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Date: Mon, 3 Dec 2012 09:02:49 -0000
From: gorry@erg.abdn.ac.uk
To: "Yuchung Cheng" <ycheng@google.com>
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Cc: tcpm <tcpm@ietf.org>
Subject: Re: [tcpm] Adopting draft-fairhurst-tcpm-newcwv
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I suggested "PS" - largely because I did not see a need for a specific
experiment if the WG successfully finished the draft (and know it can be
implemented).
Here's my thinking:

Reasons why I think it could be PS:

 - the 5 minute period is simply a judgment call - we can decide to change
this if the WG wishes, but I suggest it always will be just a constant we
choose.

- CWV was itself EXP. The present authors suggest the experiment has now
concluded - it was safe (implemented in Linux), but was not that useful -
because many apps do not use it by default. In doing this experiment we
understand more about CWV and this new-cwv follows that work.

- There has (alas) been a process flaw - the IET-specified method has not
been deployed - instead we have a mixture of TCP behaviours some of which
do no decay of cwnd at all, and since then, the default behaviour for many
apps does not follow any RFC (STD or EXP). In this respect, if the IETF
wishes to change that now deployed base, then I recommend PS.

Reasons why EXP may be best:

- We have no implementation yet of the new method (we're working on this
and would love others to join us or to do this work themselves).

- pipeACK is new (true - implementation is needed at least for PS, but
this is planned).

- We may wish to update STD TCP's tail-cwnd behaviour during fast recovery
(see draft), but we don't know whether this should be part of the draft
yet.

- There may be other new issues emerge: e.g. would we need pacing and know
what to standardise? (see draft).


To me, I just want to get it deployed, and  recognise that the final
status will only be determined at the end of the standards process. I'm
happy to add/update whatever the draft with  text the WG suggests at the
start of this draft - e.g. explaining why the draft needs to be EXP or why
the standards status could change from PS to EXP as the work progresses.
Other thoughts?

Gorry


> On Thu, Nov 29, 2012 at 8:11 PM, Pasi Sarolahti <pasi.sarolahti@iki.fi>
> wrote:
>> Hi,
>>
>> The other adoption call we did in the Atlanta meeting was about
>> draft-fairhurst-tcpm-newcwv-05. In the room there was unanimous support
>> for adopting this, but we did not discuss much about the intended status
>> of the document. I understood the authors' current preference is to go
>> for Proposed Standard, and this is what the draft header suggests. Any
>> thoughts about this? Does anyone think Experimental would be more
>> appropriate at this point (and on what basis)?
>>
> According to RFC2026 PS:
> "A Proposed Standard specification is generally stable, has resolved
>    known design choices, is believed to be well-understood, has received
>    significant community review, and appears to enjoy enough community
>    interest to be considered valuable.  However, further experience
>    might result in a change or even retraction of the specification
>    before it advances.
>
>    Usually, neither implementation nor operational experience is
>    required for the designation of a specification as a Proposed
>    Standard.  However, such experience is highly desirable, and will
>    usually represent a strong argument in favor of a Proposed Standard
>    designation."
>
> Has the idea "pipeAck" and 5min idle-time been implemented and
> evaluated in real systems/networks?
>
>
>> - Pasi
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>



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From: Yuchung Cheng <ycheng@google.com>
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Cc: tcpm <tcpm@ietf.org>
Subject: Re: [tcpm] Adopting draft-fairhurst-tcpm-newcwv
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On Mon, Dec 3, 2012 at 1:02 AM,  <gorry@erg.abdn.ac.uk> wrote:
>
> I suggested "PS" - largely because I did not see a need for a specific
> experiment if the WG successfully finished the draft (and know it can be
> implemented).
> Here's my thinking:
>
> Reasons why I think it could be PS:
>
>  - the 5 minute period is simply a judgment call - we can decide to change
> this if the WG wishes, but I suggest it always will be just a constant we
> choose.
>
> - CWV was itself EXP. The present authors suggest the experiment has now
> concluded - it was safe (implemented in Linux), but was not that useful -
> because many apps do not use it by default. In doing this experiment we
> understand more about CWV and this new-cwv follows that work.
>
> - There has (alas) been a process flaw - the IET-specified method has not
> been deployed - instead we have a mixture of TCP behaviours some of which
> do no decay of cwnd at all, and since then, the default behaviour for many
> apps does not follow any RFC (STD or EXP). In this respect, if the IETF
> wishes to change that now deployed base, then I recommend PS.
>
> Reasons why EXP may be best:
>
> - We have no implementation yet of the new method (we're working on this
> and would love others to join us or to do this work themselves).
>
> - pipeACK is new (true - implementation is needed at least for PS, but
> this is planned).
>
> - We may wish to update STD TCP's tail-cwnd behaviour during fast recovery
> (see draft), but we don't know whether this should be part of the draft
> yet.
I see the draft a good starting point to challenge some fundamental flaws of
(standard) TCP. Currently people think of TCP as bulk bit-stream and saw-tooth
behavior, but it's simply not true for the Web. Most HTTP (or SPDY) are short
bursts, including videos served with progressive download. Even video chunks
are most likely being throttled by the applications.

I would suggest forgetting about updating RFC 2861. Let's solve the bigger
problem: it's common TCP does not have enough data to fill cwnd when
receiving ACKs partly due to the overdose of persistent connections.

I'd like to develop the pipe-ACK idea to solve this problem. Neal Cardwell
and I are actually experimenting a similar idea and gathering data. But
can't promise I have something to show before the next meeting

IMO, the pipe-ACK idea, if works, should just update RFC5681 as its an
important design for modern traffic. That's why I would like to see.

Thanks.

>
> - There may be other new issues emerge: e.g. would we need pacing and know
> what to standardise? (see draft).





>
>
> To me, I just want to get it deployed, and  recognise that the final
> status will only be determined at the end of the standards process. I'm
> happy to add/update whatever the draft with  text the WG suggests at the
> start of this draft - e.g. explaining why the draft needs to be EXP or why
> the standards status could change from PS to EXP as the work progresses.
> Other thoughts?




>
> Gorry
>
>
>> On Thu, Nov 29, 2012 at 8:11 PM, Pasi Sarolahti <pasi.sarolahti@iki.fi>
>> wrote:
>>> Hi,
>>>
>>> The other adoption call we did in the Atlanta meeting was about
>>> draft-fairhurst-tcpm-newcwv-05. In the room there was unanimous support
>>> for adopting this, but we did not discuss much about the intended status
>>> of the document. I understood the authors' current preference is to go
>>> for Proposed Standard, and this is what the draft header suggests. Any
>>> thoughts about this? Does anyone think Experimental would be more
>>> appropriate at this point (and on what basis)?
>>>
>> According to RFC2026 PS:
>> "A Proposed Standard specification is generally stable, has resolved
>>    known design choices, is believed to be well-understood, has received
>>    significant community review, and appears to enjoy enough community
>>    interest to be considered valuable.  However, further experience
>>    might result in a change or even retraction of the specification
>>    before it advances.
>>
>>    Usually, neither implementation nor operational experience is
>>    required for the designation of a specification as a Proposed
>>    Standard.  However, such experience is highly desirable, and will
>>    usually represent a strong argument in favor of a Proposed Standard
>>    designation."
>>
>> Has the idea "pipeAck" and 5min idle-time been implemented and
>> evaluated in real systems/networks?
>>
>>
>>> - Pasi
>>>
>>> _______________________________________________
>>> tcpm mailing list
>>> tcpm@ietf.org
>>> https://www.ietf.org/mailman/listinfo/tcpm
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>
>
>

From magnus.westerlund@ericsson.com  Thu Dec  6 02:01:22 2012
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Subject: Re: [tcpm] Last Call: <draft-ietf-tcpm-initcwnd-06.txt> (Increasing TCP's Initial Window) to Experimental RFC
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Hi,

I have reviewed draft-ietf-tcpm-initcwnd-06 and have some questions and
comments.

1) First of all the experiments done appear to cover the impact on other
applications, like measuring the RTT variations when using IW10 compared
to IW3. If one is interested in the impact this proposal have on
real-time traffic flows over a shared bottle-neck the variations in
queue time at the bottleneck combined with that flows seen loss rates
are the most important factors. As a sudden delay spike of sufficient
magnitude will most likely result in a real-time media packet being
late, i.e. late loss rather than being lost in the network there might
be significant impact on such traffic from these IW10 packet bursts.
Have I missed any material discussing this aspect?

All the latency figures in the parts I was looking at was discussing
cases when the object transfer takes longer time. But it appear obvious
that even with a 100 ms increased one-way transfer time for a packet is
the end of the initial window the total transfer goes faster compared to
the delay caused by growing the window.

2) If I assume that most users are behind buffer-bloated links the
introduction of IW10 on wide scale will additionally disrupt interactive
applications and cause increased delay and possible late loss depending
on application. Especially in combination with domain sharding. For
example a Swedish newspaper's website front page results in that more
than 40 TCP connections are opened in a current browser. If these all
was using IW10, the amount of packets being sent initially will grow
roughly from 40*3 = 120 to 40*10 = 400. There are some distribution over
time but still this likely results in a larger delay spike.

I don't quite know how I should consider this case. One stand point is
that the interactive application is anyway buggered and IW10 effects are
not making it significantly worse. The only remedy is queue separation
so that what ever happens with the web-downloads doesn't affect the
interactive traffic. Another is that it will make the existing situation
worse and we should try to avoid making it worse.

I think I am more in the first camp, but still the second one is
worrying to me. But I dare to guess that the performance gains might be
worth the issues, but without real progress on mitigation of the buffer
bloat issues for interactive real-time traffic this will add
significantly to the issues real-time has to face.


3) The documents talks in quite loose terms about what can be done to
avoid to continue cause issues for destinations which has issues with
IW10. However I think it is a bit unspecific here. I can see that a
content deliverer can have lists for destination address blocks that
they see issues with which configures the sending server side with a
lower IW value. It also talks about the clients can configure to keep
the window low initially to prevent an IW10 sender to clobber ones link
if that is known. My question here is if the recommendation for auto
detection and configuration can be made more explicit. Fortunately the
issues with misconfiguration in this cases is not that serious. But
otherwise there is commonly need to be quite careful with such auto
configs that affects the behavior.

4) I am also worried that the document is a bit to positive in regards
to that IW10 will reduce the domain sharding practice. I think the only
thing that can do is likely a new HTTP transport strata that shows that
significantly improved performance for multiple objects from the same
domain over fewer transport flows. Maybe SPDY + IW10 can provide such
incentive, but I don't think IW10 alone will do much.

Cheers

Magnus

On 2012-11-26 22:29, The IESG wrote:
> 
> The IESG has received a request from the TCP Maintenance and Minor
> Extensions WG (tcpm) to consider the following document:
> - 'Increasing TCP's Initial Window'
>   <draft-ietf-tcpm-initcwnd-06.txt> as Experimental RFC
> 
> The IESG plans to make a decision in the next few weeks, and solicits
> final comments on this action. Please send substantive comments to the
> ietf@ietf.org mailing lists by 2012-12-10. Exceptionally, comments may be
> sent to iesg@ietf.org instead. In either case, please retain the
> beginning of the Subject line to allow automated sorting.
> 
> Abstract
> 
> 
>    This document proposes an experiment to increase the permitted TCP
>    initial window (IW) from between 2 and 4 segments, as specified in
>    RFC 3390, to 10 segments, with a fallback to the existing
>    recommendation when performance issues are detected. It discusses the
>    motivation behind the increase, the advantages and disadvantages of
>    the higher initial window, and presents results from several large
>    scale experiments showing that the higher initial window improves the
>    overall performance of many web services without resulting in a
>    congestion collapse. The document closes with a discussion of usage
>    and deployment for further experimental purpose recommended by the
>    IETF TCP Maintenance and Minor Extensions (TCPM) working group.
> 
> 
> 
> 
> The file can be obtained via
> http://datatracker.ietf.org/doc/draft-ietf-tcpm-initcwnd/
> 
> IESG discussion can be tracked via
> http://datatracker.ietf.org/doc/draft-ietf-tcpm-initcwnd/ballot/
> 
> 
> No IPR declarations have been submitted directly on this I-D.
> 
> 
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From pasi.sarolahti@iki.fi  Fri Dec  7 06:01:18 2012
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Subject: Re: [tcpm] Adopting draft-fairhurst-tcpm-newcwv
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On Dec 3, 2012, at 11:02 AM, <gorry@erg.abdn.ac.uk> =
<gorry@erg.abdn.ac.uk> wrote:

>=20
> I suggested "PS" - largely because I did not see a need for a specific
> experiment if the WG successfully finished the draft (and know it can =
be
> implemented).

RFC 2026 does not require that Experimental RFC defines any specific =
experiment.=20

> Here's my thinking:

Thanks, useful reasoning about both cases.

[...]

> To me, I just want to get it deployed, and  recognise that the final
> status will only be determined at the end of the standards process. =
I'm
> happy to add/update whatever the draft with  text the WG suggests at =
the
> start of this draft - e.g. explaining why the draft needs to be EXP or =
why
> the standards status could change from PS to EXP as the work =
progresses.
> Other thoughts?

It would be best if we could decide on the status early, because it may =
affect how people review the document. If the WG is not confident about =
going for PS (for example, if people feel the design options have not =
been sufficiently investigated), then EXP should be the default option, =
I think. On the other hand, if we had to change the status at a late =
phase, I'd be much more comfortable at downgrading from PS to EXP than =
promoting a document to PS after people have thought they have been =
looking at an experimental document.

I think it is a good idea to add some text discussing the status, for =
example based on what you had in the Email. This could be something to =
be removed from the final version, but might support us in discussing =
this issue further, for example in the next meeting.

- Pasi


From ananth@nttmcl.com  Tue Dec 11 10:13:05 2012
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From: Anantha Ramaiah <ananth@nttmcl.com>
To: Pasi Sarolahti <pasi.sarolahti@iki.fi>
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Subject: Re: [tcpm] Adopting RTO Restart
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--f46d0435bf4effb51c04d097a1d2
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I had promised to send this email to the list as requested by the authors
earlier, but better late than never.

 would like to add something more, FWIW :-

I know that Cisco IOS  had the original scheme described in -00- version of
this draft for a long time, i.e,  RTO gets adjusted as described in the
draft without taking into account the retransmit queue or the flightsize.
 (I do not know the current situation now, but I suspect nothing would have
changed in this regard)

I support adoption this draft as a WG item and eventual publication of this.

thanks,
-Anantha

On Mon, Nov 12, 2012 at 6:01 AM, Pasi Sarolahti <pasi.sarolahti@iki.fi>wrote:

> Hi,
>
> In the TCPM meeting last week there was a clear consensus for adopting RTO
> restart (draft-hurtig-tcpm-rtorestart-03) as a working group item, targeted
> for Experimental RFC. Here is a proposal for a new milestone item:
>
> Aug 2013  Submit document on restarting the RTO timer to the IESG for
> publication as an Experimental RFC
>
> Any comments or objections about the milestone?
>
> We discussed the issue of Std Track vs. Experimental a bit during the
> meeting, but did not have a clear consensus call on that. My reading of the
> discussion is that there would be room for experimentation to understand
> the effect that the proposed change might have on the likelihood of
> spurious RTOs. Therefore the proposed milestone currently is for
> Experimental. As discussed in the meeting, if during the progress it turns
> out that more experimentation is not necessary, we may, by WG consensus,
> change the intended status accordingly. Is this a fair plan for people?
>
> - Pasi
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

--f46d0435bf4effb51c04d097a1d2
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I had promised to send this email to the list as requested by the authors e=
arlier, but better late than never.=A0<div><br></div><div>=A0would like to =
add something more, FWIW :-</div>


<div><br></div><div>I know that Cisco IOS =A0had the original scheme descri=
bed in -00- version of this draft for a long time, i.e, =A0RTO gets adjuste=
d as described in the draft without taking into account the retransmit queu=
e or the flightsize. =A0(I do not know the current situation now, but I sus=
pect nothing would have changed in this regard)</div>

<div><br></div><div>I support adoption this draft as a WG item and eventual=
 publication of this.</div><div><br></div><div>thanks,</div><div>-Anantha<b=
r>

<br><div class=3D"gmail_quote">On Mon, Nov 12, 2012 at 6:01 AM, Pasi Sarola=
hti <span dir=3D"ltr">&lt;<a href=3D"mailto:pasi.sarolahti@iki.fi" target=
=3D"_blank">pasi.sarolahti@iki.fi</a>&gt;</span> wrote:<br><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">



Hi,<br>
<br>
In the TCPM meeting last week there was a clear consensus for adopting RTO =
restart (draft-hurtig-tcpm-rtorestart-03) as a working group item, targeted=
 for Experimental RFC. Here is a proposal for a new milestone item:<br>




<br>
Aug 2013 =A0Submit document on restarting the RTO timer to the IESG for pub=
lication as an Experimental RFC<br>
<br>
Any comments or objections about the milestone?<br>
<br>
We discussed the issue of Std Track vs. Experimental a bit during the meeti=
ng, but did not have a clear consensus call on that. My reading of the disc=
ussion is that there would be room for experimentation to understand the ef=
fect that the proposed change might have on the likelihood of spurious RTOs=
. Therefore the proposed milestone currently is for Experimental. As discus=
sed in the meeting, if during the progress it turns out that more experimen=
tation is not necessary, we may, by WG consensus, change the intended statu=
s accordingly. Is this a fair plan for people?<br>




<br>
- Pasi<br>
<br>
_______________________________________________<br>
tcpm mailing list<br>
<a href=3D"mailto:tcpm@ietf.org" target=3D"_blank">tcpm@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/tcpm" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/tcpm</a><br>
</blockquote></div><br></div>

--f46d0435bf4effb51c04d097a1d2--

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Subject: Re: [tcpm] Adopting RTO Restart
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On Tue, Dec 11, 2012 at 10:13 AM, Anantha Ramaiah <ananth@nttmcl.com> wrote:
> I had promised to send this email to the list as requested by the authors
> earlier, but better late than never.
>
>  would like to add something more, FWIW :-
>
> I know that Cisco IOS  had the original scheme described in -00- version of
> this draft for a long time, i.e,  RTO gets adjusted as described in the
> draft without taking into account the retransmit queue or the flightsize.
> (I do not know the current situation now, but I suspect nothing would have
> changed in this regard)
>
> I support adoption this draft as a WG item and eventual publication of this.
+1

I also suggest removing the constraint of small cwnd. just do it always.

>
> thanks,
> -Anantha
>
> On Mon, Nov 12, 2012 at 6:01 AM, Pasi Sarolahti <pasi.sarolahti@iki.fi>
> wrote:
>>
>> Hi,
>>
>> In the TCPM meeting last week there was a clear consensus for adopting RTO
>> restart (draft-hurtig-tcpm-rtorestart-03) as a working group item, targeted
>> for Experimental RFC. Here is a proposal for a new milestone item:
>>
>> Aug 2013  Submit document on restarting the RTO timer to the IESG for
>> publication as an Experimental RFC
>>
>> Any comments or objections about the milestone?
>>
>> We discussed the issue of Std Track vs. Experimental a bit during the
>> meeting, but did not have a clear consensus call on that. My reading of the
>> discussion is that there would be room for experimentation to understand the
>> effect that the proposed change might have on the likelihood of spurious
>> RTOs. Therefore the proposed milestone currently is for Experimental. As
>> discussed in the meeting, if during the progress it turns out that more
>> experimentation is not necessary, we may, by WG consensus, change the
>> intended status accordingly. Is this a fair plan for people?
>>
>> - Pasi
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From nishida@sfc.wide.ad.jp  Sat Dec 15 19:16:41 2012
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References: <20121126212940.20378.58186.idtracker@ietfa.amsl.com> <50C06CE6.40909@ericsson.com>
Date: Sat, 15 Dec 2012 19:16:35 -0800
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From: Yoshifumi Nishida <nishida@sfc.wide.ad.jp>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Cc: tcpm@ietf.org
Subject: Re: [tcpm] Last Call: <draft-ietf-tcpm-initcwnd-06.txt> (Increasing TCP's Initial Window) to Experimental RFC
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Hi Magnus,

Thanks for the comments.
I think your points are valid. But, it's not very clear to me what you
would like to suggest authors.
IW10 might have an impact on real-time traffic. But, difficult point is we
are not very sure how much it is over the whole Internet. Do we ask authors
to do experiments to address the concerns you pointed out?
There will be situations where IW10 has negative impacts as well as
positive impacts.
But, I'm feeling it might be difficult to discuss this point without large
scale experiments.

Thanks,
--
Yoshifumi

On Thu, Dec 6, 2012 at 2:01 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> Hi,
>
> I have reviewed draft-ietf-tcpm-initcwnd-06 and have some questions and
> comments.
>
> 1) First of all the experiments done appear to cover the impact on other
> applications, like measuring the RTT variations when using IW10 compared
> to IW3. If one is interested in the impact this proposal have on
> real-time traffic flows over a shared bottle-neck the variations in
> queue time at the bottleneck combined with that flows seen loss rates
> are the most important factors. As a sudden delay spike of sufficient
> magnitude will most likely result in a real-time media packet being
> late, i.e. late loss rather than being lost in the network there might
> be significant impact on such traffic from these IW10 packet bursts.
> Have I missed any material discussing this aspect?
>
> All the latency figures in the parts I was looking at was discussing
> cases when the object transfer takes longer time. But it appear obvious
> that even with a 100 ms increased one-way transfer time for a packet is
> the end of the initial window the total transfer goes faster compared to
> the delay caused by growing the window.
>
> 2) If I assume that most users are behind buffer-bloated links the
> introduction of IW10 on wide scale will additionally disrupt interactive
> applications and cause increased delay and possible late loss depending
> on application. Especially in combination with domain sharding. For
> example a Swedish newspaper's website front page results in that more
> than 40 TCP connections are opened in a current browser. If these all
> was using IW10, the amount of packets being sent initially will grow
> roughly from 40*3 =3D 120 to 40*10 =3D 400. There are some distribution o=
ver
> time but still this likely results in a larger delay spike.
>
> I don't quite know how I should consider this case. One stand point is
> that the interactive application is anyway buggered and IW10 effects are
> not making it significantly worse. The only remedy is queue separation
> so that what ever happens with the web-downloads doesn't affect the
> interactive traffic. Another is that it will make the existing situation
> worse and we should try to avoid making it worse.
>
> I think I am more in the first camp, but still the second one is
> worrying to me. But I dare to guess that the performance gains might be
> worth the issues, but without real progress on mitigation of the buffer
> bloat issues for interactive real-time traffic this will add
> significantly to the issues real-time has to face.
>
>
> 3) The documents talks in quite loose terms about what can be done to
> avoid to continue cause issues for destinations which has issues with
> IW10. However I think it is a bit unspecific here. I can see that a
> content deliverer can have lists for destination address blocks that
> they see issues with which configures the sending server side with a
> lower IW value. It also talks about the clients can configure to keep
> the window low initially to prevent an IW10 sender to clobber ones link
> if that is known. My question here is if the recommendation for auto
> detection and configuration can be made more explicit. Fortunately the
> issues with misconfiguration in this cases is not that serious. But
> otherwise there is commonly need to be quite careful with such auto
> configs that affects the behavior.
>
> 4) I am also worried that the document is a bit to positive in regards
> to that IW10 will reduce the domain sharding practice. I think the only
> thing that can do is likely a new HTTP transport strata that shows that
> significantly improved performance for multiple objects from the same
> domain over fewer transport flows. Maybe SPDY + IW10 can provide such
> incentive, but I don't think IW10 alone will do much.
>
> Cheers
>
> Magnus
>
> On 2012-11-26 22:29, The IESG wrote:
> >
> > The IESG has received a request from the TCP Maintenance and Minor
> > Extensions WG (tcpm) to consider the following document:
> > - 'Increasing TCP's Initial Window'
> >   <draft-ietf-tcpm-initcwnd-06.txt> as Experimental RFC
> >
> > The IESG plans to make a decision in the next few weeks, and solicits
> > final comments on this action. Please send substantive comments to the
> > ietf@ietf.org mailing lists by 2012-12-10. Exceptionally, comments may
> be
> > sent to iesg@ietf.org instead. In either case, please retain the
> > beginning of the Subject line to allow automated sorting.
> >
> > Abstract
> >
> >
> >    This document proposes an experiment to increase the permitted TCP
> >    initial window (IW) from between 2 and 4 segments, as specified in
> >    RFC 3390, to 10 segments, with a fallback to the existing
> >    recommendation when performance issues are detected. It discusses th=
e
> >    motivation behind the increase, the advantages and disadvantages of
> >    the higher initial window, and presents results from several large
> >    scale experiments showing that the higher initial window improves th=
e
> >    overall performance of many web services without resulting in a
> >    congestion collapse. The document closes with a discussion of usage
> >    and deployment for further experimental purpose recommended by the
> >    IETF TCP Maintenance and Minor Extensions (TCPM) working group.
> >
> >
> >
> >
> > The file can be obtained via
> > http://datatracker.ietf.org/doc/draft-ietf-tcpm-initcwnd/
> >
> > IESG discussion can be tracked via
> > http://datatracker.ietf.org/doc/draft-ietf-tcpm-initcwnd/ballot/
> >
> >
> > No IPR declarations have been submitted directly on this I-D.
> >
> >
> >
> >
>
>
> --
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>

--bcaec550b3363c36d404d0efb125
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi Magnus,<div><br></div><div>Thanks for the comments.</div><div>I think yo=
ur points are valid. But, it&#39;s not very clear to me what you would like=
 to suggest authors.</div><div>IW10 might have an impact on real-time traff=
ic. But, difficult point is we are not very sure how much it is over the wh=
ole Internet.=A0Do we ask authors to do experiments to address the concerns=
 you pointed out?=A0</div>
<div>There will be situations where IW10 has negative impacts as well as po=
sitive impacts.=A0</div><div>But, I&#39;m feeling it might be difficult to =
discuss this point without large scale experiments.</div><div><br></div><di=
v>
Thanks,</div><div>--</div><div>Yoshifumi<br><br><div class=3D"gmail_quote">=
On Thu, Dec 6, 2012 at 2:01 AM, Magnus Westerlund <span dir=3D"ltr">&lt;<a =
href=3D"mailto:magnus.westerlund@ericsson.com" target=3D"_blank">magnus.wes=
terlund@ericsson.com</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">Hi,<br>
<br>
I have reviewed draft-ietf-tcpm-initcwnd-06 and have some questions and<br>
comments.<br>
<br>
1) First of all the experiments done appear to cover the impact on other<br=
>
applications, like measuring the RTT variations when using IW10 compared<br=
>
to IW3. If one is interested in the impact this proposal have on<br>
real-time traffic flows over a shared bottle-neck the variations in<br>
queue time at the bottleneck combined with that flows seen loss rates<br>
are the most important factors. As a sudden delay spike of sufficient<br>
magnitude will most likely result in a real-time media packet being<br>
late, i.e. late loss rather than being lost in the network there might<br>
be significant impact on such traffic from these IW10 packet bursts.<br>
Have I missed any material discussing this aspect?<br>
<br>
All the latency figures in the parts I was looking at was discussing<br>
cases when the object transfer takes longer time. But it appear obvious<br>
that even with a 100 ms increased one-way transfer time for a packet is<br>
the end of the initial window the total transfer goes faster compared to<br=
>
the delay caused by growing the window.<br>
<br>
2) If I assume that most users are behind buffer-bloated links the<br>
introduction of IW10 on wide scale will additionally disrupt interactive<br=
>
applications and cause increased delay and possible late loss depending<br>
on application. Especially in combination with domain sharding. For<br>
example a Swedish newspaper&#39;s website front page results in that more<b=
r>
than 40 TCP connections are opened in a current browser. If these all<br>
was using IW10, the amount of packets being sent initially will grow<br>
roughly from 40*3 =3D 120 to 40*10 =3D 400. There are some distribution ove=
r<br>
time but still this likely results in a larger delay spike.<br>
<br>
I don&#39;t quite know how I should consider this case. One stand point is<=
br>
that the interactive application is anyway buggered and IW10 effects are<br=
>
not making it significantly worse. The only remedy is queue separation<br>
so that what ever happens with the web-downloads doesn&#39;t affect the<br>
interactive traffic. Another is that it will make the existing situation<br=
>
worse and we should try to avoid making it worse.<br>
<br>
I think I am more in the first camp, but still the second one is<br>
worrying to me. But I dare to guess that the performance gains might be<br>
worth the issues, but without real progress on mitigation of the buffer<br>
bloat issues for interactive real-time traffic this will add<br>
significantly to the issues real-time has to face.<br>
<br>
<br>
3) The documents talks in quite loose terms about what can be done to<br>
avoid to continue cause issues for destinations which has issues with<br>
IW10. However I think it is a bit unspecific here. I can see that a<br>
content deliverer can have lists for destination address blocks that<br>
they see issues with which configures the sending server side with a<br>
lower IW value. It also talks about the clients can configure to keep<br>
the window low initially to prevent an IW10 sender to clobber ones link<br>
if that is known. My question here is if the recommendation for auto<br>
detection and configuration can be made more explicit. Fortunately the<br>
issues with misconfiguration in this cases is not that serious. But<br>
otherwise there is commonly need to be quite careful with such auto<br>
configs that affects the behavior.<br>
<br>
4) I am also worried that the document is a bit to positive in regards<br>
to that IW10 will reduce the domain sharding practice. I think the only<br>
thing that can do is likely a new HTTP transport strata that shows that<br>
significantly improved performance for multiple objects from the same<br>
domain over fewer transport flows. Maybe SPDY + IW10 can provide such<br>
incentive, but I don&#39;t think IW10 alone will do much.<br>
<br>
Cheers<br>
<br>
Magnus<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
On 2012-11-26 22:29, The IESG wrote:<br>
&gt;<br>
&gt; The IESG has received a request from the TCP Maintenance and Minor<br>
&gt; Extensions WG (tcpm) to consider the following document:<br>
&gt; - &#39;Increasing TCP&#39;s Initial Window&#39;<br>
&gt; =A0 &lt;draft-ietf-tcpm-initcwnd-06.txt&gt; as Experimental RFC<br>
&gt;<br>
&gt; The IESG plans to make a decision in the next few weeks, and solicits<=
br>
&gt; final comments on this action. Please send substantive comments to the=
<br>
&gt; <a href=3D"mailto:ietf@ietf.org">ietf@ietf.org</a> mailing lists by 20=
12-12-10. Exceptionally, comments may be<br>
&gt; sent to <a href=3D"mailto:iesg@ietf.org">iesg@ietf.org</a> instead. In=
 either case, please retain the<br>
&gt; beginning of the Subject line to allow automated sorting.<br>
&gt;<br>
&gt; Abstract<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0This document proposes an experiment to increase the permitted =
TCP<br>
&gt; =A0 =A0initial window (IW) from between 2 and 4 segments, as specified=
 in<br>
&gt; =A0 =A0RFC 3390, to 10 segments, with a fallback to the existing<br>
&gt; =A0 =A0recommendation when performance issues are detected. It discuss=
es the<br>
&gt; =A0 =A0motivation behind the increase, the advantages and disadvantage=
s of<br>
&gt; =A0 =A0the higher initial window, and presents results from several la=
rge<br>
&gt; =A0 =A0scale experiments showing that the higher initial window improv=
es the<br>
&gt; =A0 =A0overall performance of many web services without resulting in a=
<br>
&gt; =A0 =A0congestion collapse. The document closes with a discussion of u=
sage<br>
&gt; =A0 =A0and deployment for further experimental purpose recommended by =
the<br>
&gt; =A0 =A0IETF TCP Maintenance and Minor Extensions (TCPM) working group.=
<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; The file can be obtained via<br>
&gt; <a href=3D"http://datatracker.ietf.org/doc/draft-ietf-tcpm-initcwnd/" =
target=3D"_blank">http://datatracker.ietf.org/doc/draft-ietf-tcpm-initcwnd/=
</a><br>
&gt;<br>
&gt; IESG discussion can be tracked via<br>
&gt; <a href=3D"http://datatracker.ietf.org/doc/draft-ietf-tcpm-initcwnd/ba=
llot/" target=3D"_blank">http://datatracker.ietf.org/doc/draft-ietf-tcpm-in=
itcwnd/ballot/</a><br>
&gt;<br>
&gt;<br>
&gt; No IPR declarations have been submitted directly on this I-D.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
<br>
<br>
</div></div><span class=3D"HOEnZb"><font color=3D"#888888">--<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287<br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079<br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
</font></span></blockquote></div><br></div>

--bcaec550b3363c36d404d0efb125--

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Subject: Re: [tcpm] Last Call: <draft-ietf-tcpm-initcwnd-06.txt> (Increasing TCP's Initial Window) to Experimental RFC
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On 2012-12-16 04:16, Yoshifumi Nishida wrote:
> Hi Magnus,
> 
> Thanks for the comments.
> I think your points are valid. But, it's not very clear to me what you
> would like to suggest authors.
> IW10 might have an impact on real-time traffic. But, difficult point is
> we are not very sure how much it is over the whole Internet. Do we ask
> authors to do experiments to address the concerns you pointed out? 
> There will be situations where IW10 has negative impacts as well as
> positive impacts. 

I agree. But I think the already performed experiments has failed to
look at the aspect of how much e2e delay has been affected by the usage
of IW10 for flow sharing the bottleneck or full path with the TCP flow.

> But, I'm feeling it might be difficult to discuss this point without
> large scale experiments.

That I am not convinced of. I would like to get some indication if they
simply failed to take this into account or that it was to difficult to
measure.

I agree that some of the below are difficult to address. Some will
require additional experimentation and evaluation. Something I think
will be required to answer when this is evaluated. Then some of it is
actually reasonably straightforward to discuss or clarify in the document.

I still expect the authors to answer my feedback.

Cheers

Magnus



> 
> Thanks,
> --
> Yoshifumi
> 
> On Thu, Dec 6, 2012 at 2:01 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> wrote:
> 
>     Hi,
> 
>     I have reviewed draft-ietf-tcpm-initcwnd-06 and have some questions and
>     comments.
> 
>     1) First of all the experiments done appear to cover the impact on other
>     applications, like measuring the RTT variations when using IW10 compared
>     to IW3. If one is interested in the impact this proposal have on
>     real-time traffic flows over a shared bottle-neck the variations in
>     queue time at the bottleneck combined with that flows seen loss rates
>     are the most important factors. As a sudden delay spike of sufficient
>     magnitude will most likely result in a real-time media packet being
>     late, i.e. late loss rather than being lost in the network there might
>     be significant impact on such traffic from these IW10 packet bursts.
>     Have I missed any material discussing this aspect?
> 
>     All the latency figures in the parts I was looking at was discussing
>     cases when the object transfer takes longer time. But it appear obvious
>     that even with a 100 ms increased one-way transfer time for a packet is
>     the end of the initial window the total transfer goes faster compared to
>     the delay caused by growing the window.
> 
>     2) If I assume that most users are behind buffer-bloated links the
>     introduction of IW10 on wide scale will additionally disrupt interactive
>     applications and cause increased delay and possible late loss depending
>     on application. Especially in combination with domain sharding. For
>     example a Swedish newspaper's website front page results in that more
>     than 40 TCP connections are opened in a current browser. If these all
>     was using IW10, the amount of packets being sent initially will grow
>     roughly from 40*3 = 120 to 40*10 = 400. There are some distribution over
>     time but still this likely results in a larger delay spike.
> 
>     I don't quite know how I should consider this case. One stand point is
>     that the interactive application is anyway buggered and IW10 effects are
>     not making it significantly worse. The only remedy is queue separation
>     so that what ever happens with the web-downloads doesn't affect the
>     interactive traffic. Another is that it will make the existing situation
>     worse and we should try to avoid making it worse.
> 
>     I think I am more in the first camp, but still the second one is
>     worrying to me. But I dare to guess that the performance gains might be
>     worth the issues, but without real progress on mitigation of the buffer
>     bloat issues for interactive real-time traffic this will add
>     significantly to the issues real-time has to face.
> 
> 
>     3) The documents talks in quite loose terms about what can be done to
>     avoid to continue cause issues for destinations which has issues with
>     IW10. However I think it is a bit unspecific here. I can see that a
>     content deliverer can have lists for destination address blocks that
>     they see issues with which configures the sending server side with a
>     lower IW value. It also talks about the clients can configure to keep
>     the window low initially to prevent an IW10 sender to clobber ones link
>     if that is known. My question here is if the recommendation for auto
>     detection and configuration can be made more explicit. Fortunately the
>     issues with misconfiguration in this cases is not that serious. But
>     otherwise there is commonly need to be quite careful with such auto
>     configs that affects the behavior.
> 
>     4) I am also worried that the document is a bit to positive in regards
>     to that IW10 will reduce the domain sharding practice. I think the only
>     thing that can do is likely a new HTTP transport strata that shows that
>     significantly improved performance for multiple objects from the same
>     domain over fewer transport flows. Maybe SPDY + IW10 can provide such
>     incentive, but I don't think IW10 alone will do much.
> 
>     Cheers
> 
>     Magnus
> 
>     On 2012-11-26 22:29, The IESG wrote:
>     >
>     > The IESG has received a request from the TCP Maintenance and Minor
>     > Extensions WG (tcpm) to consider the following document:
>     > - 'Increasing TCP's Initial Window'
>     >   <draft-ietf-tcpm-initcwnd-06.txt> as Experimental RFC
>     >
>     > The IESG plans to make a decision in the next few weeks, and solicits
>     > final comments on this action. Please send substantive comments to the
>     > ietf@ietf.org <mailto:ietf@ietf.org> mailing lists by 2012-12-10.
>     Exceptionally, comments may be
>     > sent to iesg@ietf.org <mailto:iesg@ietf.org> instead. In either
>     case, please retain the
>     > beginning of the Subject line to allow automated sorting.
>     >
>     > Abstract
>     >
>     >
>     >    This document proposes an experiment to increase the permitted TCP
>     >    initial window (IW) from between 2 and 4 segments, as specified in
>     >    RFC 3390, to 10 segments, with a fallback to the existing
>     >    recommendation when performance issues are detected. It
>     discusses the
>     >    motivation behind the increase, the advantages and disadvantages of
>     >    the higher initial window, and presents results from several large
>     >    scale experiments showing that the higher initial window
>     improves the
>     >    overall performance of many web services without resulting in a
>     >    congestion collapse. The document closes with a discussion of usage
>     >    and deployment for further experimental purpose recommended by the
>     >    IETF TCP Maintenance and Minor Extensions (TCPM) working group.
>     >
>     >
>     >
>     >
>     > The file can be obtained via
>     > http://datatracker.ietf.org/doc/draft-ietf-tcpm-initcwnd/
>     >
>     > IESG discussion can be tracked via
>     > http://datatracker.ietf.org/doc/draft-ietf-tcpm-initcwnd/ballot/
>     >
>     >
>     > No IPR declarations have been submitted directly on this I-D.
>     >
>     >
>     >
>     >
> 
> 
>     --
> 
>     Magnus Westerlund
> 
>     ----------------------------------------------------------------------
>     Multimedia Technologies, Ericsson Research EAB/TVM
>     ----------------------------------------------------------------------
>     Ericsson AB                | Phone  +46 10 7148287
>     Färögatan 6                | Mobile +46 73 0949079
>     SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>     <mailto:magnus.westerlund@ericsson.com>
>     ----------------------------------------------------------------------
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From brandon.williams@akamai.com  Thu Dec 20 09:21:09 2012
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Dear all,

A new version of this draft has been submitted that attempts to lay out 
a more clear argument for the use of both TCP and IP options, with 
references to other efforts in the same arena.

Comments are welcome.

Cheers,
--Brandon



-------- Original Message --------
Subject: New Version Notification for 
draft-williams-overlaypath-ip-tcp-rfc-03.txt
Date: Thu, 20 Dec 2012 11:55:33 -0500
From: internet-drafts@ietf.org <internet-drafts@ietf.org>
To: Williams, Brandon <bowill@akamai.com>


A new version of I-D, draft-williams-overlaypath-ip-tcp-rfc-03.txt
has been successfully submitted by Brandon Williams and posted to the
IETF repository.

Filename:	 draft-williams-overlaypath-ip-tcp-rfc
Revision:	 03
Title:		 Overlay Path Option for IP and TCP
Creation date:	 2012-12-20
WG ID:		 Individual Submission
Number of pages: 15
URL: 
http://www.ietf.org/internet-drafts/draft-williams-overlaypath-ip-tcp-rfc-03.txt
Status: 
http://datatracker.ietf.org/doc/draft-williams-overlaypath-ip-tcp-rfc
Htmlized: 
http://tools.ietf.org/html/draft-williams-overlaypath-ip-tcp-rfc-03
Diff: 
http://www.ietf.org/rfcdiff?url2=draft-williams-overlaypath-ip-tcp-rfc-03

Abstract:
    Data transport through overlay networks often uses either connection
    termination or network address translation (NAT) in such a way that
    the public IP addresses of the true endpoint machines involved in the
    data transport are invisible to each other.  This document describes
    IPv4, IPv6, and TCP options for communicating this information from
    the overlay network to the endpoint machines.

 



The IETF Secretariat




From wes@mti-systems.com  Thu Dec 20 11:16:25 2012
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On 12/20/2012 12:21 PM, Brandon Williams wrote:
> Dear all,
> 
> A new version of this draft has been submitted that attempts to lay out
> a more clear argument for the use of both TCP and IP options, with
> references to other efforts in the same arena.
> 
> Comments are welcome.

(note, I've cross-posted to INTAREA and TCPM, since similar
announcements went to each list)

Hi Brandon, *many* thanks for writing this; it does help me (at least)
to understand what you're doing with this option.

As I now understand it, instead of a tunneling approach that would
normally be applied for building overlay networks, this approach
pushes and pops IP addresses from the protocol options fields.

Can you discuss why normal tunneling protocols aren't used to build
the overlay?

Since those are easily and widely available, I wonder why they aren't
used and why something more elaborate, fragile, and not as compatible
with the Internet architecture is really needed or felt to be a good
idea?  I understand that it basically *works* ... but just am not
seeing how it makes sense?

-- 
Wes Eddy
MTI Systems


From brandon.williams@akamai.com  Thu Dec 20 12:49:30 2012
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Subject: Re: [tcpm] draft-williams-overlaypath-ip-tcp-rfc
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Hi Wes,

Thanks for your comments.

It looks like I might have managed to make the use of the proposed 
option less clear, instead of more clear. Or maybe I'm misunderstanding 
the point that you're making.

The mechanics of our system are tunnel-based, as with most overlay 
architectures that I've looked at. The tunneling starts at an overlay 
ingress machine close to one of the endpoints (i.e. the client or 
server) and ends at an overlay egress machine close to the other 
endpoint. Since the ingress and egress are on the public internet, the 
overlay does not extend all the way onto the endpoints' LANs. This means 
that standard internet routing cannot be used to drive the connections 
into the overlay. Instead, NAT is used on both sides of the overlay, 
which results in the server having no way to reliably identify the client.

The proposed options are not intended to be used as part of the 
mechanics of the overlay. The overlay is fully functional without the 
options. Instead, the options are intended to provide the client's 
connection identifying information to the server for use in 
load-balancing, diagnostics, etc.

Does this clarify things? further muddy the waters? or simply indicate 
that I missed your point?

--Brandon


PS: Thanks for cross-posting your comments. I should have done that to 
begin with. I primarily posted to TCPM for informational purposes, since 
the TCPM has not shown much interest in this or similar drafts in the 
past. The INTAREA list has been more actively engaged in discussion 
related to client identification. Still, if I was going to cross-post, I 
should have done it with a single thread.

On 12/20/2012 02:16 PM, Wesley Eddy wrote:
> On 12/20/2012 12:21 PM, Brandon Williams wrote:
>> Dear all,
>>
>> A new version of this draft has been submitted that attempts to lay out
>> a more clear argument for the use of both TCP and IP options, with
>> references to other efforts in the same arena.
>>
>> Comments are welcome.
>
> (note, I've cross-posted to INTAREA and TCPM, since similar
> announcements went to each list)
>
> Hi Brandon, *many* thanks for writing this; it does help me (at least)
> to understand what you're doing with this option.
>
> As I now understand it, instead of a tunneling approach that would
> normally be applied for building overlay networks, this approach
> pushes and pops IP addresses from the protocol options fields.
>
> Can you discuss why normal tunneling protocols aren't used to build
> the overlay?
>
> Since those are easily and widely available, I wonder why they aren't
> used and why something more elaborate, fragile, and not as compatible
> with the Internet architecture is really needed or felt to be a good
> idea?  I understand that it basically *works* ... but just am not
> seeing how it makes sense?
>

-- 
Brandon Williams; Principal Software Engineer
Cloud Engineering; Akamai Technologies Inc.

From wes@mti-systems.com  Thu Dec 20 13:05:15 2012
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On 12/20/2012 3:49 PM, Brandon Williams wrote:
> Hi Wes,
> 
> Thanks for your comments.
> 
> It looks like I might have managed to make the use of the proposed
> option less clear, instead of more clear. Or maybe I'm misunderstanding
> the point that you're making.
> 
> The mechanics of our system are tunnel-based, as with most overlay
> architectures that I've looked at. The tunneling starts at an overlay
> ingress machine close to one of the endpoints (i.e. the client or
> server) and ends at an overlay egress machine close to the other
> endpoint. Since the ingress and egress are on the public internet, the
> overlay does not extend all the way onto the endpoints' LANs. This means
> that standard internet routing cannot be used to drive the connections
> into the overlay. Instead, NAT is used on both sides of the overlay,
> which results in the server having no way to reliably identify the client.
> 
> The proposed options are not intended to be used as part of the
> mechanics of the overlay. The overlay is fully functional without the
> options. Instead, the options are intended to provide the client's
> connection identifying information to the server for use in
> load-balancing, diagnostics, etc.
> 

Ah, so are there additional devices beyond what's shown in your Figure
1?  I ask because if the overlay ingress and egress are simple tunnel
endpoints, then the endpoint addresses would be totally visible to one
another.

Your figure 1 is:

                  +------------------------------------+
                  |                                    |
                  |              INTERNET              |
                  |                                    |
   +-----------+  |  +------------+                    |
   |  HOST_1   |-----| OVRLY_IN_1 |-----------+        |
   +-----------+  |  +------------+           |        |
                  |                           |        |
   +-----------+  |  +------------+     +-----------+  |  +--------+
   |  HOST_2   |-----| OVRLY_IN_2 |-----| OVRLY_OUT |-----| SERVER |
   +-----------+  |  +------------+     +-----------+  |  +--------+
                  |                           |        |
   +-----------+  |  +------------+           |        |
   |  HOST_3   |-----| OVRLY_IN_3 |-----------+        |
   +-----------+  |  +------------+                    |
                  |                                    |
                  +------------------------------------+

                                 Figure 1

If there were tunnels between the OVRLY_IN and OVERLY_OUT boxes,
then the inner IP headers would have the HOST_X and SERVER
addresses, and the outer ones in the tunnel would have the
overlay headers.  Since the inner packets would be delivered
ultimately after egressing the tunnels, the HOST_X addresses
are totally visible to the server, and vice versa.

Your document shows instead:

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

             ip hdr contains:               ip hdr contains:
   SENDER -> src = sender   --> OVERLAY --> src = overlay2  --> RECEIVER
             dst = overlay1                 dst = receiver

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

So, this is not really showing tunnels to me ... this is rewriting
(NAT) of the destination address.

Or am I misunderstanding?

-- 
Wes Eddy
MTI Systems

From brandon.williams@akamai.com  Thu Dec 20 14:04:50 2012
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, "int-area@ietf.org" <int-area@ietf.org>
Subject: Re: [tcpm] draft-williams-overlaypath-ip-tcp-rfc
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On 12/20/2012 04:04 PM, Wesley Eddy wrote:
> On 12/20/2012 3:49 PM, Brandon Williams wrote:
>> Hi Wes,
>>
>> Thanks for your comments.
>>
>> It looks like I might have managed to make the use of the proposed
>> option less clear, instead of more clear. Or maybe I'm misunderstanding
>> the point that you're making.
>>
>> The mechanics of our system are tunnel-based, as with most overlay
>> architectures that I've looked at. The tunneling starts at an overlay
>> ingress machine close to one of the endpoints (i.e. the client or
>> server) and ends at an overlay egress machine close to the other
>> endpoint. Since the ingress and egress are on the public internet, the
>> overlay does not extend all the way onto the endpoints' LANs. This means
>> that standard internet routing cannot be used to drive the connections
>> into the overlay. Instead, NAT is used on both sides of the overlay,
>> which results in the server having no way to reliably identify the client.
>>
>> The proposed options are not intended to be used as part of the
>> mechanics of the overlay. The overlay is fully functional without the
>> options. Instead, the options are intended to provide the client's
>> connection identifying information to the server for use in
>> load-balancing, diagnostics, etc.
>>
>
> Ah, so are there additional devices beyond what's shown in your Figure
> 1?  I ask because if the overlay ingress and egress are simple tunnel
> endpoints, then the endpoint addresses would be totally visible to one
> another.

Yes. There are additional devices between the HOST and OVRLY_IN, and 
also between OVRLY_OUT and SERVER, but those devices are just the 
internet's standard routing infrastructure. There are also potential 
intermediate devices between OVRLY_IN and OVRLY_OUT that can be used for 
optimized routing between the overlay's ingress and egress.

>
> Your figure 1 is:
>
>                    +------------------------------------+
>                    |                                    |
>                    |              INTERNET              |
>                    |                                    |
>     +-----------+  |  +------------+                    |
>     |  HOST_1   |-----| OVRLY_IN_1 |-----------+        |
>     +-----------+  |  +------------+           |        |
>                    |                           |        |
>     +-----------+  |  +------------+     +-----------+  |  +--------+
>     |  HOST_2   |-----| OVRLY_IN_2 |-----| OVRLY_OUT |-----| SERVER |
>     +-----------+  |  +------------+     +-----------+  |  +--------+
>                    |                           |        |
>     +-----------+  |  +------------+           |        |
>     |  HOST_3   |-----| OVRLY_IN_3 |-----------+        |
>     +-----------+  |  +------------+                    |
>                    |                                    |
>                    +------------------------------------+
>
>                                   Figure 1
>
> If there were tunnels between the OVRLY_IN and OVERLY_OUT boxes,
> then the inner IP headers would have the HOST_X and SERVER
> addresses, and the outer ones in the tunnel would have the
> overlay headers.  Since the inner packets would be delivered
> ultimately after egressing the tunnels, the HOST_X addresses
> are totally visible to the server, and vice versa.

There are indeed tunnels between OVRLY_IN and OVRLY_OUT, and the inner 
IP headers will typically use either the client-side addresses or the 
server-side addresses. However, neither OVRLY_IN nor OVRLY_OUT can be 
assumed to be reliably in-path between HOST and SERVER, which means that 
internet routing cannot be relied upon to cause packets to arrive at the 
overlay ingress. Instead, HOST_1 must directly address OVRLY_IN_1 in 
order to send its packets into the tunnel, and SERVER must directly 
address OVRLY_OUT in order to send the return traffic into the tunnel.

>
> Your document shows instead:
>
>     ---------------------------------------------------------------------
>
>               ip hdr contains:               ip hdr contains:
>     SENDER -> src = sender   --> OVERLAY --> src = overlay2  --> RECEIVER
>               dst = overlay1                 dst = receiver
>
>     ---------------------------------------------------------------------
>
> So, this is not really showing tunnels to me ... this is rewriting
> (NAT) of the destination address.
>

As noted above, the use of tunnels and NAT in this case are not mutually 
exclusive. NAT is used to allow the overlay ingress to intercept 
packets, which are then tunneled to the overlay egress, where NAT is 
used to deliver the packets to the receiver and ensure that return 
traffic also uses the overlay.

--Brandon


PS: Sorry to double send this to you Wes. It was bounced by the ietf 
lists the first time.

-- 
Brandon Williams; Principal Software Engineer
Cloud Engineering; Akamai Technologies Inc.

From lars@netapp.com  Fri Dec 21 00:51:45 2012
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From: "Eggert, Lars" <lars@netapp.com>
To: "tcpm@ietf.org" <tcpm@ietf.org>
Thread-Topic: [Int-area] draft-williams-overlaypath-ip-tcp-rfc
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Subject: [tcpm] Fwd: [Int-area] draft-williams-overlaypath-ip-tcp-rfc
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Since a TCP option is being designed, of interest to TCPM.

Lars

Begin forwarded message:

> From: Brandon Williams <brandon.williams@akamai.com>
> Subject: [Int-area] draft-williams-overlaypath-ip-tcp-rfc
> Date: December 20, 2012 18:20:01 GMT+01:00
> To: "int-area@ietf.org" <int-area@ietf.org>
>=20
> Dear all,
>=20
> A new version of this draft has been submitted that attempts to lay out a=
 more clear argument for the use of both TCP and IP options, with reference=
s to other efforts in the same arena.
>=20
> Comments are welcome.
>=20
> Cheers,
> --Brandon
>=20
>=20
> -------- Original Message --------
> Subject: New Version Notification for draft-williams-overlaypath-ip-tcp-r=
fc-03.txt
> Date: Thu, 20 Dec 2012 11:55:33 -0500
> From: internet-drafts@ietf.org <internet-drafts@ietf.org>
> To: Williams, Brandon <bowill@akamai.com>
>=20
>=20
> A new version of I-D, draft-williams-overlaypath-ip-tcp-rfc-03.txt
> has been successfully submitted by Brandon Williams and posted to the
> IETF repository.
>=20
> Filename:	 draft-williams-overlaypath-ip-tcp-rfc
> Revision:	 03
> Title:		 Overlay Path Option for IP and TCP
> Creation date:	 2012-12-20
> WG ID:		 Individual Submission
> Number of pages: 15
> URL: http://www.ietf.org/internet-drafts/draft-williams-overlaypath-ip-tc=
p-rfc-03.txt
> Status: http://datatracker.ietf.org/doc/draft-williams-overlaypath-ip-tcp=
-rfc
> Htmlized: http://tools.ietf.org/html/draft-williams-overlaypath-ip-tcp-rf=
c-03
> Diff: http://www.ietf.org/rfcdiff?url2=3Ddraft-williams-overlaypath-ip-tc=
p-rfc-03
>=20
> Abstract:
>   Data transport through overlay networks often uses either connection
>   termination or network address translation (NAT) in such a way that
>   the public IP addresses of the true endpoint machines involved in the
>   data transport are invisible to each other.  This document describes
>   IPv4, IPv6, and TCP options for communicating this information from
>   the overlay network to the endpoint machines.
>=20
>=20
>=20
>=20
> The IETF Secretariat
>=20
>=20
>=20
> _______________________________________________
> Int-area mailing list
> Int-area@ietf.org
> https://www.ietf.org/mailman/listinfo/int-area


From c.raiciu@cs.ucl.ac.uk  Fri Dec 21 01:05:41 2012
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Subject: Re: [tcpm] Fwd: [Int-area] draft-williams-overlaypath-ip-tcp-rfc
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I have been thinking about this same problem for a project we're doing. =
The project wants to give end-systems the ability to instantiate flow =
processing (e.g. layer 4 and up) in the network - and so it needs a way =
to get packets onto these boxes.

We found that one option is to use loose source routing (LSRR). The =
intermediary hop just injects loose source route options that record all =
the relevant IPs this packet will go through; the packet's source =
address does not need to change. LSRR makes sure that the return packet =
travels the same way,=20
thus enabling TCP connectivity.

Brief wide-area tests showed that LSRR options do not get through the =
Internet (spoofing concerns lead to it being blocked a long time ago)  - =
so some sort of tunnel may be required to "hide" them from routers, =
(e.g. a UDP tunnel, IPsec). If the tunnel is MACed or encrypted then =
LSRR options are protected so on-path boxes can't modify them - thus =
alleviating the security issues relating to LSRR.

I admit that the need to tunnel isn't exactly clean either, but at least =
it supports all protocols and not just TCP.

If we want to target "just" TCP we may be able to do something more =
general than the current proposal - more about this when we have =
something fleshed out.

Best,
Costin


On 21 Dec 2012, at 10:51, Eggert, Lars wrote:

> Since a TCP option is being designed, of interest to TCPM.
>=20
> Lars
>=20
> Begin forwarded message:
>=20
>> From: Brandon Williams <brandon.williams@akamai.com>
>> Subject: [Int-area] draft-williams-overlaypath-ip-tcp-rfc
>> Date: December 20, 2012 18:20:01 GMT+01:00
>> To: "int-area@ietf.org" <int-area@ietf.org>
>>=20
>> Dear all,
>>=20
>> A new version of this draft has been submitted that attempts to lay =
out a more clear argument for the use of both TCP and IP options, with =
references to other efforts in the same arena.
>>=20
>> Comments are welcome.
>>=20
>> Cheers,
>> --Brandon
>>=20
>>=20
>> -------- Original Message --------
>> Subject: New Version Notification for =
draft-williams-overlaypath-ip-tcp-rfc-03.txt
>> Date: Thu, 20 Dec 2012 11:55:33 -0500
>> From: internet-drafts@ietf.org <internet-drafts@ietf.org>
>> To: Williams, Brandon <bowill@akamai.com>
>>=20
>>=20
>> A new version of I-D, draft-williams-overlaypath-ip-tcp-rfc-03.txt
>> has been successfully submitted by Brandon Williams and posted to the
>> IETF repository.
>>=20
>> Filename:	 draft-williams-overlaypath-ip-tcp-rfc
>> Revision:	 03
>> Title:		 Overlay Path Option for IP and TCP
>> Creation date:	 2012-12-20
>> WG ID:		 Individual Submission
>> Number of pages: 15
>> URL: =
http://www.ietf.org/internet-drafts/draft-williams-overlaypath-ip-tcp-rfc-=
03.txt
>> Status: =
http://datatracker.ietf.org/doc/draft-williams-overlaypath-ip-tcp-rfc
>> Htmlized: =
http://tools.ietf.org/html/draft-williams-overlaypath-ip-tcp-rfc-03
>> Diff: =
http://www.ietf.org/rfcdiff?url2=3Ddraft-williams-overlaypath-ip-tcp-rfc-0=
3
>>=20
>> Abstract:
>>  Data transport through overlay networks often uses either connection
>>  termination or network address translation (NAT) in such a way that
>>  the public IP addresses of the true endpoint machines involved in =
the
>>  data transport are invisible to each other.  This document describes
>>  IPv4, IPv6, and TCP options for communicating this information from
>>  the overlay network to the endpoint machines.
>>=20
>>=20
>>=20
>>=20
>> The IETF Secretariat
>>=20
>>=20
>>=20
>> _______________________________________________
>> Int-area mailing list
>> Int-area@ietf.org
>> https://www.ietf.org/mailman/listinfo/int-area
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From michael.scharf@alcatel-lucent.com  Fri Dec 21 05:35:05 2012
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From: "Scharf, Michael (Michael)" <michael.scharf@alcatel-lucent.com>
To: Brandon Williams <brandon.williams@akamai.com>, Wesley Eddy <wes@mti-systems.com>
Date: Fri, 21 Dec 2012 14:34:43 +0100
Thread-Topic: [tcpm] draft-williams-overlaypath-ip-tcp-rfc
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, "int-area@ietf.org" <int-area@ietf.org>
Subject: Re: [tcpm] draft-williams-overlaypath-ip-tcp-rfc
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Brandon,=20

> > If there were tunnels between the OVRLY_IN and OVERLY_OUT=20
> boxes, then=20
> > the inner IP headers would have the HOST_X and SERVER=20
> addresses, and=20
> > the outer ones in the tunnel would have the overlay headers.  Since=20
> > the inner packets would be delivered ultimately after egressing the=20
> > tunnels, the HOST_X addresses are totally visible to the=20
> server, and=20
> > vice versa.
>=20
> There are indeed tunnels between OVRLY_IN and OVRLY_OUT, and=20
> the inner IP headers will typically use either the=20
> client-side addresses or the server-side addresses. However,=20
> neither OVRLY_IN nor OVRLY_OUT can be assumed to be reliably=20
> in-path between HOST and SERVER, which means that internet=20
> routing cannot be relied upon to cause packets to arrive at=20
> the overlay ingress. Instead, HOST_1 must directly address=20
> OVRLY_IN_1 in order to send its packets into the tunnel, and=20
> SERVER must directly address OVRLY_OUT in order to send the=20
> return traffic into the tunnel.

Thanks for this explanation - this indeed helps to understand the architect=
ure. But actually I still don't fully understand the motivation of bypassin=
g Internet routing this way. As a non-expert on routing, it indeed looks to=
 me like reinventing source routing - but this is outside my core expertise=
.

Regarding TCPM's business: If I correctly understand the approach, OVRLY_IN=
 will "transparently" add and remove TCP options. This is kind of dangerous=
 from an end-to-end perspective... Sorry if that has been answered before, =
but I really wonder what to do if OVRLY_IN can't add this option, either be=
cause of lack of TCP option space, or because the path MTU is exceeded by t=
he resulting IP packet. (In fact, I think that this problem does not apply =
to TCP options only.)

Unless I miss something, the latter case could become much more relevant so=
on: TCPM currently works on the fast-open scheme that adds data to SYNs. Wi=
th that, I think it is possible that all data packets from a sender to a re=
ceiver are either full sized or large enough that the proposed option does =
not fit in. Given that this option can include full-sized IPv6 addresses, t=
his likelihood is much larger than for other existing TCP option, right?

In some cases, I believe that the proposed TCP option cannot be added in th=
e overlay without either IP fragmentation, which is unlikely to be a good i=
dea with NATs, or TCP segment splitting, which probably can cause harm as w=
ell. For instance, what would OVRLY_IN do if it receives an IP packet with =
a TCP SYN segment that already sums up to 1500 byte? And, to make the scena=
rio more nasty, if the same applies to the first data segments as well?

Thanks

Michael

From brandon.williams@akamai.com  Fri Dec 21 08:31:49 2012
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Cc: "tcpm@ietf.org" <tcpm@ietf.org>, "int-area@ietf.org" <int-area@ietf.org>
Subject: Re: [tcpm] draft-williams-overlaypath-ip-tcp-rfc
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Hi Michael,

Thanks for your comments.

Your are correct that the option could be problematic if added to a 
full-sized packet, or even a nearly full one. I can see that the 
document should have some discussion of this issue.

In a case like ours, where the overlay network uses tunneling, 
transparently adding the option is not a critical problem to be solved. 
It is already the case that the overlay entry point must advertise a 
reduced MSS in order to accommodate the tunnel overhead. The amount of 
space consumed by the option will always be smaller than the tunnel 
overhead, and the option can be added at OVRLY_OUT, so the two are not 
additive. That said, I can see that an overlay network that does not use 
tunnels internally, or one that in fact does apply the option on 
OVRLY_IN, would have a bigger problem, though.

The issue of the proposed fast-open scheme is one that we have not 
considered, but I don't think it adds any problems for the TCP option 
that aren't already a problem for tunneled connectivity in general. I 
will have to spend some time with that proposal and think about how they 
interrelate.

--Brandon

On 12/21/2012 08:34 AM, Scharf, Michael (Michael) wrote:
> Brandon,
>
>>> If there were tunnels between the OVRLY_IN and OVERLY_OUT
>> boxes, then
>>> the inner IP headers would have the HOST_X and SERVER
>> addresses, and
>>> the outer ones in the tunnel would have the overlay headers.  Since
>>> the inner packets would be delivered ultimately after egressing the
>>> tunnels, the HOST_X addresses are totally visible to the
>> server, and
>>> vice versa.
>>
>> There are indeed tunnels between OVRLY_IN and OVRLY_OUT, and
>> the inner IP headers will typically use either the
>> client-side addresses or the server-side addresses. However,
>> neither OVRLY_IN nor OVRLY_OUT can be assumed to be reliably
>> in-path between HOST and SERVER, which means that internet
>> routing cannot be relied upon to cause packets to arrive at
>> the overlay ingress. Instead, HOST_1 must directly address
>> OVRLY_IN_1 in order to send its packets into the tunnel, and
>> SERVER must directly address OVRLY_OUT in order to send the
>> return traffic into the tunnel.
>
> Thanks for this explanation - this indeed helps to understand the architecture. But actually I still don't fully understand the motivation of bypassing Internet routing this way. As a non-expert on routing, it indeed looks to me like reinventing source routing - but this is outside my core expertise.
>
> Regarding TCPM's business: If I correctly understand the approach, OVRLY_IN will "transparently" add and remove TCP options. This is kind of dangerous from an end-to-end perspective... Sorry if that has been answered before, but I really wonder what to do if OVRLY_IN can't add this option, either because of lack of TCP option space, or because the path MTU is exceeded by the resulting IP packet. (In fact, I think that this problem does not apply to TCP options only.)
>
> Unless I miss something, the latter case could become much more relevant soon: TCPM currently works on the fast-open scheme that adds data to SYNs. With that, I think it is possible that all data packets from a sender to a receiver are either full sized or large enough that the proposed option does not fit in. Given that this option can include full-sized IPv6 addresses, this likelihood is much larger than for other existing TCP option, right?
>
> In some cases, I believe that the proposed TCP option cannot be added in the overlay without either IP fragmentation, which is unlikely to be a good idea with NATs, or TCP segment splitting, which probably can cause harm as well. For instance, what would OVRLY_IN do if it receives an IP packet with a TCP SYN segment that already sums up to 1500 byte? And, to make the scenario more nasty, if the same applies to the first data segments as well?
>
> Thanks
>
> Michael
>

-- 
Brandon Williams; Principal Software Engineer
Cloud Engineering; Akamai Technologies Inc.

From brandon.williams@akamai.com  Fri Dec 21 08:32:52 2012
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Subject: Re: [tcpm] Fwd: [Int-area] draft-williams-overlaypath-ip-tcp-rfc
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Hi Costin,

Thanks for your comments.

The option described here is not intended to be the mechanism by which 
the overlay functions. Instead, it is intended to mitigate server-side 
issues associated with the use of NAT. This proposal is meant to meet 
some of the same requirements as other similar host identification 
drafts, as well as some other requirements that are specific to the 
overlay use case.

Is there something specific in the text of draft that appears to 
indicate that the option would be used to implement the overlay? Or is 
it perhaps simply that name that was chosen for the option?

Cheers,
--Brandon

On 12/21/2012 04:05 AM, Costin Raiciu wrote:
> I have been thinking about this same problem for a project we're doing. The project wants to give end-systems the ability to instantiate flow processing (e.g. layer 4 and up) in the network - and so it needs a way to get packets onto these boxes.
>
> We found that one option is to use loose source routing (LSRR). The intermediary hop just injects loose source route options that record all the relevant IPs this packet will go through; the packet's source address does not need to change. LSRR makes sure that the return packet travels the same way,
> thus enabling TCP connectivity.
>
> Brief wide-area tests showed that LSRR options do not get through the Internet (spoofing concerns lead to it being blocked a long time ago)  - so some sort of tunnel may be required to "hide" them from routers, (e.g. a UDP tunnel, IPsec). If the tunnel is MACed or encrypted then LSRR options are protected so on-path boxes can't modify them - thus alleviating the security issues relating to LSRR.
>
> I admit that the need to tunnel isn't exactly clean either, but at least it supports all protocols and not just TCP.
>
> If we want to target "just" TCP we may be able to do something more general than the current proposal - more about this when we have something fleshed out.
>
> Best,
> Costin
>
>
> On 21 Dec 2012, at 10:51, Eggert, Lars wrote:
>
>> Since a TCP option is being designed, of interest to TCPM.
>>
>> Lars
>>
>> Begin forwarded message:
>>
>>> From: Brandon Williams <brandon.williams@akamai.com>
>>> Subject: [Int-area] draft-williams-overlaypath-ip-tcp-rfc
>>> Date: December 20, 2012 18:20:01 GMT+01:00
>>> To: "int-area@ietf.org" <int-area@ietf.org>
>>>
>>> Dear all,
>>>
>>> A new version of this draft has been submitted that attempts to lay out a more clear argument for the use of both TCP and IP options, with references to other efforts in the same arena.
>>>
>>> Comments are welcome.
>>>
>>> Cheers,
>>> --Brandon
>>>
>>>
>>> -------- Original Message --------
>>> Subject: New Version Notification for draft-williams-overlaypath-ip-tcp-rfc-03.txt
>>> Date: Thu, 20 Dec 2012 11:55:33 -0500
>>> From: internet-drafts@ietf.org <internet-drafts@ietf.org>
>>> To: Williams, Brandon <bowill@akamai.com>
>>>
>>>
>>> A new version of I-D, draft-williams-overlaypath-ip-tcp-rfc-03.txt
>>> has been successfully submitted by Brandon Williams and posted to the
>>> IETF repository.
>>>
>>> Filename:	 draft-williams-overlaypath-ip-tcp-rfc
>>> Revision:	 03
>>> Title:		 Overlay Path Option for IP and TCP
>>> Creation date:	 2012-12-20
>>> WG ID:		 Individual Submission
>>> Number of pages: 15
>>> URL: http://www.ietf.org/internet-drafts/draft-williams-overlaypath-ip-tcp-rfc-03.txt
>>> Status: http://datatracker.ietf.org/doc/draft-williams-overlaypath-ip-tcp-rfc
>>> Htmlized: http://tools.ietf.org/html/draft-williams-overlaypath-ip-tcp-rfc-03
>>> Diff: http://www.ietf.org/rfcdiff?url2=draft-williams-overlaypath-ip-tcp-rfc-03
>>>
>>> Abstract:
>>>   Data transport through overlay networks often uses either connection
>>>   termination or network address translation (NAT) in such a way that
>>>   the public IP addresses of the true endpoint machines involved in the
>>>   data transport are invisible to each other.  This document describes
>>>   IPv4, IPv6, and TCP options for communicating this information from
>>>   the overlay network to the endpoint machines.
>>>
>>>
>>>
>>>
>>> The IETF Secretariat
>>>
>>>
>>>
>>> _______________________________________________
>>> Int-area mailing list
>>> Int-area@ietf.org
>>> https://www.ietf.org/mailman/listinfo/int-area
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

-- 
Brandon Williams; Principal Software Engineer
Cloud Engineering; Akamai Technologies Inc.

-- 
Brandon Williams; Principal Software Engineer
Cloud Engineering; Akamai Technologies Inc.

From michael.scharf@alcatel-lucent.com  Fri Dec 21 11:14:33 2012
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To: Brandon Williams <brandon.williams@akamai.com>
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Subject: Re: [tcpm] draft-williams-overlaypath-ip-tcp-rfc
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Brandon,=20

> Your are correct that the option could be problematic if=20
> added to a full-sized packet, or even a nearly full one. I=20
> can see that the document should have some discussion of this issue.

Yes.

> In a case like ours, where the overlay network uses=20
> tunneling, transparently adding the option is not a critical=20
> problem to be solved.=20
> It is already the case that the overlay entry point must=20
> advertise a reduced MSS in order to accommodate the tunnel=20
> overhead. The amount of space consumed by the option will=20
> always be smaller than the tunnel overhead, and the option=20
> can be added at OVRLY_OUT, so the two are not additive. That=20
> said, I can see that an overlay network that does not use=20
> tunnels internally, or one that in fact does apply the option=20
> on OVRLY_IN, would have a bigger problem, though.

So, the new TCP option is basically required between OVRLY_OUT and the rece=
iver/server, because the relevant information is already somehow transporte=
d in the overlay, right?

This raises another question (sorry if it is naive): Why can't the overlay =
tunnel just be extended to the server? This somehow implies that OVRLY_OUT =
would be kind of co-located with the server - obviously, there can be furth=
er routers/overlay nodes in between.

I am asking this because processing the information contained in the TCP op=
tion will require anyway a modified TCP stack in the server, i. e., the ser=
ver will not be fully backward compatible if it has to process the proposed=
 option. But if the TCP/IP stack has to be modified anyway, I could imagine=
 that one could just add to the server whatever encap/decap is required for=
 the overlay transport. Then, I have the impression that the proposed TCP o=
ption would not be needed at all.

I don't want to dig into the overlay design, because this is not really in =
scope of TCPM. But if there is a system architecture that does not require =
adding TCP options in middleboxes, thus affecting TCP end-to-end semantics,=
 it would really be important to understand why such an architecture cannot=
 be used.

Thanks

Michael


=20
> The issue of the proposed fast-open scheme is one that we=20
> have not considered, but I don't think it adds any problems=20
> for the TCP option that aren't already a problem for tunneled=20
> connectivity in general. I will have to spend some time with=20
> that proposal and think about how they interrelate.
>=20
> --Brandon
>=20
> On 12/21/2012 08:34 AM, Scharf, Michael (Michael) wrote:
> > Brandon,
> >
> >>> If there were tunnels between the OVRLY_IN and OVERLY_OUT
> >> boxes, then
> >>> the inner IP headers would have the HOST_X and SERVER
> >> addresses, and
> >>> the outer ones in the tunnel would have the overlay=20
> headers.  Since=20
> >>> the inner packets would be delivered ultimately after=20
> egressing the=20
> >>> tunnels, the HOST_X addresses are totally visible to the
> >> server, and
> >>> vice versa.
> >>
> >> There are indeed tunnels between OVRLY_IN and OVRLY_OUT, and the=20
> >> inner IP headers will typically use either the client-side=20
> addresses=20
> >> or the server-side addresses. However, neither OVRLY_IN=20
> nor OVRLY_OUT=20
> >> can be assumed to be reliably in-path between HOST and=20
> SERVER, which=20
> >> means that internet routing cannot be relied upon to cause=20
> packets to=20
> >> arrive at the overlay ingress. Instead, HOST_1 must=20
> directly address
> >> OVRLY_IN_1 in order to send its packets into the tunnel,=20
> and SERVER=20
> >> must directly address OVRLY_OUT in order to send the=20
> return traffic=20
> >> into the tunnel.
> >
> > Thanks for this explanation - this indeed helps to=20
> understand the architecture. But actually I still don't fully=20
> understand the motivation of bypassing Internet routing this=20
> way. As a non-expert on routing, it indeed looks to me like=20
> reinventing source routing - but this is outside my core expertise.
> >
> > Regarding TCPM's business: If I correctly understand the approach,=20
> > OVRLY_IN will "transparently" add and remove TCP options.=20
> This is kind=20
> > of dangerous from an end-to-end perspective... Sorry if=20
> that has been=20
> > answered before, but I really wonder what to do if OVRLY_IN=20
> can't add=20
> > this option, either because of lack of TCP option space, or because=20
> > the path MTU is exceeded by the resulting IP packet. (In=20
> fact, I think=20
> > that this problem does not apply to TCP options only.)
> >
> > Unless I miss something, the latter case could become much=20
> more relevant soon: TCPM currently works on the fast-open=20
> scheme that adds data to SYNs. With that, I think it is=20
> possible that all data packets from a sender to a receiver=20
> are either full sized or large enough that the proposed=20
> option does not fit in. Given that this option can include=20
> full-sized IPv6 addresses, this likelihood is much larger=20
> than for other existing TCP option, right?
> >
> > In some cases, I believe that the proposed TCP option=20
> cannot be added in the overlay without either IP=20
> fragmentation, which is unlikely to be a good idea with NATs,=20
> or TCP segment splitting, which probably can cause harm as=20
> well. For instance, what would OVRLY_IN do if it receives an=20
> IP packet with a TCP SYN segment that already sums up to 1500=20
> byte? And, to make the scenario more nasty, if the same=20
> applies to the first data segments as well?
> >
> > Thanks
> >
> > Michael
> >
>=20
> --
> Brandon Williams; Principal Software Engineer Cloud=20
> Engineering; Akamai Technologies Inc.
> =

From brandon.williams@akamai.com  Fri Dec 21 12:07:21 2012
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Date: Fri, 21 Dec 2012 15:07:19 -0500
From: Brandon Williams <brandon.williams@akamai.com>
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References: <20121220165533.11066.52759.idtracker@ietfa.amsl.com> <50D348FE.3060807@akamai.com>	<50D363FC.9010906@mti-systems.com> <50D379D7.609@akamai.com>	<50D37D79.20109@mti-systems.com> <50D38B80.7080409@akamai.com> <2A886F9088894347A3BE0CC5B7A85F3E9AA0F0AE17@FRMRSSXCHMBSE3.dc-m.alcatel-lucent.com> <50D48EF1.2030107@akamai.com> <2A886F9088894347A3BE0CC5B7A85F3E9AA0F0AE1E@FRMRSSXCHMBSE3.dc-m.alcatel-lucent.com>
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Cc: "int-area@ietf.org" <int-area@ietf.org>
Subject: Re: [tcpm] draft-williams-overlaypath-ip-tcp-rfc
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Michael,

Extending the overlay all the way to the application server would mean 
that existing solutions for load balancing, SSL offload, intrusion 
detection, diagnostic logging, etc. would not work. In other words, 
there are many systems in a common enterprise environment that would 
benefit from more accurate host identification, and all would require 
changes in order for the mechanism to work. On the other hand, there is 
existing middleware that can already handle an arbitrary tcp option, 
using its value for the above listed purposes. So using a TCP option for 
this purpose is deployable today, but extending the overlay is not.

At the same time, use of the option does not carry significant risk of 
breaking existing connectivity, even in cases where the option is not 
understood by the TCP stack. Testing has shown that only about 1.7% of 
the top 100,000 web servers fail to establish connections when the 
option is included (see draft-abdo-hostid-tcpopt-implementation). This 
is mostly likely a characteristic of the common TCP stacks in use today, 
and so probably extends to non-HTTP application servers, too.

--Brandon


On 12/21/2012 02:14 PM, Scharf, Michael (Michael) wrote:
> Brandon,
>
>> Your are correct that the option could be problematic if
>> added to a full-sized packet, or even a nearly full one. I
>> can see that the document should have some discussion of this issue.
>
> Yes.
>
>> In a case like ours, where the overlay network uses
>> tunneling, transparently adding the option is not a critical
>> problem to be solved.
>> It is already the case that the overlay entry point must
>> advertise a reduced MSS in order to accommodate the tunnel
>> overhead. The amount of space consumed by the option will
>> always be smaller than the tunnel overhead, and the option
>> can be added at OVRLY_OUT, so the two are not additive. That
>> said, I can see that an overlay network that does not use
>> tunnels internally, or one that in fact does apply the option
>> on OVRLY_IN, would have a bigger problem, though.
>
> So, the new TCP option is basically required between OVRLY_OUT and the receiver/server, because the relevant information is already somehow transported in the overlay, right?
>
> This raises another question (sorry if it is naive): Why can't the overlay tunnel just be extended to the server? This somehow implies that OVRLY_OUT would be kind of co-located with the server - obviously, there can be further routers/overlay nodes in between.
>
> I am asking this because processing the information contained in the TCP option will require anyway a modified TCP stack in the server, i. e., the server will not be fully backward compatible if it has to process the proposed option. But if the TCP/IP stack has to be modified anyway, I could imagine that one could just add to the server whatever encap/decap is required for the overlay transport. Then, I have the impression that the proposed TCP option would not be needed at all.
>
> I don't want to dig into the overlay design, because this is not really in scope of TCPM. But if there is a system architecture that does not require adding TCP options in middleboxes, thus affecting TCP end-to-end semantics, it would really be important to understand why such an architecture cannot be used.
>
> Thanks
>
> Michael
>
>
>
>> The issue of the proposed fast-open scheme is one that we
>> have not considered, but I don't think it adds any problems
>> for the TCP option that aren't already a problem for tunneled
>> connectivity in general. I will have to spend some time with
>> that proposal and think about how they interrelate.
>>
>> --Brandon
>>
>> On 12/21/2012 08:34 AM, Scharf, Michael (Michael) wrote:
>>> Brandon,
>>>
>>>>> If there were tunnels between the OVRLY_IN and OVERLY_OUT
>>>> boxes, then
>>>>> the inner IP headers would have the HOST_X and SERVER
>>>> addresses, and
>>>>> the outer ones in the tunnel would have the overlay
>> headers.  Since
>>>>> the inner packets would be delivered ultimately after
>> egressing the
>>>>> tunnels, the HOST_X addresses are totally visible to the
>>>> server, and
>>>>> vice versa.
>>>>
>>>> There are indeed tunnels between OVRLY_IN and OVRLY_OUT, and the
>>>> inner IP headers will typically use either the client-side
>> addresses
>>>> or the server-side addresses. However, neither OVRLY_IN
>> nor OVRLY_OUT
>>>> can be assumed to be reliably in-path between HOST and
>> SERVER, which
>>>> means that internet routing cannot be relied upon to cause
>> packets to
>>>> arrive at the overlay ingress. Instead, HOST_1 must
>> directly address
>>>> OVRLY_IN_1 in order to send its packets into the tunnel,
>> and SERVER
>>>> must directly address OVRLY_OUT in order to send the
>> return traffic
>>>> into the tunnel.
>>>
>>> Thanks for this explanation - this indeed helps to
>> understand the architecture. But actually I still don't fully
>> understand the motivation of bypassing Internet routing this
>> way. As a non-expert on routing, it indeed looks to me like
>> reinventing source routing - but this is outside my core expertise.
>>>
>>> Regarding TCPM's business: If I correctly understand the approach,
>>> OVRLY_IN will "transparently" add and remove TCP options.
>> This is kind
>>> of dangerous from an end-to-end perspective... Sorry if
>> that has been
>>> answered before, but I really wonder what to do if OVRLY_IN
>> can't add
>>> this option, either because of lack of TCP option space, or because
>>> the path MTU is exceeded by the resulting IP packet. (In
>> fact, I think
>>> that this problem does not apply to TCP options only.)
>>>
>>> Unless I miss something, the latter case could become much
>> more relevant soon: TCPM currently works on the fast-open
>> scheme that adds data to SYNs. With that, I think it is
>> possible that all data packets from a sender to a receiver
>> are either full sized or large enough that the proposed
>> option does not fit in. Given that this option can include
>> full-sized IPv6 addresses, this likelihood is much larger
>> than for other existing TCP option, right?
>>>
>>> In some cases, I believe that the proposed TCP option
>> cannot be added in the overlay without either IP
>> fragmentation, which is unlikely to be a good idea with NATs,
>> or TCP segment splitting, which probably can cause harm as
>> well. For instance, what would OVRLY_IN do if it receives an
>> IP packet with a TCP SYN segment that already sums up to 1500
>> byte? And, to make the scenario more nasty, if the same
>> applies to the first data segments as well?
>>>
>>> Thanks
>>>
>>> Michael
>>>
>>
>> --
>> Brandon Williams; Principal Software Engineer Cloud
>> Engineering; Akamai Technologies Inc.

-- 
Brandon Williams; Principal Software Engineer
Cloud Engineering; Akamai Technologies Inc.

-- 
Brandon Williams; Principal Software Engineer
Cloud Engineering; Akamai Technologies Inc.

From michawe@ifi.uio.no  Sat Dec 29 04:21:43 2012
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From: Michael Welzl <michawe@ifi.uio.no>
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Subject: [tcpm] About the SYN/ACK-timer
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Dear all,

In my opinion, the web (and with it, probably many other applications =
too) exhibits a somewhat annoying problem that we still haven't fixed, =
but just gotten used to:
sometimes, when I click something, it doesn't immediately work. So I =
stop the transfer and click again. Then it works.

Someone once told me that, when this happens, it's probably because a =
SYN or SYN/ACK was lost. I believe this is correct. Several years ago, =
this notion has prompted me to do some measuring + ugly fixing:
http://heim.ifi.uio.no/~michawe/research/tools/syn-retransmit/index.html
=85in doing so, we found that, indeed, the problem is significant (in =
terms of number of SYNs or SYN/ACKs lost). It's probably not hard to =
find more such data - I guess that Google has produced it when they =
first worked on Fast Open  (sorry for not checking right now).

When I approached TCPM with a suggestion to reduce the timers for SYN =
and SYN/ACK retransmission, I ended up with a dialogue with Joe Touch, =
in which he convinced me that reducing the SYN timer could be harmful, =
as it could cause busy servers to get hammered by even more SYNs when =
they just can't cope with them. I buy that; but IIRC he had no good =
reason against making the SYN/ACK-timer more aggressive. BTW sorry if =
I'm using a wrong name here, but I guess it's clear: I mean the timer =
used by the server to resend the SYN/ACK if it doesn't get a response =
for a while.

So, the thought occurs to me every once in a while (especially when I do =
surf the web, and something temporarily hangs), and just now I felt like =
sharing it again: why don't we really make the SYN/ACK timer more =
aggressive? This should for instance appeal to the Googlers on the list =
who are trying to improve TCP for their cause, no?

I think if it's about establishing that the problem is significant, that =
shouldn't be too hard. Measurements show that SYN/ACKs do get lost, and =
when they do, the user feels it. Sure, it's not the worst problem in the =
world, but I do believe it's worth fixing.

I'm just curious what everyone thinks=85 maybe I'm completely off track =
for one reason or another?!

Cheers,
Michael


From mellia@tlc.polito.it  Sat Dec 29 05:55:00 2012
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>
Subject: Re: [tcpm] About the SYN/ACK-timer
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Hi Michael,

below you can find some numbers, derived from passive measurements of =
about 20k ADSL customers, during 1h @peak time. It seems that the 1st =
syn lost affects mostly P2P traffic... for HTTP, it's less than 0.7%. =
Heavy hitters do not suffer for that a lot. The analysis of the most =
affected http servers seems to point to server congestion too.
[We can provide more numbers in the context of the mPlane project =
(www.ict-mplane.eu) if interested.]

My 2c is that 3s of initial timeout is possibly conservative. However, =
lowering it to 1s will not change user experience a lot.
Below 1s will cause lot of false retransmissions (especially on wireless =
access scenarios where RTT can easily grow above 1s for the first =
packet...). More complicated solutions can be engineered (I did =
something on this line 5/6y ago...). For example, choosing the initial =
RTO based on RTT history, or access network could be an option. Still, =
I'm not sure that it is worth the complexity.

Nice 2013,
Ciao
M
----------
#Protocol    #connections  #multiSYN  %multiple  SYN
UNKNOWN     524112        29299      5.59022
HTTP        1834776       12154      0.662424
P2P-plain   766858        62290      8.12276
BT-tracker  31533         1682       5.33409
SMTP        1682          3          0.178359
POP3        16107         39         0.242131
SSL/TLS     422609        2704       0.639835
P2P-obf     270055        19401      7.18409
RTMP        2255          13         0.576497
BT-obf      266951        21866      8.19102


Among HTTP connections, these are numbers for the top 10 domains =
names...
#Domain name                    #connections  #multiSYN  %multiple  SYN
apps.facebook.com              10014         34         0.339525
pagead2.googlesyndication.com  10728         22         0.205071
ad.yieldmanager.com            10944         25         0.228436
googleads.g.doubleclick.net    11976         32         0.267201
external.ak.fbcdn.net          12285         20         0.1628
s-static.ak.facebook.com       13298         39         0.293277
s-static.ak.fbcdn.net          14642         39         0.266357
www.google.com                 14740         184        1.2483
www.google-analytics.com       15755         38         0.241193
fbcdn-photos-a.akamaihd.net    15932         34         0.213407
secure-it.imrworldwide.com     16229         56         0.345061
www.google.it                  17158         75         0.437114
static.ak.fbcdn.net            22751         62         0.272515
fbcdn-profile-a.akamaihd.net   22892         65         0.283942
profile.ak.fbcdn.net           36777         116        0.315415
www.facebook.com               94317         574        0.608586

--=20
Marco Mellia - Assistant Professor
Dipartimento di Elettronica e Telecomunicazioni
Politecnico di Torino
Corso Duca Degli Abruzzi 24
10129 - Torino - IT
Tel: +39-011-090-4173
Cel: +39-331-6714789
Skype: mgmellia
Home page: http://www.tlc-networks.polito.it/mellia

Il giorno 29/dic/2012, alle ore 13:21, Michael Welzl =
<michawe@ifi.uio.no> ha scritto:

> Dear all,
>=20
> In my opinion, the web (and with it, probably many other applications =
too) exhibits a somewhat annoying problem that we still haven't fixed, =
but just gotten used to:
> sometimes, when I click something, it doesn't immediately work. So I =
stop the transfer and click again. Then it works.
>=20
> Someone once told me that, when this happens, it's probably because a =
SYN or SYN/ACK was lost. I believe this is correct. Several years ago, =
this notion has prompted me to do some measuring + ugly fixing:
> =
http://heim.ifi.uio.no/~michawe/research/tools/syn-retransmit/index.html
> =85in doing so, we found that, indeed, the problem is significant (in =
terms of number of SYNs or SYN/ACKs lost). It's probably not hard to =
find more such data - I guess that Google has produced it when they =
first worked on Fast Open  (sorry for not checking right now).
>=20
> When I approached TCPM with a suggestion to reduce the timers for SYN =
and SYN/ACK retransmission, I ended up with a dialogue with Joe Touch, =
in which he convinced me that reducing the SYN timer could be harmful, =
as it could cause busy servers to get hammered by even more SYNs when =
they just can't cope with them. I buy that; but IIRC he had no good =
reason against making the SYN/ACK-timer more aggressive. BTW sorry if =
I'm using a wrong name here, but I guess it's clear: I mean the timer =
used by the server to resend the SYN/ACK if it doesn't get a response =
for a while.
>=20
> So, the thought occurs to me every once in a while (especially when I =
do surf the web, and something temporarily hangs), and just now I felt =
like sharing it again: why don't we really make the SYN/ACK timer more =
aggressive? This should for instance appeal to the Googlers on the list =
who are trying to improve TCP for their cause, no?
>=20
> I think if it's about establishing that the problem is significant, =
that shouldn't be too hard. Measurements show that SYN/ACKs do get lost, =
and when they do, the user feels it. Sure, it's not the worst problem in =
the world, but I do believe it's worth fixing.
>=20
> I'm just curious what everyone thinks=85 maybe I'm completely off =
track for one reason or another?!
>=20
> Cheers,
> Michael
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Hi =
Michael,<div><br></div><div>below you can find some numbers, derived =
from passive measurements of about 20k ADSL customers, during 1h @peak =
time. It seems that the 1st syn lost affects mostly P2P traffic... for =
HTTP, it's less than 0.7%. Heavy hitters do not suffer for that a =
lot.&nbsp;The analysis of the most affected http servers seems to point =
to server congestion too.</div><div>[We can provide more numbers in the =
context of the mPlane project (<a =
href=3D"http://www.ict-mplane.eu">www.ict-mplane.eu</a>) if =
interested.]</div><div><br></div><div>My 2c is that 3s of initial =
timeout is possibly conservative. However, lowering it to 1s will not =
change user experience a lot.</div><div>Below 1s will cause lot of false =
retransmissions (especially on wireless access scenarios where RTT can =
easily grow above 1s for the first packet...). More complicated =
solutions can be engineered (I did something on this line 5/6y ago...). =
For example, choosing the initial RTO based on RTT history, or access =
network could be an option. Still, I'm not sure that it is worth the =
complexity.</div><div><br></div><div>Nice =
2013,</div><div>Ciao</div><div>M</div><div><font face=3D"Courier =
New">----------</font></div><div><font face=3D"Courier New">#Protocol =
&nbsp; &nbsp;#connections &nbsp;#multiSYN &nbsp;%multiple =
&nbsp;SYN</font></div><div><font face=3D"Courier New">UNKNOWN &nbsp; =
&nbsp; 524112 &nbsp; &nbsp; &nbsp; &nbsp;29299 &nbsp; &nbsp; =
&nbsp;5.59022</font></div><div><font face=3D"Courier New">HTTP &nbsp; =
&nbsp; &nbsp; &nbsp;1834776 &nbsp; &nbsp; &nbsp; 12154 &nbsp; &nbsp; =
&nbsp;0.662424</font></div><div><font face=3D"Courier New">P2P-plain =
&nbsp; 766858 &nbsp; &nbsp; &nbsp; &nbsp;62290 &nbsp; &nbsp; =
&nbsp;8.12276</font></div><div><font face=3D"Courier New">BT-tracker =
&nbsp;31533 &nbsp; &nbsp; &nbsp; &nbsp; 1682 &nbsp; &nbsp; &nbsp; =
5.33409</font></div><div><font face=3D"Courier New">SMTP &nbsp; &nbsp; =
&nbsp; &nbsp;1682 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;3 &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp;0.178359</font></div><div><font face=3D"Courier =
New">POP3 &nbsp; &nbsp; &nbsp; &nbsp;16107 &nbsp; &nbsp; &nbsp; &nbsp; =
39 &nbsp; &nbsp; &nbsp; &nbsp; 0.242131</font></div><div><font =
face=3D"Courier New">SSL/TLS &nbsp; &nbsp; 422609 &nbsp; &nbsp; &nbsp; =
&nbsp;2704 &nbsp; &nbsp; &nbsp; 0.639835</font></div><div><font =
face=3D"Courier New">P2P-obf &nbsp; &nbsp; 270055 &nbsp; &nbsp; &nbsp; =
&nbsp;19401 &nbsp; &nbsp; &nbsp;7.18409</font></div><div><font =
face=3D"Courier New">RTMP &nbsp; &nbsp; &nbsp; &nbsp;2255 &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp;13 &nbsp; &nbsp; &nbsp; &nbsp; =
0.576497</font></div><div><font face=3D"Courier New">BT-obf &nbsp; =
&nbsp; &nbsp;266951 &nbsp; &nbsp; &nbsp; &nbsp;21866 &nbsp; &nbsp; =
&nbsp;8.19102</font></div><div><br></div><div><br></div><div>Among HTTP =
connections, these are numbers for the top 10 domains =
names...</div><div><div><font face=3D"Courier New">#Domain name &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;#connections &nbsp;#multiSYN &nbsp;%multiple =
&nbsp;SYN</font></div><div><font face=3D"Courier New"><a =
href=3D"http://apps.facebook.com">apps.facebook.com</a> &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;10014 &nbsp; &nbsp; &nbsp; &nbsp; 34 =
&nbsp; &nbsp; &nbsp; &nbsp; 0.339525</font></div><div><font =
face=3D"Courier New"><a =
href=3D"http://pagead2.googlesyndication.com">pagead2.googlesyndication.co=
m</a> &nbsp;10728 &nbsp; &nbsp; &nbsp; &nbsp; 22 &nbsp; &nbsp; &nbsp; =
&nbsp; 0.205071</font></div><div><font face=3D"Courier New"><a =
href=3D"http://ad.yieldmanager.com">ad.yieldmanager.com</a> &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;10944 &nbsp; &nbsp; &nbsp; &nbsp; 25 =
&nbsp; &nbsp; &nbsp; &nbsp; 0.228436</font></div><div><font =
face=3D"Courier New"><a =
href=3D"http://googleads.g.doubleclick.net">googleads.g.doubleclick.net</a=
> &nbsp; &nbsp;11976 &nbsp; &nbsp; &nbsp; &nbsp; 32 &nbsp; &nbsp; &nbsp; =
&nbsp; 0.267201</font></div><div><font face=3D"Courier New"><a =
href=3D"http://external.ak.fbcdn.net">external.ak.fbcdn.net</a> &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp;12285 &nbsp; &nbsp; &nbsp; &nbsp; 20 &nbsp; =
&nbsp; &nbsp; &nbsp; 0.1628</font></div><div><font face=3D"Courier =
New"><a =
href=3D"http://s-static.ak.facebook.com">s-static.ak.facebook.com</a> =
&nbsp; &nbsp; &nbsp; 13298 &nbsp; &nbsp; &nbsp; &nbsp; 39 &nbsp; &nbsp; =
&nbsp; &nbsp; 0.293277</font></div><div><font face=3D"Courier New"><a =
href=3D"http://s-static.ak.fbcdn.net">s-static.ak.fbcdn.net</a> &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp;14642 &nbsp; &nbsp; &nbsp; &nbsp; 39 &nbsp; =
&nbsp; &nbsp; &nbsp; 0.266357</font></div><div><font face=3D"Courier =
New"><a href=3D"http://www.google.com">www.google.com</a> &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 14740 &nbsp; &nbsp; &nbsp; =
&nbsp; 184 &nbsp; &nbsp; &nbsp; &nbsp;1.2483</font></div><div><font =
face=3D"Courier New"><a =
href=3D"http://www.google-analytics.com">www.google-analytics.com</a> =
&nbsp; &nbsp; &nbsp; 15755 &nbsp; &nbsp; &nbsp; &nbsp; 38 &nbsp; &nbsp; =
&nbsp; &nbsp; 0.241193</font></div><div><font face=3D"Courier New"><a =
href=3D"http://fbcdn-photos-a.akamaihd.net">fbcdn-photos-a.akamaihd.net</a=
> &nbsp; &nbsp;15932 &nbsp; &nbsp; &nbsp; &nbsp; 34 &nbsp; &nbsp; &nbsp; =
&nbsp; 0.213407</font></div><div><font face=3D"Courier New"><a =
href=3D"http://secure-it.imrworldwide.com">secure-it.imrworldwide.com</a> =
&nbsp; &nbsp; 16229 &nbsp; &nbsp; &nbsp; &nbsp; 56 &nbsp; &nbsp; &nbsp; =
&nbsp; 0.345061</font></div><div><font face=3D"Courier New"><a =
href=3D"http://www.google.it">www.google.it</a> &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;17158 &nbsp; &nbsp; &nbsp; =
&nbsp; 75 &nbsp; &nbsp; &nbsp; &nbsp; 0.437114</font></div><div><font =
face=3D"Courier New"><a =
href=3D"http://static.ak.fbcdn.net">static.ak.fbcdn.net</a> &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;22751 &nbsp; &nbsp; &nbsp; &nbsp; 62 =
&nbsp; &nbsp; &nbsp; &nbsp; 0.272515</font></div><div><font =
face=3D"Courier New"><a =
href=3D"http://fbcdn-profile-a.akamaihd.net">fbcdn-profile-a.akamaihd.net<=
/a> &nbsp; 22892 &nbsp; &nbsp; &nbsp; &nbsp; 65 &nbsp; &nbsp; &nbsp; =
&nbsp; 0.283942</font></div><div><font face=3D"Courier New"><a =
href=3D"http://profile.ak.fbcdn.net">profile.ak.fbcdn.net</a> &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; 36777 &nbsp; &nbsp; &nbsp; &nbsp; 116 &nbsp; =
&nbsp; &nbsp; &nbsp;0.315415</font></div><div><font face=3D"Courier =
New"><a href=3D"http://www.facebook.com">www.facebook.com</a> &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 94317 &nbsp; &nbsp; &nbsp; =
&nbsp; 574 &nbsp; &nbsp; &nbsp; =
&nbsp;0.608586</font></div><div><br></div><div>--&nbsp;</div></div><div><d=
iv><span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
border-spacing: 0px; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; 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-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; 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-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; 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-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; 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-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; 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-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div>Marco Mellia =
-&nbsp;Assistant Professor</div><div>Dipartimento di Elettronica e =
Telecomunicazioni</div><div>Politecnico di Torino</div><div>Corso Duca =
Degli Abruzzi 24</div><div>10129 - Torino - IT</div><div>Tel: =
+39-011-090-4173</div><div>Cel: +39-331-6714789</div><div>Skype: =
mgmellia</div><div>Home page: <a =
href=3D"http://www.tlc-networks.polito.it/mellia">http://www.tlc-networks.=
polito.it/mellia</a></div></div></span></div></span></div></span></div></s=
pan></div></span></span>
</div>
<br><div><div>Il giorno 29/dic/2012, alle ore 13:21, Michael Welzl =
&lt;<a href=3D"mailto:michawe@ifi.uio.no">michawe@ifi.uio.no</a>&gt; ha =
scritto:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite">Dear all,<br><br>In my opinion, the web (and with it, =
probably many other applications too) exhibits a somewhat annoying =
problem that we still haven't fixed, but just gotten used =
to:<br>sometimes, when I click something, it doesn't immediately work. =
So I stop the transfer and click again. Then it works.<br><br>Someone =
once told me that, when this happens, it's probably because a SYN or =
SYN/ACK was lost. I believe this is correct. Several years ago, this =
notion has prompted me to do some measuring + ugly fixing:<br><a =
href=3D"http://heim.ifi.uio.no/~michawe/research/tools/syn-retransmit/inde=
x.html">http://heim.ifi.uio.no/~michawe/research/tools/syn-retransmit/inde=
x.html</a><br>=85in doing so, we found that, indeed, the problem is =
significant (in terms of number of SYNs or SYN/ACKs lost). It's probably =
not hard to find more such data - I guess that Google has produced it =
when they first worked on Fast Open &nbsp;(sorry for not checking right =
now).<br><br>When I approached TCPM with a suggestion to reduce the =
timers for SYN and SYN/ACK retransmission, I ended up with a dialogue =
with Joe Touch, in which he convinced me that reducing the SYN timer =
could be harmful, as it could cause busy servers to get hammered by even =
more SYNs when they just can't cope with them. I buy that; but IIRC he =
had no good reason against making the SYN/ACK-timer more aggressive. BTW =
sorry if I'm using a wrong name here, but I guess it's clear: I mean the =
timer used by the server to resend the SYN/ACK if it doesn't get a =
response for a while.<br><br>So, the thought occurs to me every once in =
a while (especially when I do surf the web, and something temporarily =
hangs), and just now I felt like sharing it again: why don't we really =
make the SYN/ACK timer more aggressive? This should for instance appeal =
to the Googlers on the list who are trying to improve TCP for their =
cause, no?<br><br>I think if it's about establishing that the problem is =
significant, that shouldn't be too hard. Measurements show that SYN/ACKs =
do get lost, and when they do, the user feels it. Sure, it's not the =
worst problem in the world, but I do believe it's worth =
fixing.<br><br>I'm just curious what everyone thinks=85 maybe I'm =
completely off track for one reason or =
another?!<br><br>Cheers,<br>Michael<br><br>_______________________________=
________________<br>tcpm mailing =
list<br>tcpm@ietf.org<br>https://www.ietf.org/mailman/listinfo/tcpm<br></b=
lockquote></div><br></div></body></html>=

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From: Jakob Heitz <jakob.heitz@ericsson.com>
To: Michael Welzl <michawe@ifi.uio.no>
Thread-Topic: [tcpm] About the SYN/ACK-timer
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Date: Sat, 29 Dec 2012 18:19:46 +0000
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Subject: Re: [tcpm] About the SYN/ACK-timer
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From michawe@ifi.uio.no  Sat Dec 29 10:35:10 2012
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>
Subject: Re: [tcpm] About the SYN/ACK-timer
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Hi,

The percentage matches what we found in our measurements (around 0.5%). =
However I don't think that's so small? It's not "occurrences per user =
per server", it's "occurrences in total", i.e. across all connections =
that are opened to the server=85 some websites cause tons of connections =
to be opened, so you have to multiply with the number of such =
connections to get the chance of such a problem happening to you when =
you visit a website

I agree that more complicated solutions may not be worth the complexity. =
I wonder about smaller values, like 1s as you say. Is the increased =
number of potential unnecessary retransmits a big problem? Dunno=85

Cheers,
Michael


On Dec 29, 2012, at 2:54 PM, Marco Mellia <mellia@tlc.polito.it> wrote:

> Hi Michael,
>=20
> below you can find some numbers, derived from passive measurements of =
about 20k ADSL customers, during 1h @peak time. It seems that the 1st =
syn lost affects mostly P2P traffic... for HTTP, it's less than 0.7%. =
Heavy hitters do not suffer for that a lot. The analysis of the most =
affected http servers seems to point to server congestion too.
> [We can provide more numbers in the context of the mPlane project =
(www.ict-mplane.eu) if interested.]
>=20
> My 2c is that 3s of initial timeout is possibly conservative. However, =
lowering it to 1s will not change user experience a lot.
> Below 1s will cause lot of false retransmissions (especially on =
wireless access scenarios where RTT can easily grow above 1s for the =
first packet...). More complicated solutions can be engineered (I did =
something on this line 5/6y ago...). For example, choosing the initial =
RTO based on RTT history, or access network could be an option. Still, =
I'm not sure that it is worth the complexity.
>=20
> Nice 2013,
> Ciao
> M
> ----------
> #Protocol    #connections  #multiSYN  %multiple  SYN
> UNKNOWN     524112        29299      5.59022
> HTTP        1834776       12154      0.662424
> P2P-plain   766858        62290      8.12276
> BT-tracker  31533         1682       5.33409
> SMTP        1682          3          0.178359
> POP3        16107         39         0.242131
> SSL/TLS     422609        2704       0.639835
> P2P-obf     270055        19401      7.18409
> RTMP        2255          13         0.576497
> BT-obf      266951        21866      8.19102
>=20
>=20
> Among HTTP connections, these are numbers for the top 10 domains =
names...
> #Domain name                    #connections  #multiSYN  %multiple  =
SYN
> apps.facebook.com              10014         34         0.339525
> pagead2.googlesyndication.com  10728         22         0.205071
> ad.yieldmanager.com            10944         25         0.228436
> googleads.g.doubleclick.net    11976         32         0.267201
> external.ak.fbcdn.net          12285         20         0.1628
> s-static.ak.facebook.com       13298         39         0.293277
> s-static.ak.fbcdn.net          14642         39         0.266357
> www.google.com                 14740         184        1.2483
> www.google-analytics.com       15755         38         0.241193
> fbcdn-photos-a.akamaihd.net    15932         34         0.213407
> secure-it.imrworldwide.com     16229         56         0.345061
> www.google.it                  17158         75         0.437114
> static.ak.fbcdn.net            22751         62         0.272515
> fbcdn-profile-a.akamaihd.net   22892         65         0.283942
> profile.ak.fbcdn.net           36777         116        0.315415
> www.facebook.com               94317         574        0.608586
>=20
> --=20
> Marco Mellia - Assistant Professor
> Dipartimento di Elettronica e Telecomunicazioni
> Politecnico di Torino
> Corso Duca Degli Abruzzi 24
> 10129 - Torino - IT
> Tel: +39-011-090-4173
> Cel: +39-331-6714789
> Skype: mgmellia
> Home page: http://www.tlc-networks.polito.it/mellia
>=20
> Il giorno 29/dic/2012, alle ore 13:21, Michael Welzl =
<michawe@ifi.uio.no> ha scritto:
>=20
>> Dear all,
>>=20
>> In my opinion, the web (and with it, probably many other applications =
too) exhibits a somewhat annoying problem that we still haven't fixed, =
but just gotten used to:
>> sometimes, when I click something, it doesn't immediately work. So I =
stop the transfer and click again. Then it works.
>>=20
>> Someone once told me that, when this happens, it's probably because a =
SYN or SYN/ACK was lost. I believe this is correct. Several years ago, =
this notion has prompted me to do some measuring + ugly fixing:
>> =
http://heim.ifi.uio.no/~michawe/research/tools/syn-retransmit/index.html
>> =85in doing so, we found that, indeed, the problem is significant (in =
terms of number of SYNs or SYN/ACKs lost). It's probably not hard to =
find more such data - I guess that Google has produced it when they =
first worked on Fast Open  (sorry for not checking right now).
>>=20
>> When I approached TCPM with a suggestion to reduce the timers for SYN =
and SYN/ACK retransmission, I ended up with a dialogue with Joe Touch, =
in which he convinced me that reducing the SYN timer could be harmful, =
as it could cause busy servers to get hammered by even more SYNs when =
they just can't cope with them. I buy that; but IIRC he had no good =
reason against making the SYN/ACK-timer more aggressive. BTW sorry if =
I'm using a wrong name here, but I guess it's clear: I mean the timer =
used by the server to resend the SYN/ACK if it doesn't get a response =
for a while.
>>=20
>> So, the thought occurs to me every once in a while (especially when I =
do surf the web, and something temporarily hangs), and just now I felt =
like sharing it again: why don't we really make the SYN/ACK timer more =
aggressive? This should for instance appeal to the Googlers on the list =
who are trying to improve TCP for their cause, no?
>>=20
>> I think if it's about establishing that the problem is significant, =
that shouldn't be too hard. Measurements show that SYN/ACKs do get lost, =
and when they do, the user feels it. Sure, it's not the worst problem in =
the world, but I do believe it's worth fixing.
>>=20
>> I'm just curious what everyone thinks=85 maybe I'm completely off =
track for one reason or another?!
>>=20
>> Cheers,
>> Michael
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


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Subject: Re: [tcpm] About the SYN/ACK-timer
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On Dec 29, 2012, at 7:19 PM, Jakob Heitz <jakob.heitz@ericsson.com> =
wrote:

> There is no need to fix this in TCP. The browser could fix it.
> The browser could start a new TCP connection every ~250mS.
> Then use whichever TCP connection that completes first.
> It could keep the remaining TCP sessions open, just in case it needs =
to get more objects or pages from the same web site.
>=20
> Now, there is nothing stopping browsers from doing such workarounds...
> ...except a fix in TCP. Why is a SYN retransmission timer of 100mS so =
bad, even if it does put a few more packets into the network?

Not sure if it really would be that bad, not sure at all.
Anyway, my main point was that there's probably even less harm in making =
the SYN/ACK timer more aggressive, so why not do at least that?

Cheers,
Michael


>=20
> --
> Jakob Heitz.
>=20
> On Dec 29, 2012, at 4:21 AM, "Michael Welzl" <michawe@ifi.uio.no> =
wrote:
>=20
>> Dear all,
>>=20
>> In my opinion, the web (and with it, probably many other applications =
too) exhibits a somewhat annoying problem that we still haven't fixed, =
but just gotten used to:
>> sometimes, when I click something, it doesn't immediately work. So I =
stop the transfer and click again. Then it works.
>>=20
>> Someone once told me that, when this happens, it's probably because a =
SYN or SYN/ACK was lost. I believe this is correct. Several years ago, =
this notion has prompted me to do some measuring + ugly fixing:
>> =
http://heim.ifi.uio.no/~michawe/research/tools/syn-retransmit/index.html
>> =A1=ADin doing so, we found that, indeed, the problem is significant =
(in terms of number of SYNs or SYN/ACKs lost). It's probably not hard to =
find more such data - I guess that Google has produced it when they =
first worked on Fast Open  (sorry for not checking right now).
>>=20
>> When I approached TCPM with a suggestion to reduce the timers for SYN =
and SYN/ACK retransmission, I ended up with a dialogue with Joe Touch, =
in which he convinced me that reducing the SYN timer could be harmful, =
as it could cause busy servers to get hammered by even more SYNs when =
they just can't cope with them. I buy that; but IIRC he had no good =
reason against making the SYN/ACK-timer more aggressive. BTW sorry if =
I'm using a wrong name here, but I guess it's clear: I mean the timer =
used by the server to resend the SYN/ACK if it doesn't get a response =
for a while.
>>=20
>> So, the thought occurs to me every once in a while (especially when I =
do surf the web, and something temporarily hangs), and just now I felt =
like sharing it again: why don't we really make the SYN/ACK timer more =
aggressive? This should for instance appeal to the Googlers on the list =
who are trying to improve TCP for their cause, no?
>>=20
>> I think if it's about establishing that the problem is significant, =
that shouldn't be too hard. Measurements show that SYN/ACKs do get lost, =
and when they do, the user feels it. Sure, it's not the worst problem in =
the world, but I do believe it's worth fixing.
>>=20
>> I'm just curious what everyone thinks=A1=AD maybe I'm completely off =
track for one reason or another?!
>>=20
>> Cheers,
>> Michael
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


From mellia@tlc.polito.it  Sun Dec 30 01:51:19 2012
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Date: Sun, 30 Dec 2012 10:51:10 +0100
From: Marco Mellia <mellia@tlc.polito.it>
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>, Michael Welzl <michawe@ifi.uio.no>
Subject: Re: [tcpm] About the SYN/ACK-timer
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Oh sure, you could (allow google to) do everything at the application
layer, so that your browser can open tons of connections, resolve all
hostnames, prefetech all links in a page, and download the same stuff
twice just in case a copy gets corrupted.
Who cares about fairness, efficiency, waste of resources, etc.etc. We
are living in an infinite resource world.
We already see youtube wasting 40% of the network capacity, so who cares?

Sorry for the harsh tone, but honestly if there is a (possible) problem
at the TCP layer, you should work to fix it there.
Any solution must also weight advantages and drawbacks. "few more
packets" in the network can be overkilling in some scenarios (just
thinking on slow and congested 3g network). Few more connections may
kill your NAT modem.
Lowering the initial RTO to 1s (or less, considering 95% of possible
scenarios) seems a simple and straight forward solution here.
Ciao
M

> There is no need to fix this in TCP. The browser could fix it.
> The browser could start a new TCP connection every ~250mS.
> Then use whichever TCP connection that completes first.
> It could keep the remaining TCP sessions open, just in case it needs to get more objects or pages from the same web site.
>
> Now, there is nothing stopping browsers from doing such workarounds...
> ...except a fix in TCP. Why is a SYN retransmission timer of 100mS so bad, even if it does put a few more packets into the network?
>
> --
> Jakob Heitz.
>
> On Dec 29, 2012, at 4:21 AM, "Michael Welzl" <michawe@ifi.uio.no> wrote:
>
>> Dear all,
>>
>> In my opinion, the web (and with it, probably many other applications too) exhibits a somewhat annoying problem that we still haven't fixed, but just gotten used to:
>> sometimes, when I click something, it doesn't immediately work. So I stop the transfer and click again. Then it works.
>>
>> Someone once told me that, when this happens, it's probably because a SYN or SYN/ACK was lost. I believe this is correct. Several years ago, this notion has prompted me to do some measuring + ugly fixing:
>> http://heim.ifi.uio.no/~michawe/research/tools/syn-retransmit/index.html
>> ¡­in doing so, we found that, indeed, the problem is significant (in terms of number of SYNs or SYN/ACKs lost). It's probably not hard to find more such data - I guess that Google has produced it when they first worked on Fast Open  (sorry for not checking right now).
>>
>> When I approached TCPM with a suggestion to reduce the timers for SYN and SYN/ACK retransmission, I ended up with a dialogue with Joe Touch, in which he convinced me that reducing the SYN timer could be harmful, as it could cause busy servers to get hammered by even more SYNs when they just can't cope with them. I buy that; but IIRC he had no good reason against making the SYN/ACK-timer more aggressive. BTW sorry if I'm using a wrong name here, but I guess it's clear: I mean the timer used by the server to resend the SYN/ACK if it doesn't get a response for a while.
>>
>> So, the thought occurs to me every once in a while (especially when I do surf the web, and something temporarily hangs), and just now I felt like sharing it again: why don't we really make the SYN/ACK timer more aggressive? This should for instance appeal to the Googlers on the list who are trying to improve TCP for their cause, no?
>>
>> I think if it's about establishing that the problem is significant, that shouldn't be too hard. Measurements show that SYN/ACKs do get lost, and when they do, the user feels it. Sure, it's not the worst problem in the world, but I do believe it's worth fixing.
>>
>> I'm just curious what everyone thinks¡­ maybe I'm completely off track for one reason or another?!
>>
>> Cheers,
>> Michael
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm


-- 
Ciao,                    /\/\/\rco

+-----------------------------------+
| Marco Mellia - Assistant Professor|
| Skypeid: mgmellia                 |
| Tel: +39-011-090-4173             |
| Cel: +39-331-6714789              |   /"\  .. . . . . . . . . . . . .
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| http://www.telematica.polito.it   |        .. . . . . . . . . . . . .
+-----------------------------------+
The box said "Requires Windows 95 or Better." So I installed Linux.


From touch@isi.edu  Sun Dec 30 08:46:21 2012
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>, Michael Welzl <michawe@ifi.uio.no>
Subject: Re: [tcpm] About the SYN/ACK-timer
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On 12/30/2012 1:51 AM, Marco Mellia wrote:
...
> Sorry for the harsh tone, but honestly if there is a (possible) problem
> at the TCP layer, you should work to fix it there.
> Any solution must also weight advantages and drawbacks. "few more
> packets" in the network can be overkilling in some scenarios (just
> thinking on slow and congested 3g network). Few more connections may
> kill your NAT modem.

Right now I hear the following from this discussion:

	- a *hypothesis* that connections are slow because either
	SYNs __or__ SYN/ACKs are getting lost

	- a proposed solution based on that hypothesis that
	makes connections less susceptible to timeout-based delays

What I have not seen:

	- any evidence that either SYN or SYN/ACK loss is the problem
		the former would be measured at a client;
		the latter needs to be measured at a loaded server

	- any analysis as to whether the proposed solution creates
	a problem
		e.g., if the SYN/ACKs aren't going out because of
		server overload, firing off more timers isn't going
		to help

	- any analysis of interactions between this and emerging
	uses of TCP
		e.g., one proposed SYN-loss solution is to open
		multiple connections and use the first one
		that responds; how would this interact with a
		server that also implemented lower SYN/ACK rexmits?

So, this all sounds like a very interesting area for someone to gather
data and analyze, but just because the *potential* solution is simple
doesn't mean it's warranted, safe, or appropriate at this time, IMO.

Joe

From jakob.heitz@ericsson.com  Sun Dec 30 10:16:42 2012
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From: Jakob Heitz <jakob.heitz@ericsson.com>
To: Joe Touch <touch@isi.edu>
Thread-Topic: [tcpm] About the SYN/ACK-timer
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Date: Sun, 30 Dec 2012 18:16:35 +0000
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Subject: Re: [tcpm] About the SYN/ACK-timer
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Consider:
A browser starts 8 TCP sessions
vs.
A TCP stack retransmits 8 SYN packets.

Which causes more stress on the Internet and the server?

Inaction at the TCP layer causes action at the application layer.
BTW, I think the SYN packet should include data. A simple change to the soc=
kets API should do it.

--
Jakob Heitz.

On Dec 30, 2012, at 8:46 AM, "Joe Touch" <touch@isi.edu> wrote:

>=20
>=20
> On 12/30/2012 1:51 AM, Marco Mellia wrote:
> ...
>> Sorry for the harsh tone, but honestly if there is a (possible) problem
>> at the TCP layer, you should work to fix it there.
>> Any solution must also weight advantages and drawbacks. "few more
>> packets" in the network can be overkilling in some scenarios (just
>> thinking on slow and congested 3g network). Few more connections may
>> kill your NAT modem.
>=20
> Right now I hear the following from this discussion:
>=20
>    - a *hypothesis* that connections are slow because either
>    SYNs __or__ SYN/ACKs are getting lost
>=20
>    - a proposed solution based on that hypothesis that
>    makes connections less susceptible to timeout-based delays
>=20
> What I have not seen:
>=20
>    - any evidence that either SYN or SYN/ACK loss is the problem
>        the former would be measured at a client;
>        the latter needs to be measured at a loaded server
>=20
>    - any analysis as to whether the proposed solution creates
>    a problem
>        e.g., if the SYN/ACKs aren't going out because of
>        server overload, firing off more timers isn't going
>        to help
>=20
>    - any analysis of interactions between this and emerging
>    uses of TCP
>        e.g., one proposed SYN-loss solution is to open
>        multiple connections and use the first one
>        that responds; how would this interact with a
>        server that also implemented lower SYN/ACK rexmits?
>=20
> So, this all sounds like a very interesting area for someone to gather
> data and analyze, but just because the *potential* solution is simple
> doesn't mean it's warranted, safe, or appropriate at this time, IMO.
>=20
> Joe

From swmike@swm.pp.se  Sun Dec 30 10:22:55 2012
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Subject: Re: [tcpm] About the SYN/ACK-timer
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On Sun, 30 Dec 2012, Jakob Heitz wrote:

> BTW, I think the SYN packet should include data. A simple change to the 
> sockets API should do it.

While I am all for it, I believe some security products will get upset by 
it:

<http://www.symantec.com/connect/articles/abnormal-ip-packets>

"A SYN only packet, which should only occur when a new connection is being 
initiated, should not contain any data."

So this needs to be investigated what the impact of this might be. I still 
remember security products dropping packets with ECN markings back when 
that was introduced > 10 years ago.

-- 
Mikael Abrahamsson    email: swmike@swm.pp.se

From touch@isi.edu  Sun Dec 30 10:44:15 2012
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>, Marco Mellia <mellia@tlc.polito.it>, Michael Welzl <michawe@ifi.uio.no>
Subject: Re: [tcpm] About the SYN/ACK-timer
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On 12/30/2012 10:16 AM, Jakob Heitz wrote:
> Consider:
> A browser starts 8 TCP sessions
> vs.
> A TCP stack retransmits 8 SYN packets.
>
> Which causes more stress on the Internet and the server?

Answer:

	it depends on a lot of factors:
		- is the client overloaded or memory-constrained?
		- are the 8 sessions to different servers or the same?
		- are the paths lossy?
		- are the paths congested?
		- are the servers overloaded?

If the 8 connections are to different servers, it's a completely 
different answer than if they're to the same one.

> Inaction at the TCP layer causes action at the application layer.

By that argument, your stack should send 1000 copies of every packet, in 
the hopes that at least one gets through.

See: "tragedy of the commons", however.

> BTW, I think the SYN packet should include data. A simple change to the sockets API should do it.

Please review RFC793; you would need to change more than just the API. 
This "dog" has been dragged around the block a *lot*, and it's a lot 
more complex.

Joe

>
> --
> Jakob Heitz.
>
> On Dec 30, 2012, at 8:46 AM, "Joe Touch" <touch@isi.edu> wrote:
>
>>
>>
>> On 12/30/2012 1:51 AM, Marco Mellia wrote:
>> ...
>>> Sorry for the harsh tone, but honestly if there is a (possible) problem
>>> at the TCP layer, you should work to fix it there.
>>> Any solution must also weight advantages and drawbacks. "few more
>>> packets" in the network can be overkilling in some scenarios (just
>>> thinking on slow and congested 3g network). Few more connections may
>>> kill your NAT modem.
>>
>> Right now I hear the following from this discussion:
>>
>>     - a *hypothesis* that connections are slow because either
>>     SYNs __or__ SYN/ACKs are getting lost
>>
>>     - a proposed solution based on that hypothesis that
>>     makes connections less susceptible to timeout-based delays
>>
>> What I have not seen:
>>
>>     - any evidence that either SYN or SYN/ACK loss is the problem
>>         the former would be measured at a client;
>>         the latter needs to be measured at a loaded server
>>
>>     - any analysis as to whether the proposed solution creates
>>     a problem
>>         e.g., if the SYN/ACKs aren't going out because of
>>         server overload, firing off more timers isn't going
>>         to help
>>
>>     - any analysis of interactions between this and emerging
>>     uses of TCP
>>         e.g., one proposed SYN-loss solution is to open
>>         multiple connections and use the first one
>>         that responds; how would this interact with a
>>         server that also implemented lower SYN/ACK rexmits?
>>
>> So, this all sounds like a very interesting area for someone to gather
>> data and analyze, but just because the *potential* solution is simple
>> doesn't mean it's warranted, safe, or appropriate at this time, IMO.
>>
>> Joe

From touch@isi.edu  Sun Dec 30 10:46:10 2012
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>, Marco Mellia <mellia@tlc.polito.it>, Michael Welzl <michawe@ifi.uio.no>
Subject: Re: [tcpm] About the SYN/ACK-timer
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On 12/30/2012 10:43 AM, Joe Touch wrote:
>
>
> On 12/30/2012 10:16 AM, Jakob Heitz wrote:
>> Consider:
>> A browser starts 8 TCP sessions
>> vs.
>> A TCP stack retransmits 8 SYN packets.
>>
>> Which causes more stress on the Internet and the server?
>
> Answer:
>
>      it depends on a lot of factors:
>          - is the client overloaded or memory-constrained?
>          - are the 8 sessions to different servers or the same?
>          - are the paths lossy?
>          - are the paths congested?
>          - are the servers overloaded?
>
> If the 8 connections are to different servers, it's a completely
> different answer than if they're to the same one.
>
>> Inaction at the TCP layer causes action at the application layer.
>
> By that argument, your stack should send 1000 copies of every packet, in
> the hopes that at least one gets through.
>
> See: "tragedy of the commons", however.
>
>> BTW, I think the SYN packet should include data. A simple change to
>> the sockets API should do it.
>
> Please review RFC793; you would need to change more than just the API.

...if it is to have any real effect. You can send data in the SYN now, 
but the server can't deliver it to the app layer until the three-way 
handshake completes. And it also increases the amount of state for 
pending connections, increasing the impact of DOS attacks.

Joe

From Anil.Agarwal@viasat.com  Sun Dec 30 11:11:25 2012
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From: "Agarwal, Anil" <Anil.Agarwal@viasat.com>
To: Joe Touch <touch@isi.edu>, Jakob Heitz <jakob.heitz@ericsson.com>
Thread-Topic: [tcpm] About the SYN/ACK-timer
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>, Michael Welzl <michawe@ifi.uio.no>, Marco Mellia <mellia@tlc.polito.it>
Subject: Re: [tcpm] About the SYN/ACK-timer
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Michael et al,

I tend to agree that reducing the TCP SYN-ACK timeout value would be
beneficial.

A typical Linux TCP implementation retransmits SYN-ACKs 5 times
at times shown below, assuming the first SYN-ACK is sent at time 0 -
    3, 9, 21, 45, 93 seconds
and finally give up at time 189 seconds.

If we start with an initial timer value of 1 second, then TCP would
Retransmit SYN-ACK segments at times -
    1, 3, 7, 15, 31 seconds
and finally give up at time 63 seconds.

At first blush, this looks better, for the vast variety of network conditio=
ns today.

Some expected objections:
1. This would cause unnecessary retransmissions in some cases,=20
   when RTT is > 1 second.
   Firstly, these unnecessary retransmissions would not break anything;
   TCP knows how to handle duplicate SYNs and SYN-ACKs.
   If we are concerned with causing network congestion, then keep
   in mind this injects < 64 bytes in a network per new connection.
   Surely, most networks can afford to carry an additional < 512 bps for
   a new connection, given that established connections consume much more
   bandwidth than that.

2. This cause additional processing overhead at servers
   Not really.
   The server is going to retransmit up to 5 times for a new connection.
   Whether it does it in 93 seconds or 31 seconds should not make much
   difference.
   Quicker establishment and completion of connections is better for server=
s
   compared to keeping connections in SYN-Recieved state for extended
   periods of time.

Other benefits:
This enables servers to free up resources quicker, when connections fail
to complete or under SYN flood attacks.

Also, TCP should allow the SYN-ACK timer value and the number of
SYN Retransmissions value to be configurable and perhaps should allow=20
history-based RTT learning, to handle cases where RTT is known to be=20
greater than 1 second. RFC 5482 "TCP User Timeout Option" would also be use=
ful.
We could recommend to implementers that reduced SYN-ACK timer value
always be supplemented by history-based RTT learning.

One could argue that the reduced initial timer value would be beneficial fo=
r
SYN packets as well.

I don't think we should encourage proposals that open multiple connections
to the same server and select the first one that succeeds. It consumes more
resources at the client, in the network and at the server. The technique is
probably a reaction to the "sluggishness" of TCP connection establishment
when SYN or SYN-ACK segments are lost.

Yes, we need to "collect" more data on current Internet traffic and
quantify the benefits of the proposed change, but I think it an idea worth
pursuing. It has similarities to the proposals to increase the initial TCP
congestion window size.


Cheers,
Anil Agarwal

-----Original Message-----
From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of Joe=
 Touch
Sent: Sunday, December 30, 2012 1:46 PM
To: Jakob Heitz
Cc: tcpm (tcpm@ietf.org); Marco Mellia; Michael Welzl
Subject: Re: [tcpm] About the SYN/ACK-timer



On 12/30/2012 10:43 AM, Joe Touch wrote:
>
>
> On 12/30/2012 10:16 AM, Jakob Heitz wrote:
>> Consider:
>> A browser starts 8 TCP sessions
>> vs.
>> A TCP stack retransmits 8 SYN packets.
>>
>> Which causes more stress on the Internet and the server?
>
> Answer:
>
>      it depends on a lot of factors:
>          - is the client overloaded or memory-constrained?
>          - are the 8 sessions to different servers or the same?
>          - are the paths lossy?
>          - are the paths congested?
>          - are the servers overloaded?
>
> If the 8 connections are to different servers, it's a completely
> different answer than if they're to the same one.
>
>> Inaction at the TCP layer causes action at the application layer.
>
> By that argument, your stack should send 1000 copies of every packet, in
> the hopes that at least one gets through.
>
> See: "tragedy of the commons", however.
>
>> BTW, I think the SYN packet should include data. A simple change to
>> the sockets API should do it.
>
> Please review RFC793; you would need to change more than just the API.

...if it is to have any real effect. You can send data in the SYN now,=20
but the server can't deliver it to the app layer until the three-way=20
handshake completes. And it also increases the amount of state for=20
pending connections, increasing the impact of DOS attacks.

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

From jakob.heitz@ericsson.com  Sun Dec 30 11:23:09 2012
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From: Jakob Heitz <jakob.heitz@ericsson.com>
To: Joe Touch <touch@isi.edu>
Thread-Topic: [tcpm] About the SYN/ACK-timer
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Subject: Re: [tcpm] About the SYN/ACK-timer
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On Dec 30, 2012, at 10:46 AM, "Joe Touch" <touch@isi.edu> wrote:
>=20
> ...if it is to have any real effect. You can send data in the SYN now, bu=
t the server can't deliver it to the app layer until the three-way handshak=
e completes. And it also increases the amount of state for pending connecti=
ons, increasing the impact of DOS attacks.
>=20
> Joe

It could, with a change to the sockets API. Of course, the app will need to=
 participate in SYN flood protection, but that's not impossible given the r=
ight architecture.

--
Jakob Heitz.


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Subject: Re: [tcpm] About the SYN/ACK-timer
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On Dec 30, 2012, at 11:23 AM, Jakob Heitz <jakob.heitz@ericsson.com> wrote:

> On Dec 30, 2012, at 10:46 AM, "Joe Touch" <touch@isi.edu> wrote:
>>=20
>> ...if it is to have any real effect. You can send data in the SYN now, bu=
t the server can't deliver it to the app layer until the three-way handshake=
 completes. And it also increases the amount of state for pending connection=
s, increasing the impact of DOS attacks.
>>=20
>> Joe
>=20
> It could, with a change to the sockets API.

No, it cannot without a change in CTO semantics, which affects the state dia=
gram and its processing on both ends - which also changes the meaning of an a=
ck and the concept of reliability.=20

There is a lot required, and for the first connection to a site it won't wor=
k to allow the server to act on data before a connection has been establishe=
d.=20

This is well- trod ground.=20

Joe

> Of course, the app will need to participate in SYN flood protection, but t=
hat's not impossible given the right architecture.
>=20
> --
> Jakob Heitz.
>=20

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Subject: Re: [tcpm] About the SYN/ACK-timer
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Um.  So, I just skimmed this thread.  I see people suggesting a "SYN-ACK
timer" of 1sec---even though it seems that the original note is that it
be "reduced" without providing any concreteness to that.  But,
subsequently we see things like this ...

> A typical Linux TCP implementation retransmits SYN-ACKs 5 times
> at times shown below, assuming the first SYN-ACK is sent at time 0 -
>     3, 9, 21, 45, 93 seconds
> and finally give up at time 189 seconds.
> 
> If we start with an initial timer value of 1 second, then TCP would
> Retransmit SYN-ACK segments at times -
>     1, 3, 7, 15, 31 seconds
> and finally give up at time 63 seconds.

Well, um, err, AHEM ... this working group produced RFC 6298, which says
the initial RTO SHOULD be 1sec.

So, "typical Linux" should either be using the latter progression given
above or someone has decided there is a good reason to be more
conservative.

Or, is someone proposing we lower the initial RTO further?

allman




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From swmike@swm.pp.se  Mon Dec 31 01:23:26 2012
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From: Mikael Abrahamsson <swmike@swm.pp.se>
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Subject: Re: [tcpm] About the SYN/ACK-timer
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On Sun, 30 Dec 2012, Agarwal, Anil wrote:

> quantify the benefits of the proposed change, but I think it an idea 
> worth pursuing. It has similarities to the proposals to increase the 
> initial TCP congestion window size.

I agree. Whilst some users still are on 2G connections which might be very 
slow (12-14 kilobit/s), they still can handle both an initial larger 
congestion window (there is sufficient buffering) and a lower SYN+ACK 
retransmit timer. For most users around the world now and more going 
forward, RTT is going to be less than 1 second.

I wholeheartedly support any work on TCP that will make it more adaptable 
to user conditions. TCP was designed to handle everything from a low bit/s 
access connection to tens of gigabit/s, but these initial conditions 
before the connection is up and running with sufficiently sized congestion 
window for the high bandwidth access can still be improved. One help with 
this might be heuritics to understand what the general access latency is 
for the user connection, and how it behaves.

A FTTH connection is vastly different in behaviour from a UMTS/3G 
connection which is vastly different from a 64 kilobit/s fixed connection.

If we want TCP to behave more optimized for each access scenario, I 
believe TCP needs to know about it over time and behave differently 
depending on detected access type, and most likely also communicate the 
access it has to the other end of the sessions being initiated.

-- 
Mikael Abrahamsson    email: swmike@swm.pp.se

From michawe@ifi.uio.no  Mon Dec 31 01:53:02 2012
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Subject: Re: [tcpm] About the SYN/ACK-timer
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Hi,

Not sure about everyone else, but the way I interpreted (and meant) the =
1s bit was to say "a new value, such as 1s, as opposed to a complex =
adaptive mechanism". It wasn't about the exact value of 1s per se.

What would be a good new value? No idea=85 as Joe has pointed out, there =
is still some work to be done here, before proposing a change. I agree =
that my story of web surfing hiccups (where you have to press reload to =
get the whole thing that you're waiting for) is merely a hypothesis, and =
e.g. investigating lost SYNs / SYN-ACKs are really the root of the =
problem, and how severe that problem is, could be quite interesting.

A part of the problem as I described it involves the user pressing the =
browser's "refresh" button. The effect of that could be seen on the =
wire=85 HTTP requests would be sent again, at user-determined times=85 =
does it happen often? Does it coincide with lost SYNs or SYN-ACKs? =
Interesting stuff to look into, IMO.

About lowering the initial RTO further: I, for one, hadn't thought of =
that, and am not fully aware of all the pro's and con's. But yeah, it =
might be worth considering?

Cheers,
Michael


On Dec 31, 2012, at 4:50 AM, Mark Allman <mallman@icir.org> wrote:

>=20
> Um.  So, I just skimmed this thread.  I see people suggesting a =
"SYN-ACK
> timer" of 1sec---even though it seems that the original note is that =
it
> be "reduced" without providing any concreteness to that.  But,
> subsequently we see things like this ...
>=20
>> A typical Linux TCP implementation retransmits SYN-ACKs 5 times
>> at times shown below, assuming the first SYN-ACK is sent at time 0 -
>>    3, 9, 21, 45, 93 seconds
>> and finally give up at time 189 seconds.
>>=20
>> If we start with an initial timer value of 1 second, then TCP would
>> Retransmit SYN-ACK segments at times -
>>    1, 3, 7, 15, 31 seconds
>> and finally give up at time 63 seconds.
>=20
> Well, um, err, AHEM ... this working group produced RFC 6298, which =
says
> the initial RTO SHOULD be 1sec.
>=20
> So, "typical Linux" should either be using the latter progression =
given
> above or someone has decided there is a good reason to be more
> conservative.
>=20
> Or, is someone proposing we lower the initial RTO further?
>=20
> allman
>=20
>=20
>=20


From Anil.Agarwal@viasat.com  Mon Dec 31 06:11:56 2012
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From: "Agarwal, Anil" <Anil.Agarwal@viasat.com>
To: "mallman@icir.org" <mallman@icir.org>
Thread-Topic: [tcpm] About the SYN/ACK-timer 
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Date: Mon, 31 Dec 2012 14:11:50 +0000
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Cc: "tcpm \(tcpm@ietf.org\)" <tcpm@ietf.org>, Marco Mellia <mellia@tlc.polito.it>, Michael Welzl <michawe@ifi.uio.no>, Joe Touch <touch@isi.edu>
Subject: Re: [tcpm] About the SYN/ACK-timer
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Mark,

I stand corrected; forgot about RFC 6298.
RFC 6298 clearly states that until an RTT measurement is made, "the sender =
SHOULD set RTO <- 1 second". RFC 6298 contains rationale, supporting data a=
nd additional rules/guidelines for use of this shortened timer.

Looks like the initial timeout was reduced to 1 second in Linux 3.1, both f=
or SYN and SYN-ACK segments. Did a quick test with Linux 3.2.14; the SYN re=
transmission times are -
     1, 3, 7, 15, 31 seconds
with a final give up time of ~63 seconds.

Anil


-----Original Message-----
From: mallman@icir.org [mailto:mallman@icir.org]=20
Sent: Sunday, December 30, 2012 10:51 PM
To: Agarwal, Anil
Cc: Joe Touch; Jakob Heitz; tcpm (tcpm@ietf.org); Michael Welzl; Marco Mell=
ia
Subject: Re: [tcpm] About the SYN/ACK-timer=20


Um.  So, I just skimmed this thread.  I see people suggesting a "SYN-ACK
timer" of 1sec---even though it seems that the original note is that it
be "reduced" without providing any concreteness to that.  But,
subsequently we see things like this ...

> A typical Linux TCP implementation retransmits SYN-ACKs 5 times
> at times shown below, assuming the first SYN-ACK is sent at time 0 -
>     3, 9, 21, 45, 93 seconds
> and finally give up at time 189 seconds.
>=20
> If we start with an initial timer value of 1 second, then TCP would
> Retransmit SYN-ACK segments at times -
>     1, 3, 7, 15, 31 seconds
> and finally give up at time 63 seconds.

Well, um, err, AHEM ... this working group produced RFC 6298, which says
the initial RTO SHOULD be 1sec.

So, "typical Linux" should either be using the latter progression given
above or someone has decided there is a good reason to be more
conservative.

Or, is someone proposing we lower the initial RTO further?

allman




From jakob.heitz@ericsson.com  Mon Dec 31 11:24:31 2012
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From: Jakob Heitz <jakob.heitz@ericsson.com>
To: Michael Welzl <michawe@ifi.uio.no>, "tcpm (tcpm@ietf.org)" <tcpm@ietf.org>
Thread-Topic: [tcpm] About the SYN/ACK-timer
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Subject: Re: [tcpm] About the SYN/ACK-timer
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I have an alternative to cookies.

Send data with the SYN.
When retransmission of the SYN is required, retransmit it without the data.
Retransmit the SYN quickly: 100 to 200 mS.

It works with Middleboxes.
SYN flood protection works as normal, just bias it against SYN with data.

Cheers,
Jakob.

On Dec 31, 2012, at 1:44 AM, "Michael Welzl" <michawe@ifi.uio.no> wrote:

> Hi,
>=20
> About data-in-the-SYN: I agree with Joe that this has been debated a lot =
already - see https://tools.ietf.org/html/draft-ietf-tcpm-fastopen-02 and t=
he discussions surrounding it. FWIW this draft is a (IMHO, good) proposal t=
o do what you want, in the right way
>=20
> Cheers,
> Michael
>=20
>=20
> On Dec 30, 2012, at 10:17 PM, Joe Touch <touch@isi.edu> wrote:
>=20
>>=20
>>=20
>> On Dec 30, 2012, at 11:23 AM, Jakob Heitz <jakob.heitz@ericsson.com> wro=
te:
>>=20
>>> On Dec 30, 2012, at 10:46 AM, "Joe Touch" <touch@isi.edu> wrote:
>>>>=20
>>>> ...if it is to have any real effect. You can send data in the SYN now,=
 but the server can't deliver it to the app layer until the three-way hands=
hake completes. And it also increases the amount of state for pending conne=
ctions, increasing the impact of DOS attacks.
>>>>=20
>>>> Joe
>>>=20
>>> It could, with a change to the sockets API.
>>=20
>> No, it cannot without a change in CTO semantics, which affects the state=
 diagram and its processing on both ends - which also changes the meaning o=
f an ack and the concept of reliability.=20
>>=20
>> There is a lot required, and for the first connection to a site it won't=
 work to allow the server to act on data before a connection has been estab=
lished.=20
>>=20
>> This is well- trod ground.=20
>>=20
>> Joe
>>=20
>>> Of course, the app will need to participate in SYN flood protection, bu=
t that's not impossible given the right architecture.
>>>=20
>>> --
>>> Jakob Heitz.
>>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>=20
