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From nobody Tue Jul  1 00:26:44 2014
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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
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Subject: Re: [tsvwg] I-D Action: draft-ietf-tsvwg-sctp-failover-04.txt
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--001a11349488e61b4004fd1cb16b
Content-Type: text/plain; charset=UTF-8

Hi,
We've submitted a new version of SCTP quick failover draft.
We've added some changes in order to reflect the previous comments we've
received. The main parts of changes are the followings.

1: Restructuring of some sections
    We've moved some sections that are not directly related to the spec for
quick failover (related works, some discussions for side effects) to
Appendix
2: Clarifying the logics for quick failover with standards recommendations
    We've added explicit recommendations in the behavior of quick failover
(Section 4)
3: Updating abstract to provide clearer view of the document

If you have any questions, comments or suggestions, please let us know.

Regards,
--
Yoshi

On Mon, Jun 30, 2014 at 10:37 PM,  <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 Working
Group of the IETF.
>
>         Title           : Quick Failover Algorithm in SCTP
>         Authors         : Yoshifumi Nishida
>                           Preethi Natarajan
>                           Armando Caro
>                           Paul D. Amer
>                           Karen E. E. Nielsen
>         Filename        : draft-ietf-tsvwg-sctp-failover-04.txt
>         Pages           : 17
>         Date            : 2014-06-30
>
> Abstract:
>    One of the major advantages of SCTP is that it supports multi-homed
>    communication.  A multi-homed SCTP end-point has the ability to
>    withstand network failures by migrating the traffic from an inactive
>    network to an active one.  However, if the [RFC4960] specified
>    failover operation is followed there can be a significant delay in
>    the migration to the active destination addresses, thus severely
>    reducing the effectiveness of SCTP multi-homed operation.
>
>    The memo complements RFC4960 by the introduction of the Potentially
>    Failed state and associated new Quick Failover operation to apply
>    during network failure and specifies for SCTP senders to support this
>    more performance optimal failover procedure as an add-on to the
>    [RFC4960] failover operation.  The memo in addition updates [RFC4960]
>    by clarifying a number of issues related to the path management
>    operation during periods of partial or complete network failures.
>    Finally the memo complements [RFC4960] by introduction of alternative
>    switchover operation modes for the data transfer path management
>    after a failover.  These operation modes offer for more performance
>    optimal operation in some network environments.  From the perspective
>    of this memo the implementation of the additional switchover
>    operation modes is considered optional.
>
>    The procedures defined require only minimal modifications to the
>    current specification.  The procedures are sender-side only and do
>    not impact the SCTP receiver.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-failover/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-failover-04
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-sctp-failover-04
>
>
> Please note that it may take a couple of minutes from the time of
submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> 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

--001a11349488e61b4004fd1cb16b
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi,<br>We&#39;ve submitted a new version of SCTP quick fai=
lover draft.<br>We&#39;ve added some changes in order to reflect the previo=
us comments we&#39;ve received. The main parts of changes are the following=
s.<br>
<br>1: Restructuring of some sections<br>=C2=A0 =C2=A0 We&#39;ve moved some=
 sections that are not directly related to the spec for quick failover (rel=
ated works, some discussions for side effects) to Appendix<br>2: Clarifying=
 the logics for quick failover with standards recommendations<br>
=C2=A0 =C2=A0 We&#39;ve added explicit recommendations in the behavior of q=
uick failover (Section 4)<div>3: Updating abstract to provide clearer view =
of the document</div><div><br></div><div>If you have any questions, comment=
s or suggestions, please let us know.</div>
<div><br></div><div>Regards,</div><div>--</div><div>Yoshi<br><br>On Mon, Ju=
n 30, 2014 at 10:37 PM, =C2=A0&lt;<a href=3D"mailto:internet-drafts@ietf.or=
g">internet-drafts@ietf.org</a>&gt; wrote:<br>&gt;<br>&gt; A New Internet-D=
raft is available from the on-line Internet-Drafts directories.<br>
&gt; =C2=A0This draft is a work item of the Transport Area Working Group Wo=
rking Group of the IETF.<br>&gt;<br>&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 Title =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Quick Failover Algorithm in SCTP<br>&g=
t; =C2=A0 =C2=A0 =C2=A0 =C2=A0 Authors =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Yoshif=
umi Nishida<br>
&gt; =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 Preethi Natarajan<br>&gt; =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 Armando Caro=
<br>&gt; =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 Paul D. Amer<br>&gt; =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 Karen E. E. =
Nielsen<br>&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename =C2=A0 =C2=A0 =C2=A0 =
=C2=A0: draft-ietf-tsvwg-sctp-failover-04.txt<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 17<br>&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0 Date =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0: 2014-06-30<br>&gt;<br>&gt; Abstract:<br>&gt; =C2=A0 =C2=A0On=
e of the major advantages of SCTP is that it supports multi-homed<br>&gt; =
=C2=A0 =C2=A0communication. =C2=A0A multi-homed SCTP end-point has the abil=
ity to<br>
&gt; =C2=A0 =C2=A0withstand network failures by migrating the traffic from =
an inactive<br>&gt; =C2=A0 =C2=A0network to an active one. =C2=A0However, i=
f the [RFC4960] specified<br>&gt; =C2=A0 =C2=A0failover operation is follow=
ed there can be a significant delay in<br>
&gt; =C2=A0 =C2=A0the migration to the active destination addresses, thus s=
everely<br>&gt; =C2=A0 =C2=A0reducing the effectiveness of SCTP multi-homed=
 operation.<br>&gt;<br>&gt; =C2=A0 =C2=A0The memo complements RFC4960 by th=
e introduction of the Potentially<br>
&gt; =C2=A0 =C2=A0Failed state and associated new Quick Failover operation =
to apply<br>&gt; =C2=A0 =C2=A0during network failure and specifies for SCTP=
 senders to support this<br>&gt; =C2=A0 =C2=A0more performance optimal fail=
over procedure as an add-on to the<br>
&gt; =C2=A0 =C2=A0[RFC4960] failover operation. =C2=A0The memo in addition =
updates [RFC4960]<br>&gt; =C2=A0 =C2=A0by clarifying a number of issues rel=
ated to the path management<br>&gt; =C2=A0 =C2=A0operation during periods o=
f partial or complete network failures.<br>
&gt; =C2=A0 =C2=A0Finally the memo complements [RFC4960] by introduction of=
 alternative<br>&gt; =C2=A0 =C2=A0switchover operation modes for the data t=
ransfer path management<br>&gt; =C2=A0 =C2=A0after a failover. =C2=A0These =
operation modes offer for more performance<br>
&gt; =C2=A0 =C2=A0optimal operation in some network environments. =C2=A0Fro=
m the perspective<br>&gt; =C2=A0 =C2=A0of this memo the implementation of t=
he additional switchover<br>&gt; =C2=A0 =C2=A0operation modes is considered=
 optional.<br>&gt;<br>&gt; =C2=A0 =C2=A0The procedures defined require only=
 minimal modifications to the<br>
&gt; =C2=A0 =C2=A0current specification. =C2=A0The procedures are sender-si=
de only and do<br>&gt; =C2=A0 =C2=A0not impact the SCTP receiver.<br>&gt;<b=
r>&gt;<br>&gt; The IETF datatracker status page for this draft is:<br>&gt; =
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-failover/=
">https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-failover/</a><br>
&gt;<br>&gt; There&#39;s also a htmlized version available at:<br>&gt; <a h=
ref=3D"http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-failover-04">http:/=
/tools.ietf.org/html/draft-ietf-tsvwg-sctp-failover-04</a><br>&gt;<br>&gt; =
A diff from the previous version is available at:<br>
&gt; <a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-sctp-fa=
ilover-04">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-sctp-failove=
r-04</a><br>&gt;<br>&gt;<br>&gt; Please note that it may take a couple of m=
inutes from the time of submission<br>
&gt; until the htmlized version and diff are available at <a href=3D"http:/=
/tools.ietf.org">tools.ietf.org</a>.<br>&gt;<br>&gt; Internet-Drafts are al=
so available by anonymous FTP at:<br>&gt; <a href=3D"ftp://ftp.ietf.org/int=
ernet-drafts/">ftp://ftp.ietf.org/internet-drafts/</a><br>
&gt;<br>&gt; _______________________________________________<br>&gt; I-D-An=
nounce mailing list<br>&gt; <a href=3D"mailto:I-D-Announce@ietf.org">I-D-An=
nounce@ietf.org</a><br>&gt; <a href=3D"https://www.ietf.org/mailman/listinf=
o/i-d-announce">https://www.ietf.org/mailman/listinfo/i-d-announce</a><br>
&gt; Internet-Draft directories: <a href=3D"http://www.ietf.org/shadow.html=
">http://www.ietf.org/shadow.html</a><br>&gt; or <a href=3D"ftp://ftp.ietf.=
org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</a></=
div>
</div>

--001a11349488e61b4004fd1cb16b--


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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 Working Group of the IETF.

        Title           : Generic UDP Encapsulation for IP Tunneling
        Authors         : Edward Crabbe
                          Lucy Yong
                          Xiaohu Xu
	Filename        : draft-ietf-tsvwg-gre-in-udp-encap-02.txt
	Pages           : 13
	Date            : 2014-07-01

Abstract:
   This document describes a method of encapsulating arbitrary
   protocols within GRE and UDP headers.  In this encapsulation, the
   source UDP port may be used as an entropy field for purposes of load
   balancing while the payload protocol may be identified by the GRE
   Protocol Type.


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:
http://tools.ietf.org/html/draft-ietf-tsvwg-gre-in-udp-encap-02

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


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From nobody Wed Jul  2 14:55:26 2014
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Subject: Re: [tsvwg] WGLC Follow-Up: draft-ietf-tsvwg-sctp-dtls-encaps
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On 24 Jun 2014, at 15:28, Gorry Fairhurst <gorry@erg.abdn.ac.uk> wrote:

> The WGLC for the above document ended 28th February 2014, and =
follow-up discussions resulted in a revised draft.
>=20
> I am now starting to prepare for submission of the above document for =
requested publication. Please find detailed comments/questions about the =
text below.
Thank you very much for your comments. See my answers in-line.

Best regards
Michael
>=20
> Best wishes,
>=20
> Gorry Fairhurst.
> (TSVWG Co-Chair)
>=20
> ---
> ABSTRACT:
>   "Using
>   encapsulation method described in this document, SCTP is..."
>  ^
> - insert /the/ ?
Done.
>=20
>=20
> ---
> ABSTRACT:
>  "As a consequence, the SCTP associations are single homed."
> - This final statement is not clear, although true. Please add text to =
explain to the reader why this is needed in the abstract. Could you say =
something that follows this below, and also explain why in section 6.1?
Not sure why the conclusion is not clear. How would you setup a =
multihomed
SCTP association without using IP addresses in control chunks =
(INIT/INIT-ACK/ASCONF)?
> "The security properties of this encapsulation limit usage to =
single-homed SCTP associations."
> ---
> Section 5:
>   "a safe value for the path MTU has to be used by the SCTP stack."
>   - How do you choose? - please specify a default or a method.
The data channel document specifies 1280 for IPv6 and 1200 for IPv4.
The IPv4 values was chosen based on experience by some WG members.
I can put the numbers in this document, if you want.
> ---
> Section 5:
>   "Differentiated services code point (DSCP) "
>   - reference needed.
I added a reference to RFC 2474.
> ---
> Section 5:
>   "explicit congestion notifications (ECN)"
>   - reference needed.
Added a reference to RFC 3168.
> ---
> Section 5:
>   "SCTP performs segmentation and reassembly based on the path MTU.
>   Therefore the DTLS layer MUST NOT use any compression algorithm.
>=20
>   The DTLS MUST support sending messages larger than the current path
>   MTU.  This might result in sending IP level fragmented messages."
>   - I wonder if some reordering of paras within teh section could
>   gather all the size-related topics in one part of the section?
Makes sense. I moved them up.
> ---
> Section 6:
>   - I am surprised this section does not mention how to handle
>   any path changes, should PMTUD be triggered following known changes,
>   for instance? Does this affect the SCTP CC state?
The data channel document contains:

The ICE/UDP layer can handle IP address changes during a session without =
needing
interaction with the DTLS and SCTP layers.
However, SCTP SHOULD be notified when an address changes has happened.
In this case SCTP SHOULD retest the Path MTU and reset the congestion
state to the initial state.
In case of a window based congestion control like the one specified in
<xref target=3D'RFC4960'/>, this means setting the congestion window and
slow start threshold to its initial values.</t>

So I added a bullet:

<t>If the SCTP is notified about a path change by its lower layers,
   SCTP SHOULD retest the Path MTU and reset the congestion state to the
   initial state. In case of a window based congestion control like the =
one
   specified in <xref target=3D'RFC4960'/>, this means setting the =
congestion
   window and slow start threshold to its initial values.</t>

Does this address your issue?
> ---
> Section 6.1:
>    /doesn't deal/does not deal/
>    - please change.
Done.
> ---
> Section 6.1:
>   "All SCTP associations are single-homed."
>   Could you add something like:
>   ", because DTLS does not expose any address
>   management to its upper layer."
Done.
> ---
> Section 6.2:
>  "The padding extension defined in [RFC4820] MUST be supported and =
used
>   for probe packets when performing path MTU discovery as specified in
>   [RFC4821]."
> - Is this section relevant when DTLS implements PMTUD, please explain?
I added "by the SCTP layer".
> ---
> References:
> A later version (-10) exists ofdraft-ietf-rtcweb-overview-09
> A later version (-10) exists of draft-ietf-rtcweb-data-channel-08
They will be updated when submitting a new revision.
> ---
> The requirement to use SCTP/DTLS appears to also be cited in =
draft-ietf-rtcweb-overview-09, section 5.5, which requires normative use =
of SCTP/DTLS, but this document is not cited in the current document, I =
think it should be, if not why not?
It is cited as the first ID in
=
http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-dtls-encaps-04#section-10=
.2
>=20
>=20
>=20
>=20
>=20


From nobody Wed Jul  2 22:51:21 2014
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From: "Fred Baker (fred)" <fred@cisco.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: [OPS-DIR] Early Ops-Dir review for rtcweb-qos draft
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Subject: [tsvwg] Fwd: [OPS-DIR] Early Ops-Dir review for rtcweb-qos draft
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David Black asked me to forward my Ops Directorate =93early review=94 of =
draft-ietf-tsvwg-rtcweb-qos to the list.

As an additional comment, reflecting on comments in other quarters =
(dart), it may be worthwhile pointing out in the draft that the DSCP =
indicates a minimum acceptable service level; if a network thinks it can =
achieve that service level within some larger scheme (such as a traffic =
class described in Ruediger=92s draft, RFC 5127, or simply by =
oversupplying bandwidth), that is at the operator=92s discretion.

Begin forwarded message:

> From: "Fred Baker (fred)" <fred@cisco.com>
> Subject: Re: [OPS-DIR] Early Ops-Dir review for rtcweb-qos draft
> Date: July 2, 2014 at 11:07:43 AM PDT
> To: "Benoit Claise (bclaise)" <bclaise@cisco.com>, "ops-dir@ietf.org" =
<ops-dir@ietf.org>
> Cc: "tsv-ads@tools.ietf.org" <tsv-ads@tools.ietf.org>, =
"tsvwg-chairs@tools.ietf.org" <tsvwg-chairs@tools.ietf.org>, =
"draft-ietf-tsvwg-rtcweb-qos.all@tools.ietf.org" =
<draft-ietf-tsvwg-rtcweb-qos.all@tools.ietf.org>
>=20
>=20
> On Jul 2, 2014, at 9:40 AM, Benoit Claise <bclaise@cisco.com> wrote:
>=20
>> Hi David,
>>=20
>> Fred Baker accepted to review this document for the early Ops-DIR
>> Thanks Fred.
>> Gunter, can you please mark it as such in the in OPS DIR review tool.
>>=20
>> Regards, Benoit
>>=20
>>> Greetings,
>>>=20
>>> On behalf of the tsvwg WG, I'd like to request an early OPS-Dir =
review of draft-ietf-tsvwg-rtcweb-qos from a network operations =
perspective:
>>>=20
>>> 	http://datatracker.ietf.org/doc/draft-ietf-tsvwg-rtcweb-qos/
>>>=20
>>> Although I'm an OPS-Dir member, I'm looking for a review from =
someone w/actual network operations expertise, which I freely admit is =
not me ;-).  The concern that motivates this review is that this draft =
recommends use of a significant number of Diffserv DSCPs in Section 5:
>>>=20
>>> 	=
http://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-02#section-5
>>>=20
>>> I have input from one large network operator (DT) that this is well =
off (by a factor of 2 or more) from the number of DSCPs that are (and =
are likely to be) supported on their network (and other similar =
networks) for non-internal (e.g., subscriber/customer) traffic, and =
hence much of the differentiation is likely to vanish on ingress to such =
a network, if not before.
>>>=20
>>> Input from someone else familiar with the QoS aspects of network =
operations would be rather helpful at this stage.
>>>=20
>>> Thanks,
>>> =97David
>=20
> In fairness, I=92m not an operator either. To directly answer David's =
question, you need an operator to respond.
>=20
> I am, however, one of the authors of RFC 4594, which this note quotes, =
and RFC 5127, which addresses aggregation of those classes in high speed =
networks.=20
>=20
> https://tools.ietf.org/html/rfc4594
> 4594 Configuration Guidelines for DiffServ Service Classes. J.
>     Babiarz, K. Chan, F. Baker. August 2006. (Format: TXT=3D144044 =
bytes)
>     (Updated by RFC5865) (Status: INFORMATIONAL)
>=20
> https://tools.ietf.org/html/rfc5127
> 5127 Aggregation of Diffserv Service Classes. K. Chan, J. Babiarz, F.
>     Baker. February 2008. (Format: TXT=3D43751 bytes) (Status:
>     INFORMATIONAL)
>=20
> The heart of the document is section 5, in which it makes specific =
DSCP recommendations. Were I to try to implement it, I think I would =
find it confusing and difficult. Color me pedantic, but let me give you =
the questions that flow through my head as I read it.
>=20
> An important statement made here is:
>=20
>>   =
+---------------------------+-------+------+------------+-----------+
>>   |         Data Type         |  Very | Low  |   Medium   |    High   =
|
>>   |                           |  Low  |      |            |           =
|
>>   =
+---------------------------+-------+------+------------+-----------+
>>   |           Audio           |  CS1  |  BE  |  EF (46)   |  EF (46)  =
|
>>   |                           |  (8)  | (0)  |            |           =
|
>>          <various other classes and code points>
>>   =
+---------------------------+-------+------+------------+-----------+
>>=20
>>                                  Table 1
>>=20
>>   The columns "very low", "low", "Medium" and "high" are the priority
>>   levels.   The browser SHOULD first select the data type of the =
media
>>   flow.  Within the data type, the priority of the media flow SHOULD =
be
>>   selected.  All packets within a media flow SHOULD have the same
>>   priority.  [snip]
>=20
> RFC 4594 has no corollary concept of =93priority=94 of the data =
traffic. It focuses on the type of application and that application=92s =
requirements; if it needs to minimize network-induced jitter, it might =
recommend using a priority scheduler, transferring jitter from EF to =
other traffic classes, and if it targets TCP traffic, it might recommend =
the use of AQM technologies. For it, the question is not the relative =
importance of the session to the user or some other outside observer, =
but the technology required to deliver the QoS services that the =
application requires. By comparison, RFC 5865 does discuss precedence =
classes for sessions; in that case, the question is a service policy =
that might admit a new session to the network if it is deemed =
=93important=94 via some authenticated authorization process when it =
would not admit a new session from just anyone. So I struggle to =
determine how the browser would decide whether a given session was =93very=
 low=94, =93low=94, =93medium=94, or =93high=94 priority, and what the =
concept of =93priority=94 means in the context.
>=20
> I also struggle to understand the meaning of the reference to =93BE=94. =
I presume it means =93Best Effort=94. RFC 4594 does not describe a =93Best=
 Effort=94 DSCP or traffic class; it does describe =93Default=94, which =
I think this is referring to. Similarly, definitions of the =
Differentiated Services architecture (I haven=92t checked all of the 272 =
specifications that include the text string =93Differentiated Services=94,=
 but I have reviewed the documents from RFC 2474 through 2999) make no =
such reference. =93Best Effort=94 is a term describing a characteristic =
of any datagram network service; packets in flight may be dropped, =
duplicated, or reordered, as compared with services that enforce =
in-order and in some cases reliable delivery. I would prefer to see the =
appropriate traffic class, which I think might be =93Default=94, used in =
this document.
>=20
> The document also makes a statement that I think is unnecessary:
>=20
>   The above table assumes that packets marked with CS1 is treated as
>   "less than best effort".  However, the treatment of CS1 is
>   implementation dependent.  If an implementation treats CS1 as other
>   than "less than best effort", then the priority of the packets may =
be
>   changed from what is intended.
>=20
> With respect, the treatment of every traffic class is =
implementation-dependent, and every mark is fungible. The document =
should describe how traffic should be marked, and what traffic so marked =
expects from the network. The network will then do whatever it thinks is =
appropriate to satisfy that requirement.
>=20
> I would not recommend the adoption or publication of the draft in its =
present form.
> _______________________________________________
> OPS-DIR mailing list
> OPS-DIR@ietf.org
> https://www.ietf.org/mailman/listinfo/ops-dir


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From nobody Thu Jul  3 01:45:55 2014
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From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: [tsvwg] WGLC for draft-ietf-tsvwg-sctp-prpolicies-03 - due to conclude Monday, July 21st, 2014
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This is the start of the WGLC on "Additional Policies for the Partial 
Reliability Extension of the Stream Control Transmission Protocol" for 
publication intended as a PS.

https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-prpolicies/

This WGLC will last until Monday, July 21st, 2014 (longer since this 
includes time prior to an IETF meeting).

Could people please send *any* comments to the list if they think this 
is ready to publish?

A good note is "I have read this version of the draft,
and I approve (or support) its progression to RFC".

James/Gorry/David
TSVWG chairs


From nobody Thu Jul  3 02:37:39 2014
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From: Weixinpeng <weixinpeng@huawei.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: draft-wei-tsvwg-tunnel-congestion-feedback-02
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Hi all,

A new version of draft-wei-tsvwg-tunnel-congestion-feedback-02 is submitted=
.

Comments are welcomed!

Thanks.



Best Regards,

-Xinpeng





Name:               draft-wei-tsvwg-tunnel-congestion-feedback

Revision:  02

Title:                  Tunnel Congestion Feedback

Document date:       2014-07-03

Group:               Individual Submission

Pages:               18

URL:            http://www.ietf.org/internet-drafts/draft-wei-tsvwg-tunnel-=
congestion-feedback-02.txt

Status:         https://datatracker.ietf.org/doc/draft-wei-tsvwg-tunnel-con=
gestion-feedback/

Htmlized:       http://tools.ietf.org/html/draft-wei-tsvwg-tunnel-congestio=
n-feedback-02

Diff:           http://www.ietf.org/rfcdiff?url2=3Ddraft-wei-tsvwg-tunnel-c=
ongestion-feedback-02



Abstract:

   This document describes a mechanism to calculate congestion of a

   tunnel segment based on RFC 6040 recommendations, and a feedback

   protocol by which to send the measured congestion of the tunnel from

   egress to ingress router. A basic  model for measuring tunnel

   congestion and feedback is described, and a protocol for carrying the

   feedback data is outlined.


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<p class=3D"MsoPlainText"><span lang=3D"EN-US">Hi all,<o:p></o:p></span></p=
>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">A new version of draft-wei-t=
svwg-tunnel-congestion-feedback-02 is submitted.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Comments are welcomed!<o:p><=
/o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Thanks.<o:p></o:p></span></p=
>
<p class=3D"MsoPlainText"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Best Regards,<o:p></o:p></sp=
an></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">-Xinpeng<o:p></o:p></span></=
p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Name:&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; draft-wei-tsv=
wg-tunnel-congestion-feedback<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Revision:&nbsp; 02<o:p></o:p=
></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Title:&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; Tunnel Congestion Feedback<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Document date:&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp; 2014-07-03<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Group:&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Individual S=
ubmission<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Pages:&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 18<o:p></o:p=
></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">URL:&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D"http://www.ietf.org/i=
nternet-drafts/draft-wei-tsvwg-tunnel-congestion-feedback-02.txt">
http://www.ietf.org/internet-drafts/draft-wei-tsvwg-tunnel-congestion-feedb=
ack-02.txt</a><o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Status:&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D"https://datatracker.ietf.org/doc/dra=
ft-wei-tsvwg-tunnel-congestion-feedback/">
https://datatracker.ietf.org/doc/draft-wei-tsvwg-tunnel-congestion-feedback=
/</a><o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Htmlized:&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp; <a href=3D"http://tools.ietf.org/html/draft-wei-tsvwg-tun=
nel-congestion-feedback-02">
http://tools.ietf.org/html/draft-wei-tsvwg-tunnel-congestion-feedback-02</a=
><o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Diff:&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D"http://www.ietf.org/rfcdif=
f?url2=3Ddraft-wei-tsvwg-tunnel-congestion-feedback-02">
http://www.ietf.org/rfcdiff?url2=3Ddraft-wei-tsvwg-tunnel-congestion-feedba=
ck-02</a><o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">Abstract:<o:p></o:p></span><=
/p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp; This document d=
escribes a mechanism to calculate congestion of a<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp; tunnel segment =
based on RFC 6040 recommendations, and a feedback<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp; protocol by whi=
ch to send the measured congestion of the tunnel from<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp; egress to ingre=
ss router. A basic&nbsp; model for measuring tunnel<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp; congestion and =
feedback is described, and a protocol for carrying the<o:p></o:p></span></p=
>
<p class=3D"MsoPlainText"><span lang=3D"EN-US">&nbsp;&nbsp; feedback data i=
s outlined.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
</div>
</body>
</html>

--_000_C5C3BB522B1DDF478AA09545169155B46D81AA01nkgeml507mbxchi_--


From nobody Thu Jul  3 06:01:49 2014
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Date: Thu, 3 Jul 2014 14:01:02 +0100
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Subject: [tsvwg] TSVWG - Request for Agenda Items for Toronto meetings
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Hi tsvwg'ers,

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

I've upload a very initial agenda here:
http://www.ietf.org/proceedings/90/agenda/agenda-90-tsvwg

Many of the talks are marked TBC (editors please confirm). We do have
space to discuss additional drafts.

Please send all requests and any corrections to the chairs asap.

Gorry, David and James
(tsvwg co-chairs)



From nobody Thu Jul  3 13:26:47 2014
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Date: Fri, 04 Jul 2014 08:26:32 +1200
From: Brian E Carpenter <brian.e.carpenter@gmail.com>
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Subject: Re: [tsvwg] Fwd: [OPS-DIR] Early Ops-Dir review for rtcweb-qos draft
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Hi Fred,

I agree with your review. Just two remarks inserted below:

On 03/07/2014 17:50, Fred Baker (fred) wrote:
> David Black asked me to forward my Ops Directorate =E2=80=9Cearly revie=
w=E2=80=9D of draft-ietf-tsvwg-rtcweb-qos to the list.
>=20
> As an additional comment, reflecting on comments in other quarters (dar=
t), it may be worthwhile pointing out in the draft that the DSCP indicate=
s a minimum acceptable service level; if a network thinks it can achieve =
that service level within some larger scheme (such as a traffic class des=
cribed in Ruediger=E2=80=99s draft, RFC 5127, or simply by oversupplying =
bandwidth), that is at the operator=E2=80=99s discretion.

Yes. However, EF is a little bit strange in that respect, as I understand=

the formal definition. EF is supposed to provide a configured maximum
capacity, and traffic that exceeds that capacity is presumed to be
dropped. So the excess traffic, at least in theory, gets the worst
treatment of all, and the capacity in excess of that configured for
EF is left free for non-EF traffic. That is a bit counter-intuitive
for people who think of EF as the "best" class.

> Begin forwarded message:
>=20
>> From: "Fred Baker (fred)" <fred@cisco.com>
>> Subject: Re: [OPS-DIR] Early Ops-Dir review for rtcweb-qos draft
>> Date: July 2, 2014 at 11:07:43 AM PDT
>> To: "Benoit Claise (bclaise)" <bclaise@cisco.com>, "ops-dir@ietf.org" =
<ops-dir@ietf.org>
>> Cc: "tsv-ads@tools.ietf.org" <tsv-ads@tools.ietf.org>, "tsvwg-chairs@t=
ools.ietf.org" <tsvwg-chairs@tools.ietf.org>, "draft-ietf-tsvwg-rtcweb-qo=
s.all@tools.ietf.org" <draft-ietf-tsvwg-rtcweb-qos.all@tools.ietf.org>
>>
>>
>> On Jul 2, 2014, at 9:40 AM, Benoit Claise <bclaise@cisco.com> wrote:
>>
>>> Hi David,
>>>
>>> Fred Baker accepted to review this document for the early Ops-DIR
>>> Thanks Fred.
>>> Gunter, can you please mark it as such in the in OPS DIR review tool.=

>>>
>>> Regards, Benoit
>>>
>>>> Greetings,
>>>>
>>>> On behalf of the tsvwg WG, I'd like to request an early OPS-Dir revi=
ew of draft-ietf-tsvwg-rtcweb-qos from a network operations perspective:
>>>>
>>>> 	http://datatracker.ietf.org/doc/draft-ietf-tsvwg-rtcweb-qos/
>>>>
>>>> Although I'm an OPS-Dir member, I'm looking for a review from someon=
e w/actual network operations expertise, which I freely admit is not me ;=
-).  The concern that motivates this review is that this draft recommends=
 use of a significant number of Diffserv DSCPs in Section 5:
>>>>
>>>> 	http://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-02#section-5=

>>>>
>>>> I have input from one large network operator (DT) that this is well =
off (by a factor of 2 or more) from the number of DSCPs that are (and are=
 likely to be) supported on their network (and other similar networks) fo=
r non-internal (e.g., subscriber/customer) traffic, and hence much of the=
 differentiation is likely to vanish on ingress to such a network, if not=
 before.
>>>>
>>>> Input from someone else familiar with the QoS aspects of network ope=
rations would be rather helpful at this stage.
>>>>
>>>> Thanks,
>>>> =E2=80=94David
>> In fairness, I=E2=80=99m not an operator either. To directly answer Da=
vid's question, you need an operator to respond.
>>
>> I am, however, one of the authors of RFC 4594, which this note quotes,=
 and RFC 5127, which addresses aggregation of those classes in high speed=
 networks.=20
>>
>> https://tools.ietf.org/html/rfc4594
>> 4594 Configuration Guidelines for DiffServ Service Classes. J.
>>     Babiarz, K. Chan, F. Baker. August 2006. (Format: TXT=3D144044 byt=
es)
>>     (Updated by RFC5865) (Status: INFORMATIONAL)
>>
>> https://tools.ietf.org/html/rfc5127
>> 5127 Aggregation of Diffserv Service Classes. K. Chan, J. Babiarz, F.
>>     Baker. February 2008. (Format: TXT=3D43751 bytes) (Status:
>>     INFORMATIONAL)
>>
>> The heart of the document is section 5, in which it makes specific DSC=
P recommendations. Were I to try to implement it, I think I would find it=
 confusing and difficult. Color me pedantic, but let me give you the ques=
tions that flow through my head as I read it.
>>
>> An important statement made here is:
>>
>>>   +---------------------------+-------+------+------------+----------=
-+
>>>   |         Data Type         |  Very | Low  |   Medium   |    High  =
 |
>>>   |                           |  Low  |      |            |          =
 |
>>>   +---------------------------+-------+------+------------+----------=
-+
>>>   |           Audio           |  CS1  |  BE  |  EF (46)   |  EF (46) =
 |
>>>   |                           |  (8)  | (0)  |            |          =
 |
>>>          <various other classes and code points>
>>>   +---------------------------+-------+------+------------+----------=
-+
>>>
>>>                                  Table 1
>>>
>>>   The columns "very low", "low", "Medium" and "high" are the priority=

>>>   levels.   The browser SHOULD first select the data type of the medi=
a
>>>   flow.  Within the data type, the priority of the media flow SHOULD =
be
>>>   selected.  All packets within a media flow SHOULD have the same
>>>   priority.  [snip]
>> RFC 4594 has no corollary concept of =E2=80=9Cpriority=E2=80=9D of the=
 data traffic. It focuses on the type of application and that application=
=E2=80=99s requirements; if it needs to minimize network-induced jitter, =
it might recommend using a priority scheduler, transferring jitter from E=
F to other traffic classes, and if it targets TCP traffic, it might recom=
mend the use of AQM technologies. For it, the question is not the relativ=
e importance of the session to the user or some other outside observer, b=
ut the technology required to deliver the QoS services that the applicati=
on requires. By comparison, RFC 5865 does discuss precedence classes for =
sessions; in that case, the question is a service policy that might admit=
 a new session to the network if it is deemed =E2=80=9Cimportant=E2=80=9D=
 via some authenticated authorization process when it would not admit a n=
ew session from just anyone. So I struggle to determine how the browser w=
ould decide whether a given session was =E2=80=9Cvery low=E2=80=9D, =E2=80=
=9Clow=E2=80=9D, =E2=80=9C
medium=E2=80=9D, or =E2=80=9Chigh=E2=80=9D priority, and what the concept=
 of =E2=80=9Cpriority=E2=80=9D means in the context.
>>
>> I also struggle to understand the meaning of the reference to =E2=80=9C=
BE=E2=80=9D. I presume it means =E2=80=9CBest Effort=E2=80=9D. RFC 4594 d=
oes not describe a =E2=80=9CBest Effort=E2=80=9D DSCP or traffic class; i=
t does describe =E2=80=9CDefault=E2=80=9D, which I think this is referrin=
g to. Similarly, definitions of the Differentiated Services architecture =
(I haven=E2=80=99t checked all of the 272 specifications that include the=
 text string =E2=80=9CDifferentiated Services=E2=80=9D, but I have review=
ed the documents from RFC 2474 through 2999) make no such reference. =E2=80=
=9CBest Effort=E2=80=9D is a term describing a characteristic of any data=
gram network service; packets in flight may be dropped, duplicated, or re=
ordered, as compared with services that enforce in-order and in some case=
s reliable delivery. I would prefer to see the appropriate traffic class,=
 which I think might be =E2=80=9CDefault=E2=80=9D, used in this document.=

>>
>> The document also makes a statement that I think is unnecessary:
>>
>>   The above table assumes that packets marked with CS1 is treated as
>>   "less than best effort".  However, the treatment of CS1 is
>>   implementation dependent.  If an implementation treats CS1 as other
>>   than "less than best effort", then the priority of the packets may b=
e
>>   changed from what is intended.
>>
>> With respect, the treatment of every traffic class is implementation-d=
ependent, and every mark is fungible. The document should describe how tr=
affic should be marked, and what traffic so marked expects from the netwo=
rk. The network will then do whatever it thinks is appropriate to satisfy=
 that requirement.

Yes. And specifically, diffserv carefully avoids the primitive idea
of "priority", because that implies simple preemption, which is not
how diffserv works. The text about CS1 presumably intends to refer to
RFC 3662 (A Lower Effort Per-Domain Behavior (PDB) for Differentiated
Services) which is only Informational, but at least gives a complete
description of a specific behaviour called LE, not LBE.

   Brian

>> I would not recommend the adoption or publication of the draft in its =
present form.
>> _______________________________________________
>> OPS-DIR mailing list
>> OPS-DIR@ietf.org
>> https://www.ietf.org/mailman/listinfo/ops-dir
>=20


From nobody Thu Jul  3 18:34:10 2014
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Subject: Re: [tsvwg] WGLC for draft-ietf-tsvwg-sctp-prpolicies-03 - due to conclude Monday, July 21st, 2014
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Gorry,

Please see comments below:

Gorry Fairhurst wrote:                           (Thu, 03 Jul 2014 09:45:46)
> This is the start of the WGLC on "Additional Policies for the Partial  
> Reliability Extension of the Stream Control Transmission Protocol" for  
> publication intended as a PS.
>
> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-prpolicies/
>
> This WGLC will last until Monday, July 21st, 2014 (longer since this  
> includes time prior to an IETF meeting).
>
> Could people please send *any* comments to the list if they think this  
> is ready to publish?
>
> A good note is "I have read this version of the draft,
> and I approve (or support) its progression to RFC".
>
> James/Gorry/David
> TSVWG chairs


I have read this version of the draft; however, I have some
comments:

It appears that the document only has about 3 (or 4) normative
paragraphs.  Of those three (or 4) paragraphs the key word
"REQUIRED" is used once with reference to RFC3758; the key word
"MUST" is used once with regard to abandoning a user message that
would exceed a retransmission limit.  There are two "MAY" and one
"OPTIONAL" key words.  The rest of the document is informational.

I question the sole "MUST" in section 3.1.  The most I think it
should be is a "SHOULD" and it should probably just be a "MAY".
Other considerations could go into whether to abandon or simply
postpone a retransmission that achieve the same effect.  Perhaps
what is attempted here is to say that "after a message has reached
its retransmission limit, the sender SHOULD NOT retransmit the
message again at the exclusion of resources to other messages
available for transmission or retransmission."  I do not see why you
would want to abandon retransmissions of messages on otherwise idle
associations: it would unnecessarily lower the reliability of the
association.

Also, when the sender is to abandon the message, it should likely
only be on the event of a timer-based retransmission request.  A
sender can get all kinds of spurious fast-retransmission requests
resulting in the discard of an excessive number of messages under
the current policy.  In fact, abandoning messages _in_itself_ is
likely to cause additional spurious gap reports and
fast-retransmission.  However, abandoning on a timer-based
retransmission would make the policy hardly distinguishable on the
wire from the existing lifetimes-based PR policy.

In summary, it seems to me that the document simply asks us to
implement some loose definition of a sender-side only API that does
not describe anything that "is required for interoperation or to
limit the behavior which has potential for causing harm" as required
by RFC 2119.  It seems to abuse the only normative MUST in the
document applied to a sender-side procedure that appears to not to
have been given much careful consideration.

The entire document appears more focused on the INFORMATIONAL
section 4 than on the NORMATIVE section 3.

As such, I object to it being published as a PS.  I would support
the draft being published as an INFORMATIONAL RFC if the procedure
described in section 3.1 is properly revised.

--brian

-- 
Brian F. G. Bidulock
bidulock@openss7.org
http://www.openss7.org/


From nobody Thu Jul  3 18:37:49 2014
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From: "Huangyihong (Rachel)" <rachel.huang@huawei.com>
To: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: [tsvwg] TSVWG - Request for Agenda Items for Toronto meetings
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Date: Fri, 4 Jul 2014 01:37:18 +0000
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Subject: Re: [tsvwg] TSVWG - Request for Agenda Items for Toronto meetings
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Hi Chairs,

I'd like to request a slot for my draft http://www.ietf.org/internet-drafts=
/draft-huang-tsvwg-tr-tcp-ps-00.txt. 10 minutes would be fine. Thank you so=
 much.

BR,
Rachel

> -----Original Message-----
> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of
> gorry@erg.abdn.ac.uk
> Sent: Thursday, July 03, 2014 9:01 PM
> To: tsvwg@ietf.org
> Subject: [tsvwg] TSVWG - Request for Agenda Items for Toronto meetings
>=20
>=20
> Hi tsvwg'ers,
>=20
> This is a request for people to offer presentations on drafts to be prese=
nted to
> the tsvwg working group at the Toronto IETF.
>=20
> I've upload a very initial agenda here:
> http://www.ietf.org/proceedings/90/agenda/agenda-90-tsvwg
>=20
> Many of the talks are marked TBC (editors please confirm). We do have spa=
ce
> to discuss additional drafts.
>=20
> Please send all requests and any corrections to the chairs asap.
>=20
> Gorry, David and James
> (tsvwg co-chairs)
>=20


From nobody Fri Jul  4 01:50:04 2014
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From: gorry@erg.abdn.ac.uk
To: "Huangyihong (Rachel)" <rachel.huang@huawei.com>
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Subject: Re: [tsvwg] TSVWG - Request for Agenda Items for Toronto meetings
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Thanks for submitting this draft.

This may be an interesting topic, however, we don't normally present
initial drafts until there has been discussion at a WG meeting or on the
list.

It is best to start by asking the list comments on the mailing list. If
you haven't already done so - you should probably post a  note separately
to the TAPS group and ask them for feedback on the proposed topics.

I'll consult my co-chairs about the Agenda, and note your interest in
presenting.

Gorry

> Hi Chairs,
>
> I'd like to request a slot for my draft
> http://www.ietf.org/internet-drafts/draft-huang-tsvwg-tr-tcp-ps-00.txt. 10
> minutes would be fine. Thank you so much.
>
> BR,
> Rachel
>
>> -----Original Message-----
>> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of
>> gorry@erg.abdn.ac.uk
>> Sent: Thursday, July 03, 2014 9:01 PM
>> To: tsvwg@ietf.org
>> Subject: [tsvwg] TSVWG - Request for Agenda Items for Toronto meetings
>>
>>
>> Hi tsvwg'ers,
>>
>> This is a request for people to offer presentations on drafts to be
>> presented to
>> the tsvwg working group at the Toronto IETF.
>>
>> I've upload a very initial agenda here:
>> http://www.ietf.org/proceedings/90/agenda/agenda-90-tsvwg
>>
>> Many of the talks are marked TBC (editors please confirm). We do have
>> space
>> to discuss additional drafts.
>>
>> Please send all requests and any corrections to the chairs asap.
>>
>> Gorry, David and James
>> (tsvwg co-chairs)
>>
>



From nobody Fri Jul  4 02:10:19 2014
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: "Huangyihong (Rachel)" <rachel.huang@huawei.com>
Thread-Topic: [tsvwg] TSVWG - Request for Agenda Items for Toronto meetings
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Subject: Re: [tsvwg] TSVWG - Request for Agenda Items for Toronto meetings
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If the intention of this document is to discuss a modification of TCP param=
eters, the TCPM mailing list might also be an appropriate place to get feed=
back (albeit most contributors to TCPM also follow TSVWG).

I'd recommend to have a look at RFC 6077 first.

Michael


> -----Original Message-----
> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of
> gorry@erg.abdn.ac.uk
> Sent: Friday, July 04, 2014 10:50 AM
> To: Huangyihong (Rachel)
> Cc: gorry@erg.abdn.ac.uk; tsvwg@ietf.org
> Subject: Re: [tsvwg] TSVWG - Request for Agenda Items for Toronto
> meetings
>=20
> Thanks for submitting this draft.
>=20
> This may be an interesting topic, however, we don't normally present
> initial drafts until there has been discussion at a WG meeting or on
> the
> list.
>=20
> It is best to start by asking the list comments on the mailing list. If
> you haven't already done so - you should probably post a  note
> separately
> to the TAPS group and ask them for feedback on the proposed topics.
>=20
> I'll consult my co-chairs about the Agenda, and note your interest in
> presenting.
>=20
> Gorry
>=20
> > Hi Chairs,
> >
> > I'd like to request a slot for my draft
> > http://www.ietf.org/internet-drafts/draft-huang-tsvwg-tr-tcp-ps-
> 00.txt. 10
> > minutes would be fine. Thank you so much.
> >
> > BR,
> > Rachel
> >
> >> -----Original Message-----
> >> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of
> >> gorry@erg.abdn.ac.uk
> >> Sent: Thursday, July 03, 2014 9:01 PM
> >> To: tsvwg@ietf.org
> >> Subject: [tsvwg] TSVWG - Request for Agenda Items for Toronto
> meetings
> >>
> >>
> >> Hi tsvwg'ers,
> >>
> >> This is a request for people to offer presentations on drafts to be
> >> presented to
> >> the tsvwg working group at the Toronto IETF.
> >>
> >> I've upload a very initial agenda here:
> >> http://www.ietf.org/proceedings/90/agenda/agenda-90-tsvwg
> >>
> >> Many of the talks are marked TBC (editors please confirm). We do
> have
> >> space
> >> to discuss additional drafts.
> >>
> >> Please send all requests and any corrections to the chairs asap.
> >>
> >> Gorry, David and James
> >> (tsvwg co-chairs)
> >>
> >
>=20


From nobody Fri Jul  4 02:35:59 2014
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From: Weixinpeng <weixinpeng@huawei.com>
To: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: [tsvwg] TSVWG - Request for Agenda Items for Toronto meetings
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Date: Fri, 4 Jul 2014 09:35:42 +0000
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Subject: Re: [tsvwg] TSVWG - Request for Agenda Items for Toronto meetings
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Hi gorry,
I would like to apply a 10min time slot for a presentation of http://tools.=
ietf.org/html/draft-wei-tsvwg-tunnel-congestion-feedback-02,
since the last meeting we have finished a simulation work for congestion fe=
edback mechanism and would like to present it.
Thanks.

Best Regards,
Xinpeng

>-----Original Message-----
>From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of
>gorry@erg.abdn.ac.uk
>Sent: Thursday, July 03, 2014 9:01 PM
>To: tsvwg@ietf.org
>Subject: [tsvwg] TSVWG - Request for Agenda Items for Toronto meetings
>
>
>Hi tsvwg'ers,
>
>This is a request for people to offer presentations on drafts to be presen=
ted to
>the tsvwg working group at the Toronto IETF.
>
>I've upload a very initial agenda here:
>http://www.ietf.org/proceedings/90/agenda/agenda-90-tsvwg
>
>Many of the talks are marked TBC (editors please confirm). We do have spac=
e
>to discuss additional drafts.
>
>Please send all requests and any corrections to the chairs asap.
>
>Gorry, David and James
>(tsvwg co-chairs)
>


From nobody Fri Jul  4 04:41:53 2014
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From: <Ruediger.Geib@telekom.de>
To: <david.black@emc.com>, <gorry@erg.abdn.ac.uk>, <jmpolk@cisco.com>
Date: Fri, 4 Jul 2014 13:41:42 +0200
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From nobody Fri Jul  4 10:55:17 2014
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Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-sctp-dtls-encaps-05.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 Working Group of the IETF.

        Title           : DTLS Encapsulation of SCTP Packets
        Authors         : Michael Tuexen
                          Randall R. Stewart
                          Randell Jesup
                          Salvatore Loreto
	Filename        : draft-ietf-tsvwg-sctp-dtls-encaps-05.txt
	Pages           : 8
	Date            : 2014-07-04

Abstract:
   The Stream Control Transmission Protocol (SCTP) is a transport
   protocol originally defined to run on top of the network protocols
   IPv4 or IPv6.  This document specifies how SCTP can be used on top of
   the Datagram Transport Layer Security (DTLS) protocol.  Using the
   encapsulation method described in this document, SCTP is agnostic
   about the protocols being used below DTLS, explicit IP addresses can
   not be used in the SCTP control chunks.  As a consequence, the SCTP
   associations are single homed.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-dtls-encaps-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-sctp-dtls-encaps-05


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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From nobody Fri Jul  4 12:54:38 2014
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Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-rsvp-app-id-vv-profiles-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 Working Group of the IETF.

        Title           : Resource Reservation Protocol (RSVP) Application-ID Profiles for Voice and Video Streams
        Authors         : James Polk
                          Subha Dhesikan
	Filename        : draft-ietf-tsvwg-rsvp-app-id-vv-profiles-02.txt
	Pages           : 15
	Date            : 2014-07-04

Abstract:
   RFC 2872 defines an Resource Reservation Protocol (RSVP) object for
   application identifiers.  This document uses that App-ID and gives
   implementers specific guidelines for differing voice and video
   stream identifications to nodes along a reservation path, creating
   specific profiles for voice and video session identification.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rsvp-app-id-vv-profiles/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tsvwg-rsvp-app-id-vv-profiles-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-rsvp-app-id-vv-profiles-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.

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


From nobody Fri Jul  4 13:42:23 2014
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On 05 Nov 2013, at 00:51, Karen E. Egede Nielsen =
<karen.nielsen@tieto.com> wrote:

> Hi,
> =20
> Please accept the folowing comments to =
draft-stewart-tsvwg-sctp-ndata-03.txt.
> Please use as fit.
Hi Karen,

thank you very much for your comments. I'm sorry for not responding =
earlier...
See my comments in-line.

Best regards
Michael
> =20
> General comments:
> -----------------------
> =20
> The draft draft-stewart-tsvwg-sctp-ndata-03.txt, in my opinion, =
addresses an important limitation of SCTP message fragmentation The
> limitation is important to have removed in order to support large =
message transfer by SCTP without detroying concurrent transmission of
> data with real-time characteristics.
> =20
> The advanced stream scheduling modes, suggested in the draft, also =
seems very instrumental in order to instantiate different real time =
characteristics on
> different streams.
> =20
> Title:  Given the scheduling aspects included in the draft, one should =
consider to adjust the title so that it better accommodate for the =
scheduling parts. Actually stream scheduling seems to be main topic of =
the draft with the new (or extended) data chunk format just a mean to =
support more flexible (interleaved) scheduling of fragments.
The is correct.
> =20
> Something else:  The =93N=94 as =93New=94 in N-DATA seems not so =
future-proof. Perhaps E for word =93Extended=94 (E-DATA) would be a =
better naming ?
Possibly. But we are using N-DATA in several places... Will think about =
it.
> =20
> It should be clarified in the Introduction that the draft makes use of =
the Supported Extension Parameter as defined in RFC5061.
Make sense. Done.
> =20
> Detailed Comments:
> ---------------------------
> =20
> Section 1.1:
> 1.       It should be clarified that the draft defines a new data =
chunk option as well as that it defines an advanced stream scheduler =
function and that these are independent functions.
Correct. I agree.
> 2.       It could be clarified that the intention is for the new data =
chunk to be used for all data transmitted on a particular association =
(i.e., not only fragments).
Done.
> 3.       It could be clarified that the usage of the new data chunk =
will differentiable on a per association basis and that an SCTP =
implementation as such shall support coexistence of associations which =
use the existing data chunk format and associations which use the new =
data chunk format.
Sure. Done.
> =20
> Section 3.1:
> 1.       In the third paragraph that starts with =93Sender side =
usage.. =94, it should be clarified that the new data chunk is used for =
all data chunks and not for only data chunks that need fragmentation.
Fixed.
> 2.       The situation, when the peer does not support N-DATA but the =
user request it, should preferably be elaborated on in this section or =
somewhere else in the draft. Presumably then in this situation the SCTP =
layer simply uses the legacy data chunk with its in-build limitations. =
However this should be described.
Fixed.
> 3.       Information should preferably be provided to UL about whether =
peer supports usage of N-DATA. For a socket api compliant implementation =
such information could possibly be provided in an extension of =
SCTP_STATUS ?
I would prefer to signal it in the SCTP_ASSOC_CHANGE indicating =
SCTP_COMM_UP or SCTP_RESTART
by using SCTP_SUPPORTS_INTERLEAVING similar to SCTP_ASSOC_SUPPORTS_PR, =
SCTP_ASSOC_SUPPORTS_AUTH, ...
However, this goes into the socket API section.
>=20
> Section 4.1 =96 minor/editorial:
> =20
> 1.       The text could be clarified so that it more clearly refers to =
=93one-to-many=94 and to=20
> =93one-to-one=94 sockets. The text could also better describe the =
behaviour of the socket option on each type.
Could you be more specific where information is missing?
> 2.       The second paragraph sentence: =93An N-DATA chunk aware =
application should also set the fragment interleave level to 2=94 =96 =
should refer to appropriate option of RFC6458.

> =20
> Section 4.1 =96 substance:
> =20
> 1.       I would like to press for that the ability to change the =
N-DATA setting on current (established) associations is removed. It =
looks to open up for a number of complexities both in SCTP layer as well =
as in O&M layer;  E.g.,  if N-DATA feature is disabled, what about =
retransmissions where original was send as N-DATA. When the O&M =
operation is made, then how to control when it takes effect in the SCTP =
layer in respect to data already in SND buffer. Further the disabling =
only effectively can control the sending side. These details would have =
to be described in draft and, secondly, implemented, both of which could =
be somewhat of a challenge. Are there some important use-cases that =
demands for this =93flexibility=94 ?
This functionality isn't there. I agree completely with you. Isn't the =
text in 4.1.1 on
SCTP_NDATA_ENABLE clear on this?
> 2.       I believe that one NEED to have the possibility, on a per =
association basis, to read whether the peer supports N-DATA or not. This =
regardless of whether it be an association belonging
Isn't the notification good enough? You can't query ASCONF support, =
Stream Reset, and so on.
This is currently only done via the notification. I'm open to discuss a =
generic method...
> to a one-to-many socket or a one-to-one socket. Support for read via =
socket api should be described in the draft. This may be provided via an =
extension of SCTP_STATUS (?). Optionally one may also prescribe for such =
information to be returned as part of an extension of RFC3873, =
sctpAssocTable object.
Let's talk about MIB stuff in general in a separate document. We are =
lacking MIB support for
all SCTP extensions. If there is a real need in this, we should think =
about updating the
MIB RFC. Years ago, when I implemented it for FreeBSD, I had a bunch of =
comments.
> 3.       Situation with association restart on SCTP level should =
preferably be described.
You get a new notification...
> =20
> Section 4.1.2:
> =20
> =20
> 1.       Would it be reasonable to specify a subset of the listed =
scheduling options as something that an implementation that support =
N-DATA MUST support and others as MAY or optional.  Alternatively one =
may specify that at least one of the options need to be supported ?
I would just look which constants are defined... We can make a statement =
that not all implementation
support all.
> 2.       The SCTP_SS_FAIR_BANDWIDTH could be demanding from an =
implementation perspective.
As will be the SCTP_SS_WEIGHTED_FAIR_QUEUEING...
> 3.       It should be clarified that these scheduling options are =
applicable for sending normal DATA chunks also when peer does not =
support usage of N-DATA.
Yepp. I'll work on this while polishing/testing the implementation..
>=20
> =20
> Possible addition to the draft content: (?)
> =20
> Possibly one could contemplate to include improvement of the receiver =
side operation also. As follows:
> =20
> =B7         The SCTP receive operation could be extended to allow for =
specification of read stream scheduling  - possibly options =96 to start =
with (RS: Read Scheduling):
> o   SCTP_RS_FIRST_COME: The simple first-come, first-serve algorithm
> is selected by using this value.  It just queues messages that are =
readable in the order in which they have been received by the SCTP =
layer.
> o   SCTP_RS_PRIORITY:  SCTP operates with a number of receive queues =
per association according to a number of priorities set. For each =
priority queue simple first-come, first-serve algorithm is used .I.e., =
for each priority queue messages are queued as readable in the order in =
which they have been received by the SCTP layer .The priority can be =
assigned on a per stream basis. Value 0 is default priority, higher =
values are higher priority (or lower), lower values are lower priority =
(or higher).
> Select, poll (BSD) and epoll (Linux) mechanism must be extended to =
support specification, and notification, of receiving readable event on =
a per priority RCV queue basis.
As we discussed, I think this could go into a separate document...
>=20
> =20
> Editorial:
> =20
> Abstract: which which =E0 which
> Section 2: FSN's are unsigned number =E0 FSN's are unsigned numbers
Fixed.
> =20
> BR, Karen


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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 Working Group of the IETF.

        Title           : Stream Schedulers and a New Data Chunk for the Stream Control Transmission Protocol
        Authors         : Randall R. Stewart
                          Michael Tuexen
                          Salvatore Loreto
                          Robin Seggelmann
	Filename        : draft-ietf-tsvwg-sctp-ndata-01.txt
	Pages           : 11
	Date            : 2014-07-04

Abstract:
   The Stream Control Transmission Protocol (SCTP) is a message oriented
   transport protocol supporting arbitrary large user messages.
   However, the sender can not interleave different user messages which
   causes head of line blocking at the sender side.  To overcome this
   limitation, this document adds a new data chunk to SCTP.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-ndata-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-sctp-ndata-01


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

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


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Michael,

I had some interest in this draft after having identified and
discussed this issue so many years ago.  I have read this version
of your draft and have some comments:

| 1.  Introduction
| 
| 1.1.  Overview
| 
|    When SCTP [RFC4960] was initially designed it was mainly envisioned
|    for transport of small signaling messages.  Late in the design stage
|    it was decided to add support for fragmentation and reassembly of
|    larger messages with the thought that someday Session Initiation
|    Protocol (SIP) [RFC3261] style signaling messages may also need to
|    use SCTP and a single MTU sized message would be too small.
|    Unfortunately this design decision, though valid at the time, did not
|    account for other applications which might send very large messages
|    over SCTP.  When such large messages are now sent over SCTP a form of
|    sender side head of line blocking becomes created within the
|    protocol.  This head of line blocking is caused by the use of the
|    Transmission Sequence Number (TSN) for two different purposes:
| 
|    1.  As an identifier for DATA chunks to provide a reliable transfer.
| 
|    2.  As an identifier for the sequence of fragments to allow
|        reassembly.
| 
|    The protocol requires all fragments of a user message to have
|    consecutive TSNs.  Therefore the sender can not interleave different
|    messages.

I think that is a quite fictional rewrite of history.  These issues
were addressed long before RFC 3960 issued and some even vocally
requested the postponement of the RFC until related issues could be
resolved (no, not me, that was a different issue).

|    The protocol requires all fragments of a user message to have
|    consecutive TSNs.  Therefore the sender can not interleave different
|    messages.

The first statement is correct; the inference is false.  SCTP places
no total ordering constraint on transmission of data chunks.

I will say that again: SCTP places no total ordering constraint on
transmission of data chunks.   Multi-homing would be impractical if
it did: it would be impractical in a modern operating system for a
sender at the transport level to ensure that a TSN was not emitted
from one interface before another.  It would be pointless anyway
given that the underlying network gives no guarantees of total
ordering.

Therefore, given that there is no total ordering constraint imposed
by SCTP on the transmission of TSNs, the fact that TSNs are assigned
in numerical sequence for a message does not prohibit a sender from
interleaving data chunks in transmission.  Therefore, there is no
head of line blocking.

So, for example,  As a sender, I get an large ordered message that I
chunk up into 2000 data chunks.  I assign the data chunks TSNs 1
through 2000 and start sending the first few, say 1, 2, 3, until I
use up my cwnd.  The rest I queue for transmission in a normal
queue.  I then get a small unordered message.  I want to expedite
this message (because SIGTRAN uses unordered messages for expedited
requests).  I assign it TSN 2001 and queue it in an expedited queue
for transmission.  When the cwnd opens I send TSNs 2001, 4, 5, 6.
SCTP does not prohibit me from doing this.  Yes, I will likely get a
gap ack, but so what?  I know I didn't send the messages in the gap
yet.  Also, creating a situation that could happen at any moment
anyway cannot be harmful.

Interleaving of transmission was never the issue.

The issue was with the interface between SCTP and the ULP and the
needs of the ULP with regard to message composition.  For the sender
to do what I just described, it must be able to sequentially assign
all of the large message's TSNs before it can assign a TSN to the
small message.  To assign TSNs to the large message, it needs to
know the total size of the message so it can calculate chunk sizes
and determine the total number of data chunks (and TSNs) required.
The problem with this is that it places the onus on the ULP to know
the total message size before passing any of it to SCTP, requiring
it to compose the entire message before writing the first byte.
This, of course, hampers stream-based compression, encoding schemes,
block cypher and a whole bunch of useful stuff that might want to
put the data into a big SCTP message.  However, the most direct way
around that is to fragment the stream in the ULP and wrap it with an
application header giving information like the byte offset within
the stream.  We discussed this concerning RDDP back in 2000.

My annoyance with having to know the total message size was for
directly using a standard Sockets (sendmsg) or STREAMS interface
(XTI) as an interface to SCTP where the caller just tells the
transport that there is "MORE" data making up the record that will
be presented later: it precluded me from sending the first part
until the later parts arrived and potentially blocking up the write
buffer.  No other transport had this problem.

What I wanted to see happen, and I discussed this back in 2000 and
went over it in detail with you and Randal at the 2006 interop was
to relax the sequential TSN assignment to simply an ordered TSN
assignment.  I will use the example again:

As a sender, I get the beginning of a large unordered message that
I chunk up into 4 data chunks.  I add these 4 data chunks to the
send queue for the stream.  I perform a transmission when cwnd
becomes available and assign TSNs 1, 2 and 3, and send the first 3
chunks in a packet and use up my cwnd.   I then get the small
unordered message which I chunk into one chunk and place on the
expedited queue.  I then get another portion of the large unordered
message which I chunk up into 4 more data chunks.  My cwnd opens
and I assign TSN 4 to the expedited chunk, and TSNs 5, 6 and 7 to
chunks from the large message.

Under the relaxed constraint on assigning TSNs, I can easily do what
is necessary and even avoid unnecessary gap acks.

The relaxed constraint does not require major modifications at the
receiver: the receiver cannot deliver the large message past a
missing TSN in either case.  Besides, a receiver implementation
should not attempt to enforce sender requirements, nor unnecessarily
rely heavily on sender behaviour.  When all of the TSNs are in
(between B and E), there is always sufficient information to
reassemble messages regardless of whether the sender interleaves
TSNs or not.


The position where HOL blocking could be reintroduced is where a
number of large messages with interleaved TSNs are being transferred
and one TSN in the middle of theme is lost, blocking them all until
it can be recovered (under some APIs).  This is not due to sequential
TSN assignment, but comes instead from relaxing the constraint and
deficiencies of the receiver API.

Your frame sequence number could assist the receiver in determining
in the event of a missing TSN, not to which message the missing TSN
belongs, but to which it does not belong: allowing the complete parts
of the other messages to be released.  But because of that, I don't
see why the FSN cannot simply be added to the DATA chunk header and
let the receiver that doesn't understand it ignore it.

| 3.2.  Receiver Side Considerations
| 
|    Upon reception of an SCTP packet containing a N-DATA chunk if the
|    message needs to be reassembled, then the receiver MUST use the FSN
|    for reassembly of the message and not the TSN.  Note that a non-
|    fragmented messages is indicated by the fact that both the 'E' and
|    'B' bits are set.  An ordered or unordered fragmented message is thus
|    identified with any message not having both bits set.

There is no reason to require the receiver to not use the TSN.  If all
TSNs have been received, the FSN is not even useful.  If the
receiver ignores the FSN even when a TSN is missing, it simply might
not deliver messages to its ULP as quickly as it might.  But there
are other considerations on delivery of messages to the ULP and
reassembly is not even necessary until the implementation goes to
deliver a message.  Hey, some API might even reassemble two of three
scatter-gather buffers and deliver the two before the missing one
even arrives.  I think it is an implementation decision.

I think this should say "may use the FSN in reassembly of messages".

| 3.1.  Sender Side Considerations
| 
|    A sender MUST NOT use the N-DATA chunk unless the user has requested
|    that use via the socket API (see Section 4).  This constraint is made

Please don't tell me I MUST use an INFORMATIONAL section referring
to an INFORMATIONAL RFC for anything.  End the sentence after "that
use".

I would consider that a ULP writing a message with the "MORE" bit
set is requesting the service, or that is simply using an API where
it can set a "MORE" bit as requesting the service, where "setting
a MORE bit" means opening any API where the ULP can send the first
portion of a message without specifying the total length of the
message.

| 4.  Socket API Considerations
| 
|    This section describes how the socket API defined in [RFC6458] is
|    extended to allow applications to use the extension described in this
|    document.
| 
|    Please note that this section is informational only.

Hey, can you put all this non-standard INFORMATIONAL stuff in a
different document?  It is much longer than the NORMATIVE part and
really doesn't contribute anything to the document.

Well, that's enough for now.  I'm sure some flurry will result from
these comments anyway so we can discuss more up the ladder.

I have one question (just to check that you actually read this far):
what caused you to try to address this issue now, 14 years after the
fact?

--brian

-- 
Brian F. G. Bidulock
bidulock@openss7.org
http://www.openss7.org/


From nobody Sun Jul  6 05:16:13 2014
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On 04 Jul 2014, at 19:55, internet-drafts@ietf.org wrote:

Dear WG,

I just wanted to let you know that this version contain changes based =
feedback
from the WG chairs and Eric Rescorla regarding DTLS.

Eric pointed out that WebRTC at the time of the review only supported =
DTLS 1.0,
whereas the previous version of the documents required DTLS 1.2 or =
higher.
We agreed on using DTLS in a consistent way is important. So I brought =
the
issue up at the rtcweb mailing list and changed the version in the =
document
to 1.0. However, the discussion on the rtcweb mailing is ongoing and we =
might
need another revision to get the DTLS version synced. The current state =
is
to require support for DTLS 1.0 and DTSL 1.2.

Best regards
Michael
>=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 Working =
Group of the IETF.
>=20
>        Title           : DTLS Encapsulation of SCTP Packets
>        Authors         : Michael Tuexen
>                          Randall R. Stewart
>                          Randell Jesup
>                          Salvatore Loreto
> 	Filename        : draft-ietf-tsvwg-sctp-dtls-encaps-05.txt
> 	Pages           : 8
> 	Date            : 2014-07-04
>=20
> Abstract:
>   The Stream Control Transmission Protocol (SCTP) is a transport
>   protocol originally defined to run on top of the network protocols
>   IPv4 or IPv6.  This document specifies how SCTP can be used on top =
of
>   the Datagram Transport Layer Security (DTLS) protocol.  Using the
>   encapsulation method described in this document, SCTP is agnostic
>   about the protocols being used below DTLS, explicit IP addresses can
>   not be used in the SCTP control chunks.  As a consequence, the SCTP
>   associations are single homed.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-dtls-encaps/
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-dtls-encaps-05
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-sctp-dtls-encaps-05
>=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
>=20


From nobody Sun Jul  6 12:05:19 2014
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Subject: Re: [tsvwg] WGLC for draft-ietf-tsvwg-sctp-prpolicies-03 - due to conclude Monday, July 21st, 2014
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On 04 Jul 2014, at 03:34, Brian F. G. Bidulock <bidulock@openss7.org> =
wrote:

> Gorry,
>=20
> Please see comments below:
>=20
> Gorry Fairhurst wrote:                           (Thu, 03 Jul 2014 =
09:45:46)
>> This is the start of the WGLC on "Additional Policies for the Partial =
=20
>> Reliability Extension of the Stream Control Transmission Protocol" =
for =20
>> publication intended as a PS.
>>=20
>> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-prpolicies/
>>=20
>> This WGLC will last until Monday, July 21st, 2014 (longer since this =20=

>> includes time prior to an IETF meeting).
>>=20
>> Could people please send *any* comments to the list if they think =
this =20
>> is ready to publish?
>>=20
>> A good note is "I have read this version of the draft,
>> and I approve (or support) its progression to RFC".
>>=20
>> James/Gorry/David
>> TSVWG chairs
>=20
>=20
> I have read this version of the draft; however, I have some
> comments:
Hi Brian,

thanks for reading the document and providing feedback.
See my comments in-line.

Best regards
Michael
>=20
> It appears that the document only has about 3 (or 4) normative
> paragraphs.  Of those three (or 4) paragraphs the key word
> "REQUIRED" is used once with reference to RFC3758; the key word
> "MUST" is used once with regard to abandoning a user message that
MUST appears two times...
> would exceed a retransmission limit.  There are two "MAY" and one
> "OPTIONAL" key words.  The rest of the document is informational.
>=20
> I question the sole "MUST" in section 3.1.  The most I think it
> should be is a "SHOULD" and it should probably just be a "MAY".
There are two MUSTs here. This first one is the one you refer to,
I think. The second one needs to be a MUST even if you downgrade the
first one to SHOULD.
> Other considerations could go into whether to abandon or simply
> postpone a retransmission that achieve the same effect.  Perhaps
> what is attempted here is to say that "after a message has reached
> its retransmission limit, the sender SHOULD NOT retransmit the
> message again at the exclusion of resources to other messages
> available for transmission or retransmission."  I do not see why you
> would want to abandon retransmissions of messages on otherwise idle
> associations: it would unnecessarily lower the reliability of the
> association.
The user point is what the upper layer expects if he limits the
number of retransmissions. Your interpretation would be:
* When the number of retransmissions exceeds the threshold, the
  SCTP implementation is free is abandon the message.
The intended semantic used in this document is
* When the number of retransmissions exceeds the threshold, the
  SCTP implementation abandons the message.
This is in-tune with the semantic used by the RTCWeb WG.
> Also, when the sender is to abandon the message, it should likely
> only be on the event of a timer-based retransmission request.  A
> sender can get all kinds of spurious fast-retransmission requests
> resulting in the discard of an excessive number of messages under
> the current policy.  In fact, abandoning messages _in_itself_ is
> likely to cause additional spurious gap reports and
> fast-retransmission.  However, abandoning on a timer-based
> retransmission would make the policy hardly distinguishable on the
> wire from the existing lifetimes-based PR policy.
I think one of the most important limits is 0. Here the upper layer does
not distinguish between timer based or fast retransmissions.
>=20
> In summary, it seems to me that the document simply asks us to
> implement some loose definition of a sender-side only API that does
> not describe anything that "is required for interoperation or to
> limit the behavior which has potential for causing harm" as required
> by RFC 2119.  It seems to abuse the only normative MUST in the
> document applied to a sender-side procedure that appears to not to
> have been given much careful consideration.
The intended status was changed from Informational to PS because
* the second MUST as described above.
* without normative language you can make sure that the upper
  gets what he expects.
This might not affect interoperability at the SCTP layer, but does
influence the upper layer.
>=20
> The entire document appears more focused on the INFORMATIONAL
> section 4 than on the NORMATIVE section 3.
Section 4 is longer, that is right. But the focus is not there.
If you don't case about the socket layer, section 4 is not important
to you.
>=20
> As such, I object to it being published as a PS.  I would support
> the draft being published as an INFORMATIONAL RFC if the procedure
> described in section 3.1 is properly revised.
>=20
> --brian
>=20
> --=20
> Brian F. G. Bidulock
> bidulock@openss7.org
> http://www.openss7.org/
>=20
>=20


From nobody Sun Jul  6 19:08:34 2014
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Thank you, Gorry. I will do it.

BR,
Rachel

> -----Original Message-----
> From: gorry@erg.abdn.ac.uk [mailto:gorry@erg.abdn.ac.uk]
> Sent: Friday, July 04, 2014 4:50 PM
> To: Huangyihong (Rachel)
> Cc: gorry@erg.abdn.ac.uk; tsvwg@ietf.org
> Subject: RE: [tsvwg] TSVWG - Request for Agenda Items for Toronto meeting=
s
>=20
> Thanks for submitting this draft.
>=20
> This may be an interesting topic, however, we don't normally present init=
ial
> drafts until there has been discussion at a WG meeting or on the list.
>=20
> It is best to start by asking the list comments on the mailing list. If y=
ou haven't
> already done so - you should probably post a  note separately to the TAPS
> group and ask them for feedback on the proposed topics.
>=20
> I'll consult my co-chairs about the Agenda, and note your interest in pre=
senting.
>=20
> Gorry
>=20
> > Hi Chairs,
> >
> > I'd like to request a slot for my draft
> > http://www.ietf.org/internet-drafts/draft-huang-tsvwg-tr-tcp-ps-00.txt
> > . 10 minutes would be fine. Thank you so much.
> >
> > BR,
> > Rachel
> >
> >> -----Original Message-----
> >> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of
> >> gorry@erg.abdn.ac.uk
> >> Sent: Thursday, July 03, 2014 9:01 PM
> >> To: tsvwg@ietf.org
> >> Subject: [tsvwg] TSVWG - Request for Agenda Items for Toronto
> >> meetings
> >>
> >>
> >> Hi tsvwg'ers,
> >>
> >> This is a request for people to offer presentations on drafts to be
> >> presented to the tsvwg working group at the Toronto IETF.
> >>
> >> I've upload a very initial agenda here:
> >> http://www.ietf.org/proceedings/90/agenda/agenda-90-tsvwg
> >>
> >> Many of the talks are marked TBC (editors please confirm). We do have
> >> space to discuss additional drafts.
> >>
> >> Please send all requests and any corrections to the chairs asap.
> >>
> >> Gorry, David and James
> >> (tsvwg co-chairs)
> >>
> >
>=20


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From nobody Sun Jul  6 20:04:12 2014
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From: "Huangyihong (Rachel)" <rachel.huang@huawei.com>
To: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
Thread-Topic: [tsvwg] TSVWG - Request for Agenda Items for Toronto meetings
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Cc: "gorry@erg.abdn.ac.uk" <gorry@erg.abdn.ac.uk>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Subject: Re: [tsvwg] TSVWG - Request for Agenda Items for Toronto meetings
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Hi Michael,

Thanks for the suggestion. I'll do it.=20
Our motivation is to discuss the problems when traditional TCP meets some s=
cenarios, e.g., different use cases may need different requirements to TCP =
(different congestion control algorithms, different TCP parameters). Tradit=
ional TCP is not flexible enough to deal with these problems. For example, =
some congestion control algorithms are application layer, some parameters a=
re fixed which can't be changes easily. Maybe a new method/solution is help=
ful. We have an OPEN TCP solution that tries to open TCP enough so that dif=
ferent applications can choose their different parameters or proper congest=
ion control algorithms in transport layer. Currently, we just want to discu=
ss with the people in IETF to make sure that such problems do exist and we =
do have the requirements to change them.
Thanks for pointing out RFC 6077. I'm not familiar with it. But I'll read i=
t immediately ^^.

BR,
Rachel

> -----Original Message-----
> From: Scharf, Michael (Michael) [mailto:michael.scharf@alcatel-lucent.com=
]
> Sent: Friday, July 04, 2014 5:10 PM
> To: Huangyihong (Rachel)
> Cc: tsvwg@ietf.org; gorry@erg.abdn.ac.uk
> Subject: RE: [tsvwg] TSVWG - Request for Agenda Items for Toronto meeting=
s
>=20
> If the intention of this document is to discuss a modification of TCP par=
ameters,
> the TCPM mailing list might also be an appropriate place to get feedback =
(albeit
> most contributors to TCPM also follow TSVWG).
>=20
> I'd recommend to have a look at RFC 6077 first.
>=20
> Michael
>=20
>=20
> > -----Original Message-----
> > From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of
> > gorry@erg.abdn.ac.uk
> > Sent: Friday, July 04, 2014 10:50 AM
> > To: Huangyihong (Rachel)
> > Cc: gorry@erg.abdn.ac.uk; tsvwg@ietf.org
> > Subject: Re: [tsvwg] TSVWG - Request for Agenda Items for Toronto
> > meetings
> >
> > Thanks for submitting this draft.
> >
> > This may be an interesting topic, however, we don't normally present
> > initial drafts until there has been discussion at a WG meeting or on
> > the list.
> >
> > It is best to start by asking the list comments on the mailing list.
> > If you haven't already done so - you should probably post a  note
> > separately to the TAPS group and ask them for feedback on the proposed
> > topics.
> >
> > I'll consult my co-chairs about the Agenda, and note your interest in
> > presenting.
> >
> > Gorry
> >
> > > Hi Chairs,
> > >
> > > I'd like to request a slot for my draft
> > > http://www.ietf.org/internet-drafts/draft-huang-tsvwg-tr-tcp-ps-
> > 00.txt. 10
> > > minutes would be fine. Thank you so much.
> > >
> > > BR,
> > > Rachel
> > >
> > >> -----Original Message-----
> > >> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of
> > >> gorry@erg.abdn.ac.uk
> > >> Sent: Thursday, July 03, 2014 9:01 PM
> > >> To: tsvwg@ietf.org
> > >> Subject: [tsvwg] TSVWG - Request for Agenda Items for Toronto
> > meetings
> > >>
> > >>
> > >> Hi tsvwg'ers,
> > >>
> > >> This is a request for people to offer presentations on drafts to be
> > >> presented to the tsvwg working group at the Toronto IETF.
> > >>
> > >> I've upload a very initial agenda here:
> > >> http://www.ietf.org/proceedings/90/agenda/agenda-90-tsvwg
> > >>
> > >> Many of the talks are marked TBC (editors please confirm). We do
> > have
> > >> space
> > >> to discuss additional drafts.
> > >>
> > >> Please send all requests and any corrections to the chairs asap.
> > >>
> > >> Gorry, David and James
> > >> (tsvwg co-chairs)
> > >>
> > >
> >


From nobody Mon Jul  7 12:46:53 2014
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Subject: Re: [tsvwg] draft-stewart-tsvwg-sctp-ndata-03.txt
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On 05 Jul 2014, at 08:20, Brian F. G. Bidulock <bidulock@openss7.org> =
wrote:

> Michael,
>=20
> I had some interest in this draft after having identified and
> discussed this issue so many years ago.  I have read this version
> of your draft and have some comments:
Hi Brian,

thanks for reading it and providing feedback.
See my comments in-line.

Best regards
Michael
>=20
> | 1.  Introduction
> |=20
> | 1.1.  Overview
> |=20
> |    When SCTP [RFC4960] was initially designed it was mainly =
envisioned
> |    for transport of small signaling messages.  Late in the design =
stage
> |    it was decided to add support for fragmentation and reassembly of
> |    larger messages with the thought that someday Session Initiation
> |    Protocol (SIP) [RFC3261] style signaling messages may also need =
to
> |    use SCTP and a single MTU sized message would be too small.
> |    Unfortunately this design decision, though valid at the time, did =
not
> |    account for other applications which might send very large =
messages
> |    over SCTP.  When such large messages are now sent over SCTP a =
form of
> |    sender side head of line blocking becomes created within the
> |    protocol.  This head of line blocking is caused by the use of the
> |    Transmission Sequence Number (TSN) for two different purposes:
> |=20
> |    1.  As an identifier for DATA chunks to provide a reliable =
transfer.
> |=20
> |    2.  As an identifier for the sequence of fragments to allow
> |        reassembly.
> |=20
> |    The protocol requires all fragments of a user message to have
> |    consecutive TSNs.  Therefore the sender can not interleave =
different
> |    messages.
>=20
> I think that is a quite fictional rewrite of history.  These issues
> were addressed long before RFC 3960 issued and some even vocally
> requested the postponement of the RFC until related issues could be
> resolved (no, not me, that was a different issue).
As far as I remember, the focus was on support for small message, not
for large ones. The limitation was know at the time of RFC 2960, but
the decision at that time was to live with the limitation... For
SIGTRAN is was not a real issue.
>=20
> |    The protocol requires all fragments of a user message to have
> |    consecutive TSNs.  Therefore the sender can not interleave =
different
> |    messages.
>=20
> The first statement is correct; the inference is false.  SCTP places
> no total ordering constraint on transmission of data chunks.
Formally you are right in the sense that SCTP will work as long
no matter in which sequence the DATA chunks will arrive at the
receiver. They can be reordered by the network or even be sent
by the sender in any order. This can't be distinguished by the
receiver. However:
* You can't assign a TSN for a DATA chunk of message before
  knowing how many TSNs you need for the messages already started
  to send (and assigned a TSN).
* The receiver will report TSNs not received in order, so you
  will get lot's of gap reports which might or might not trigger
  fast retransmissions and therefore having an impact on performance.
>=20
> I will say that again: SCTP places no total ordering constraint on
> transmission of data chunks.   Multi-homing would be impractical if
> it did: it would be impractical in a modern operating system for a
> sender at the transport level to ensure that a TSN was not emitted
> from one interface before another.  It would be pointless anyway
> given that the underlying network gives no guarantees of total
> ordering.
>=20
> Therefore, given that there is no total ordering constraint imposed
> by SCTP on the transmission of TSNs, the fact that TSNs are assigned
> in numerical sequence for a message does not prohibit a sender from
> interleaving data chunks in transmission.  Therefore, there is no
> head of line blocking.
Really? Think about sending a 1 GB message on stream 0, having =
transferred
100MB of it. Now you send a 10 byte message on stream 1. What if you
don't know the size of the big messages yet (using the explicit EOR =
mode)?
>=20
> So, for example,  As a sender, I get an large ordered message that I
> chunk up into 2000 data chunks.  I assign the data chunks TSNs 1
> through 2000 and start sending the first few, say 1, 2, 3, until I
So you do early TSN assignment. This means that if the MTU decreases
during sending TSN up to 2000, you need to rely on IP-fragmentation.
This also only works, if you know how many chunks you need in total.
This does not work if you use the explicit EOR mode, so you can't
send messages larger than your send buffer.
> use up my cwnd.  The rest I queue for transmission in a normal
> queue.  I then get a small unordered message.  I want to expedite
> this message (because SIGTRAN uses unordered messages for expedited
> requests).  I assign it TSN 2001 and queue it in an expedited queue
> for transmission.  When the cwnd opens I send TSNs 2001, 4, 5, 6.
> SCTP does not prohibit me from doing this.  Yes, I will likely get a
> gap ack, but so what?  I know I didn't send the messages in the gap
> yet.  Also, creating a situation that could happen at any moment
> anyway cannot be harmful.
It depends on harmful. You are right, all messages will be delivered
within the ordering constraints of the sender. However, there can
be performance penalties if you can't differentiate anymore between
never send or probably lost.
>=20
> Interleaving of transmission was never the issue.
>=20
> The issue was with the interface between SCTP and the ULP and the
> needs of the ULP with regard to message composition.  For the sender
> to do what I just described, it must be able to sequentially assign
> all of the large message's TSNs before it can assign a TSN to the
> small message.  To assign TSNs to the large message, it needs to
> know the total size of the message so it can calculate chunk sizes
> and determine the total number of data chunks (and TSNs) required.
It also needs the PMTU, which can change over time...
> The problem with this is that it places the onus on the ULP to know
> the total message size before passing any of it to SCTP, requiring
> it to compose the entire message before writing the first byte.
... and the assumption that the MTU doesn't change while sending
the message.
> This, of course, hampers stream-based compression, encoding schemes,
> block cypher and a whole bunch of useful stuff that might want to
> put the data into a big SCTP message.  However, the most direct way
Not sure. If you run DTLS over SCTP, for example, the message size
limited anyway to about 16KB. However, RTCWeb uses SCTP over DTLS and
therefore the "upper layer" is the JS layer.
> around that is to fragment the stream in the ULP and wrap it with an
> application header giving information like the byte offset within
> the stream.  We discussed this concerning RDDP back in 2000.
>=20
> My annoyance with having to know the total message size was for
> directly using a standard Sockets (sendmsg) or STREAMS interface
> (XTI) as an interface to SCTP where the caller just tells the
> transport that there is "MORE" data making up the record that will
> be presented later: it precluded me from sending the first part
> until the later parts arrived and potentially blocking up the write
> buffer.  No other transport had this problem.
>=20
> What I wanted to see happen, and I discussed this back in 2000 and
> went over it in detail with you and Randal at the 2006 interop was
> to relax the sequential TSN assignment to simply an ordered TSN
> assignment.  I will use the example again:
I have to say that I can't remember. Sorry. So we need to go through
it again...
>=20
> As a sender, I get the beginning of a large unordered message that
> I chunk up into 4 data chunks.  I add these 4 data chunks to the
> send queue for the stream.  I perform a transmission when cwnd
> becomes available and assign TSNs 1, 2 and 3, and send the first 3
> chunks in a packet and use up my cwnd.   I then get the small
> unordered message which I chunk into one chunk and place on the
> expedited queue.  I then get another portion of the large unordered
> message which I chunk up into 4 more data chunks.  My cwnd opens
> and I assign TSN 4 to the expedited chunk, and TSNs 5, 6 and 7 to
> chunks from the large message.
>=20
> Under the relaxed constraint on assigning TSNs, I can easily do what
> is necessary and even avoid unnecessary gap acks.
You are focussing on the "handle small unordered as expedited" issue.
The problem addressed here is the generic one:
You might even have a large message on stream 0, and a higher priority
smaller message, which still needs fragmentation, on a different stream.

I agree that you probably can come up with a solution for sending
smaller unordered messages "within" larger ordered ones, but this
doesn't solve the general problem. I think the base problem is that
the TSN is used for two things. Decoupling these will get clean
solution.
>=20
> The relaxed constraint does not require major modifications at the
> receiver: the receiver cannot deliver the large message past a
> missing TSN in either case.  Besides, a receiver implementation
> should not attempt to enforce sender requirements, nor unnecessarily
> rely heavily on sender behaviour.  When all of the TSNs are in
> (between B and E), there is always sufficient information to
> reassemble messages regardless of whether the sender interleaves
> TSNs or not.
>=20
>=20
> The position where HOL blocking could be reintroduced is where a
> number of large messages with interleaved TSNs are being transferred
> and one TSN in the middle of theme is lost, blocking them all until
> it can be recovered (under some APIs).  This is not due to sequential
> TSN assignment, but comes instead from relaxing the constraint and
> deficiencies of the receiver API.
>=20
> Your frame sequence number could assist the receiver in determining
> in the event of a missing TSN, not to which message the missing TSN
> belongs, but to which it does not belong: allowing the complete parts
> of the other messages to be released.  But because of that, I don't
> see why the FSN cannot simply be added to the DATA chunk header and
> let the receiver that doesn't understand it ignore it.
How can you add something to the DATA chunk header and get a receiver
to ignore it in case it doesn't support the extension?
>=20
> | 3.2.  Receiver Side Considerations
> |=20
> |    Upon reception of an SCTP packet containing a N-DATA chunk if the
> |    message needs to be reassembled, then the receiver MUST use the =
FSN
> |    for reassembly of the message and not the TSN.  Note that a non-
> |    fragmented messages is indicated by the fact that both the 'E' =
and
> |    'B' bits are set.  An ordered or unordered fragmented message is =
thus
> |    identified with any message not having both bits set.
>=20
> There is no reason to require the receiver to not use the TSN.  If all
> TSNs have been received, the FSN is not even useful.  If the
> receiver ignores the FSN even when a TSN is missing, it simply might
> not deliver messages to its ULP as quickly as it might.  But there
> are other considerations on delivery of messages to the ULP and
> reassembly is not even necessary until the implementation goes to
> deliver a message.  Hey, some API might even reassemble two of three
> scatter-gather buffers and deliver the two before the missing one
> even arrives.  I think it is an implementation decision.
>=20
> I think this should say "may use the FSN in reassembly of messages".
I'm not sure what you are arguing here... A receiver supporting the =
N-DATA
chunk but not using the FSN for reassembly? So you want to reuse the
old code, but use the new packet format?
>=20
> | 3.1.  Sender Side Considerations
> |=20
> |    A sender MUST NOT use the N-DATA chunk unless the user has =
requested
> |    that use via the socket API (see Section 4).  This constraint is =
made
>=20
> Please don't tell me I MUST use an INFORMATIONAL section referring
> to an INFORMATIONAL RFC for anything.  End the sentence after "that
> use".
No, this is not intended. Good point. Changed to

<t>A sender MUST NOT use the N-DATA chunk unless the user
has requested that use (e.g. via the socket API, see <xref =
target=3D'api'/>).

It is the upper layer which has to say "use interleaving" since it also
must handle the higher interleaving level on the receiving side. See
http://tools.ietf.org/html/rfc6458#section-8.1.20
>=20
> I would consider that a ULP writing a message with the "MORE" bit
> set is requesting the service, or that is simply using an API where
> it can set a "MORE" bit as requesting the service, where "setting
> a MORE bit" means opening any API where the ULP can send the first
> portion of a message without specifying the total length of the
> message.
This is possible with the socket API:
http://tools.ietf.org/html/rfc6458#section-8.1.26
>=20
> | 4.  Socket API Considerations
> |=20
> |    This section describes how the socket API defined in [RFC6458] is
> |    extended to allow applications to use the extension described in =
this
> |    document.
> |=20
> |    Please note that this section is informational only.
>=20
> Hey, can you put all this non-standard INFORMATIONAL stuff in a
> different document?  It is much longer than the NORMATIVE part and
> really doesn't contribute anything to the document.
After RFC 6458 was published, we started to added a socket API section
to the documents describing extensions, which describe how you could
implement an API for the extension based on the socket API. I wasn't
to bad to have both in one document. However, the API section is always
purely informational (similar to =
http://tools.ietf.org/html/rfc4960#section-10),
so if you don't implement the socket API, you don't need to care. If
you implement it, you can even do it differently. However, up to now,
the SCTP Linux Kernel implementation, the Solaris implementation and
the FreeBSD implementation use this API.
>=20
> Well, that's enough for now.  I'm sure some flurry will result from
> these comments anyway so we can discuss more up the ladder.
Sure. I know that the document is not done yet and I hope that
we can improve it based on the discussion on the list.
>=20
> I have one question (just to check that you actually read this far):
So I passed that test, at least...
> what caused you to try to address this issue now, 14 years after the
> fact?
RTCWeb is using SCTP for potentially large messages and needs support
for interleaving.
>=20
> --brian
>=20
> --=20
> Brian F. G. Bidulock
> bidulock@openss7.org
> http://www.openss7.org/
>=20
>=20


From nobody Tue Jul  8 08:38:08 2014
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Date: Tue, 08 Jul 2014 16:38:01 +0100
From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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I have some comments on this draft as an individual, please see below:

* First, I think the new revision helps explain some of the questions 
that were highlighted, thanks for preparing the update.

* section 3
"randomly selected constant value, e.g. zero,"
- I don't understand this in two ways:
(1) I don't think it should be zero, I think you could find a hash of 
some internal state to generate a ramdom number. For instance this is 
used in the PIM routing (RFC4601) to specify the genID which is simply a 
unique number each time the system reboots or the session is restarted 
(this has the advantage of reseting state after a reboot).
(2) setting it to zero means you wont have any entropy for ECMP, so why 
is this mode specified in this draft - or is for more general use?

* Section 3 para 3 - "packets destined
    to this assigned UDP destination port [TBD] MAY have their UDP
    checksum set to zero."
I think the first specification here applies only to IPv4, sicne IPv6 
rules are stated later, could you insert "IPv4" before "packets" ?

* Section 3 para 3, says SHOULD use IPv6 UDP Checksum, which is fine, 
but then it leaves open the possibility that an implementation may not 
(i.es.this is more than the base IPv6 sepcification). I think if you 
choose SHOULD rather MUST there should at least be some statement that 
points to RFC6936 (e.g. section 2 explains the function of this check, 
and section 5 decsribes the additional functions that would need to be 
defined to allow this encapsulation to use a zero IPv6 UDP Checksum.)

* Section 3 para 5, The Congestion Control statement is true if the IP 
traffic implements Congestion Control, so I'm not sure I buy "no need to 
apply", and would prefer the statement in RFC 5405, which carefully says:

   "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."

- Citing RFC 5405 (or successor) may help clarify this.

* The current revision syas: " In the absence of either, a congestion	
control mechanism SHOULD be implemented at the tunnel ingress and	
egress."
- The "Circuit-Breaker" draft is not really a form of congestion 
control, as currently written. A circuit-breaker is intended primarily 
to terminate traffic that is persistently thought to cause congestion. I 
think this is actually what is intended.

* I am dubious whether it is sufficient to say you need to do 
"something", personally I'd much rather see something that says you MUST 
implement XXXX, and you SHOULD use it under YYY conditions. It would be 
fine to explain that well-provisioned or controlled 
environments...etc... provide other equivalent mechanisms that would 
allow an operator to not use this. I don't see the point in writing a 
spec that says you need to invent another mechanism to deploy it.

- How hard can it be to identify the best type of circuit breaker for 
this case and require the sender and receiver implementers counters per 
source address/source port pair?

* Section 4.
/ Tunnel ingress SHOULD/
^ Insert "the"

* SHOULD perform fragmentation ... and if it does not, then what 
happens, please specify what happens when the recommendation is not applied.

* "impact network performance and user experience."
- true, but I think you should be more precise, e.g. refer to section 
1.3 of RFC6936, where various limitations were explained.

* Section 5
    "Upon receiving this ICMP message, the tunnel
    ingress MUST NOT continue to use GRE-in-UDP encapsulation toward
    this tunnel egress without management intervention."
- but also the ingress MUST validate the IP address originating the ICMP 
message is an address of an active tunnel and MUST evaluate the packet 
header returned in the ICMP payload to ensure the source port is the one 
used for this tunnel. The draft should also note that how to handle 
tunnels carrying other tunnels :-)

Gorry

On 02/07/2014 14:32, Lucy yong wrote:
> Hello,
>
> The new version is uploaded. It addresses the comments in London's meeting, i.e. congestion control, checksum, etc. Please let us know if the description is sufficient in this version. Other comments are welcome.
>
> Thanks,
> Lucy
>
> -----Original Message-----
> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of internet-drafts@ietf.org
> Sent: Tuesday, July 01, 2014 2:26 PM
> To: i-d-announce@ietf.org
> Cc: tsvwg@ietf.org
> Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-gre-in-udp-encap-02.txt
>
>
> 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 Working Group of the IETF.
>
>          Title           : Generic UDP Encapsulation for IP Tunneling
>          Authors         : Edward Crabbe
>                            Lucy Yong
>                            Xiaohu Xu
> 	Filename        : draft-ietf-tsvwg-gre-in-udp-encap-02.txt
> 	Pages           : 13
> 	Date            : 2014-07-01
>
> Abstract:
>     This document describes a method of encapsulating arbitrary
>     protocols within GRE and UDP headers.  In this encapsulation, the
>     source UDP port may be used as an entropy field for purposes of load
>     balancing while the payload protocol may be identified by the GRE
>     Protocol Type.
>
>
> 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:
> http://tools.ietf.org/html/draft-ietf-tsvwg-gre-in-udp-encap-02
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-gre-in-udp-encap-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 Tue Jul  8 10:45:42 2014
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From: Edward Crabbe <edc@google.com>
Date: Tue, 8 Jul 2014 10:44:52 -0700
Message-ID: <CACKN6JHcZBAfTGGSCFGvgecnSPmiJ8fc=nQReq-9+4QDtdZfkg@mail.gmail.com>
To: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
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Subject: Re: [tsvwg] FW: I-D Action: draft-ietf-tsvwg-gre-in-udp-encap-02.txt
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Gorry, thank you for the comments.  Implicit agreement where not noted,
with explicit disagreements inline:


On Tue, Jul 8, 2014 at 8:38 AM, Gorry Fairhurst <gorry@erg.abdn.ac.uk>
wrote:

> I have some comments on this draft as an individual, please see below:
>
> * First, I think the new revision helps explain some of the questions that
> were highlighted, thanks for preparing the update.
>
> * section 3
> "randomly selected constant value, e.g. zero,"
> - I don't understand this in two ways:
> (1) I don't think it should be zero, I think you could find a hash of some
> internal state to generate a ramdom number. For instance this is used in
> the PIM routing (RFC4601) to specify the genID which is simply a unique
> number each time the system reboots or the session is restarted (this has
> the advantage of reseting state after a reboot).
> (2) setting it to zero means you wont have any entropy for ECMP, so why is
> this mode specified in this draft - or is for more general use?
>
> * Section 3 para 3 - "packets destined
>    to this assigned UDP destination port [TBD] MAY have their UDP
>    checksum set to zero."
> I think the first specification here applies only to IPv4, sicne IPv6
> rules are stated later, could you insert "IPv4" before "packets" ?
>
> * Section 3 para 3, says SHOULD use IPv6 UDP Checksum, which is fine, but
> then it leaves open the possibility that an implementation may not
> (i.es.this is more than the base IPv6 sepcification). I think if you choose
> SHOULD rather MUST there should at least be some statement that points to
> RFC6936 (e.g. section 2 explains the function of this check, and section 5
> decsribes the additional functions that would need to be defined to allow
> this encapsulation to use a zero IPv6 UDP Checksum.)
>
> * Section 3 para 5, The Congestion Control statement is true if the IP
> traffic implements Congestion Control, so I'm not sure I buy "no need to
> apply", and would prefer the statement in RFC 5405, which carefully says:
>
>   "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."
>
> - Citing RFC 5405 (or successor) may help clarify this.
>
> * The current revision syas: " In the absence of either, a congestion
> control mechanism SHOULD be implemented at the tunnel ingress and
> egress."
> - The "Circuit-Breaker" draft is not really a form of congestion control,
> as currently written. A circuit-breaker is intended primarily to terminate
> traffic that is persistently thought to cause congestion. I think this is
> actually what is intended.
>
> * I am dubious whether it is sufficient to say you need to do "something",
> personally I'd much rather see something that says you MUST implement XXXX,
> and you SHOULD use it under YYY conditions. It would be fine to explain
> that well-provisioned or controlled environments...etc... provide other
> equivalent mechanisms that would allow an operator to not use this. I don't
> see the point in writing a spec that says you need to invent another
> mechanism to deploy it.
>
>
[ed] we had discussed this in reference to the as the transport area
meeting and come to the agreement that SHOULD was sufficient.
 Specifically:  if someone is operating this only on an intradomain basis
without interdomain traversal, they should not have to implement 'the
circuit breaker'.

Regardless, is it sufficient to say that for an interdomain use case where
the traffic is not congestion controlled at the source, a circuitbreaker
MUST be implemented?



> - How hard can it be to identify the best type of circuit breaker for this
> case and require the sender and receiver implementers counters per source
> address/source port pair?
>
> * Section 4.
> / Tunnel ingress SHOULD/
> ^ Insert "the"
>
> * SHOULD perform fragmentation ... and if it does not, then what happens,
> please specify what happens when the recommendation is not applied.
>
> * "impact network performance and user experience."
> - true, but I think you should be more precise, e.g. refer to section 1.3
> of RFC6936, where various limitations were explained.
>
> * Section 5
>    "Upon receiving this ICMP message, the tunnel
>    ingress MUST NOT continue to use GRE-in-UDP encapsulation toward
>    this tunnel egress without management intervention."
> - but also the ingress MUST validate the IP address originating the ICMP
> message is an address of an active tunnel and MUST evaluate the packet
> header returned in the ICMP payload to ensure the source port is the one
> used for this tunnel. The draft should also note that how to handle tunnels
> carrying other tunnels :-)
>
> Gorry
>
>
> On 02/07/2014 14:32, Lucy yong wrote:
>
>> Hello,
>>
>> The new version is uploaded. It addresses the comments in London's
>> meeting, i.e. congestion control, checksum, etc. Please let us know if the
>> description is sufficient in this version. Other comments are welcome.
>>
>> Thanks,
>> Lucy
>>
>> -----Original Message-----
>> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of
>> internet-drafts@ietf.org
>> Sent: Tuesday, July 01, 2014 2:26 PM
>> To: i-d-announce@ietf.org
>> Cc: tsvwg@ietf.org
>> Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-gre-in-udp-encap-02.txt
>>
>>
>> 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 Working
>> Group of the IETF.
>>
>>          Title           : Generic UDP Encapsulation for IP Tunneling
>>          Authors         : Edward Crabbe
>>                            Lucy Yong
>>                            Xiaohu Xu
>>         Filename        : draft-ietf-tsvwg-gre-in-udp-encap-02.txt
>>         Pages           : 13
>>         Date            : 2014-07-01
>>
>> Abstract:
>>     This document describes a method of encapsulating arbitrary
>>     protocols within GRE and UDP headers.  In this encapsulation, the
>>     source UDP port may be used as an entropy field for purposes of load
>>     balancing while the payload protocol may be identified by the GRE
>>     Protocol Type.
>>
>>
>> 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:
>> http://tools.ietf.org/html/draft-ietf-tsvwg-gre-in-udp-encap-02
>>
>> A diff from the previous version is available at:
>> http://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-gre-in-udp-encap-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/
>>
>>
>

--089e01228d46813a6904fdb22828
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr">Gorry, thank you for the comments. =A0Implicit agreement w=
here not noted, with explicit disagreements inline:<div class=3D"gmail_extr=
a"><br><br><div class=3D"gmail_quote">On Tue, Jul 8, 2014 at 8:38 AM, Gorry=
 Fairhurst <span dir=3D"ltr">&lt;<a href=3D"mailto:gorry@erg.abdn.ac.uk" ta=
rget=3D"_blank">gorry@erg.abdn.ac.uk</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">I have some comments on this draft as an ind=
ividual, please see below:<br>
<br>
* First, I think the new revision helps explain some of the questions that =
were highlighted, thanks for preparing the update.<br>
<br>
* section 3<br>
&quot;randomly selected constant value, e.g. zero,&quot;<br>
- I don&#39;t understand this in two ways:<br>
(1) I don&#39;t think it should be zero, I think you could find a hash of s=
ome internal state to generate a ramdom number. For instance this is used i=
n the PIM routing (RFC4601) to specify the genID which is simply a unique n=
umber each time the system reboots or the session is restarted (this has th=
e advantage of reseting state after a reboot).<br>


(2) setting it to zero means you wont have any entropy for ECMP, so why is =
this mode specified in this draft - or is for more general use?<br>
<br>
* Section 3 para 3 - &quot;packets destined<br>
=A0 =A0to this assigned UDP destination port [TBD] MAY have their UDP<br>
=A0 =A0checksum set to zero.&quot;<br>
I think the first specification here applies only to IPv4, sicne IPv6 rules=
 are stated later, could you insert &quot;IPv4&quot; before &quot;packets&q=
uot; ?<br>
<br>
* Section 3 para 3, says SHOULD use IPv6 UDP Checksum, which is fine, but t=
hen it leaves open the possibility that an implementation may not (i.es.thi=
s is more than the base IPv6 sepcification). I think if you choose SHOULD r=
ather MUST there should at least be some statement that points to RFC6936 (=
e.g. section 2 explains the function of this check, and section 5 decsribes=
 the additional functions that would need to be defined to allow this encap=
sulation to use a zero IPv6 UDP Checksum.)<br>


<br>
* Section 3 para 5, The Congestion Control statement is true if the IP traf=
fic implements Congestion Control, so I&#39;m not sure I buy &quot;no need =
to apply&quot;, and would prefer the statement in RFC 5405, which carefully=
 says:<br>


<br>
=A0 &quot;IP-based traffic is generally assumed to be congestion-controlled=
,<br>
=A0 =A0i.e., it is assumed that the transport protocols generating IP-based=
<br>
=A0 =A0traffic at the sender already employ mechanisms that are sufficient<=
br>
=A0 =A0to address congestion on the path. =A0Consequently, a tunnel carryin=
g<br>
=A0 =A0IP-based traffic should already interact appropriately with other<br=
>
=A0 =A0traffic sharing the path, and specific congestion control mechanisms=
<br>
=A0 =A0for the tunnel are not necessary.&quot;<br>
<br>
- Citing RFC 5405 (or successor) may help clarify this.<br>
<br>
* The current revision syas: &quot; In the absence of either, a congestion =
=A0 <br>
control mechanism SHOULD be implemented at the tunnel ingress and =A0 =A0 =
=A0 <br>
egress.&quot;<br>
- The &quot;Circuit-Breaker&quot; draft is not really a form of congestion =
control, as currently written. A circuit-breaker is intended primarily to t=
erminate traffic that is persistently thought to cause congestion. I think =
this is actually what is intended.<br>


<br>
* I am dubious whether it is sufficient to say you need to do &quot;somethi=
ng&quot;, personally I&#39;d much rather see something that says you MUST i=
mplement XXXX, and you SHOULD use it under YYY conditions. It would be fine=
 to explain that well-provisioned or controlled environments...etc... provi=
de other equivalent mechanisms that would allow an operator to not use this=
. I don&#39;t see the point in writing a spec that says you need to invent =
another mechanism to deploy it.<br>


<br></blockquote><div><br></div><div>[ed] we had discussed this in referenc=
e to the as the transport area meeting and come to the agreement that SHOUL=
D was sufficient. =A0Specifically: =A0if someone is operating this only on =
an intradomain basis without interdomain traversal, they should not have to=
 implement &#39;the circuit breaker&#39;. =A0</div>

<div>=A0<br></div><div>Regardless, is it sufficient to say that for an inte=
rdomain use case where the traffic is not congestion controlled at the sour=
ce, a circuitbreaker MUST be implemented?</div><div><br></div><div>=A0</div=
>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
- How hard can it be to identify the best type of circuit breaker for this =
case and require the sender and receiver implementers counters per source a=
ddress/source port pair?<br>
<br>
* Section 4.<br>
/ Tunnel ingress SHOULD/<br>
^ Insert &quot;the&quot;<br>
<br>
* SHOULD perform fragmentation ... and if it does not, then what happens, p=
lease specify what happens when the recommendation is not applied.<br>
<br>
* &quot;impact network performance and user experience.&quot;<br>
- true, but I think you should be more precise, e.g. refer to section 1.3 o=
f RFC6936, where various limitations were explained.<br>
<br>
* Section 5<br>
=A0 =A0&quot;Upon receiving this ICMP message, the tunnel<br>
=A0 =A0ingress MUST NOT continue to use GRE-in-UDP encapsulation toward<br>
=A0 =A0this tunnel egress without management intervention.&quot;<br>
- but also the ingress MUST validate the IP address originating the ICMP me=
ssage is an address of an active tunnel and MUST evaluate the packet header=
 returned in the ICMP payload to ensure the source port is the one used for=
 this tunnel. The draft should also note that how to handle tunnels carryin=
g other tunnels :-)<span class=3D"HOEnZb"><font color=3D"#888888"><br>


<br>
Gorry</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
On 02/07/2014 14:32, Lucy yong wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Hello,<br>
<br>
The new version is uploaded. It addresses the comments in London&#39;s meet=
ing, i.e. congestion control, checksum, etc. Please let us know if the desc=
ription is sufficient in this version. Other comments are welcome.<br>


<br>
Thanks,<br>
Lucy<br>
<br>
-----Original Message-----<br>
From: tsvwg [mailto:<a href=3D"mailto:tsvwg-bounces@ietf.org" target=3D"_bl=
ank">tsvwg-bounces@ietf.org</a><u></u>] On Behalf Of <a href=3D"mailto:inte=
rnet-drafts@ietf.org" target=3D"_blank">internet-drafts@ietf.org</a><br>
Sent: Tuesday, July 01, 2014 2:26 PM<br>
To: <a href=3D"mailto:i-d-announce@ietf.org" target=3D"_blank">i-d-announce=
@ietf.org</a><br>
Cc: <a href=3D"mailto:tsvwg@ietf.org" target=3D"_blank">tsvwg@ietf.org</a><=
br>
Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-gre-in-udp-<u></u>encap-02.tx=
t<br>
<br>
<br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=A0 This draft is a work item of the Transport Area Working Group Working G=
roup of the IETF.<br>
<br>
=A0 =A0 =A0 =A0 =A0Title =A0 =A0 =A0 =A0 =A0 : Generic UDP Encapsulation fo=
r IP Tunneling<br>
=A0 =A0 =A0 =A0 =A0Authors =A0 =A0 =A0 =A0 : Edward Crabbe<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Lucy Yong<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Xiaohu Xu<br>
=A0 =A0 =A0 =A0 Filename =A0 =A0 =A0 =A0: draft-ietf-tsvwg-gre-in-udp-<u></=
u>encap-02.txt<br>
=A0 =A0 =A0 =A0 Pages =A0 =A0 =A0 =A0 =A0 : 13<br>
=A0 =A0 =A0 =A0 Date =A0 =A0 =A0 =A0 =A0 =A0: 2014-07-01<br>
<br>
Abstract:<br>
=A0 =A0 This document describes a method of encapsulating arbitrary<br>
=A0 =A0 protocols within GRE and UDP headers. =A0In this encapsulation, the=
<br>
=A0 =A0 source UDP port may be used as an entropy field for purposes of loa=
d<br>
=A0 =A0 balancing while the payload protocol may be identified by the GRE<b=
r>
=A0 =A0 Protocol Type.<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-tsvwg-gre-in-udp-enc=
ap/" target=3D"_blank">https://datatracker.ietf.org/<u></u>doc/draft-ietf-t=
svwg-gre-in-<u></u>udp-encap/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-tsvwg-gre-in-udp-encap-02"=
 target=3D"_blank">http://tools.ietf.org/html/<u></u>draft-ietf-tsvwg-gre-i=
n-udp-<u></u>encap-02</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-gre-in-udp-e=
ncap-02" target=3D"_blank">http://www.ietf.org/rfcdiff?<u></u>url2=3Ddraft-=
ietf-tsvwg-gre-in-<u></u>udp-encap-02</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n until the htmlized version and diff are available at <a href=3D"http://to=
ols.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-<u></u>drafts/</a><br>
<br>
</blockquote>
<br>
</div></div></blockquote></div><br></div></div>

--089e01228d46813a6904fdb22828--


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From: Lucy yong <lucy.yong@huawei.com>
To: Edward Crabbe <edc@google.com>, Gorry Fairhurst <gorry@erg.abdn.ac.uk>
Thread-Topic: [tsvwg] FW: I-D Action: draft-ietf-tsvwg-gre-in-udp-encap-02.txt
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Hi Gorry,

Yes, thank you for the review and comments. Please see inline.

From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Edward Crabbe
Sent: Tuesday, July 08, 2014 12:45 PM
To: Gorry Fairhurst
Cc: draft-ietf-tsvwg-gre-in-udp-encap@tools.ietf.org; tsvwg WG
Subject: Re: [tsvwg] FW: I-D Action: draft-ietf-tsvwg-gre-in-udp-encap-02.t=
xt

Gorry, thank you for the comments.  Implicit agreement where not noted, wit=
h explicit disagreements inline:

On Tue, Jul 8, 2014 at 8:38 AM, Gorry Fairhurst <gorry@erg.abdn.ac.uk<mailt=
o:gorry@erg.abdn.ac.uk>> wrote:
I have some comments on this draft as an individual, please see below:

* First, I think the new revision helps explain some of the questions that =
were highlighted, thanks for preparing the update.

* section 3
"randomly selected constant value, e.g. zero,"
- I don't understand this in two ways:
(1) I don't think it should be zero, I think you could find a hash of some =
internal state to generate a random number. For instance this is used in th=
e PIM routing (RFC4601) to specify the genID which is simply a unique numbe=
r each time the system reboots or the session is restarted (this has the ad=
vantage of reseting state after a reboot).
(2) setting it to zero means you wont have any entropy for ECMP, so why is =
this mode specified in this draft - or is for more general use?
[Lucy] how about "a randomly selected constant value"? The reason we mentio=
n this method here is to address the case when ingress is unable to differe=
ntiate the flows from the received packets. In this case, the ingress sets =
a same value for all the packets. If ingress is able to generate the entrop=
y value from the flows, the entropy value is used at source UDP.
Regarding the method for selecting the value, it simply can use the time va=
lue at boot up time. However I prefer to leave the algorithm for selecting =
the value outside scope of the draft. Will that be acceptable?


* Section 3 para 3 - "packets destined
   to this assigned UDP destination port [TBD] MAY have their UDP
   checksum set to zero."
I think the first specification here applies only to IPv4, sicne IPv6 rules=
 are stated later, could you insert "IPv4" before "packets" ?
[Lucy] OK.


* Section 3 para 3, says SHOULD use IPv6 UDP Checksum, which is fine, but t=
hen it leaves open the possibility that an implementation may not (i.es.thi=
s is more than the base IPv6 sepcification). I think if you choose SHOULD r=
ather MUST there should at least be some statement that points to RFC6936 (=
e.g. section 2 explains the function of this check, and section 5 decsribes=
 the additional functions that would need to be defined to allow this encap=
sulation to use a zero IPv6 UDP Checksum.)
[Lucy] will use "MUST" here. Thus no need to refer to RFC6936.


* Section 3 para 5, The Congestion Control statement is true if the IP traf=
fic implements Congestion Control, so I'm not sure I buy "no need to apply"=
, and would prefer the statement in RFC 5405, which carefully says:

  "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."


- Citing RFC 5405 (or successor) may help clarify this.
[Lucy] Thanks for the suggestion. We will take your suggestion.



* The current revision syas: " In the absence of either, a congestion
control mechanism SHOULD be implemented at the tunnel ingress and
egress."
- The "Circuit-Breaker" draft is not really a form of congestion control, a=
s currently written. A circuit-breaker is intended primarily to terminate t=
raffic that is persistently thought to cause congestion. I think this is ac=
tually what is intended.
[Lucy] We will not describe Circuit-breaker in this draft but cite the CB d=
raft. Terminating traffic is one way to reduce the congestion. Will "conges=
tion handling mechanism" is acceptable?
Lucy


* I am dubious whether it is sufficient to say you need to do "something", =
personally I'd much rather see something that says you MUST implement XXXX,=
 and you SHOULD use it under YYY conditions. It would be fine to explain th=
at well-provisioned or controlled environments...etc... provide other equiv=
alent mechanisms that would allow an operator to not use this. I don't see =
the point in writing a spec that says you need to invent another mechanism =
to deploy it.

[ed] we had discussed this in reference to the as the transport area meetin=
g and come to the agreement that SHOULD was sufficient.  Specifically:  if =
someone is operating this only on an intradomain basis without interdomain =
traversal, they should not have to implement 'the circuit breaker'.

Regardless, is it sufficient to say that for an interdomain use case where =
the traffic is not congestion controlled at the source, a circuitbreaker MU=
ST be implemented?


- How hard can it be to identify the best type of circuit breaker for this =
case and require the sender and receiver implementers counters per source a=
ddress/source port pair?

* Section 4.
/ Tunnel ingress SHOULD/
^ Insert "the"

* SHOULD perform fragmentation ... and if it does not, then what happens, p=
lease specify what happens when the recommendation is not applied.

* "impact network performance and user experience."
- true, but I think you should be more precise, e.g. refer to section 1.3 o=
f RFC6936, where various limitations were explained.

* Section 5
   "Upon receiving this ICMP message, the tunnel
   ingress MUST NOT continue to use GRE-in-UDP encapsulation toward
   this tunnel egress without management intervention."
- but also the ingress MUST validate the IP address originating the ICMP me=
ssage is an address of an active tunnel and MUST evaluate the packet header=
 returned in the ICMP payload to ensure the source port is the one used for=
 this tunnel. The draft should also note that how to handle tunnels carryin=
g other tunnels :-)

Gorry


On 02/07/2014 14:32, Lucy yong wrote:
Hello,

The new version is uploaded. It addresses the comments in London's meeting,=
 i.e. congestion control, checksum, etc. Please let us know if the descript=
ion is sufficient in this version. Other comments are welcome.

Thanks,
Lucy

-----Original Message-----
From: tsvwg [mailto:tsvwg-bounces@ietf.org<mailto:tsvwg-bounces@ietf.org>] =
On Behalf Of internet-drafts@ietf.org<mailto:internet-drafts@ietf.org>
Sent: Tuesday, July 01, 2014 2:26 PM
To: i-d-announce@ietf.org<mailto:i-d-announce@ietf.org>
Cc: tsvwg@ietf.org<mailto:tsvwg@ietf.org>
Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-gre-in-udp-encap-02.txt


A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
  This draft is a work item of the Transport Area Working Group Working Gro=
up of the IETF.

         Title           : Generic UDP Encapsulation for IP Tunneling
         Authors         : Edward Crabbe
                           Lucy Yong
                           Xiaohu Xu
        Filename        : draft-ietf-tsvwg-gre-in-udp-encap-02.txt
        Pages           : 13
        Date            : 2014-07-01

Abstract:
    This document describes a method of encapsulating arbitrary
    protocols within GRE and UDP headers.  In this encapsulation, the
    source UDP port may be used as an entropy field for purposes of load
    balancing while the payload protocol may be identified by the GRE
    Protocol Type.


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:
http://tools.ietf.org/html/draft-ietf-tsvwg-gre-in-udp-encap-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-gre-in-udp-encap-02


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

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



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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi Gorry,<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Yes, thank you for the re=
view and comments. Please see inline.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> tsvwg [m=
ailto:tsvwg-bounces@ietf.org]
<b>On Behalf Of </b>Edward Crabbe<br>
<b>Sent:</b> Tuesday, July 08, 2014 12:45 PM<br>
<b>To:</b> Gorry Fairhurst<br>
<b>Cc:</b> draft-ietf-tsvwg-gre-in-udp-encap@tools.ietf.org; tsvwg WG<br>
<b>Subject:</b> Re: [tsvwg] FW: I-D Action: draft-ietf-tsvwg-gre-in-udp-enc=
ap-02.txt<o:p></o:p></span></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">Gorry, thank you for the comments. &nbsp;Implicit ag=
reement where not noted, with explicit disagreements inline:<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Tue, Jul 8, 2014 at 8:38 AM, Gorry Fairhurst &lt;=
<a href=3D"mailto:gorry@erg.abdn.ac.uk" target=3D"_blank">gorry@erg.abdn.ac=
.uk</a>&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">I have some comments =
on this draft as an individual, please see below:<br>
<br>
* First, I think the new revision helps explain some of the questions that =
were highlighted, thanks for preparing the update.<br>
<br>
* section 3<br>
&quot;randomly selected constant value, e.g. zero,&quot;<br>
- I don't understand this in two ways:<br>
(1) I don't think it should be zero, I think you could find a hash of some =
internal state to generate a random number. For instance this is used in th=
e PIM routing (RFC4601) to specify the genID which is simply a unique numbe=
r each time the system reboots or
 the session is restarted (this has the advantage of reseting state after a=
 reboot).<br>
(2) setting it to zero means you wont have any entropy for ECMP, so why is =
this mode specified in this draft - or is for more general use?<span style=
=3D"color:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><i><span style=3D"=
font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;col=
or:#1F497D">[Lucy] how about &#8220;a randomly selected constant value&#822=
1;? The reason we mention this method here is to address the case when
 ingress is unable to differentiate the flows from the received packets. In=
 this case, the ingress sets a same value for all the packets. If ingress i=
s able to generate the entropy value from the flows, the entropy value is u=
sed at source UDP.<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><i><span style=3D"=
font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;col=
or:#1F497D">Regarding the method for selecting the value, it simply can use=
 the time value at boot up time. However I prefer to leave
 the algorithm for selecting the value outside scope of the draft. Will tha=
t be acceptable?<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
* Section 3 para 3 - &quot;packets destined<br>
&nbsp; &nbsp;to this assigned UDP destination port [TBD] MAY have their UDP=
<br>
&nbsp; &nbsp;checksum set to zero.&quot;<br>
I think the first specification here applies only to IPv4, sicne IPv6 rules=
 are stated later, could you insert &quot;IPv4&quot; before &quot;packets&q=
uot; ?<span style=3D"color:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><i><span style=3D"=
font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;col=
or:#1F497D">[Lucy] OK.<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
* Section 3 para 3, says SHOULD use IPv6 UDP Checksum, which is fine, but t=
hen it leaves open the possibility that an implementation may not (i.es.thi=
s is more than the base IPv6 sepcification). I think if you choose SHOULD r=
ather MUST there should at least
 be some statement that points to RFC6936 (e.g. section 2 explains the func=
tion of this check, and section 5 decsribes the additional functions that w=
ould need to be defined to allow this encapsulation to use a zero IPv6 UDP =
Checksum.)<span style=3D"color:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><i><span style=3D"=
font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;col=
or:#1F497D">[Lucy] will use &#8220;MUST&#8221; here. Thus no need to refer =
to RFC6936.<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
* Section 3 para 5, The Congestion Control statement is true if the IP traf=
fic implements Congestion Control, so I'm not sure I buy &quot;no need to a=
pply&quot;, and would prefer the statement in RFC 5405, which carefully say=
s:<br>
<br>
&nbsp; &quot;IP-based traffic is generally assumed to be congestion-control=
led,<br>
&nbsp; &nbsp;i.e., it is assumed that the transport protocols generating IP=
-based<br>
&nbsp; &nbsp;traffic at the sender already employ mechanisms that are suffi=
cient<br>
&nbsp; &nbsp;to address congestion on the path. &nbsp;Consequently, a tunne=
l carrying<br>
&nbsp; &nbsp;IP-based traffic should already interact appropriately with ot=
her<br>
&nbsp; &nbsp;traffic sharing the path, and specific congestion control mech=
anisms<br>
&nbsp; &nbsp;for the tunnel are not necessary.&quot;<span style=3D"color:#1=
F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
- Citing RFC 5405 (or successor) may help clarify this.<span style=3D"color=
:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><i><span style=3D"=
font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;col=
or:#1F497D">[Lucy] Thanks for the suggestion. We will take your suggestion.=
<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><i><span style=3D"=
font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;col=
or:#1F497D"><o:p>&nbsp;</o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
* The current revision syas: &quot; In the absence of either, a congestion =
&nbsp; <br>
control mechanism SHOULD be implemented at the tunnel ingress and &nbsp; &n=
bsp; &nbsp; <br>
egress.&quot;<br>
- The &quot;Circuit-Breaker&quot; draft is not really a form of congestion =
control, as currently written. A circuit-breaker is intended primarily to t=
erminate traffic that is persistently thought to cause congestion. I think =
this is actually what is intended.<span style=3D"color:#1F497D"><o:p></o:p>=
</span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><i><span style=3D"=
font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;col=
or:#1F497D">[Lucy] We will not describe Circuit-breaker in this draft but c=
ite the CB draft. Terminating traffic is one way to reduce
 the congestion. Will &#8220;congestion handling mechanism&#8221; is accept=
able? &nbsp;<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><i><span style=3D"=
font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;col=
or:#1F497D">Lucy<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
* I am dubious whether it is sufficient to say you need to do &quot;somethi=
ng&quot;, personally I'd much rather see something that says you MUST imple=
ment XXXX, and you SHOULD use it under YYY conditions. It would be fine to =
explain that well-provisioned or controlled
 environments...etc... provide other equivalent mechanisms that would allow=
 an operator to not use this. I don't see the point in writing a spec that =
says you need to invent another mechanism to deploy it.<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">[ed] we had discussed this in reference to the as th=
e transport area meeting and come to the agreement that SHOULD was sufficie=
nt. &nbsp;Specifically: &nbsp;if someone is operating this only on an intra=
domain basis without interdomain traversal,
 they should not have to implement 'the circuit breaker'. &nbsp;<o:p></o:p>=
</p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Regardless, is it sufficient to say that for an inte=
rdomain use case where the traffic is not congestion controlled at the sour=
ce, a circuitbreaker MUST be implemented?<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0in;margin-=
bottom:5.0pt">
<p class=3D"MsoNormal">- How hard can it be to identify the best type of ci=
rcuit breaker for this case and require the sender and receiver implementer=
s counters per source address/source port pair?<br>
<br>
* Section 4.<br>
/ Tunnel ingress SHOULD/<br>
^ Insert &quot;the&quot;<br>
<br>
* SHOULD perform fragmentation ... and if it does not, then what happens, p=
lease specify what happens when the recommendation is not applied.<br>
<br>
* &quot;impact network performance and user experience.&quot;<br>
- true, but I think you should be more precise, e.g. refer to section 1.3 o=
f RFC6936, where various limitations were explained.<br>
<br>
* Section 5<br>
&nbsp; &nbsp;&quot;Upon receiving this ICMP message, the tunnel<br>
&nbsp; &nbsp;ingress MUST NOT continue to use GRE-in-UDP encapsulation towa=
rd<br>
&nbsp; &nbsp;this tunnel egress without management intervention.&quot;<br>
- but also the ingress MUST validate the IP address originating the ICMP me=
ssage is an address of an active tunnel and MUST evaluate the packet header=
 returned in the ICMP payload to ensure the source port is the one used for=
 this tunnel. The draft should also
 note that how to handle tunnels carrying other tunnels :-)<span style=3D"c=
olor:#888888"><br>
<br>
<span class=3D"hoenzb">Gorry</span></span><o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><br>
<br>
On 02/07/2014 14:32, Lucy yong wrote:<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Hello,<br>
<br>
The new version is uploaded. It addresses the comments in London's meeting,=
 i.e. congestion control, checksum, etc. Please let us know if the descript=
ion is sufficient in this version. Other comments are welcome.<br>
<br>
Thanks,<br>
Lucy<br>
<br>
-----Original Message-----<br>
From: tsvwg [mailto:<a href=3D"mailto:tsvwg-bounces@ietf.org" target=3D"_bl=
ank">tsvwg-bounces@ietf.org</a>] On Behalf Of
<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">internet-draf=
ts@ietf.org</a><br>
Sent: Tuesday, July 01, 2014 2:26 PM<br>
To: <a href=3D"mailto:i-d-announce@ietf.org" target=3D"_blank">i-d-announce=
@ietf.org</a><br>
Cc: <a href=3D"mailto:tsvwg@ietf.org" target=3D"_blank">tsvwg@ietf.org</a><=
br>
Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-gre-in-udp-encap-02.txt<br>
<br>
<br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
&nbsp; This draft is a work item of the Transport Area Working Group Workin=
g Group of the IETF.<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Title &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
: Generic UDP Encapsulation for IP Tunneling<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Authors &nbsp; &nbsp; &nbsp; &nbsp; : Edw=
ard Crabbe<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
; &nbsp; &nbsp; &nbsp;Lucy Yong<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
; &nbsp; &nbsp; &nbsp;Xiaohu Xu<br>
&nbsp; &nbsp; &nbsp; &nbsp; Filename &nbsp; &nbsp; &nbsp; &nbsp;: draft-iet=
f-tsvwg-gre-in-udp-encap-02.txt<br>
&nbsp; &nbsp; &nbsp; &nbsp; Pages &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : 13<b=
r>
&nbsp; &nbsp; &nbsp; &nbsp; Date &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;:=
 2014-07-01<br>
<br>
Abstract:<br>
&nbsp; &nbsp; This document describes a method of encapsulating arbitrary<b=
r>
&nbsp; &nbsp; protocols within GRE and UDP headers. &nbsp;In this encapsula=
tion, the<br>
&nbsp; &nbsp; source UDP port may be used as an entropy field for purposes =
of load<br>
&nbsp; &nbsp; balancing while the payload protocol may be identified by the=
 GRE<br>
&nbsp; &nbsp; Protocol Type.<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-tsvwg-gre-in-udp-enc=
ap/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-tsvwg-gr=
e-in-udp-encap/</a><br>
<br>
There's also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-tsvwg-gre-in-udp-encap-02"=
 target=3D"_blank">http://tools.ietf.org/html/draft-ietf-tsvwg-gre-in-udp-e=
ncap-02</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-tsvwg-gre-in-udp-e=
ncap-02" target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-ts=
vwg-gre-in-udp-encap-02</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n until the htmlized version and diff are available at
<a href=3D"http://tools.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</body>
</html>

--_000_2691CE0099834E4A9C5044EEC662BB9D453BD8E3dfweml701chmchi_--


From nobody Tue Jul  8 12:56:37 2014
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References: <2691CE0099834E4A9C5044EEC662BB9D453B0EFA@dfweml701-chm.china.huawei.com> <53BC1059.4000905@erg.abdn.ac.uk> <CACKN6JHcZBAfTGGSCFGvgecnSPmiJ8fc=nQReq-9+4QDtdZfkg@mail.gmail.com> <2691CE0099834E4A9C5044EEC662BB9D453BD8E3@dfweml701-chm.china.huawei.com>
Date: Tue, 8 Jul 2014 20:56:31 +0100
From: gorry@erg.abdn.ac.uk
To: "Lucy yong" <lucy.yong@huawei.com>
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Cc: Gorry Fairhurst <gorry@erg.abdn.ac.uk>, "draft-ietf-tsvwg-gre-in-udp-encap@tools.ietf.org" <draft-ietf-tsvwg-gre-in-udp-encap@tools.ietf.org>, tsvwg WG <tsvwg@ietf.org>
Subject: Re: [tsvwg] FW: I-D Action: draft-ietf-tsvwg-gre-in-udp-encap-02.txt
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Thanks, See some responses in-line,

Gorry

> Hi Gorry,
>
> Yes, thank you for the review and comments. Please see inline.
>
> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Edward Crabbe
> Sent: Tuesday, July 08, 2014 12:45 PM
> To: Gorry Fairhurst
> Cc: draft-ietf-tsvwg-gre-in-udp-encap@tools.ietf.org; tsvwg WG
> Subject: Re: [tsvwg] FW: I-D Action:
> draft-ietf-tsvwg-gre-in-udp-encap-02.txt
>
> Gorry, thank you for the comments.  Implicit agreement where not noted,
> with explicit disagreements inline:
>
> On Tue, Jul 8, 2014 at 8:38 AM, Gorry Fairhurst
> <gorry@erg.abdn.ac.uk<mailto:gorry@erg.abdn.ac.uk>> wrote:
> I have some comments on this draft as an individual, please see below:
>
> * First, I think the new revision helps explain some of the questions that
> were highlighted, thanks for preparing the update.
>
> * section 3
> "randomly selected constant value, e.g. zero,"
> - I don't understand this in two ways:
> (1) I don't think it should be zero, I think you could find a hash of some
> internal state to generate a random number. For instance this is used in
> the PIM routing (RFC4601) to specify the genID which is simply a unique
> number each time the system reboots or the session is restarted (this has
> the advantage of reseting state after a reboot).
> (2) setting it to zero means you wont have any entropy for ECMP, so why is
> this mode specified in this draft - or is for more general use?
> [Lucy] how about "a randomly selected constant value"? The reason we
> mention this method here is to address the case when ingress is unable to
> differentiate the flows from the received packets. In this case, the
> ingress sets a same value for all the packets. If ingress is able to
> generate the entropy value from the flows, the entropy value is used at
> source UDP.
> Regarding the method for selecting the value, it simply can use the time
> value at boot up time. However I prefer to leave the algorithm for
> selecting the value outside scope of the draft. Will that be acceptable?
>
GF: That makes sense - I agree, you could give an example, but I wouldn't
expect the ID to specify the algorithm for initialising the value.

>
> * Section 3 para 3 - "packets destined
>    to this assigned UDP destination port [TBD] MAY have their UDP
>    checksum set to zero."
> I think the first specification here applies only to IPv4, sicne IPv6
> rules are stated later, could you insert "IPv4" before "packets" ?
> [Lucy] OK.
>
>
> * Section 3 para 3, says SHOULD use IPv6 UDP Checksum, which is fine, but
> then it leaves open the possibility that an implementation may not
> (i.es.this is more than the base IPv6 sepcification). I think if you
> choose SHOULD rather MUST there should at least be some statement that
> points to RFC6936 (e.g. section 2 explains the function of this check, and
> section 5 decsribes the additional functions that would need to be defined
> to allow this encapsulation to use a zero IPv6 UDP Checksum.)
> [Lucy] will use "MUST" here. Thus no need to refer to RFC6936.
>
GF: Either way works for me. MUST for IPv6 is obviously easy to get
agreement upon.

>
> * Section 3 para 5, The Congestion Control statement is true if the IP
> traffic implements Congestion Control, so I'm not sure I buy "no need to
> apply", and would prefer the statement in RFC 5405, which carefully says:
>
>   "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."
>
>
> - Citing RFC 5405 (or successor) may help clarify this.
> [Lucy] Thanks for the suggestion. We will take your suggestion.
>
>
>
> * The current revision syas: " In the absence of either, a congestion
> control mechanism SHOULD be implemented at the tunnel ingress and
> egress."
> - The "Circuit-Breaker" draft is not really a form of congestion control,
> as currently written. A circuit-breaker is intended primarily to terminate
> traffic that is persistently thought to cause congestion. I think this is
> actually what is intended.
>
> [Lucy] We will not describe Circuit-breaker in this draft but cite the CB
> draft. Terminating traffic is one way to reduce the congestion.
> Will  "congestion handling mechanism" be acceptable?
>
GF: I think that is better.
>
>
> * I am dubious whether it is sufficient to say you need to do "something",
> personally I'd much rather see something that says you MUST implement
> XXXX, and you SHOULD use it under YYY conditions. It would be fine to
> explain that well-provisioned or controlled environments...etc... provide
> other equivalent mechanisms that would allow an operator to not use this.
> I don't see the point in writing a spec that says you need to invent
> another mechanism to deploy it.
>
> [ed] we had discussed this in reference to the as the transport area
> meeting and come to the agreement that SHOULD was sufficient.
> Specifically:  if someone is operating this only on an intradomain basis
> without interdomain traversal, they should not have to implement 'the
> circuit breaker'.
>
GF: That's where we differ, but we need to be precise - I'm not saying an
operator MUST use the Circuit breaker, because clearly there are
operational characteristics of a network that may mean the operator
already has mechanisms to deal with overload.

However. the question is whether the implementor needs to provides the
mechanism, to allow the operator to enable or disable the feature. This is
about what needs to be specified to conform - assuming that a lightweight
circuit breaker is possible at design time, but we know from other
experience is really hard to introduce afterwards to a deployed protocol.

> Regardless, is it sufficient to say that for an interdomain use case where
> the traffic is not congestion controlled at the source, a circuitbreaker
> MUST be implemented?
>
GF: Probably correct. But see above. It is possible to use words such as
MUST be supported and SHOULD be enabled by default.
>
> - How hard can it be to identify the best type of circuit breaker for this
> case and require the sender and receiver implementers counters per source
> address/source port pair?
>
> * Section 4.
> / Tunnel ingress SHOULD/
> ^ Insert "the"
>
> * SHOULD perform fragmentation ... and if it does not, then what happens,
> please specify what happens when the recommendation is not applied.
>
> * "impact network performance and user experience."
> - true, but I think you should be more precise, e.g. refer to section 1.3
> of RFC6936, where various limitations were explained.
>
> * Section 5
>    "Upon receiving this ICMP message, the tunnel
>    ingress MUST NOT continue to use GRE-in-UDP encapsulation toward
>    this tunnel egress without management intervention."
> - but also the ingress MUST validate the IP address originating the ICMP
> message is an address of an active tunnel and MUST evaluate the packet
> header returned in the ICMP payload to ensure the source port is the one
> used for this tunnel. The draft should also note that how to handle
> tunnels carrying other tunnels :-)
>
> Gorry
>
>
> On 02/07/2014 14:32, Lucy yong wrote:
> Hello,
>
> The new version is uploaded. It addresses the comments in London's
> meeting, i.e. congestion control, checksum, etc. Please let us know if the
> description is sufficient in this version. Other comments are welcome.
>
> Thanks,
> Lucy
>
> -----Original Message-----
> From: tsvwg [mailto:tsvwg-bounces@ietf.org<mailto:tsvwg-bounces@ietf.org>]
> On Behalf Of internet-drafts@ietf.org<mailto:internet-drafts@ietf.org>
> Sent: Tuesday, July 01, 2014 2:26 PM
> To: i-d-announce@ietf.org<mailto:i-d-announce@ietf.org>
> Cc: tsvwg@ietf.org<mailto:tsvwg@ietf.org>
> Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-gre-in-udp-encap-02.txt
>
>
> 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 Working
> Group of the IETF.
>
>          Title           : Generic UDP Encapsulation for IP Tunneling
>          Authors         : Edward Crabbe
>                            Lucy Yong
>                            Xiaohu Xu
>         Filename        : draft-ietf-tsvwg-gre-in-udp-encap-02.txt
>         Pages           : 13
>         Date            : 2014-07-01
>
> Abstract:
>     This document describes a method of encapsulating arbitrary
>     protocols within GRE and UDP headers.  In this encapsulation, the
>     source UDP port may be used as an entropy field for purposes of load
>     balancing while the payload protocol may be identified by the GRE
>     Protocol Type.
>
>
> 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:
> http://tools.ietf.org/html/draft-ietf-tsvwg-gre-in-udp-encap-02
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-gre-in-udp-encap-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<http://tools.ietf.org>.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
>
>



From nobody Sun Jul 13 15:15:22 2014
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Hi,

This is just a heads-up, that Gorry & I will briefly present =
draft=E2=80=90welzl=E2=80=90ecn=E2=80=90benefits-01 at TSVWG.

Cheers,
Michael


From nobody Sun Jul 20 06:54:25 2014
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From: "Karen E. Egede Nielsen" <karen.nielsen@tieto.com>
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Subject: Re: [tsvwg] WGLC for draft-ietf-tsvwg-sctp-prpolicies-03 - due to conclude Monday, July 21st, 2014
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 HI,

I support the extensions of the PR function as defined in the document.
I support the work as moving forward as PS. RFC3758 is PS and I do not see
that the new policies
defined pose any new issues compared to the policy already defined in
RFC3758.

(I actually do have some petty issues with the text of  RFC3758,
but these issues are equally applicable to the existing policy of
RFC3758 as to the new proposed ones. I will send these comments later.)

In respect to the technical scope presently in the document, then I think
that the text of the document is in a state ready to publish.
I have the following 2 comments to the scope of function as it is right
now.

#1: Given that we now have multiple PR policies, potentially applicable
within the same
SCTP association (but likely same policy on all messages within same
stream, e.g., media stream/data channel),
then I think that it would be relevant to have the stats in the
sctp_prstatus  be reported on a per policy basis.
Meaning that the socket option sctp_prstatus  should be extended to make
this differentiation.

#2: I note that the socket api handing specified in the document for
 control of whether to attempt to negotiate the function or not, does
 not adequately handle how to control the operation during restarts
(INIT from remote side) of an existing association. However this is a
general issue of the socket api  as specified in RFC6458 and as it is
not particular to the PR function, but general to all negotiable
features.  I thus recognize that be better addresses in an possible
follow up of RFC6458. Whether it be handled as a special purpose
 function to control the operation via-a-vis negotiable features
generally at remote restart of existing association or whether it
comes in as a general extension of all the particular functions
controlling a negotiable feature.
The issue is relevant for SCTP implementations in deployment and is
 today addressed  (at least by some implementations I know of) by
 proprietary means outside of (but somewhat in interworking with) the
socket api.

 BR, Karen


> -----Original Message-----
> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Gorry Fairhurst
> Sent: 3. juli 2014 10:46
> To: tsvwg-chairs@ietf.org; tsvwg@ietf.org
> Subject: [tsvwg] WGLC for draft-ietf-tsvwg-sctp-prpolicies-03 - due to
> conclude Monday, July 21st, 2014
>
> This is the start of the WGLC on "Additional Policies for the Partial
Reliability
> Extension of the Stream Control Transmission Protocol" for publication
> intended as a PS.
>
> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-prpolicies/
>
> This WGLC will last until Monday, July 21st, 2014 (longer since this
includes
> time prior to an IETF meeting).
>
> Could people please send *any* comments to the list if they think this
is
> ready to publish?
>
> A good note is "I have read this version of the draft, and I approve (or
> support) its progression to RFC".
>
> James/Gorry/David
> TSVWG chairs


From nobody Sun Jul 20 07:18:58 2014
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Am 20.07.2014 um 15:54 schrieb Karen E. Egede Nielsen <karen.nielsen@tieto.=
com>:

> HI,
>=20
> I support the extensions of the PR function as defined in the document.

+1

Best,=20
Martin


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--f46d043c7c4cf1aedb04fea0e1c3
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HI,



Having studied RFC3758 in a little detail (in connection with
draft-ietf-tsvwg-sctp-prpolicies)

then there is an issue with the present document text, which I think
deserves

clarification. Or at least I am not sure about how to interpret the text.



I hope that the wg may help guide on what the right
clarification/understanding is.



The issue relate to handling of the flightsize and the calculated

receiver window rwnd when abandoning a message and/or receiving a (S)ACK of
the Forward TSN, respectively.



The rule A2) (section 3.5) says:





       When a data chunk is "abandoned", the sender MUST treat the data
       chunk as being finally acked and no longer outstanding.



This MUST clearly must not be taken literally as it indeed is being
disclaimed by a number of following

recommendations given:



Section 3,5, A2)



       The sender MUST NOT credit an "abandoned" data chunk to the
       partial_bytes_acked as defined in Section 7.2.2 of RFC 2960 [2],
       and MUST NOT advance the cwnd based on this "abandoned" data
       chunk.



Section 3,5 F5)



       If the decision to "abandon" a chunk is made, no matter how such
       a decision is made, the appropriate congestion adjustment MUST be
       made as specified in RFC 2960 if the chunk would have been marked
       for retransmission later (e.g., either by T3-Timeout or by Fast
       Retransmit).





What I would have like to see, however, would be some disclaimers and/or
more explicit descriptions of what

to do/or not do with with the flight-size and the calculated rwnd when a
message is abandoned.



Here there are clearly 2 options in terms of how to interpret the =E2=80=9C=
MUST
treat as no longer outstanding=E2=80=9D:



>From a conservative congestion control perspective then I would opt for
that the data chunk should not be taken out of the flightsize and sliding
window protocol variables before a SACK of the TSN/respectively of the
FORWARD TSN has been received (conditional to F5).



Alternatively if the intend of RFC3758 is to have an abandoned TSN be taken
out of flight-size immediately, then I first and foremost think that

the implication via-a-vis congestion control should be explicitly described
in the document (if the message is in-flight one is in fact overshooting
the cwnd) and secondly then for the sender not to send outside of rwnd, one
would then want to more stringently specify that a delayed FORWARD TSN MUST
be sent no later than with the first data chunk after the abandon of a
message. This may be the intend of C3) and F2) of the document, but it is
not written explicitly.



Thanks in advance.



BR, Karen

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=C2=A0</p><p class=3D"MsoNormal"><span lang=3D"EN-US">Having studied RFC375=
8 in a little detail (in connection with draft-ietf-tsvwg-sctp-prpolicies) =
</span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">then there is an issue with the=
 present document text, which I think deserves </span></p><p class=3D"MsoNo=
rmal"><span lang=3D"EN-US">clarification. Or at least I am not sure about h=
ow to interpret the text.</span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0</span></p><p class=3D"Ms=
oNormal"><span lang=3D"EN-US">I hope that the wg may help guide on what the=
 right clarification/understanding is.</span></p><p class=3D"MsoNormal"><sp=
an lang=3D"EN-US">=C2=A0</span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">The issue relate to handling of=
 the flightsize and the calculated </span></p><p class=3D"MsoNormal"><span =
lang=3D"EN-US">receiver window rwnd when abandoning a message and/or receiv=
ing a (S)ACK of the Forward TSN, respectively.</span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0</span></p><p class=3D"Ms=
oNormal"><span lang=3D"EN-US">The rule A2) (section 3.5) says:</span></p><p=
 class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0</span></p><p class=3D"MsoN=
ormal"><span lang=3D"EN-US">=C2=A0</span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US" style=3D"font-family:&quot;C=
ourier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 When a data chunk is=
 &quot;abandoned&quot;, the sender MUST treat the data<br>=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0 chunk as being finally acked and no longer outstandin=
g.</span><span lang=3D"EN-US"></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0</span></p><p class=3D"Ms=
oNormal"><span lang=3D"EN-US">This MUST clearly must not be taken literally=
 as it indeed is being disclaimed by a number of following </span></p><p cl=
ass=3D"MsoNormal">
<span lang=3D"EN-US">recommendations given:</span></p><p class=3D"MsoNormal=
"><span lang=3D"EN-US">=C2=A0</span></p><p class=3D"MsoNormal"><span lang=
=3D"EN-US">Section 3,5, A2)</span></p><p class=3D"MsoNormal"><span lang=3D"=
EN-US">=C2=A0</span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US" style=3D"font-family:&quot;C=
ourier New&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 The sender MUST NOT =
credit an &quot;abandoned&quot; data chunk to the<br>=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 partial_bytes_acked as defined in Section 7.2.2 of RFC 2960=
 [2],<br>
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 and MUST NOT advance the cwnd based on=
 this &quot;abandoned&quot; data<br>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 ch=
unk.</span></p><p class=3D"MsoPlainText"><span lang=3D"EN-US" style=3D"font=
-family:&quot;Courier New&quot;">=C2=A0</span></p><p class=3D"MsoPlainText"=
>
<span lang=3D"EN-US" style=3D"font-family:&quot;Courier New&quot;">Section =
3,5 F5)</span></p><p class=3D"MsoPlainText"><span lang=3D"EN-US" style=3D"f=
ont-family:&quot;Courier New&quot;">=C2=A0</span></p><p class=3D"MsoPlainTe=
xt"><span lang=3D"EN-US" style=3D"font-family:&quot;Courier New&quot;">=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0 =C2=A0If the decision to &quot;abandon&quot; a =
chunk is made, no matter how such<br>
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 a decision is made, the appropriate co=
ngestion adjustment MUST be<br>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 made as=
 specified in RFC 2960 if the chunk would have been marked<br>=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0 for retransmission later (e.g., either by T3-Timeo=
ut or by Fast<br>
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 Retransmit).</span><span lang=3D"EN-US=
"></span></p><p class=3D"MsoPlainText"><span lang=3D"EN-US">=C2=A0</span></=
p><p class=3D"MsoPlainText"><span lang=3D"EN-US">=C2=A0</span></p><p class=
=3D"MsoNormal"><span lang=3D"EN-US">What I would have like to see, however,=
 would be some disclaimers and/or more explicit descriptions of what </span=
></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">to do/or not do with with the f=
light-size and the calculated rwnd when a message is abandoned.</span></p><=
p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0</span></p><p class=3D"Mso=
Normal"><span lang=3D"EN-US">Here there are clearly 2 options in terms of h=
ow to interpret the =E2=80=9CMUST treat as no longer outstanding=E2=80=9D:<=
/span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0</span></p><p class=3D"Ms=
oNormal"><span lang=3D"EN-US">From a conservative congestion control perspe=
ctive then I would opt for that the data chunk should not be taken out of t=
he flightsize and sliding window protocol variables before a SACK of the TS=
N/respectively of the FORWARD TSN has been received (conditional to F5).</s=
pan></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0</span></p><p class=3D"Ms=
oNormal"><span lang=3D"EN-US">Alternatively if the intend of RFC3758 is to =
have an abandoned TSN be taken out of flight-size immediately, then I first=
 and foremost think that</span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">the implication via-a-vis conge=
stion control should be explicitly described in the document (if the messag=
e is in-flight one is in fact overshooting the cwnd) and secondly then for =
the sender not to send outside of rwnd, one would then want to more stringe=
ntly specify that a delayed FORWARD TSN MUST be sent no later than with the=
 first data chunk after the abandon of a message. This may be the intend of=
 C3) and F2) of the document, but it is not written explicitly.</span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0</span></p><p class=3D"Ms=
oNormal"><span lang=3D"EN-US">Thanks in advance.</span></p><p class=3D"MsoN=
ormal"><span lang=3D"EN-US">=C2=A0</span></p><p class=3D"MsoNormal"><span l=
ang=3D"EN-US">BR, Karen</span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">=C2=A0</span></p><p class=3D"Ms=
oNormal"><span lang=3D"EN-US">=C2=A0</span></p><p class=3D"MsoNormal"><span=
 lang=3D"EN-US">=C2=A0</span></p><p class=3D"MsoNormal"><span lang=3D"EN-US=
">=C2=A0</span></p></div></body></html>

--f46d043c7c4cf1aedb04fea0e1c3--


From nobody Sun Jul 20 09:00:17 2014
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From: "Karen E. Egede Nielsen" <karen.nielsen@tieto.com>
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Subject: Re: [tsvwg] I-D Action: draft-ietf-tsvwg-sctp-ndata-01.txt
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HI,

I have read this draft version. I think that the draft still requires more
work.
Most importantly:

 I think that the stream scheduling functions proposed need to be described
in the core parts of
the document. Right now they are only described in the socket api
consideration part off the document.
Furthermore I think that the draft should single out the subset of stream
scheduling policies that an implementation following
this work must support and the subset that it could support.

I think that the title of the document (as well as the abstract) should be
altered to better reflect that this document does not
only define a new data chunk format to amend some issues with head of line
blocking by large messages,
but indeed prescribes for more general head-of-line controlling functions
such as the stream scheduling functions.

A number of the detailed comments given to a previous version of this draft
still applies as far as I can see.
Please refer to
http://www.ietf.org/mail-archive/web/tsvwg/current/msg12303.html.

BR, Karen



> -----Original Message-----
> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of internet-
> drafts@ietf.org
> Sent: 4. juli 2014 22:52
> To: i-d-announce@ietf.org
> Cc: tsvwg@ietf.org
> Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-sctp-ndata-01.txt
>
>
> 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 Working
> Group of the IETF.
>
>         Title           : Stream Schedulers and a New Data Chunk for the
> Stream
> Control Transmission Protocol
>         Authors         : Randall R. Stewart
>                           Michael Tuexen
>                           Salvatore Loreto
>                           Robin Seggelmann
> 	Filename        : draft-ietf-tsvwg-sctp-ndata-01.txt
> 	Pages           : 11
> 	Date            : 2014-07-04
>
> Abstract:
>    The Stream Control Transmission Protocol (SCTP) is a message oriented
>    transport protocol supporting arbitrary large user messages.
>    However, the sender can not interleave different user messages which
>    causes head of line blocking at the sender side.  To overcome this
>    limitation, this document adds a new data chunk to SCTP.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-ndata/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-ndata-01
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-sctp-ndata-01
>
>
> Please note that it may take a couple of minutes from the time of
> submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/


From nobody Sun Jul 20 12:22:30 2014
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From: "Black, David" <david.black@emc.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>
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The tsvwg agenda for Tuesday (both sessions) was posted a while back:

	https://datatracker.ietf.org/meeting/90/agenda/tsvwg/

Those presenters who haven't already sent slides (thank you to those who
have) should send slides ASAP, by mid-day Monday at the latest please.

If the tsvwg meeting starts on Tuesday w/o slides for an agenda item,
the chairs may impose a 3-5min time limit for a brief verbal summary + Q&A.

Thanks,
--David
----------------------------------------------------
David L. Black, Distinguished Engineer
EMC Corporation, 176 South St., Hopkinton, MA=A0 01748
+1 (508) 293-7953=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 FAX: +1 (508) 293-778=
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david.black@emc.com=A0=A0=A0=A0=A0=A0=A0 Mobile: +1 (978) 394-7754
----------------------------------------------------


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Subject: Re: [tsvwg] I-D Action: draft-ietf-tsvwg-rtcweb-qos-02.txt
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Hi,

Please accept the following comments:

I wonder if the recommendations of this document should not relate to the
viability (or rather the opposite) of
handling different RTCWEB data channels individually from a DSCP markings
perspective. Or more generally
relate to the fact that media streams within the same congestion control
context cannot be handled individually.
Right now such considerations are not in the document. If the intention is
for such consideration to come
in documents from the dart wg, then it would be very important to  have a
reference to this dependency - or ?

Something else:

Right now the document stipulates that packets within the same media stream
may be marked with different DSCP codepoints, i.e.,
it is said:

   One may select difference drop precedences for the
   different packets in the media flow.  Therefore, all packets in the
   stream SHOULD be marked with the same priority but can have
   difference drop precedences.

which indeed is in compliance with the recommendations put forward in
draft-york-dart-dscp-rtp-00 in that
it ensures that the DSCP markings  do not result in re-ordering  within the
media flow, but has it been ascertained, from a CC
perspective that usage of different drop precedence within the same CC
context is viable ?

>From the RTCWEB data channel perspective this certainly isn't viable (being
SCTP based),
whether it is viable from an RMCAT CC perspective is something that we in
the RMCAT wg would need to
make more clear.

BR, Karen

> -----Original Message-----
> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of internet-
> drafts@ietf.org
> Sent: 23. juni 2014 21:12
> To: i-d-announce@ietf.org
> Cc: tsvwg@ietf.org
> Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-rtcweb-qos-02.txt
>
>
> 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 Working
> Group of the IETF.
>
>         Title           : DSCP and other packet markings for RTCWeb QoS
>         Authors         : Subha Dhesikan
>                           Cullen Jennings
>                           Dan Druta
>                           Paul Jones
>                           James Polk
> 	Filename        : draft-ietf-tsvwg-rtcweb-qos-02.txt
> 	Pages           : 7
> 	Date            : 2014-06-23
>
> Abstract:
>    Many networks, such as service provider and enterprise networks, can
>    provide per packet treatments based on Differentiated Services Code
>    Points (DSCP) on a per-hop basis.  This document provides the
>    recommended DSCP values for browsers to use for various classes of
>    traffic.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rtcweb-qos/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-02
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-rtcweb-qos-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 Sun Jul 20 14:38:28 2014
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From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: "Karen E. Egede Nielsen" <karen.nielsen@tieto.com>
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Subject: Re: [tsvwg] comments to RFC3758
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Karen,

Please see comments below...

Karen E. Egede Nielsen wrote:         (Sun, 20 Jul 2014 16:34:06)
> 

(1)
>    When a data chunk is "abandoned", the sender MUST treat the data
>    chunk as being finally acked and no longer outstanding.
> 

This pertains to acknowledgement only: the packet is acked; plain
and simple.  Do not read more into it than that.

The other passages are explicit on how to treat congestion.

(2)
>    The sender MUST NOT credit an "abandoned" data chunk to the
>    partial_bytes_acked as defined in Section 7.2.2 of RFC 2960 [2],
>    and MUST NOT advance the cwnd based on this "abandoned" data chunk.

(3)
>    If the decision to "abandon" a chunk is made, no matter how such a
>    decision is made, the appropriate congestion adjustment MUST be
>    made as specified in RFC 2960 if the chunk would have been marked
>    for retransmission later (e.g., either by T3-Timeout or by Fast
>    Retransmit).


>    What I would have like to see, however, would be some disclaimers
>    and/or more explicit descriptions of what to do/or not do with with
>    the flight-size and the calculated rwnd when a message is
>    abandoned.

I think all three passages are quite clear.

>    Here there are clearly 2 options in terms of how to interpret the
>    MUST treat as no longer outstandingâ:
> 
>    From a conservative congestion control perspective then I would opt
>    for that the data chunk should not be taken out of the flightsize
>    and sliding window protocol variables before a SACK of the
>    TSN/respectively of the FORWARD TSN has been received (conditional
>    to F5).

That is what is said in (3) above.

>    Alternatively if the intend of RFC3758 is to have an abandoned TSN
>    be taken out of flight-size immediately, then I first and foremost
>    think that the implication via-a-vis congestion control should be
>    explicitly described in the document (if the message is in-flight
>    one is in fact overshooting the cwnd) and secondly then for the
>    sender not to send outside of rwnd, one would then want to more
>    stringently specify that a delayed FORWARD TSN MUST be sent no
>    later than with the first data chunk after the abandon of a
>    message. This may be the intend of C3) and F2) of the document, but
>    it is not written explicitly.

No, (2) says it cannot contribute to partially acked bytes and (3) says
it cannot be removed from flight size until a T3-timeout or
acknowledgement.

--brian


-- 
Brian F. G. Bidulock
bidulock@openss7.org
http://www.openss7.org/


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Hi,

>
> (1)
> >    When a data chunk is "abandoned", the sender MUST treat the data
> >    chunk as being finally acked and no longer outstanding.
> >
>
> This pertains to acknowledgement only: the packet is acked; plain and
> simple.  Do not read more into it than that.
>
> The other passages are explicit on how to treat congestion.
>
> (2)
> >    The sender MUST NOT credit an "abandoned" data chunk to the
> >    partial_bytes_acked as defined in Section 7.2.2 of RFC 2960 [2],
> >    and MUST NOT advance the cwnd based on this "abandoned" data chunk.
>
> (3)
> >    If the decision to "abandon" a chunk is made, no matter how such a
> >    decision is made, the appropriate congestion adjustment MUST be
> >    made as specified in RFC 2960 if the chunk would have been marked
> >    for retransmission later (e.g., either by T3-Timeout or by Fast
> >    Retransmit).
>
>
> >    What I would have like to see, however, would be some disclaimers
> >    and/or more explicit descriptions of what to do/or not do with with
> >    the flight-size and the calculated rwnd when a message is
> >    abandoned.
>
> I think all three passages are quite clear.
>
> >    Here there are clearly 2 options in terms of how to interpret the
> >    MUST treat as no longer outstanding :
> >
> >    From a conservative congestion control perspective then I would opt
> >    for that the data chunk should not be taken out of the flightsize
> >    and sliding window protocol variables before a SACK of the
> >    TSN/respectively of the FORWARD TSN has been received (conditional
> >    to F5).
>
> That is what is said in (3) above.
>
> >    Alternatively if the intend of RFC3758 is to have an abandoned TSN
> >    be taken out of flight-size immediately, then I first and foremost
> >    think that the implication via-a-vis congestion control should be
> >    explicitly described in the document (if the message is in-flight
> >    one is in fact overshooting the cwnd) and secondly then for the
> >    sender not to send outside of rwnd, one would then want to more
> >    stringently specify that a delayed FORWARD TSN MUST be sent no
> >    later than with the first data chunk after the abandon of a
> >    message. This may be the intend of C3) and F2) of the document, but
> >    it is not written explicitly.
>
> No, (2) says it cannot contribute to partially acked bytes and (3) says it
> cannot
> be removed from flight size until a T3-timeout or acknowledgement.
>
[Karen]  I cannot see that 3 says this. Such an interpretation of (3)
presumes that
a message is only abandoned at a T3 or when marked for fast retransmission.
Furthermore 3 speaks
about congestion control adjustments not about inflight or rwnd.

BR, Karen

The document does not presume that
> --brian
>
>
> --
> Brian F. G. Bidulock
> bidulock@openss7.org
> http://www.openss7.org/


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From: "Black, David" <david.black@emc.com>
To: "Karen E. Egede Nielsen" <karen.nielsen@tieto.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Date: Sun, 20 Jul 2014 20:51:12 -0400
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From: "Karen E. Egede Nielsen" <karen.nielsen@tieto.com>
References: <20140623191132.21904.23978.idtracker@ietfa.amsl.com> <1373b5f3f2f88c06aafc0deb45287f61@mail.gmail.com> <8D3D17ACE214DC429325B2B98F3AE71207783F6386@MX15A.corp.emc.com>
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Date: Mon, 21 Jul 2014 03:20:16 +0200
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To: "Black, David" <david.black@emc.com>, tsvwg@ietf.org
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Cc: rmcat WG <rmcat@ietf.org>, dart@ietf.org
Subject: Re: [tsvwg] I-D Action: draft-ietf-tsvwg-rtcweb-qos-02.txt
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Hi,

> -----Original Message-----
> From: Black, David [mailto:david.black@emc.com]
> Sent: 21. juli 2014 02:51
> To: Karen E. Egede Nielsen; tsvwg@ietf.org
> Cc: rmcat WG; dart@ietf.org; Black, David
> Subject: RE: [tsvwg] I-D Action: draft-ietf-tsvwg-rtcweb-qos-02.txt
>
> Karen,
>
> <WG chair hat off>, [+dart list]
>
> > I wonder if the recommendations of this document should not relate to
> > the viability (or rather the opposite) of handling different RTCWEB
> > data channels individually from a DSCP markings perspective. Or more
> > generally relate to the fact that media streams within the same
> > congestion control context cannot be handled individually.
> > Right now such considerations are not in the document. If the
> > intention is for such consideration to come in documents from the dart
> > wg, then it would be very important to  have a reference to this
> > dependency - or ?
>
> IMHO, such a reference is necessary, although the dart WG draft will be
> informational.  The reference should be added once there is an official
> dart
> WG draft (likely to happen this week).
>
> > Something else:
> >
> > Right now the document stipulates that packets within the same media
> > stream may be marked with different DSCP codepoints, i.e., it is said:
> >
> >    One may select difference drop precedences for the
> >    different packets in the media flow.  Therefore, all packets in the
> >    stream SHOULD be marked with the same priority but can have
> >    difference drop precedences.
> >
> > which indeed is in compliance with the recommendations put forward in
> > draft-york-dart-dscp-rtp-00 in that
> > it ensures that the DSCP markings  do not result in re-ordering
> > within the media flow, but has it been ascertained, from a CC
> > perspective that usage of different drop precedence within the same CC
> > context is viable ?
>
> Good question - I will include that in my slides for the dart WG meeting.
>
> > >From the RTCWEB data channel perspective this certainly isn't viable
> > >(being
> > SCTP based),
> > whether it is viable from an RMCAT CC perspective is something that we
> > in the RMCAT wg would need to make more clear.
>
> What's the rationale for "certainly isn't viable" regarding different drop
> precedences within a single SCTP session?
[Karen] Ok - "certainly not viable" is too strong formulation.
Having different drop precedences within the same SCTP CC context would,
given that SCTP CC presently is drop driven only,
results in that the whole data channel is impacted, from a CC perspective,
by the highest drop precedence. But I realize that whether this is
acceptable is a question indeed.

>
> Thanks,
> --David (as an editor of draft-york, *not* as a tsvwg WG chair)
>
> > -----Original Message-----
> > From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Karen E.
> > Egede Nielsen
> > Sent: Sunday, July 20, 2014 4:49 PM
> > To: tsvwg@ietf.org
> > Cc: rmcat WG
> > Subject: Re: [tsvwg] I-D Action: draft-ietf-tsvwg-rtcweb-qos-02.txt
> >
> > Hi,
> >
> > Please accept the following comments:
> >
> > I wonder if the recommendations of this document should not relate to
> > the viability (or rather the opposite) of handling different RTCWEB
> > data channels individually from a DSCP markings perspective. Or more
> > generally relate to the fact that media streams within the same
> > congestion control context cannot be handled individually.
> > Right now such considerations are not in the document. If the
> > intention is for such consideration to come in documents from the dart
> > wg, then it would be very important to  have a reference to this
> > dependency - or ?
> >
> > Something else:
> >
> > Right now the document stipulates that packets within the same media
> > stream may be marked with different DSCP codepoints, i.e., it is said:
> >
> >    One may select difference drop precedences for the
> >    different packets in the media flow.  Therefore, all packets in the
> >    stream SHOULD be marked with the same priority but can have
> >    difference drop precedences.
> >
> > which indeed is in compliance with the recommendations put forward in
> > draft-york-dart-dscp-rtp-00 in that
> > it ensures that the DSCP markings  do not result in re-ordering
> > within the media flow, but has it been ascertained, from a CC
> > perspective that usage of different drop precedence within the same CC
> > context is viable ?
> >
> > >From the RTCWEB data channel perspective this certainly isn't viable
> > >(being
> > SCTP based),
> > whether it is viable from an RMCAT CC perspective is something that we
> > in the RMCAT wg would need to make more clear.
> >
> > BR, Karen
> >
> > > -----Original Message-----
> > > From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of internet-
> > > drafts@ietf.org
> > > Sent: 23. juni 2014 21:12
> > > To: i-d-announce@ietf.org
> > > Cc: tsvwg@ietf.org
> > > Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-rtcweb-qos-02.txt
> > >
> > >
> > > 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
> > > Working Group of the IETF.
> > >
> > >         Title           : DSCP and other packet markings for RTCWeb
> > > QoS
> > >         Authors         : Subha Dhesikan
> > >                           Cullen Jennings
> > >                           Dan Druta
> > >                           Paul Jones
> > >                           James Polk
> > > 	Filename        : draft-ietf-tsvwg-rtcweb-qos-02.txt
> > > 	Pages           : 7
> > > 	Date            : 2014-06-23
> > >
> > > Abstract:
> > >    Many networks, such as service provider and enterprise networks,
> > > can
> > >    provide per packet treatments based on Differentiated Services Code
> > >    Points (DSCP) on a per-hop basis.  This document provides the
> > >    recommended DSCP values for browsers to use for various classes of
> > >    traffic.
> > >
> > >
> > > The IETF datatracker status page for this draft is:
> > > https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rtcweb-qos/
> > >
> > > There's also a htmlized version available at:
> > > http://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-02
> > >
> > > A diff from the previous version is available at:
> > > http://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-rtcweb-qos-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/


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From: "Black, David" <david.black@emc.com>
To: "Karen E. Egede Nielsen" <karen.nielsen@tieto.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Date: Sun, 20 Jul 2014 22:00:28 -0400
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From nobody Sun Jul 20 21:54:44 2014
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Date: Sun, 20 Jul 2014 22:54:37 -0600
From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: "Karen E. Egede Nielsen" <karen.nielsen@tieto.com>
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Subject: Re: [tsvwg] comments to RFC3758
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Karen,

Please see comments below...

Karen E. Egede Nielsen wrote:           (Mon, 21 Jul 2014 01:03:59)
> >
> > (3)
> > >    If the decision to "abandon" a chunk is made, no matter how such a
> > >    decision is made, the appropriate congestion adjustment MUST be
> > >    made as specified in RFC 2960 if the chunk would have been marked
> > >    for retransmission later (e.g., either by T3-Timeout or by Fast
> > >    Retransmit).
> >
> [Karen]  I cannot see that 3 says this. Such an interpretation of (3)
> presumes that a message is only abandoned at a T3 or when marked for
> fast retransmission.  Furthermore 3 speaks about congestion control
> adjustments not about inflight or rwnd.
> 
> BR, Karen
> 
> The document does not presume that

If "the chunk would have been marked for retransmission later", then it
would have been considered in flight had it not been "abandoned".
Therefore, (3) effectively says that the abandoned chunk must continue
to be considered in flight for as long as it would have been had it not
been abandoned, which would be until the next T3-expiry or an ack.

--brian

-- 
Brian F. G. Bidulock
bidulock@openss7.org
http://www.openss7.org/


From nobody Sun Jul 20 22:40:22 2014
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Date: Sun, 20 Jul 2014 22:39:28 -0700
From: Joe Touch <touch@isi.edu>
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Subject: Re: [tsvwg] Toronto Tue agenda posted, send slides!!
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Hi, David (et al.),

There are no slides for the port use doc. I did not request a slot, and 
am not attending.

The next step is for me to post a summary of the issues raised during 
WGLC and resolve those issues in an updated document, but that was not 
ready in time for the draft deadline, so I had not planned on having 
anything presented.

Joe

On 7/20/2014 12:19 PM, Black, David wrote:
> The tsvwg agenda for Tuesday (both sessions) was posted a while back:
>
> 	https://datatracker.ietf.org/meeting/90/agenda/tsvwg/
>
> Those presenters who haven't already sent slides (thank you to those who
> have) should send slides ASAP, by mid-day Monday at the latest please.
>
> If the tsvwg meeting starts on Tuesday w/o slides for an agenda item,
> the chairs may impose a 3-5min time limit for a brief verbal summary + Q&A.
>
> Thanks,
> --David
> ----------------------------------------------------
> David L. Black, Distinguished Engineer
> EMC Corporation, 176 South St., Hopkinton, MA  01748
> +1 (508) 293-7953             FAX: +1 (508) 293-7786
> david.black@emc.com        Mobile: +1 (978) 394-7754
> ----------------------------------------------------
>


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Hi,

I have read this version of the draft and support its progression to =
RFC.

Best regards

Irene

Am 03.07.2014 um 10:45 schrieb Gorry Fairhurst <gorry@erg.abdn.ac.uk>:

> This is the start of the WGLC on "Additional Policies for the Partial =
Reliability Extension of the Stream Control Transmission Protocol" for =
publication intended as a PS.
>=20
> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-prpolicies/
>=20
> This WGLC will last until Monday, July 21st, 2014 (longer since this =
includes time prior to an IETF meeting).
>=20
> Could people please send *any* comments to the list if they think this =
is ready to publish?
>=20
> A good note is "I have read this version of the draft,
> and I approve (or support) its progression to RFC".
>=20
> James/Gorry/David
> TSVWG chairs
>=20

*****************************************

Dr. Irene R=FCngeler=20
Fachhochschule M=FCnster, Abt. Steinfurt=20
FB Elektrotechnik und Informatik=20
Labor f=FCr Netzwerkprogrammierung, Datenbanklabor=20
Bismarckstr. 11=20
Tel.: 02551-962561=20
Fax: 02551-962563=20
EMail: I.Ruengeler@fh-muenster.de






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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Diso-8859-1"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;">Hi,<div><br></div><div>I have read this version of =
the draft and support its progression to =
RFC.</div><div><br></div><div>Best =
regards</div><div><br></div><div>Irene</div><div><br><div><div>Am =
03.07.2014 um 10:45 schrieb Gorry Fairhurst &lt;<a =
href=3D"mailto:gorry@erg.abdn.ac.uk">gorry@erg.abdn.ac.uk</a>&gt;:</div><b=
r class=3D"Apple-interchange-newline"><blockquote type=3D"cite">This is =
the start of the WGLC on "Additional Policies for the Partial =
Reliability Extension of the Stream Control Transmission Protocol" for =
publication intended as a PS.<br><br><a =
href=3D"https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-prpolicies/=
">https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-prpolicies/</a><b=
r><br>This WGLC will last until Monday, July 21st, 2014 (longer since =
this includes time prior to an IETF meeting).<br><br>Could people please =
send *any* comments to the list if they think this is ready to =
publish?<br><br>A good note is "I have read this version of the =
draft,<br>and I approve (or support) its progression to =
RFC".<br><br>James/Gorry/David<br>TSVWG =
chairs<br><br></blockquote></div><br><div>
<div style=3D"color: rgb(0, 0, 0); font-family: Helvetica;  font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; orphans: 2; text-align: -webkit-auto; =
text-indent: 0px; text-transform: none; white-space: normal; widows: 2; =
word-spacing: 0px; -webkit-text-size-adjust: auto; =
-webkit-text-stroke-width: 0px; word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><div><p><font =
size=3D"2">*****************************************</font></p><p><font =
size=3D"2">Dr. Irene R=FCngeler</font>&nbsp;<br><font =
size=3D"2">Fachhochschule M=FCnster, Abt. =
Steinfurt&nbsp;</font><br><font size=3D"2">FB Elektrotechnik und =
Informatik&nbsp;</font><br><font size=3D"2">Labor f=FCr =
Netzwerkprogrammierung, Datenbanklabor&nbsp;</font><br><font =
size=3D"2">Bismarckstr. 11</font>&nbsp;<br><font size=3D"2">Tel.: =
02551-962561</font>&nbsp;<br><font size=3D"2">Fax: =
02551-962563</font>&nbsp;<br><font size=3D"2">EMail:&nbsp;<a =
href=3D"mailto:I.Ruengeler@fh-muenster.de">I.Ruengeler@fh-muenster.de</a><=
/font></p></div><div><br></div></div><br =
class=3D"Apple-interchange-newline"><br =
class=3D"Apple-interchange-newline">
</div>
<br></div></body></html>=

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From nobody Mon Jul 21 04:08:53 2014
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Date: Mon, 21 Jul 2014 23:08:33 +1200
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Subject: Re: [tsvwg] I-D Action: draft-ietf-tsvwg-rtcweb-qos-02.txt
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On 21/07/2014 14:00, Black, David wrote:
> Karen,
> 
>>> What's the rationale for "certainly isn't viable" regarding different drop
>>> precedences within a single SCTP session?
>> [Karen] Ok - "certainly not viable" is too strong formulation.
>> Having different drop precedences within the same SCTP CC context would,
>> given that SCTP CC presently is drop driven only,
>> results in that the whole data channel is impacted, from a CC perspective,
>> by the highest drop precedence. But I realize that whether this is
>> acceptable is a question indeed.
> 
> I think you're on to something here.  If the drop precedences only came into
> play after the decision to drop a packet had been made, their effect would
> probably be minimal, as they would only affect what to drop in a session, not
> how much to drop.  OTOH, DiffServ AF is not intended to operate in that fashion
> - the rationale for the 3 drop precedences per AF class was based on envisioning
> the output of two-threshold rate shaper:
> 	AFx1 - Within base or committed traffic profile
> 	AFx2 - Beyond base/committed, but within excess or burst traffic profile
> 	AFx3 - Completely out of profile
> 
> Hence the relative mix of those drop precedences can be expected to affect
> the overall drop rate/probability seen in a session where those precedences
> are mixed.

Yes. And it was clearly aimed at layered video, such that there was
a benefit to the end user in dropping the least important layer
during lossy periods. With anything like a TCP-like congestion
control algorithm in sight, the effects may be completely bizarre.

   Brian

> 
> Thanks,
> --David
> 
>> -----Original Message-----
>> From: Karen E. Egede Nielsen [mailto:karen.nielsen@tieto.com]
>> Sent: Sunday, July 20, 2014 9:20 PM
>> To: Black, David; tsvwg@ietf.org
>> Cc: rmcat WG; dart@ietf.org
>> Subject: RE: [tsvwg] I-D Action: draft-ietf-tsvwg-rtcweb-qos-02.txt
>>
>> Hi,
>>
>>> -----Original Message-----
>>> From: Black, David [mailto:david.black@emc.com]
>>> Sent: 21. juli 2014 02:51
>>> To: Karen E. Egede Nielsen; tsvwg@ietf.org
>>> Cc: rmcat WG; dart@ietf.org; Black, David
>>> Subject: RE: [tsvwg] I-D Action: draft-ietf-tsvwg-rtcweb-qos-02.txt
>>>
>>> Karen,
>>>
>>> <WG chair hat off>, [+dart list]
>>>
>>>> I wonder if the recommendations of this document should not relate to
>>>> the viability (or rather the opposite) of handling different RTCWEB
>>>> data channels individually from a DSCP markings perspective. Or more
>>>> generally relate to the fact that media streams within the same
>>>> congestion control context cannot be handled individually.
>>>> Right now such considerations are not in the document. If the
>>>> intention is for such consideration to come in documents from the dart
>>>> wg, then it would be very important to  have a reference to this
>>>> dependency - or ?
>>> IMHO, such a reference is necessary, although the dart WG draft will be
>>> informational.  The reference should be added once there is an official
>>> dart
>>> WG draft (likely to happen this week).
>>>
>>>> Something else:
>>>>
>>>> Right now the document stipulates that packets within the same media
>>>> stream may be marked with different DSCP codepoints, i.e., it is said:
>>>>
>>>>    One may select difference drop precedences for the
>>>>    different packets in the media flow.  Therefore, all packets in the
>>>>    stream SHOULD be marked with the same priority but can have
>>>>    difference drop precedences.
>>>>
>>>> which indeed is in compliance with the recommendations put forward in
>>>> draft-york-dart-dscp-rtp-00 in that
>>>> it ensures that the DSCP markings  do not result in re-ordering
>>>> within the media flow, but has it been ascertained, from a CC
>>>> perspective that usage of different drop precedence within the same CC
>>>> context is viable ?
>>> Good question - I will include that in my slides for the dart WG meeting.
>>>
>>>> >From the RTCWEB data channel perspective this certainly isn't viable
>>>>> (being
>>>> SCTP based),
>>>> whether it is viable from an RMCAT CC perspective is something that we
>>>> in the RMCAT wg would need to make more clear.
>>> What's the rationale for "certainly isn't viable" regarding different drop
>>> precedences within a single SCTP session?
>> [Karen] Ok - "certainly not viable" is too strong formulation.
>> Having different drop precedences within the same SCTP CC context would,
>> given that SCTP CC presently is drop driven only,
>> results in that the whole data channel is impacted, from a CC perspective,
>> by the highest drop precedence. But I realize that whether this is
>> acceptable is a question indeed.
>>
>>> Thanks,
>>> --David (as an editor of draft-york, *not* as a tsvwg WG chair)
>>>
>>>> -----Original Message-----
>>>> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Karen E.
>>>> Egede Nielsen
>>>> Sent: Sunday, July 20, 2014 4:49 PM
>>>> To: tsvwg@ietf.org
>>>> Cc: rmcat WG
>>>> Subject: Re: [tsvwg] I-D Action: draft-ietf-tsvwg-rtcweb-qos-02.txt
>>>>
>>>> Hi,
>>>>
>>>> Please accept the following comments:
>>>>
>>>> I wonder if the recommendations of this document should not relate to
>>>> the viability (or rather the opposite) of handling different RTCWEB
>>>> data channels individually from a DSCP markings perspective. Or more
>>>> generally relate to the fact that media streams within the same
>>>> congestion control context cannot be handled individually.
>>>> Right now such considerations are not in the document. If the
>>>> intention is for such consideration to come in documents from the dart
>>>> wg, then it would be very important to  have a reference to this
>>>> dependency - or ?
>>>>
>>>> Something else:
>>>>
>>>> Right now the document stipulates that packets within the same media
>>>> stream may be marked with different DSCP codepoints, i.e., it is said:
>>>>
>>>>    One may select difference drop precedences for the
>>>>    different packets in the media flow.  Therefore, all packets in the
>>>>    stream SHOULD be marked with the same priority but can have
>>>>    difference drop precedences.
>>>>
>>>> which indeed is in compliance with the recommendations put forward in
>>>> draft-york-dart-dscp-rtp-00 in that
>>>> it ensures that the DSCP markings  do not result in re-ordering
>>>> within the media flow, but has it been ascertained, from a CC
>>>> perspective that usage of different drop precedence within the same CC
>>>> context is viable ?
>>>>
>>>> >From the RTCWEB data channel perspective this certainly isn't viable
>>>>> (being
>>>> SCTP based),
>>>> whether it is viable from an RMCAT CC perspective is something that we
>>>> in the RMCAT wg would need to make more clear.
>>>>
>>>> BR, Karen
>>>>
>>>>> -----Original Message-----
>>>>> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of internet-
>>>>> drafts@ietf.org
>>>>> Sent: 23. juni 2014 21:12
>>>>> To: i-d-announce@ietf.org
>>>>> Cc: tsvwg@ietf.org
>>>>> Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-rtcweb-qos-02.txt
>>>>>
>>>>>
>>>>> 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
>>>>> Working Group of the IETF.
>>>>>
>>>>>         Title           : DSCP and other packet markings for RTCWeb
>>>>> QoS
>>>>>         Authors         : Subha Dhesikan
>>>>>                           Cullen Jennings
>>>>>                           Dan Druta
>>>>>                           Paul Jones
>>>>>                           James Polk
>>>>> 	Filename        : draft-ietf-tsvwg-rtcweb-qos-02.txt
>>>>> 	Pages           : 7
>>>>> 	Date            : 2014-06-23
>>>>>
>>>>> Abstract:
>>>>>    Many networks, such as service provider and enterprise networks,
>>>>> can
>>>>>    provide per packet treatments based on Differentiated Services Code
>>>>>    Points (DSCP) on a per-hop basis.  This document provides the
>>>>>    recommended DSCP values for browsers to use for various classes of
>>>>>    traffic.
>>>>>
>>>>>
>>>>> The IETF datatracker status page for this draft is:
>>>>> https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rtcweb-qos/
>>>>>
>>>>> There's also a htmlized version available at:
>>>>> http://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-02
>>>>>
>>>>> A diff from the previous version is available at:
>>>>> http://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-rtcweb-qos-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 Mon Jul 21 05:14:21 2014
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Subject: Re: [tsvwg] I-D Action: draft-ietf-tsvwg-rtcweb-qos-02.txt
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On 21. juli 2014, at 13:08, Brian E Carpenter =
<brian.e.carpenter@gmail.com> wrote:

> On 21/07/2014 14:00, Black, David wrote:
>> Karen,
>>=20
>>>> What's the rationale for "certainly isn't viable" regarding =
different drop
>>>> precedences within a single SCTP session?
>>> [Karen] Ok - "certainly not viable" is too strong formulation.
>>> Having different drop precedences within the same SCTP CC context =
would,
>>> given that SCTP CC presently is drop driven only,
>>> results in that the whole data channel is impacted, from a CC =
perspective,
>>> by the highest drop precedence. But I realize that whether this is
>>> acceptable is a question indeed.
>>=20
>> I think you're on to something here.  If the drop precedences only =
came into
>> play after the decision to drop a packet had been made, their effect =
would
>> probably be minimal, as they would only affect what to drop in a =
session, not
>> how much to drop.  OTOH, DiffServ AF is not intended to operate in =
that fashion
>> - the rationale for the 3 drop precedences per AF class was based on =
envisioning
>> the output of two-threshold rate shaper:
>> 	AFx1 - Within base or committed traffic profile
>> 	AFx2 - Beyond base/committed, but within excess or burst traffic =
profile
>> 	AFx3 - Completely out of profile
>>=20
>> Hence the relative mix of those drop precedences can be expected to =
affect
>> the overall drop rate/probability seen in a session where those =
precedences
>> are mixed.
>=20
> Yes. And it was clearly aimed at layered video, such that there was
> a benefit to the end user in dropping the least important layer
> during lossy periods. With anything like a TCP-like congestion
> control algorithm in sight, the effects may be completely bizarre.

Reordering could make this even worse. Could different drop precedences =
cause reordering?

Cheers,
Michael


From nobody Mon Jul 21 05:25:39 2014
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Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-sctp-failover-05.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 Working Group of the IETF.

        Title           : Quick Failover Algorithm in SCTP
        Authors         : Yoshifumi Nishida
                          Preethi Natarajan
                          Armando Caro
                          Paul D. Amer
                          Karen E. E. Nielsen
	Filename        : draft-ietf-tsvwg-sctp-failover-05.txt
	Pages           : 17
	Date            : 2014-07-21

Abstract:
   One of the major advantages of SCTP is that it supports multi-homed
   communication.  A multi-homed SCTP end-point has the ability to
   withstand network failures by migrating the traffic from an inactive
   network to an active one.  However, if the [RFC4960] specified
   failover operation is followed there can be a significant delay in
   the migration to the active destination addresses, thus severely
   reducing the effectiveness of SCTP multi-homed operation.

   The memo complements RFC4960 by the introduction of the Potentially
   Failed state and associated new Quick Failover operation to apply
   during network failure and specifies for SCTP senders to support this
   more performance optimal failover procedure as an add-on to the
   [RFC4960] failover operation.  The memo in addition complements
   [RFC4960] by introduction of alternative switchover operation modes
   for the data transfer path management after a failover.  These
   operation modes offer for more performance optimal operation in some
   network environments.  From the perspective of this memo the
   implementation of the additional switchover operation modes is
   considered optional.

   The procedures defined require only minimal modifications to the
   current specification.  The procedures are sender-side only and do
   not impact the SCTP receiver.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-failover-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-tsvwg-sctp-failover-05


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 Mon Jul 21 06:02:49 2014
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From: "Black, David" <david.black@emc.com>
To: Michael Welzl <michawe@ifi.uio.no>, Brian E Carpenter <brian.e.carpenter@gmail.com>
Date: Mon, 21 Jul 2014 09:02:17 -0400
Thread-Topic: [tsvwg] I-D Action: draft-ietf-tsvwg-rtcweb-qos-02.txt
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> >> Hence the relative mix of those drop precedences can be expected to af=
fect
> >> the overall drop rate/probability seen in a session where those preced=
ences
> >> are mixed.
> >
> > Yes. And it was clearly aimed at layered video, such that there was
> > a benefit to the end user in dropping the least important layer
> > during lossy periods. With anything like a TCP-like congestion
> > control algorithm in sight, the effects may be completely bizarre.
>=20
> Reordering could make this even worse. Could different drop precedences c=
ause
> reordering?

No, packets MUST NOT be reordered among the three drop precedences within
an AF class (e.g., AF2x).  See RFC 2597 and draft-york.

Thanks,
--David

> -----Original Message-----
> From: Michael Welzl [mailto:michawe@ifi.uio.no]
> Sent: Monday, July 21, 2014 8:14 AM
> To: Brian E Carpenter
> Cc: Black, David; Karen E. Egede Nielsen; rmcat WG; dart@ietf.org;
> tsvwg@ietf.org
> Subject: Re: [tsvwg] I-D Action: draft-ietf-tsvwg-rtcweb-qos-02.txt
>=20
>=20
> On 21. juli 2014, at 13:08, Brian E Carpenter <brian.e.carpenter@gmail.co=
m>
> wrote:
>=20
> > On 21/07/2014 14:00, Black, David wrote:
> >> Karen,
> >>
> >>>> What's the rationale for "certainly isn't viable" regarding differen=
t
> drop
> >>>> precedences within a single SCTP session?
> >>> [Karen] Ok - "certainly not viable" is too strong formulation.
> >>> Having different drop precedences within the same SCTP CC context wou=
ld,
> >>> given that SCTP CC presently is drop driven only,
> >>> results in that the whole data channel is impacted, from a CC perspec=
tive,
> >>> by the highest drop precedence. But I realize that whether this is
> >>> acceptable is a question indeed.
> >>
> >> I think you're on to something here.  If the drop precedences only cam=
e
> into
> >> play after the decision to drop a packet had been made, their effect w=
ould
> >> probably be minimal, as they would only affect what to drop in a sessi=
on,
> not
> >> how much to drop.  OTOH, DiffServ AF is not intended to operate in tha=
t
> fashion
> >> - the rationale for the 3 drop precedences per AF class was based on
> envisioning
> >> the output of two-threshold rate shaper:
> >> 	AFx1 - Within base or committed traffic profile
> >> 	AFx2 - Beyond base/committed, but within excess or burst traffic prof=
ile
> >> 	AFx3 - Completely out of profile
> >>
> >> Hence the relative mix of those drop precedences can be expected to af=
fect
> >> the overall drop rate/probability seen in a session where those preced=
ences
> >> are mixed.
> >
> > Yes. And it was clearly aimed at layered video, such that there was
> > a benefit to the end user in dropping the least important layer
> > during lossy periods. With anything like a TCP-like congestion
> > control algorithm in sight, the effects may be completely bizarre.
>=20
> Reordering could make this even worse. Could different drop precedences c=
ause
> reordering?
>=20
> Cheers,
> Michael


From nobody Mon Jul 21 08:04:06 2014
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From: Jamal Hadi Salim <hadi@mojatatu.com>
Date: Sat, 19 Jul 2014 15:06:32 -0400
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Subject: Re: [tsvwg] WGLC for draft-ietf-tsvwg-sctp-prpolicies-03
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I have read the draft and happy with the outcome. I believe it is
ready for publication.

cheers,
jamal


From nobody Tue Jul 22 05:56:54 2014
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Subject: [tsvwg] Progressing foo-over-UDP
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In London I took an action to write up suggested text for how to handle the
concerns of congestion and checksum in UDP-over-foo.

I think we had some agreement of what was needed (look at the minutes of TSVWG -
or was it TSV Area? - from London).

Of course, I have let you all down. I have about 75% of the necessary text for
MPLS-over-UDP (as a concrete example), but I am not complete. You don't need to
know my excuses.

Attached below (so you can laugh at me) is the text I have so far. it is work in
progress, but since I am currently getting in the way, some of you might want to
take the pen (if no-one wants to, I will try to continue and complete this).

Cheers,
Adrian

==================

x.  Applicability and Deployment of MPLS-in-UDP

   It is helpful to categorise the type of networks in which this
   solution might be deployed.  That is, it is useful to look at the
   type of network over which the UDP tunnel might run, and the type of
   network that may be running MPLS.  An understanding of these 
   categories will lead to the realization that deploying MPLS-in-UDP in
   some environments is entirely safe, in other scenarios might cause 
   issues to the networks involved but is a local choice for the 
   operators of those networks, while in other situations the deployment
   might be significantly harmful to other services or traffic outside
   the responsibility of the operator deploying MPLS-in-UDP.

   In broad terms these deployment categories can be listed as follows:

   - Tunneling across the Internet
   - Tunneling over a provider network in a customer/provider
     relationship
   - Tunneling in a client/server relationship owned by a single
     operator
   - Tunneling within a "flat" network

   The remainder of this section describes the deployment of MPLS-in-UDP
   in these networks, and gives warnings, advice, and caveats
   appropriate to those scenarios with special reference to congestion
   avoidance and the use of checksums.  The use made of MPLS-in-UDP by
   applications must be cognisant of the underlying network deployment.
   However, it needs to be understood that an end-point application is
   generally unaware of the services or connectivity over which it is
   sending data.  That is, an end-point application is sending IP data
   possibly using TCP or UDP, and that data is encapsulated in MPLS
   somewhere within the network, and then those MPLS packets are
   themselves encapsulated in UDP as described in this document.  That
   means that the decision points for the concerns and choices described
   in the sections that follow are at the point of encapsulation of MPLS
   in UDP, and at the time of deployment of the encapsulation function.
   While there are possibilities to apply back-pressure to applications,
   when there is tunneling within tunnels and a significant amount of
   multiplexing of flows within flows (as is the case with MPLS-in-UDP)
   these techniques are hard to apply and may have less immediate
   effect.

   Implementers are advised to note that widely deployable
   implementations have to provide the function set that is needed in
   each of the deployment scenarios discussed below.  That is, those
   implementations that only provide a subset of the functionality are
   not suitable for deployment in the full set of scenarios.

   The text in these sections is not intended to be, and should not be
   taken as, broader guidance for the issues of tunneling different
   protocols over UDP.

x.1.  Congestion Considerations

   End-to-end traffic flows may be multiplexed into MPLS LSPs, and those
   LSPs may be multiplexed into a single UDP tunnel.  These UDP tunnels
   appear as single flows in the core IP network over which the tunnel
   flows.  Thus, any congestion issues related to the core network and 
   caused by end-to-end traffic flows may be hard to report back to the
   head ends of the flows causing the congestion.

   The impact of congestion must be considered both in terms of the
   effect on the rest of the network of a UDP tunnel that is consuming
   excessive capacity, and in terms of the effect on the flows using the
   UDP tunnels in particular if packets are queued (causing delay or 
   jitter) or dropped.

   Various mechanisms exist to propagate congestion information to the
   source of flows either when congestion has occurred or when it is
   perceived to be imminent.  It should be noted that in the case of a
   tunnel (such as a UDP tunnel), the source of a flow is the head end
   of the tunnel.  That is, any congestion notification from the core
   network will be propagated to the head end of the tunnel, not to the
   original source of the traffic.  

   The head end of a UDP tunnel can take a number of actions when it is
   informed about congestion.  No comment is made here about which
   action is best or even appropriate, but section x.3 discusses these
   actions in the context of the different deployment scenarios.

   - Do nothing assuming that the burden of congestion will be carried 
     by other flows, or that the fact of congestion on the UDP tunnel
     will cause back-pressure in the payload flows.

   - Pass on the congestion information to the payload flows.  However,
     in this instance the payload of the UDP tunnel is MPLS flows, and
     the mechanisms for congestion notification in MPLS flows are
     lacking.

   - Throttle the data being placed onto the UDP tunnel by queuing or
     dropping packets from the payload flows.

   - Change the flows that are in a tunnel by putting some of them in a
     different tunnel.  This may have no effect on the congestion, but
     in some networks with ECMP may result in an improvement.

   - Completely fail one or more flows carried in the tunnel, or even
     fail the entire tunnel.  This mechanism is sometimes known as
     "circuit breaker".

x.2.  UDP Checksum Considerations

is checksum an issue
what can be done about it

- use it or not

x.3.  Deployment Scenarios

x.3.1.  MPLS-in-UDP over the Public Internet
 

   - Tunneling across the Internet
   The use of MPLS-in-UDP over the public Internet is NOT RECOMMENDED.

x.3.2. MPLS-in-UDP in Managed Networks

   - Tunneling over a provider network in a customer/provider
     relationship
   - Tunneling in a client/server relationship owned by a single
     operator

x.3.3. MPLS-in-UDP in Private Networks

   - Tunneling in a client/server relationship owned by a single
     operator
   - Tunneling within a "flat" network


From nobody Tue Jul 22 07:18:26 2014
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From: "Fred Baker (fred)" <fred@cisco.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: Text from RFC 3246 regarding excess traffic
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   The EF PHB is intended to be a building block for low loss services.
   However, under sufficiently high load of EF traffic (including
   unexpectedly large bursts from many inputs at once), any device with
   finite buffers may need to discard packets.  Thus, it must be
   possible to establish whether a device conforms to the EF definition
   even when some packets are lost.  This is done by performing an
   "off-line" test of conformance to equations 1 through 4.  After
   observing a sequence of packets entering and leaving the node, the
   packets which did not leave are assumed lost and are notionally
   removed from the input stream.  The remaining packets now constitute
   the arrival stream (the a_j's) and the packets which left the node
   constitute the departure stream (the d_j's).  Conformance to the
   equations can thus be verified by considering only those packets that
   successfully passed through the node.

   If the EF PHB is implemented by a mechanism that allows unlimited
   preemption of other traffic (e.g., a priority queue), the
   implementation MUST include some means to limit the damage EF traffic
   could inflict on other traffic (e.g., a token bucket rate limiter).
   Traffic that exceeds this limit MUST be discarded.  This maximum EF
   rate, and burst size if appropriate, MUST be settable by a network
   administrator (using whatever mechanism the node supports for non-
   volatile configuration).

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From nobody Tue Jul 22 07:21:38 2014
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Subject: [tsvwg] Name of the new DATA chunk
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Dear all,

it was suggested to use I-DATA chunk for Interleaving DATA chunk as a better
name for the chunk defined in
http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-ndata-01

Any opinons?

Best regards
Michael


From nobody Tue Jul 22 08:10:36 2014
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On 07/22/2014 08:56 AM, Adrian Farrel wrote:
> In London I took an action to write up suggested text for how to handle the
> concerns of congestion and checksum in UDP-over-foo.
>
> I think we had some agreement of what was needed (look at the minutes of TSVWG -
> or was it TSV Area? - from London).
>
> Of course, I have let you all down. I have about 75% of the necessary text for
> MPLS-over-UDP (as a concrete example), but I am not complete. You don't need to
> know my excuses.
>
> Attached below (so you can laugh at me) is the text I have so far. it is work in
> progress, but since I am currently getting in the way, some of you might want to
> take the pen (if no-one wants to, I will try to continue and complete this).

Adrian, thank you for sending this.

It's important (as we gather what seems to be an inexhaustible supply of 
foos to encapsulate with UDP), and David and Gorry were able to point to 
your e-mail during TSVWG this morning.

Spencer

> Cheers,
> Adrian
>
> ==================
>
> x.  Applicability and Deployment of MPLS-in-UDP
>
>     It is helpful to categorise the type of networks in which this
>     solution might be deployed.  That is, it is useful to look at the
>     type of network over which the UDP tunnel might run, and the type of
>     network that may be running MPLS.  An understanding of these
>     categories will lead to the realization that deploying MPLS-in-UDP in
>     some environments is entirely safe, in other scenarios might cause
>     issues to the networks involved but is a local choice for the
>     operators of those networks, while in other situations the deployment
>     might be significantly harmful to other services or traffic outside
>     the responsibility of the operator deploying MPLS-in-UDP.
>
>     In broad terms these deployment categories can be listed as follows:
>
>     - Tunneling across the Internet
>     - Tunneling over a provider network in a customer/provider
>       relationship
>     - Tunneling in a client/server relationship owned by a single
>       operator
>     - Tunneling within a "flat" network
>
>     The remainder of this section describes the deployment of MPLS-in-UDP
>     in these networks, and gives warnings, advice, and caveats
>     appropriate to those scenarios with special reference to congestion
>     avoidance and the use of checksums.  The use made of MPLS-in-UDP by
>     applications must be cognisant of the underlying network deployment.
>     However, it needs to be understood that an end-point application is
>     generally unaware of the services or connectivity over which it is
>     sending data.  That is, an end-point application is sending IP data
>     possibly using TCP or UDP, and that data is encapsulated in MPLS
>     somewhere within the network, and then those MPLS packets are
>     themselves encapsulated in UDP as described in this document.  That
>     means that the decision points for the concerns and choices described
>     in the sections that follow are at the point of encapsulation of MPLS
>     in UDP, and at the time of deployment of the encapsulation function.
>     While there are possibilities to apply back-pressure to applications,
>     when there is tunneling within tunnels and a significant amount of
>     multiplexing of flows within flows (as is the case with MPLS-in-UDP)
>     these techniques are hard to apply and may have less immediate
>     effect.
>
>     Implementers are advised to note that widely deployable
>     implementations have to provide the function set that is needed in
>     each of the deployment scenarios discussed below.  That is, those
>     implementations that only provide a subset of the functionality are
>     not suitable for deployment in the full set of scenarios.
>
>     The text in these sections is not intended to be, and should not be
>     taken as, broader guidance for the issues of tunneling different
>     protocols over UDP.
>
> x.1.  Congestion Considerations
>
>     End-to-end traffic flows may be multiplexed into MPLS LSPs, and those
>     LSPs may be multiplexed into a single UDP tunnel.  These UDP tunnels
>     appear as single flows in the core IP network over which the tunnel
>     flows.  Thus, any congestion issues related to the core network and
>     caused by end-to-end traffic flows may be hard to report back to the
>     head ends of the flows causing the congestion.
>
>     The impact of congestion must be considered both in terms of the
>     effect on the rest of the network of a UDP tunnel that is consuming
>     excessive capacity, and in terms of the effect on the flows using the
>     UDP tunnels in particular if packets are queued (causing delay or
>     jitter) or dropped.
>
>     Various mechanisms exist to propagate congestion information to the
>     source of flows either when congestion has occurred or when it is
>     perceived to be imminent.  It should be noted that in the case of a
>     tunnel (such as a UDP tunnel), the source of a flow is the head end
>     of the tunnel.  That is, any congestion notification from the core
>     network will be propagated to the head end of the tunnel, not to the
>     original source of the traffic.
>
>     The head end of a UDP tunnel can take a number of actions when it is
>     informed about congestion.  No comment is made here about which
>     action is best or even appropriate, but section x.3 discusses these
>     actions in the context of the different deployment scenarios.
>
>     - Do nothing assuming that the burden of congestion will be carried
>       by other flows, or that the fact of congestion on the UDP tunnel
>       will cause back-pressure in the payload flows.
>
>     - Pass on the congestion information to the payload flows.  However,
>       in this instance the payload of the UDP tunnel is MPLS flows, and
>       the mechanisms for congestion notification in MPLS flows are
>       lacking.
>
>     - Throttle the data being placed onto the UDP tunnel by queuing or
>       dropping packets from the payload flows.
>
>     - Change the flows that are in a tunnel by putting some of them in a
>       different tunnel.  This may have no effect on the congestion, but
>       in some networks with ECMP may result in an improvement.
>
>     - Completely fail one or more flows carried in the tunnel, or even
>       fail the entire tunnel.  This mechanism is sometimes known as
>       "circuit breaker".
>
> x.2.  UDP Checksum Considerations
>
> is checksum an issue
> what can be done about it
>
> - use it or not
>
> x.3.  Deployment Scenarios
>
> x.3.1.  MPLS-in-UDP over the Public Internet
>   
>
>     - Tunneling across the Internet
>     The use of MPLS-in-UDP over the public Internet is NOT RECOMMENDED.
>
> x.3.2. MPLS-in-UDP in Managed Networks
>
>     - Tunneling over a provider network in a customer/provider
>       relationship
>     - Tunneling in a client/server relationship owned by a single
>       operator
>
> x.3.3. MPLS-in-UDP in Private Networks
>
>     - Tunneling in a client/server relationship owned by a single
>       operator
>     - Tunneling within a "flat" network
>


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Please review the draft notes and send any comments/corrections, they are
available at:
http://www.ietf.org/proceedings/90/minutes/minutes-90-tsvwg

Gorry


