>> (based on drops and ECN-CE marks), and that 10x factor is the trip =
threshold for the
>> circuit breaker, which is the focus of this discussion.=20
>>=20
>> I would think that the "regime where the ECN-CE marking rate =
indicates excessive
>> congestion, before the queue overflows and drops packets?" would not =
extend as
>> far as RTP running at 10x the TCP equivalent rate, which is where the =
RTP circuit
>> breaker trips.
>=20
> Seems reasonable to me=E2=80=A6.

Does anyone have data to let us find out either way?=20

>> This is all a thought exercise - I'm happy to be shown to be wrong =
based on actual
>> data and experience with "running code" ...
>>=20
>> To which Michael responded:=20
>>=20
>>> Shouldn=E2=80=99t a congestion control mechanism react well before =
that?
>>=20
>> Yes, if there is one ;-).  This is RTP =E2=80=A6
>=20
> Earlier, in the same conversation, so - I assume - the same context, =
Colin wrote:
> "Assuming, in all cases, that there=E2=80=99s a parallel congestion =
control algorithm running=E2=80=9D

If the congestion control is working, then it should adapt the rate so =
the circuit breaker never triggers. Yes, there should be a parallel =
congestion control, but for the circuit breaker to be relevant, that =
congestion control is either not working properly, or not implemented.=20=


--=20
Colin Perkins
https://csperkins.org/





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From: "De Schepper, Koen (Nokia - BE)" <koen.de_schepper@nokia-bell-labs.com>
To: "Black, David" <david.black@emc.com>
Thread-Topic: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, tsvwg <tsvwg@ietf.org>, IETF AVTCore WG <avt@ietf.org>
Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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Subject: Re: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: draft-ietf-avtcore-rtp-circuit-breakers-16
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Hi,


> On 30. jun. 2016, at 17.58, De Schepper, Koen (Nokia - BE) =
<koen.de_schepper@nokia-bell-labs.com> wrote:
>=20
>=20
>> -----Original Message-----
>> From: Black, David [mailto:david.black@emc.com]
>> Sent: maandag 27 juni 2016 22:10
>> To: De Schepper, Koen (Nokia - BE)
>> Cc: tsvwg; IETF AVTCore WG; rtcweb@ietf.org; Black, David
>> Subject: RE: [tsvwg] [rtcweb] [AVTCORE] WG Last Call on changes: =
draft-
>> ietf-avtcore-rtp-circuit-breakers-16
>>=20
>>> As long as an AQM is marking at the same rate as dropping
>>=20
>> That's an interesting assumption - it should be true for AQMs vetted
>> here in the past,
>=20
> And I hope for ect(0) the network will keep doing this in the future, =
at=20
> least not to deviate too far from this.
> L4S proposes to use ect(1) just because it deviates too much, so the =
network=20
> can detect the different end-system behavior and can couple it =
correctly to=20
> what it is doing for drop and ect(0).
>=20
>=20
>> but there are easy ways for it not to hold (e.g., if
>> dropping
>> or marking is based on queue occupancy, it is possible that dropping
>> reduces queue occupancy in a fashion that marking does not).
>=20
> Do you mean that this results in different mark and drop probabilities=20=

> for competing ECN and non-ECN flows?
>=20
> The impact on the dropping/marking probability *value* applied by an =
AQM can change=20
> due to presence of ECN flows, but I don't think this results in a =
different=20
> drop compared to mark probability. Up to now all classic AQMs smooth =
their p=20
> over sufficient long time to remove short fluctuations in queue sizes.=20=

>=20
> =46rom an end system I think for classic ECN it is safe to assume that =
if you=20
> detect high levels of drop, it means there is a high level of=20
> congestion AND the network takes care of it. If you detect a high =
level
> of marking, it means lots of congestion and nobody is currently taking =
care=20
> of it. Everybody is looking at you as a sender to take care of it. =
That's why
> I think you should not use ECN if you don't use congestion control.
>=20
> It also means that the competing drop flows are having a high level of =
matching
> drop. That's why I think a classic circuit breaker should also treat =
drop and=20
> classic marking similar.
>=20
> I understood ABE tries to modify the end system behavior slightly for =
ECN. If=20
> this can be done safely with benefits that outweighs the =
disadvantages,=20
> than that's no problem, but I think nobody is in favor to modify the =
AQM=20
> behavior to ect(0).

I pretty much agree with everything here=E2=80=A6


> Based on DualQ compatibility experiments with classic ECN, I also =
believe=20
> that, if we want to further promote classic ECN too, we need some =
extra=20
> safety measures for classic AQMs using ECN to avoid high CE marking =
levels,=20
> where ect(0) flows start to push away too much the non-ect flows. For=20=

> instance PIE has (for good reasons) a max of 10% marking, and then =
switches=20
> to drop. As far as I know such measures are not mandatory in other =
ECN/AQM=20
> related drafts, is it?

=E2=80=A6 but here I=E2=80=99d like to put a note of warning: see the =
appendix here:
http://heim.ifi.uio.no/michawe/research/publications/CAIA-TR-150710A.pdf

here, we found that this behavior is detrimental (the 10% threshold was =
too small).
=3D> I agree with the general notion of such a threshold, but the =
threshold shouldn=E2=80=99t be too low (it can eliminate ECN=E2=80=99s =
benefits).

Cheers,
Michael

