
From jsaldana@unizar.es  Mon Apr  2 00:26:46 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
To: <gorry@erg.abdn.ac.uk>, <tsvwg@ietf.org>
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Subject: RE: TSV draft notes Paris (Draft)
Date: Mon, 2 Apr 2012 09:26:24 +0200
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The notes are ok for me,

Jose

-----Mensaje original-----
De: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] En nombre de
gorry@erg.abdn.ac.uk
Enviado el: viernes, 30 de marzo de 2012 10:54
Para: tsvwg@ietf.org
CC: tsvwg-chairs@ietf.org
Asunto: TSV draft notes Paris (Draft)


The draft notes from this week's TSVWG meeting have been uploaded here:

http://www.ietf.org/proceedings/83/minutes/minutes-83-tsvwg.txt

- Please send comment/corrections to the chairs or mailing list.

Gorry & James
(tsvwg co-chairs)


From jsaldana@unizar.es  Mon Apr  2 01:15:20 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
To: "'Wesley Eddy'" <wes@mti-systems.com>, "Gorry Fairhurst" <gorry@erg.abdn.ac.uk>, "James Polk" <jmpolk@cisco.com>, "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>, "Muthu Arul Mozhi Perumal \(mperumal\)" <mperumal@cisco.com>, "Dan Wing" <dwing@cisco.com>, =?iso-8859-1?Q?Juli=E1n_Fern=E1ndez_Navajas?= <navajas@unizar.es>, =?iso-8859-1?Q?Jos=E9_Ruiz_Mas?= <jruiz@unizar.es>, "Michael Ramalho \(mramalho\)" <mramalho@cisco.com>
Subject: Some thoughts about draft-saldana-tsvwg-tcmtf-02
Date: Mon, 2 Apr 2012 10:14:58 +0200
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In this message I try to answer some of the questions that have been sent to
the list. I think the main questions are:

1) Does the draft intend to obsolete RFC4170?
2) The name of the draft
3) Agreement between the ends


Let's go with question 1)

Wes says:
>Hi Jose; by my understanding, your proposal is significantly different from
4170, and not just a set of changes to it, 
>so I don't think it would strictly be a "bis" document to 4170.
>This is just my opinion :)

James says:
>"bis" named documents are intended to replace existing RFC(s) (i.e., make
an existing RFC obsolete (if, and) when
> this new draft is published as an RFC. This -rfcXXXbis- designation is
intended to give context to what
>this new draft is about, but does not indicate any changes in
solution/direction/whatever this new draft is attempting to achieve.

>A good example from TSVWG is when the original SCTP base spec RFC
>(2960) was rewritten to update and correct any known errors (mistakes or
omissions) that draft was draft-*-tsvwg-
>2960bis-... when in WG draft form. This draft reached consensus within this
WG and the community to become RFC 
>4960, which is a proposed standard currently. 
>If, and when, SCTP needs to progress towards Internet Standard (IS) along
the standards track, it (likely) will start 
>with  a filename with "...-tsvwg-4960bis-..." in it.

>Does this make sense?

>If you follow that (and no one disagrees with it), do you believe your
draft qualifies as an intended rewrite of RFC
>4170? It is important what you as an author believes about this, as well as
what the WG believes about this. If 
>consensus is reached whether (or not) this draft should be considered a
rewrite of RFC 4170, the filename can 
>change then (usually at the direction of the WG chairs). The draft title
does not need to change, and in fact often
>does not.

My opinion:
The idea of the draft is to widen the scope of RFC4170 (TCRTP), which only
includes RTP traffic, and a single option for each layer
- RTP traffic
- ECRTP compression
- PPPMux multiplexing
- L2TPv3 tunneling

The new draft would include
- RTP, UDP and TCP traffic
- ECRTP, ROHC, IPHC, no compression
- PPPMux or other multiplexing options. In this layer I don't know if there
exists any other suitable multiplexing protocol
- L2TPv3 of GRE tunneling

So the draft includes all the options of RFC4170, as one of its possible
uses. Would it be necessary to obsolete RFC4170?

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

Question 2): The name of the draft

In my opinion, we can leave the name of the draft as it is now. The name is
not the most important thing.

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

Question 3) Signaling 

Wes says:
>I have a related question on the document.  Since it allows for multiple
options at each "layer" of the tunneling and 
>compression, how do you get agreement at each end about what it's okay to
use, and what actually is used?  Is it 
>just assumed to be a matter of management out of band?

My opinion:
The main idea is to do something similar to RFC4170. If you look at section
4 (http://tools.ietf.org/html/rfc4170#section-4), there is a "Example
implementation of TCRTP". the ways to deploy the PPP and L2TP negotiations
are suggested there: LCP negotiation, IPCP negotiation, tunnel
establishment, session establishment and tunnel tear down. The negotiations
are the ones described on each of the RFCs of the protocols corresponding of
each layer. The idea of the new draft could be similar.

So there are two options:
a) First, the ends would have to negotiate which protocols are they going to
use at each layer. And then, each of the protocols would have its own
negotiation. In this case, the new draft would have to consider a first
negotiation of each layer.

b) Other option is that in each scenario only one option is considered: if
the best option for my scenario is e.g. UDP-ROHC-PPPMux-GRE, then I will
always use it, and I do not need to negotiate anything. In this case, the
draft would not have to include a specific negotiation of the protocols to
use for each layer.


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

I would appreciate if you could send me some feedback.

Best regards, and thank you for your questions,

Jose


From pasi.sarolahti@iki.fi  Mon Apr  2 02:47:36 2012
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From: Pasi Sarolahti <pasi.sarolahti@iki.fi>
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Subject: Fwd: [dccp] WGLC for draft-ietf-dccp-udpencap-10
Date: Mon, 2 Apr 2012 12:47:19 +0300
References: <20731_1333359735_4F797477_20731_2375_1_EC220417-FEDF-443A-88F1-9DF9CA60158D@iki.fi>
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Hi,

For information, because this, and similar topics have also been =
discussed in TSVWG: the DCCP working group has started a new WGLC on =
draft-ietf-dccp-udpencap-10. The mail below gives more information. =
Please send comments to dccp mailing list.

Thanks!

- Pasi


Begin forwarded message:

> From: Pasi Sarolahti <pasi.sarolahti@iki.fi>
> Subject: [dccp] WGLC for draft-ietf-dccp-udpencap-10
> Date: April 2, 2012 12:41:46 PM GMT+03:00
> To: DCCP WG <dccp@ietf.org>
>=20
> Hi,
>=20
> As people have noticed, not much has happened in DCCP WG for the =
several past months. However, we still should finish the last (recently =
updated) WG document, after which the working group is expected to be =
done.
>=20
> So: this Email starts the working group last call for "Datagram =
Congestion Control Protocol (DCCP) Encapsulation for NAT Traversal =
(DCCP-UDP)" (draft-ietf-dccp-udpencap-10). The WGLC lasts two weeks and =
ends on Monday, April 16th. Please send possible comments to the dccp =
mailing list.
>=20
> For the udpencap document the last remaining issue, from the previous =
WGLC, has related to the text that speaks about negotiating the use of =
DCCP-UDP encapsulation, particularly in ICE. After discussing this with =
the authors, we concluded that coming up with a solid negotiation =
specification would still require significant amount of work and review =
cycles, and this is not worth delaying the document (and the WG) any =
further. Therefore, ICE negotiation is left for future work. If there is =
interest and energy, negotiation can be specified in a separate =
document, probably in a different working group.
>=20
> - Pasi
>=20


From jsaldana@unizar.es  Mon Apr  2 03:17:19 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
To: "James Polk" <jmpolk@cisco.com>
Subject: About draft-polk-tsvwg-rfc4594-update-00.txt
Date: Mon, 2 Apr 2012 12:17:04 +0200
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James,

 

After returning home from IETF 83, I have been reading your draft. I find it
interesting to standardize the service classes. Otherwise, if each one
classifies each traffic according to different criteria, Diffserv becomes
less useful.

 

Regarding "Real-time interactive" service class, which refers to online
gaming (my main research field), I find it really interesting: online games
are at least as real-time as VoIP is. In most cases, the gamers are a very
difficult people to deal with. If the latency is high, they leave and never
return (this statement is not mine, it can be corroborated in the references
I put below). This has been studied by many research groups, with empirical
tests using real gamers and adding delay, jitter, packet loss.

 

Other question: in the draft it is said that online games use RTP/UDP
streams. This can be a little confusing. In fact, the vast majority of
online games do not use RTP, but bare UDP or even TCP. In
http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.pdf, the
authors proposed the use of an adaptation of RTP in order to manage gaming
traffic. But as far as I know, this approach is not used by commercial
games.

 

 

A typical classification of today's games is:

 

-          FPS (First Person Shooters) e.g. Counter Strike, Halo, Call of
Duty: They use UDP

-          MMORPG (Massively Multiplayer Online Role-playing Games), e.g.
World of Warcraft: They normally use TCP, even though some of them use UDP

-          RTS (Real-time strategy) e.g. Age of Empires: They may use TCP or
UDP

-          Sports games: Since there is a big variety, they have not been
deeply studied in the literature

 

Some research papers supporting this classification, and also the real-time
requirements of the games:

 

First Person Shooters

 

-          A survey of traffic models for 17 First Person Shooters can be
found here. All of them use UDP
http://www.discover.uottawa.ca/publications/files/MMOG-Survey_IEEE-IC.pdf

 

-          In this paper from my research group, 8 FPS games were analyzed
in order to compress and multiplex their traffic. Figure 5 shows their
histograms (inter-packet time and packet size). It can be seen that they
generate real-time flows, with characteristics very similar to VoIP (packets
of 60-80 bytes every 15-40 ms):
http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana.pdf

 

-          Another paper about FPS using UDP. Section V presents some
interesting conclusions about routing infrastructure.
http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf

 

-          In this paper the network factors annoying the gamers are
analyzed. There are also some calculations of the MOS. Above 150 ms the
quality is bad (something similar to what happens with VoIP)
http://www.research.ibm.com/netgames2005/papers/dick.pdf

 

-          In this one, a MOS model for a FPS (Quake IV) is deployed. It is
only based on delay and jitter, since packet loss has vevy little influence
(for this concrete game). Delay above 140 ms produces MOS below 3
http://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf

 

-          In this paper, it is found that some FPS work properly even with
30% packet loss, whereas others stop working with 4%. RTT above 300 ms (150
OWD) makes MOS go above 3
http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pdf

 

MMORPGs:

-          A traffic model for World of Warcraft:
http://publik.tuwien.ac.at/files/pub-et_12119.pdf

 

-          This one presents a MOS for World of Warcraft. When added delay
is above 160ms, the MOS is roughly 3.:
http://publik.tuwien.ac.at/files/PubDat_170202.pdf

 

Best regards,

 

Jose Saldana


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</o:shapelayout></xml><![endif]--></head><body lang=3DES link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US>James,<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>After returning home from IETF 83, I have been reading your =
draft. I find it interesting to standardize the service classes. =
Otherwise, if each one classifies each traffic according to different =
criteria, Diffserv becomes less useful.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>Regarding &#8220;Real-time =
interactive&#8221; service class, which refers to online gaming (my main =
research field), I find it really interesting: online games are at least =
as real-time as VoIP is. In most cases, the gamers are a very difficult =
people to deal with. If the latency is high, they leave and never return =
(this statement is not mine, it can be corroborated in the references I =
put below). This has been studied by many research groups, with =
empirical tests using real gamers and adding delay, jitter, packet =
loss.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>Other question: in the draft it is said that online games =
use RTP/UDP streams. This can be a little confusing. In fact, the vast =
majority of online games do not use RTP, but bare UDP or even TCP. In <a =
href=3D"http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.=
pdf">http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.pdf=
</a>, the authors proposed the use of an adaptation of RTP in order to =
manage gaming traffic. But as far as I know, this approach is not used =
by commercial games.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>A typical classification of today&#8217;s games =
is:<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>FPS (First Person =
Shooters) e.g. Counter Strike, Halo, Call of Duty: They use =
UDP<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>MMORPG (Massively =
Multiplayer Online Role-playing Games), e.g. World of Warcraft: They =
normally use TCP, even though some of them use =
UDP<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>RTS (Real-time =
strategy) e.g. Age of Empires: They may use TCP or =
UDP<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>Sports games: Since =
there is a big variety, they have not been deeply studied in the =
literature<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>Some research papers supporting this classification, and =
also the real-time requirements of the games:<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>First Person =
Shooters<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>A survey of traffic =
models for 17 First Person Shooters can be found here. All of them use =
UDP <a =
href=3D"http://www.discover.uottawa.ca/publications/files/MMOG-Survey_IEE=
E-IC.pdf">http://www.discover.uottawa.ca/publications/files/MMOG-Survey_I=
EEE-IC.pdf</a><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>In this paper from my =
research group, 8 FPS games were analyzed in order to compress and =
multiplex their traffic. Figure 5 shows their histograms (inter-packet =
time and packet size). It can be seen that they generate real-time =
flows, with characteristics very similar to VoIP (packets of 60-80 bytes =
every 15-40 ms): <a =
href=3D"http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana=
.pdf">http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana.p=
df</a><o:p></o:p></span></p><p class=3DMsoListParagraph><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>Another paper about =
FPS using UDP. Section V presents some interesting conclusions about =
routing infrastructure. <a =
href=3D"http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf">h=
ttp://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf</a><o:p></o=
:p></span></p><p class=3DMsoListParagraph><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>In this paper the =
network factors annoying the gamers are analyzed. There are also some =
calculations of the MOS. Above 150 ms the quality is bad (something =
similar to what happens with VoIP) <a =
href=3D"http://www.research.ibm.com/netgames2005/papers/dick.pdf">http://=
www.research.ibm.com/netgames2005/papers/dick.pdf</a><o:p></o:p></span></=
p><p class=3DMsoListParagraph><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>In this one, a MOS =
model for a FPS (Quake IV) is deployed. It is only based on delay and =
jitter, since packet loss has vevy little influence (for this concrete =
game). Delay above 140 ms produces MOS below 3 <a =
href=3D"http://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf">htt=
p://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf</a><o:p></o:p><=
/span></p><p class=3DMsoListParagraph><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>In this paper, it is =
found that some FPS work properly even with 30% packet loss, whereas =
others stop working with 4%. RTT above 300 ms (150 OWD) makes MOS go =
above 3 <a =
href=3D"http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pd=
f">http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pdf</a>=
<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>MMORPGs:<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>A traffic model for =
World of Warcraft: <a =
href=3D"http://publik.tuwien.ac.at/files/pub-et_12119.pdf">http://publik.=
tuwien.ac.at/files/pub-et_12119.pdf</a><o:p></o:p></span></p><p =
class=3DMsoListParagraph><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>This one presents a =
MOS for World of Warcraft. When added delay is above 160ms, the MOS is =
roughly 3.: <a =
href=3D"http://publik.tuwien.ac.at/files/PubDat_170202.pdf">http://publik=
.tuwien.ac.at/files/PubDat_170202.pdf</a><o:p></o:p></span></p><p =
class=3DMsoListParagraph><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>Best regards,<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>Jose =
Saldana<o:p></o:p></span></p></div></body></html>
------=_NextPart_000_007C_01CD10CA.85552380--


From philip.eardley@bt.com  Mon Apr  2 09:49:38 2012
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Date: Mon, 2 Apr 2012 17:49:46 +0100
Subject: RE: Note: WGLC Announcement for draft-ietf-tsvwg-byte-pkt-congest-07
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Hi,
Sorry these comments are late.

S2.2 I thought the numbered bullets could be better phrased, something like=
 this:-

  1.  AQM algorithms such as RED SHOULD use packet-mode drop, ie they SHOUL=
D NOT use byte-mode drop. The latter=20
is more complex,
       it creates the perverse incentive to fragment segments into tiny
       pieces and it is vulnerable to floods of small-
       packets.

   2.  If a vendor has implemented byte-mode drop, and an operator has
       turned it on, it is RECOMMENDED to turn it off, after establishing i=
f there are any implications on=20
the relative performance of
       applications using different packet sizes.

  3.  RED SHOULD NOT be turned off. Without RED, a drop tail
       queue biases against large packets. =20

Nit - the following paragraph shouldn't be indented (Note well that RED's..=
.)

S3.3
The section could be slightly clearer. Starting with the second para, where=
 you start talking about something you don't recommend (ie you have to reme=
mber that you're talking about the consequences of not following your recom=
mendation).

S3.3
   o  Also, if the network biases congestion notification by packet size
      it has to assume a baseline packet size--all proposed algorithms
      use the local MTU.  Then transports have to guess which link was
      congested and what its local MTU was, in order to know how to
      tailor their congestion response to that link.
So you're saying that for byte-mode drop, the network element has to compar=
e the size of the current pkt with the MTU, and if it's 25* smaller then re=
duce the pkt loss probably by 25*? And if the MTU was bigger, then the curr=
ent pkt would be dropped with greater probability? And both of these things=
 are hard to do.

S3.4 (& knock-on effect on 3.3)
I think this could be better.=20
I agree with your arguments in S3.1 & 3.2 that dropping or marking should b=
e independent of pkt size.=20
You now go into a couple of pages, which I think could be boiled down to a =
paragraph, saying something like:-=20
* the overall system (network + transport) has to be packet-size dependent =
in order to drive out the right amount of traffic so the resource becomes u=
ncongested (since we assume all resources are in practice byte rather than =
packet limited)
* from S3.1 and S3.2 the IETF is convinced that the network should notify i=
ndependent of packet-size.
* therefore the transport must act dependent on packet-size.

It seems to me this is a necessary consequence of your earlier assumptions =
/arguments.

[On the other hand, if someone disagrees with the arguments and thinks that=
 the network should notify dependent on packet-size, then it follows that t=
he transport can act independent of packet-size.]

I think in S3.3 you're trying to make another argument (?):-
* we can either do packet-size biasing in the network element or on the end=
 host transport (see argument above)
* packet-size biasing in the network is quite tricky, because then both the=
 network element & the transport have to make some guesses about MTU - wher=
eas with packet-size biasing in the transport, neither network element nor =
transport needs to know what the MTU is.
* it is also easier to implement packet-size biasing in the transport than =
in the network.

On the last bullet, I didn't really follow your argument in S3.5. Might be =
easier to follow if S3.5 also talked about how a network element would take=
 account of pkt size? Also, does the following argument also apply?:=20
* if pkt-size biasing is done in the transport, then the work (to do the bi=
asing) is spread over many end hosts,=20
* whereas if pkt-size biasing is done in the network, then the work is conc=
entrated in just the congested network element, which has to operate on all=
 the traffic at line speed.
=09
Nits
Page 5, line 3 - less control packets /fewer control packets
Page 7, RED terminology - In RED /In RED [ref]


Best wishes,
phil
-----Original Message-----
From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of G=
orry Fairhurst
Sent: 28 March 2012 14:02
To: tsvwg@ietf.org
Subject: Note: WGLC Announcement for draft-ietf-tsvwg-byte-pkt-congest-07


Please respond to the WGLC that ends on Friday 30th March 2012.

Notes to the list would be helpful to complete the WGCC.

Gorry & James

-------- Original Message --------
Subject: WGLC Announcement for draft-ietf-tsvwg-byte-pkt-congest-07


This email announces the start of a working group last call for
draft-ietf-tsvwg-byte-pkt-congest-07, "Byte and Packet Congestion
Notification". This document is now thought to be ready
to proceed to be published as a BCP. Please send any comments to the
TSVWG list.

The draft is available at:
http://tools.ietf.org/html/draft-ietf-tsvwg-byte-pkt-congest-07

The last call will run for TWO weeks, ending Friday 30th March 2012.

Emails saying "I support" or "I don't support" publication are also most=20
helpful in judging the WG consensus. Please let us know if this draft is=20
useful and seems to provide the correct advice.

James and Gorry
(TSVWG Chairs)
tsvwg-chairs@ietf.org







From p.ohanlon@gmail.com  Tue Apr  3 08:16:24 2012
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Subject: Re: TSV draft notes Paris (Draft)
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From: Piers O'Hanlon <p.ohanlon@gmail.com>
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Date: Tue, 3 Apr 2012 16:16:19 +0100
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Hi Gorry,

The minutes look fine in general. A few comments:

1) "Georgios - RSVP support for PCN"
 Garbled text at end of first paragraph

2) "Randy - ECN for SCTP"
Although this error originated in the schedule I guess that it should be =
"Randall" not "Randy"

3) "James - Proposed update to RFC 4594"
Rudediger -> Ruediger

4) "Piers - Reactions to Signaling from ECN Support for RTP/RTCP"
- A small nit - my name was spelled correctly in the title but not in =
the minuted discussions.
- Data-limited discussion addition:
"Piers: Can Gorry clarify his comments on the list regarding DCCP and =
the newer version of TFRC RFC?
Gorry: I was mainly talking about the data-limited behaviours that were =
improved in the latest version of TFRC RFC. These benefit periods of =
temporarily low rate operation, for example as a result of silence or =
comfort noise.
Piers: Right ok we were aware of these changes but will add more =
information into the draft."


Piers




On 30 Mar 2012, at 09:54, gorry@erg.abdn.ac.uk wrote:

>=20
> The draft notes from this week's TSVWG meeting have been uploaded =
here:
>=20
> http://www.ietf.org/proceedings/83/minutes/minutes-83-tsvwg.txt
>=20
> - Please send comment/corrections to the chairs or mailing list.
>=20
> Gorry & James
> (tsvwg co-chairs)


From jsaldana@unizar.es  Tue Apr  3 23:19:05 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
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James,

 

I have been reading the draft, and I have found some small errors, which may
be corrected in the next version.

 

Best regards,

 

Jose Saldana

 

--------

 

- Page 3, first paragraph of the introduction: There is something wrong in
this sentence:

>Local administrators are who configuration each router for which
Differentiated Services Code 

>Points (DSCP) are to be treated differently, ...

 

 

- Page 4: There is something missing in this sentence:

>[ID-DSCP] goes further to explain that may layer 2 technologies fewer bits
for marking and prioritization.

 

 

- Page 15: It is said that 

>Conversational service group consists of five service classes:

 

But only three classes appear

 

 

 

- Page 16, Real-time paragraph. Games do not use RTP, but only bare UDP an
TCP.

>Real-Time Interactive, which includes both 'admitted' and 'unadmitted'
Realtime-Interactive

>service class, is for bidirectional variable rate inelastic applications
that require low 

>jitter and loss and very low delay, such as interactive gaming applications
that use 

>RTP/UDP streams for game control commands,

 

 

- Page 20: You say that real-time interactive service has to work with very
low loss rate. This may be true or not. In a reference I sent yesterday, it
was shown that Microsoft's game "Halo" stops working with 4% packet loss,
whereas Quake III works well with 30%, and players do not notice

 

 

- Page 42, first paragraph: 

>Interactive gaming applications that do not have the ability to change
encoding rates 

>or to mark packets with different importance indications is one good
example of such an 

>application.

 

I think that talking about encoding rates for games is a bit confusing. The
games I have studied do not modify encoding rates, since they do not
transmit audio or video. They only transmit the game status or the actions
of the player. The game is rendered in the local machine. One exception:
games including a voice channel in order to coordinate the different
players.

 


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
lang=3DEN-US>James,<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US>I have been reading the =
draft, and I have found some small errors, which may be corrected in the =
next version.<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US>Best =
regards,<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US>Jose =
Saldana<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
lang=3DEN-US>--------<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US>- Page 3, first =
paragraph of the introduction: There is something wrong in this =
sentence:<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier;color:#1F497D'>&gt;</span><=
span lang=3DEN-US style=3D'font-size:10.0pt;font-family:Courier'>Local =
administrators are who configuration each router for which =
Differentiated Services Code <span =
style=3D'color:#1F497D'><o:p></o:p></span></span></p><p =
class=3DMsoNormal style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier;color:#1F497D'>&gt;</span><=
span lang=3DEN-US style=3D'font-size:10.0pt;font-family:Courier'>Points =
(DSCP) are to be treated differently, ...</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US>- Page 4: There is =
something missing in this sentence:<o:p></o:p></span></p><p =
class=3DMsoNormal style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier;color:#1F497D'>&gt;</span><=
span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier'>[ID-DSCP] goes further to =
explain that may layer 2 technologies fewer bits for marking and =
prioritization.<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier'><o:p>&nbsp;</o:p></span></=
p><p class=3DMsoNormal style=3D'text-autospace:none'><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US>- Page 15: It is said =
that <o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier;color:#1F497D'>&gt;</span><=
span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier'>Conversational service =
group consists of five service classes:<o:p></o:p></span></p><p =
class=3DMsoNormal style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier'><o:p>&nbsp;</o:p></span></=
p><p class=3DMsoNormal style=3D'text-autospace:none'><span =
lang=3DEN-US>But only three classes appear<o:p></o:p></span></p><p =
class=3DMsoNormal style=3D'text-autospace:none'><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US>- Page 16, Real-time =
paragraph. Games do not use RTP, but only bare UDP an =
TCP.<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier;color:#1F497D'>&gt;</span><=
span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier'>Real-Time Interactive, =
which includes both &#8217;admitted&#8217; and &#8217;unadmitted&#8217; =
Realtime-Interactive<span =
style=3D'color:#1F497D'><o:p></o:p></span></span></p><p =
class=3DMsoNormal style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier;color:#1F497D'>&gt;</span><=
span lang=3DEN-US style=3D'font-size:10.0pt;font-family:Courier'>service =
class, is for bidirectional variable rate inelastic applications that =
require low <span =
style=3D'color:#1F497D'><o:p></o:p></span></span></p><p =
class=3DMsoNormal style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier;color:#1F497D'>&gt;</span><=
span lang=3DEN-US style=3D'font-size:10.0pt;font-family:Courier'>jitter =
and loss and very low delay, such as interactive gaming applications =
that use <span style=3D'color:#1F497D'><o:p></o:p></span></span></p><p =
class=3DMsoNormal style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier;color:#1F497D'>&gt;</span><=
span lang=3DEN-US style=3D'font-size:10.0pt;font-family:Courier'>RTP/UDP =
streams for game control commands,</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US>- Page 20: You say that =
rea<span style=3D'color:#1F497D'>l</span>-time interactive service has =
to work with very low loss rate. This may be true or not. In a reference =
I sent yesterday, it was shown that Microsoft&#8217;s game =
&#8220;Halo&#8221; stops working with 4% packet loss, whereas Quake III =
works well with 30%, and players do not notice<o:p></o:p></span></p><p =
class=3DMsoNormal style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US>- Page 42, first =
paragraph: <o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier;color:#1F497D'>&gt;</span><=
span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier'>Interactive gaming =
applications that do not have the ability to change encoding rates <span =
style=3D'color:#1F497D'><o:p></o:p></span></span></p><p =
class=3DMsoNormal style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier;color:#1F497D'>&gt;</span><=
span lang=3DEN-US style=3D'font-size:10.0pt;font-family:Courier'>or to =
mark packets with different importance indications is one good example =
of such an <span style=3D'color:#1F497D'><o:p></o:p></span></span></p><p =
class=3DMsoNormal style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier;color:#1F497D'>&gt;</span><=
span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier'>application.<span =
style=3D'color:#1F497D'><o:p></o:p></span></span></p><p =
class=3DMsoNormal style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier'><o:p>&nbsp;</o:p></span></=
p><p class=3DMsoNormal style=3D'text-autospace:none'><span =
lang=3DEN-US>I think that talking about encoding rates for games is a =
bit confusing. The games I have studied do not modify encoding rates, =
since they do not transmit audio or video. They only transmit the game =
status or the actions of the player. The game is rendered in the local =
machine. One exception: games including a voice channel in order to =
coordinate the different players.<o:p></o:p></span></p><p =
class=3DMsoNormal style=3D'text-autospace:none'><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:Courier'><o:p>&nbsp;</o:p></span></=
p></div></body></html>
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From jmpolk@cisco.com  Wed Apr  4 14:35:37 2012
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Date: Wed, 04 Apr 2012 16:35:35 -0500
To: "Jose Saldana" <jsaldana@unizar.es>, <tsvwg@ietf.org>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: Some comments about draft-polk-tsvwg-rfc4594-update-00
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Jose

Thanks for catching these, I'll fix each of them in the upcoming revision.

James

At 01:18 AM 4/4/2012, Jose Saldana wrote:
>James,
>
>I have been reading the draft, and I have found some small errors, 
>which may be corrected in the next version.
>
>Best regards,
>
>Jose Saldana
>
>--------
>
>- Page 3, first paragraph of the introduction: There is something 
>wrong in this sentence:
> >Local administrators are who configuration each router for which 
> Differentiated Services Code
> >Points (DSCP) are to be treated differently, ...
>
>
>- Page 4: There is something missing in this sentence:
> >[ID-DSCP] goes further to explain that may layer 2 technologies 
> fewer bits for marking and prioritization.
>
>
>- Page 15: It is said that
> >Conversational service group consists of five service classes:
>
>But only three classes appear
>
>
>
>- Page 16, Real-time paragraph. Games do not use RTP, but only bare 
>UDP an TCP.
> >Real-Time Interactive, which includes both 'admitted' and 
> 'unadmitted' Realtime-Interactive
> >service class, is for bidirectional variable rate inelastic 
> applications that require low
> >jitter and loss and very low delay, such as interactive gaming 
> applications that use
> >RTP/UDP streams for game control commands,
>
>
>- Page 20: You say that real-time interactive service has to work 
>with very low loss rate. This may be true or not. In a reference I 
>sent yesterday, it was shown that Microsoft's game "Halo" stops 
>working with 4% packet loss, whereas Quake III works well with 30%, 
>and players do not notice
>
>
>- Page 42, first paragraph:
> >Interactive gaming applications that do not have the ability to 
> change encoding rates
> >or to mark packets with different importance indications is one 
> good example of such an
> >application.
>
>I think that talking about encoding rates for games is a bit 
>confusing. The games I have studied do not modify encoding rates, 
>since they do not transmit audio or video. They only transmit the 
>game status or the actions of the player. The game is rendered in 
>the local machine. One exception: games including a voice channel in 
>order to coordinate the different players.
>


From Ruediger.Geib@telekom.de  Thu Apr  5 00:48:47 2012
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From: <Ruediger.Geib@telekom.de>
To: <jsaldana@unizar.es>
Date: Thu, 5 Apr 2012 09:48:37 +0200
Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
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Jose,

you raise an interesting point:

draft-polk-tsvwg-rfc4594-update suggests to standardise application to code=
point mapping as well as codepoint to class properties. You observe that so=
me games are using TCP while others use UDP, which I think is a useful leve=
l of QoS requirement abstraction. There's no need to have like 25+ QoS (sub=
)classes, I think. But one class optimised to transport  "UDP" and one for =
"TCP" are agreeable.

The feedback that I had on carrier class QoS usage is that there's a point =
in standardising class properties, and certainly not to standardise applica=
tion mappings to classes (recommending them may however be useful). My take=
 would be to standardise four classes/properties (in brackets example usage=
):

a) for traffic which doesn't support re-transmits, requiring lowest delay a=
nd jitter, having limited volume and packet
   size, but being loss tolerant (example usage VoIP)
b) for any other traffic which doesn't support re-transmits (let's say traf=
fic with real time like properties, but any
   packet size and possibly high volume, UDP transport, like IPTV)
c) for any other traffic which supports re-transmits (QoS traffic using TCP=
, like application signalling and network management traffic)
d) a default class, for any traffic supporting re-transmits (Best Effort) -=
 this class may not be regarded as QoS (or as lowest CoS)

What I can tell ist that many carriers and standards on QoS support 3-4 cla=
sses, having properties as described above. It is useful to recommend codep=
oints matching to these classes. Standardising these codepoints requires ag=
reement between those having deplyed them. Straightforward for a) EF and d)=
 000000, probably easy for c) being used to transport Network Management (C=
S 6). I'm not aware of common ground for anything else. My favourites are:

b) AF4 (GSMA supports this one, RFC4594 also)
c) AF3 (MEF and GSMA support this one)

Gaming from my point of view would then match to an AF3 or AF4 class, this =
may depend on the implemented application transport. Further details may be=
 discussed.

Everything else could be reserved, but shouldn't be recommended. Carriers h=
ave deployed differing schemes on codepoints and classes. They need coding =
reserverves in 3 bit space (MPLS TC, IP Precedence and Ethernet P-Bits) as =
well as in DSCP space if they should converge to a standardised codepoint/c=
lass scheme.

I object to any approach standardising a match of classes and applications =
with codepoints as a requirement and I also object to standards leaving no =
spare codepoints in 3 bit or 6 bit QoS codepoint space (including 3 bit IP =
precedence).

Bob Briscoe mentioned the difference between classes and queues and above p=
roperties can be mapped to queues. If end to end QoS is desired, the same c=
lass properties must be negotiated end to end. The lower the number of clas=
ses and codepoints being deployed, the higher the probability of getting en=
d to end QoS. I don't support an approach requiring 8 three bit classes and=
 25 or more 6 bit classes and codepoints.

Regards,

Ruediger

________________________________

From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of J=
ose Saldana
Sent: Monday, April 02, 2012 12:17 PM
To: James Polk
Cc: tsvwg@ietf.org
Subject: About draft-polk-tsvwg-rfc4594-update-00.txt



James,



After returning home from IETF 83, I have been reading your draft. I find i=
t interesting to standardize the service classes. Otherwise, if each one cl=
assifies each traffic according to different criteria, Diffserv becomes les=
s useful.



Regarding "Real-time interactive" service class, which refers to online gam=
ing (my main research field), I find it really interesting: online games ar=
e at least as real-time as VoIP is. In most cases, the gamers are a very di=
fficult people to deal with. If the latency is high, they leave and never r=
eturn (this statement is not mine, it can be corroborated in the references=
 I put below). This has been studied by many research groups, with empirica=
l tests using real gamers and adding delay, jitter, packet loss.



Other question: in the draft it is said that online games use RTP/UDP strea=
ms. This can be a little confusing. In fact, the vast majority of online ga=
mes do not use RTP, but bare UDP or even TCP. In http://www.cn.uni-duesseld=
orf.de/publications/library/Mauve2001a.pdf, the authors proposed the use of=
 an adaptation of RTP in order to manage gaming traffic. But as far as I kn=
ow, this approach is not used by commercial games.





A typical classification of today's games is:



-          FPS (First Person Shooters) e.g. Counter Strike, Halo, Call of D=
uty: They use UDP

-          MMORPG (Massively Multiplayer Online Role-playing Games), e.g. W=
orld of Warcraft: They normally use TCP, even though some of them use UDP

-          RTS (Real-time strategy) e.g. Age of Empires: They may use TCP o=
r UDP

-          Sports games: Since there is a big variety, they have not been d=
eeply studied in the literature



Some research papers supporting this classification, and also the real-time=
 requirements of the games:



First Person Shooters



-          A survey of traffic models for 17 First Person Shooters can be f=
ound here. All of them use UDP http://www.discover.uottawa.ca/publications/=
files/MMOG-Survey_IEEE-IC.pdf



-          In this paper from my research group, 8 FPS games were analyzed =
in order to compress and multiplex their traffic. Figure 5 shows their hist=
ograms (inter-packet time and packet size). It can be seen that they genera=
te real-time flows, with characteristics very similar to VoIP (packets of 6=
0-80 bytes every 15-40 ms): http://diec.unizar.es/~jsaldana/personal/commag=
_nov_2011_jsaldana.pdf



-          Another paper about FPS using UDP. Section V presents some inter=
esting conclusions about routing infrastructure. http://www.thefengs.com/wu=
chang/work/cstrike/ton05_traffic.pdf



-          In this paper the network factors annoying the gamers are analyz=
ed. There are also some calculations of the MOS. Above 150 ms the quality i=
s bad (something similar to what happens with VoIP) http://www.research.ibm=
.com/netgames2005/papers/dick.pdf



-          In this one, a MOS model for a FPS (Quake IV) is deployed. It is=
 only based on delay and jitter, since packet loss has vevy little influenc=
e (for this concrete game). Delay above 140 ms produces MOS below 3 http://=
www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf



-          In this paper, it is found that some FPS work properly even with=
 30% packet loss, whereas others stop working with 4%. RTT above 300 ms (15=
0 OWD) makes MOS go above 3 http://caia.swin.edu.au/pubs/ATNAC04/zander-arm=
itage-ATNAC2004.pdf



MMORPGs:

-          A traffic model for World of Warcraft: http://publik.tuwien.ac.a=
t/files/pub-et_12119.pdf



-          This one presents a MOS for World of Warcraft. When added delay =
is above 160ms, the MOS is roughly 3.: http://publik.tuwien.ac.at/files/Pub=
Dat_170202.pdf



Best regards,



Jose Saldana


From jsaldana@unizar.es  Thu Apr  5 02:28:43 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
To: <Ruediger.Geib@telekom.de>
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Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
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Ruediger,

I have read your message about draft-polk-tsvwg-rfc4594-update. As you may
know, I presented another draft in Paris (draft-saldana-tsvwg-tcmtf-02), and
after talking to James, I thought that the idea of standardizing the traffic
classification fits well with my proposal. I would like to know your opinion
about it, if possible.

The idea is to use tunnels in order to compress and multiplex real-time
traffic flows. Since these flows tend to use very small packets, they have
very bad efficiency. So perhaps tunnels could be used in order to group
these flows. The drawback is that they add small delays, i.e. the waiting
time in the multiplexer. So another use is to provide flexibility: if you
are a game provider, you may over provision your network in order to be able
to manage traffic peaks. But perhaps using this, you may be able to have a
smaller network, and use compression when necessary (rush hour, first days
after releasing a new game, etc). When traffic is small, you do not
multiplex, in order to reduce latency.

So if traffic classification is a standard, everyone in the way from the
origin to the destination will handle the traffic with the correct priority.

I attach a figure in order to clarify.

Thank you,

Jose

PS: In
http://community.virginmedia.com/t5/Fibre-optic-broadband-cable/World-of-War
craft-high-latency-issues-with-virgin-media-users/td-p/44498 you can find
the history of the problems of World of Warcraft for Virgin Media users. The
problem was caused by a modification in the traffic, and the provider did
not notice about it, so it treated the packets as best-effort.

-----Mensaje original-----
De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de] 
Enviado el: jueves, 05 de abril de 2012 9:49
Para: jsaldana@unizar.es
CC: tsvwg@ietf.org
Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

Jose,

you raise an interesting point:

draft-polk-tsvwg-rfc4594-update suggests to standardise application to
codepoint mapping as well as codepoint to class properties. You observe that
some games are using TCP while others use UDP, which I think is a useful
level of QoS requirement abstraction. There's no need to have like 25+ QoS
(sub)classes, I think. But one class optimised to transport  "UDP" and one
for "TCP" are agreeable.

The feedback that I had on carrier class QoS usage is that there's a point
in standardising class properties, and certainly not to standardise
application mappings to classes (recommending them may however be useful).
My take would be to standardise four classes/properties (in brackets example
usage):

a) for traffic which doesn't support re-transmits, requiring lowest delay
and jitter, having limited volume and packet
   size, but being loss tolerant (example usage VoIP)
b) for any other traffic which doesn't support re-transmits (let's say
traffic with real time like properties, but any
   packet size and possibly high volume, UDP transport, like IPTV)
c) for any other traffic which supports re-transmits (QoS traffic using TCP,
like application signalling and network management traffic)
d) a default class, for any traffic supporting re-transmits (Best Effort) -
this class may not be regarded as QoS (or as lowest CoS)

What I can tell ist that many carriers and standards on QoS support 3-4
classes, having properties as described above. It is useful to recommend
codepoints matching to these classes. Standardising these codepoints
requires agreement between those having deplyed them. Straightforward for a)
EF and d) 000000, probably easy for c) being used to transport Network
Management (CS 6). I'm not aware of common ground for anything else. My
favourites are:

b) AF4 (GSMA supports this one, RFC4594 also)
c) AF3 (MEF and GSMA support this one)

Gaming from my point of view would then match to an AF3 or AF4 class, this
may depend on the implemented application transport. Further details may be
discussed.

Everything else could be reserved, but shouldn't be recommended. Carriers
have deployed differing schemes on codepoints and classes. They need coding
reserverves in 3 bit space (MPLS TC, IP Precedence and Ethernet P-Bits) as
well as in DSCP space if they should converge to a standardised
codepoint/class scheme.

I object to any approach standardising a match of classes and applications
with codepoints as a requirement and I also object to standards leaving no
spare codepoints in 3 bit or 6 bit QoS codepoint space (including 3 bit IP
precedence).

Bob Briscoe mentioned the difference between classes and queues and above
properties can be mapped to queues. If end to end QoS is desired, the same
class properties must be negotiated end to end. The lower the number of
classes and codepoints being deployed, the higher the probability of getting
end to end QoS. I don't support an approach requiring 8 three bit classes
and 25 or more 6 bit classes and codepoints.

Regards,

Ruediger

________________________________

From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of
Jose Saldana
Sent: Monday, April 02, 2012 12:17 PM
To: James Polk
Cc: tsvwg@ietf.org
Subject: About draft-polk-tsvwg-rfc4594-update-00.txt



James,



After returning home from IETF 83, I have been reading your draft. I find it
interesting to standardize the service classes. Otherwise, if each one
classifies each traffic according to different criteria, Diffserv becomes
less useful.



Regarding "Real-time interactive" service class, which refers to online
gaming (my main research field), I find it really interesting: online games
are at least as real-time as VoIP is. In most cases, the gamers are a very
difficult people to deal with. If the latency is high, they leave and never
return (this statement is not mine, it can be corroborated in the references
I put below). This has been studied by many research groups, with empirical
tests using real gamers and adding delay, jitter, packet loss.



Other question: in the draft it is said that online games use RTP/UDP
streams. This can be a little confusing. In fact, the vast majority of
online games do not use RTP, but bare UDP or even TCP. In
http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.pdf, the
authors proposed the use of an adaptation of RTP in order to manage gaming
traffic. But as far as I know, this approach is not used by commercial
games.





A typical classification of today's games is:



-          FPS (First Person Shooters) e.g. Counter Strike, Halo, Call of
Duty: They use UDP

-          MMORPG (Massively Multiplayer Online Role-playing Games), e.g.
World of Warcraft: They normally use TCP, even though some of them use UDP

-          RTS (Real-time strategy) e.g. Age of Empires: They may use TCP or
UDP

-          Sports games: Since there is a big variety, they have not been
deeply studied in the literature



Some research papers supporting this classification, and also the real-time
requirements of the games:



First Person Shooters



-          A survey of traffic models for 17 First Person Shooters can be
found here. All of them use UDP
http://www.discover.uottawa.ca/publications/files/MMOG-Survey_IEEE-IC.pdf



-          In this paper from my research group, 8 FPS games were analyzed
in order to compress and multiplex their traffic. Figure 5 shows their
histograms (inter-packet time and packet size). It can be seen that they
generate real-time flows, with characteristics very similar to VoIP (packets
of 60-80 bytes every 15-40 ms):
http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana.pdf



-          Another paper about FPS using UDP. Section V presents some
interesting conclusions about routing infrastructure.
http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf



-          In this paper the network factors annoying the gamers are
analyzed. There are also some calculations of the MOS. Above 150 ms the
quality is bad (something similar to what happens with VoIP)
http://www.research.ibm.com/netgames2005/papers/dick.pdf



-          In this one, a MOS model for a FPS (Quake IV) is deployed. It is
only based on delay and jitter, since packet loss has vevy little influence
(for this concrete game). Delay above 140 ms produces MOS below 3
http://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf



-          In this paper, it is found that some FPS work properly even with
30% packet loss, whereas others stop working with 4%. RTT above 300 ms (150
OWD) makes MOS go above 3
http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pdf



MMORPGs:

-          A traffic model for World of Warcraft:
http://publik.tuwien.ac.at/files/pub-et_12119.pdf



-          This one presents a MOS for World of Warcraft. When added delay
is above 160ms, the MOS is roughly 3.:
http://publik.tuwien.ac.at/files/PubDat_170202.pdf



Best regards,



Jose Saldana


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From Ruediger.Geib@telekom.de  Thu Apr  5 03:22:52 2012
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From: <Ruediger.Geib@telekom.de>
To: <jsaldana@unizar.es>
Date: Thu, 5 Apr 2012 12:22:47 +0200
Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
Thread-Topic: About draft-polk-tsvwg-rfc4594-update-00.txt
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Jose,

regarding QoS, I'm happy to agree on 4 standard classes with properties as
mentioned in my last email. If there's consensus on some codepoints for the=
m,
I'll support that.

If IETF recommends how and which applications to use these classes/codepoin=
ts,
that's again agreeable. I don't think, that by now requiring a specific
application to carry a specific DSCP makes sense, apart from few exceptions
(like VoIP payload, network management traffic or Best Effort traffic).
IETF needs consensus to do that.
Note that the number of AF classes and codepoints are troublesome. Too many
with too little explanation how to use them, nothing is aligned with user
demand. EF/VoIP payload and network management are examples how to proceed:
just specify what's required, no extras.


Then regarding tunnels and header compression
Yes, that's smart and there may be more use cases than gaming. I'm a backbo=
ne
engineer however and your proposal is rather application specific. What I
can do is ask some colleagues on their view. But there's no guarantee, that
they will respond.

Some initial thoughts:
To me a key point seems to be what can be reached at the bottleneck. There =
the
compression should save bandwidth with little cost and no negative
service impact. Any kind of shared access from which traffic is transmitted
to the same destination seems to be relevant. The application/tunnel awaren=
ess of
network elements may not be feasible, a network terminal at a UNI may be ab=
le to
handle that, or any service specific device in a network.

Regards,

Ruediger


-----Original Message-----
From: Jose Saldana [mailto:jsaldana@unizar.es]
Sent: Thursday, April 05, 2012 11:29 AM
To: Geib, R=FCdiger
Cc: tsvwg@ietf.org
Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

Ruediger,

I have read your message about draft-polk-tsvwg-rfc4594-update. As you may
know, I presented another draft in Paris (draft-saldana-tsvwg-tcmtf-02), an=
d
after talking to James, I thought that the idea of standardizing the traffi=
c
classification fits well with my proposal. I would like to know your opinio=
n
about it, if possible.

The idea is to use tunnels in order to compress and multiplex real-time
traffic flows. Since these flows tend to use very small packets, they have
very bad efficiency. So perhaps tunnels could be used in order to group
these flows. The drawback is that they add small delays, i.e. the waiting
time in the multiplexer. So another use is to provide flexibility: if you
are a game provider, you may over provision your network in order to be abl=
e
to manage traffic peaks. But perhaps using this, you may be able to have a
smaller network, and use compression when necessary (rush hour, first days
after releasing a new game, etc). When traffic is small, you do not
multiplex, in order to reduce latency.

So if traffic classification is a standard, everyone in the way from the
origin to the destination will handle the traffic with the correct priority=
.

I attach a figure in order to clarify.

Thank you,

Jose

PS: In
http://community.virginmedia.com/t5/Fibre-optic-broadband-cable/World-of-Wa=
r
craft-high-latency-issues-with-virgin-media-users/td-p/44498 you can find
the history of the problems of World of Warcraft for Virgin Media users. Th=
e
problem was caused by a modification in the traffic, and the provider did
not notice about it, so it treated the packets as best-effort.

-----Mensaje original-----
De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]
Enviado el: jueves, 05 de abril de 2012 9:49
Para: jsaldana@unizar.es
CC: tsvwg@ietf.org
Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

Jose,

you raise an interesting point:

draft-polk-tsvwg-rfc4594-update suggests to standardise application to
codepoint mapping as well as codepoint to class properties. You observe tha=
t
some games are using TCP while others use UDP, which I think is a useful
level of QoS requirement abstraction. There's no need to have like 25+ QoS
(sub)classes, I think. But one class optimised to transport  "UDP" and one
for "TCP" are agreeable.

The feedback that I had on carrier class QoS usage is that there's a point
in standardising class properties, and certainly not to standardise
application mappings to classes (recommending them may however be useful).
My take would be to standardise four classes/properties (in brackets exampl=
e
usage):

a) for traffic which doesn't support re-transmits, requiring lowest delay
and jitter, having limited volume and packet
   size, but being loss tolerant (example usage VoIP)
b) for any other traffic which doesn't support re-transmits (let's say
traffic with real time like properties, but any
   packet size and possibly high volume, UDP transport, like IPTV)
c) for any other traffic which supports re-transmits (QoS traffic using TCP=
,
like application signalling and network management traffic)
d) a default class, for any traffic supporting re-transmits (Best Effort) -
this class may not be regarded as QoS (or as lowest CoS)

What I can tell ist that many carriers and standards on QoS support 3-4
classes, having properties as described above. It is useful to recommend
codepoints matching to these classes. Standardising these codepoints
requires agreement between those having deplyed them. Straightforward for a=
)
EF and d) 000000, probably easy for c) being used to transport Network
Management (CS 6). I'm not aware of common ground for anything else. My
favourites are:

b) AF4 (GSMA supports this one, RFC4594 also)
c) AF3 (MEF and GSMA support this one)

Gaming from my point of view would then match to an AF3 or AF4 class, this
may depend on the implemented application transport. Further details may be
discussed.

Everything else could be reserved, but shouldn't be recommended. Carriers
have deployed differing schemes on codepoints and classes. They need coding
reserverves in 3 bit space (MPLS TC, IP Precedence and Ethernet P-Bits) as
well as in DSCP space if they should converge to a standardised
codepoint/class scheme.

I object to any approach standardising a match of classes and applications
with codepoints as a requirement and I also object to standards leaving no
spare codepoints in 3 bit or 6 bit QoS codepoint space (including 3 bit IP
precedence).

Bob Briscoe mentioned the difference between classes and queues and above
properties can be mapped to queues. If end to end QoS is desired, the same
class properties must be negotiated end to end. The lower the number of
classes and codepoints being deployed, the higher the probability of gettin=
g
end to end QoS. I don't support an approach requiring 8 three bit classes
and 25 or more 6 bit classes and codepoints.

Regards,

Ruediger

________________________________

From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of
Jose Saldana
Sent: Monday, April 02, 2012 12:17 PM
To: James Polk
Cc: tsvwg@ietf.org
Subject: About draft-polk-tsvwg-rfc4594-update-00.txt



James,



After returning home from IETF 83, I have been reading your draft. I find i=
t
interesting to standardize the service classes. Otherwise, if each one
classifies each traffic according to different criteria, Diffserv becomes
less useful.



Regarding "Real-time interactive" service class, which refers to online
gaming (my main research field), I find it really interesting: online games
are at least as real-time as VoIP is. In most cases, the gamers are a very
difficult people to deal with. If the latency is high, they leave and never
return (this statement is not mine, it can be corroborated in the reference=
s
I put below). This has been studied by many research groups, with empirical
tests using real gamers and adding delay, jitter, packet loss.



Other question: in the draft it is said that online games use RTP/UDP
streams. This can be a little confusing. In fact, the vast majority of
online games do not use RTP, but bare UDP or even TCP. In
http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.pdf, the
authors proposed the use of an adaptation of RTP in order to manage gaming
traffic. But as far as I know, this approach is not used by commercial
games.





A typical classification of today's games is:



-          FPS (First Person Shooters) e.g. Counter Strike, Halo, Call of
Duty: They use UDP

-          MMORPG (Massively Multiplayer Online Role-playing Games), e.g.
World of Warcraft: They normally use TCP, even though some of them use UDP

-          RTS (Real-time strategy) e.g. Age of Empires: They may use TCP o=
r
UDP

-          Sports games: Since there is a big variety, they have not been
deeply studied in the literature



Some research papers supporting this classification, and also the real-time
requirements of the games:



First Person Shooters



-          A survey of traffic models for 17 First Person Shooters can be
found here. All of them use UDP
http://www.discover.uottawa.ca/publications/files/MMOG-Survey_IEEE-IC.pdf



-          In this paper from my research group, 8 FPS games were analyzed
in order to compress and multiplex their traffic. Figure 5 shows their
histograms (inter-packet time and packet size). It can be seen that they
generate real-time flows, with characteristics very similar to VoIP (packet=
s
of 60-80 bytes every 15-40 ms):
http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana.pdf



-          Another paper about FPS using UDP. Section V presents some
interesting conclusions about routing infrastructure.
http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf



-          In this paper the network factors annoying the gamers are
analyzed. There are also some calculations of the MOS. Above 150 ms the
quality is bad (something similar to what happens with VoIP)
http://www.research.ibm.com/netgames2005/papers/dick.pdf



-          In this one, a MOS model for a FPS (Quake IV) is deployed. It is
only based on delay and jitter, since packet loss has vevy little influence
(for this concrete game). Delay above 140 ms produces MOS below 3
http://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf



-          In this paper, it is found that some FPS work properly even with
30% packet loss, whereas others stop working with 4%. RTT above 300 ms (150
OWD) makes MOS go above 3
http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pdf



MMORPGs:

-          A traffic model for World of Warcraft:
http://publik.tuwien.ac.at/files/pub-et_12119.pdf



-          This one presents a MOS for World of Warcraft. When added delay
is above 160ms, the MOS is roughly 3.:
http://publik.tuwien.ac.at/files/PubDat_170202.pdf



Best regards,



Jose Saldana


From jsaldana@unizar.es  Thu Apr  5 05:14:17 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
To: <Ruediger.Geib@telekom.de>
Subject: About the tunnels: packets per second savings
Date: Thu, 5 Apr 2012 14:14:07 +0200
Message-ID: <000601cd1325$9a1afcb0$ce50f610$@unizar.es>
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Ruediger,

Thank you for your fast response.  I will separately answer the two =
parts of
your email. First, this message about the tunnels. This afternoon or
tomorrow I will send to the list an idea about QoS classes, but I have =
to
think more before sending it.

I'm glad to hear that you consider the "tunnels" proposal interesting. =
There
is another issue that I discussed with Gorry in Paris: the saving in =
terms
of packets per second. I attach a graph of the saving in terms of this
parameter using Counter Strike. If you multiplex all the packets you =
have
received e.g. every 20ms, then you will send  50 pps (the inverse of the
multiplexing period). So you can save a lot of pps.

It is clear that bandwidth saving is interesting, but I have read some
papers in which a "packets per second" upper bound in the routers was
reported:

Section V.A of
W. Feng, F. Chang, W. Feng, J. Walpole: =93A Traffic Characterization of
Popular On-Line Games=94. IEEE/ACM Trans. Netw., pp. 488-500, 2005.
http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf

Section IV.3 of
J. Yu, I. Al-Ajarmeh. =93Call Admission Control and Traffic Engineering =
of
VoIP=94, In Proc. Second Intenational Conference on Digital
Telecommunications, IEEE ICDT 2007.
http://140.192.40.4:8001/~imad/voip/YuJ-CallAdmin.pdf

They not only talk about the bandwidth limitation of routers, but also =
the
packets per second limit. Gorry told me that high-end routers may not =
have
this problem. But low-end routers are not able to back-to-back send =
small
packets, due to the processing limitation.

So, do you think this saving is interesting? Since you work in the =
backbone,
your opinion is important for me, taking into account that I am an =
Academia
researcher, and sometimes we are a little far from real things. Perhaps =
you
may ask some of your colleagues.

Thanks,

Jose=20

-----Mensaje original-----
De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]=20
Enviado el: jueves, 05 de abril de 2012 12:23
Para: jsaldana@unizar.es
CC: tsvwg@ietf.org
Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

Jose,

(...) I have removed the first part of the message for clarity.

Then regarding tunnels and header compression Yes, that's smart and =
there
may be more use cases than gaming. I'm a backbone engineer however and =
your
proposal is rather application specific. What I can do is ask some
colleagues on their view. But there's no guarantee, that they will =
respond.

Some initial thoughts:
To me a key point seems to be what can be reached at the bottleneck. =
There
the compression should save bandwidth with little cost and no negative
service impact. Any kind of shared access from which traffic is =
transmitted
to the same destination seems to be relevant. The application/tunnel
awareness of network elements may not be feasible, a network terminal at =
a
UNI may be able to handle that, or any service specific device in a =
network.

Regards,

Ruediger


-----Original Message-----
From: Jose Saldana [mailto:jsaldana@unizar.es]
Sent: Thursday, April 05, 2012 11:29 AM
To: Geib, R=FCdiger
Cc: tsvwg@ietf.org
Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

Ruediger,

I have read your message about draft-polk-tsvwg-rfc4594-update. As you =
may
know, I presented another draft in Paris (draft-saldana-tsvwg-tcmtf-02), =
and
after talking to James, I thought that the idea of standardizing the =
traffic
classification fits well with my proposal. I would like to know your =
opinion
about it, if possible.

The idea is to use tunnels in order to compress and multiplex real-time
traffic flows. Since these flows tend to use very small packets, they =
have
very bad efficiency. So perhaps tunnels could be used in order to group
these flows. The drawback is that they add small delays, i.e. the =
waiting
time in the multiplexer. So another use is to provide flexibility: if =
you
are a game provider, you may over provision your network in order to be =
able
to manage traffic peaks. But perhaps using this, you may be able to have =
a
smaller network, and use compression when necessary (rush hour, first =
days
after releasing a new game, etc). When traffic is small, you do not
multiplex, in order to reduce latency.

So if traffic classification is a standard, everyone in the way from the
origin to the destination will handle the traffic with the correct =
priority.

I attach a figure in order to clarify.

Thank you,

Jose

PS: In
http://community.virginmedia.com/t5/Fibre-optic-broadband-cable/World-of-=
War
craft-high-latency-issues-with-virgin-media-users/td-p/44498 you can =
find
the history of the problems of World of Warcraft for Virgin Media users. =
The
problem was caused by a modification in the traffic, and the provider =
did
not notice about it, so it treated the packets as best-effort.

-----Mensaje original-----
De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de] Enviado =
el:
jueves, 05 de abril de 2012 9:49
Para: jsaldana@unizar.es
CC: tsvwg@ietf.org
Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

Jose,

you raise an interesting point:

draft-polk-tsvwg-rfc4594-update suggests to standardise application to
codepoint mapping as well as codepoint to class properties. You observe =
that
some games are using TCP while others use UDP, which I think is a useful
level of QoS requirement abstraction. There's no need to have like 25+ =
QoS
(sub)classes, I think. But one class optimised to transport  "UDP" and =
one
for "TCP" are agreeable.

The feedback that I had on carrier class QoS usage is that there's a =
point
in standardising class properties, and certainly not to standardise
application mappings to classes (recommending them may however be =
useful).
My take would be to standardise four classes/properties (in brackets =
example
usage):

a) for traffic which doesn't support re-transmits, requiring lowest =
delay
and jitter, having limited volume and packet
   size, but being loss tolerant (example usage VoIP)
b) for any other traffic which doesn't support re-transmits (let's say
traffic with real time like properties, but any
   packet size and possibly high volume, UDP transport, like IPTV)
c) for any other traffic which supports re-transmits (QoS traffic using =
TCP,
like application signalling and network management traffic)
d) a default class, for any traffic supporting re-transmits (Best =
Effort) -
this class may not be regarded as QoS (or as lowest CoS)

What I can tell ist that many carriers and standards on QoS support 3-4
classes, having properties as described above. It is useful to recommend
codepoints matching to these classes. Standardising these codepoints
requires agreement between those having deplyed them. Straightforward =
for a)
EF and d) 000000, probably easy for c) being used to transport Network
Management (CS 6). I'm not aware of common ground for anything else. My
favourites are:

b) AF4 (GSMA supports this one, RFC4594 also)
c) AF3 (MEF and GSMA support this one)

Gaming from my point of view would then match to an AF3 or AF4 class, =
this
may depend on the implemented application transport. Further details may =
be
discussed.

Everything else could be reserved, but shouldn't be recommended. =
Carriers
have deployed differing schemes on codepoints and classes. They need =
coding
reserverves in 3 bit space (MPLS TC, IP Precedence and Ethernet P-Bits) =
as
well as in DSCP space if they should converge to a standardised
codepoint/class scheme.

I object to any approach standardising a match of classes and =
applications
with codepoints as a requirement and I also object to standards leaving =
no
spare codepoints in 3 bit or 6 bit QoS codepoint space (including 3 bit =
IP
precedence).

Bob Briscoe mentioned the difference between classes and queues and =
above
properties can be mapped to queues. If end to end QoS is desired, the =
same
class properties must be negotiated end to end. The lower the number of
classes and codepoints being deployed, the higher the probability of =
getting
end to end QoS. I don't support an approach requiring 8 three bit =
classes
and 25 or more 6 bit classes and codepoints.

Regards,

Ruediger

________________________________

From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf =
Of
Jose Saldana
Sent: Monday, April 02, 2012 12:17 PM
To: James Polk
Cc: tsvwg@ietf.org
Subject: About draft-polk-tsvwg-rfc4594-update-00.txt



James,



After returning home from IETF 83, I have been reading your draft. I =
find it
interesting to standardize the service classes. Otherwise, if each one
classifies each traffic according to different criteria, Diffserv =
becomes
less useful.



Regarding "Real-time interactive" service class, which refers to online
gaming (my main research field), I find it really interesting: online =
games
are at least as real-time as VoIP is. In most cases, the gamers are a =
very
difficult people to deal with. If the latency is high, they leave and =
never
return (this statement is not mine, it can be corroborated in the =
references
I put below). This has been studied by many research groups, with =
empirical
tests using real gamers and adding delay, jitter, packet loss.



Other question: in the draft it is said that online games use RTP/UDP
streams. This can be a little confusing. In fact, the vast majority of
online games do not use RTP, but bare UDP or even TCP. In
http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.pdf, =
the
authors proposed the use of an adaptation of RTP in order to manage =
gaming
traffic. But as far as I know, this approach is not used by commercial
games.





A typical classification of today's games is:



-          FPS (First Person Shooters) e.g. Counter Strike, Halo, Call =
of
Duty: They use UDP

-          MMORPG (Massively Multiplayer Online Role-playing Games), =
e.g.
World of Warcraft: They normally use TCP, even though some of them use =
UDP

-          RTS (Real-time strategy) e.g. Age of Empires: They may use =
TCP or
UDP

-          Sports games: Since there is a big variety, they have not =
been
deeply studied in the literature



Some research papers supporting this classification, and also the =
real-time
requirements of the games:



First Person Shooters



-          A survey of traffic models for 17 First Person Shooters can =
be
found here. All of them use UDP
http://www.discover.uottawa.ca/publications/files/MMOG-Survey_IEEE-IC.pdf=




-          In this paper from my research group, 8 FPS games were =
analyzed
in order to compress and multiplex their traffic. Figure 5 shows their
histograms (inter-packet time and packet size). It can be seen that they
generate real-time flows, with characteristics very similar to VoIP =
(packets
of 60-80 bytes every 15-40 ms):
http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana.pdf



-          Another paper about FPS using UDP. Section V presents some
interesting conclusions about routing infrastructure.
http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf



-          In this paper the network factors annoying the gamers are
analyzed. There are also some calculations of the MOS. Above 150 ms the
quality is bad (something similar to what happens with VoIP)
http://www.research.ibm.com/netgames2005/papers/dick.pdf



-          In this one, a MOS model for a FPS (Quake IV) is deployed. It =
is
only based on delay and jitter, since packet loss has vevy little =
influence
(for this concrete game). Delay above 140 ms produces MOS below 3
http://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf



-          In this paper, it is found that some FPS work properly even =
with
30% packet loss, whereas others stop working with 4%. RTT above 300 ms =
(150
OWD) makes MOS go above 3
http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pdf



MMORPGs:

-          A traffic model for World of Warcraft:
http://publik.tuwien.ac.at/files/pub-et_12119.pdf



-          This one presents a MOS for World of Warcraft. When added =
delay
is above 160ms, the MOS is roughly 3.:
http://publik.tuwien.ac.at/files/PubDat_170202.pdf



Best regards,



Jose Saldana


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------=_NextPart_000_0007_01CD1336.5DA49000--


From cbenson@adax.com  Thu Apr  5 13:26:04 2012
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Date: Thu, 5 Apr 2012 13:26:03 -0700 (PDT)
From: Chris Benson <cbenson@adax.com>
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To: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
Subject: Re: WGLC for draft-ietf-tsvwg-sctp-udp-encaps-03, ending Friday 20th April 2012
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James and Gorry,

I support this.

I have read and followed development of this draft for 
encapsulating SCTP over UDP.  I recognize the need in
this area, and approve of its content as useful, correct
and complete.

With thanks, from Chris Benson, Adax Inc.

On Wed, 28 Mar 2012, Gorry Fairhurst wrote:

>>  Date: Wed, 28 Mar 2012 15:25:39 +0200
>>  From: Gorry Fairhurst <gorry@erg.abdn.ac.uk>
>>  To:  <tsvwg@ietf.org>
>>  Cc:  <tsvwg-chairs@ietf.org>
>>  Subject: WGLC for draft-ietf-tsvwg-sctp-udp-encaps-03,
>>      ending Friday 20th April 2012
>>  
>>  This email announces the start of a working group last call for
>>  draft-ietf-tsvwg-sctp-udp-encaps, "UDP Encapsulation of SCTP Packets".
>>  
>>  Following the meeting this week, this document is now thought to be ready to
>>  proceed to be published as a BCP. Please send any comments to the TSVWG
>>  list.
>>  
>>  The draft is available at:
>>  http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-udp-encaps-03
>>  
>>  The last call will run for THREE weeks, ending Friday 20th April 2012.
>>  
>>  Emails saying "I support" or "I don't support" publication are also most
>>  helpful in judging the WG consensus. Please let us know if this draft is
>>  useful and seems to provide the correct advice.
>>  
>>  James and Gorry
>>  (TSVWG Chairs)
>>  tsvwg-chairs@ietf.org
>>  

From jsaldana@unizar.es  Fri Apr  6 04:07:16 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
To: "'James M. Polk'" <jmpolk@cisco.com>
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In-Reply-To: <201204042135.q34LZZW0022804@mtv-core-4.cisco.com>
Subject: RE: Some comments about draft-polk-tsvwg-rfc4594-update-00
Date: Fri, 6 Apr 2012 13:07:08 +0200
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One errata:

Page 16: "conversational signaling" service appears twice in the list of
services.

Regards,

Jose

-----Mensaje original-----
De: James M. Polk [mailto:jmpolk@cisco.com]=20
Enviado el: mi=E9rcoles, 04 de abril de 2012 23:36
Para: Jose Saldana; tsvwg@ietf.org
Asunto: Re: Some comments about draft-polk-tsvwg-rfc4594-update-00

Jose

Thanks for catching these, I'll fix each of them in the upcoming =
revision.

James

At 01:18 AM 4/4/2012, Jose Saldana wrote:
>James,
>
>I have been reading the draft, and I have found some small errors,=20
>which may be corrected in the next version.
>
>Best regards,
>
>Jose Saldana
>
>--------
>
>- Page 3, first paragraph of the introduction: There is something wrong =

>in this sentence:
> >Local administrators are who configuration each router for which
> Differentiated Services Code
> >Points (DSCP) are to be treated differently, ...
>
>
>- Page 4: There is something missing in this sentence:
> >[ID-DSCP] goes further to explain that may layer 2 technologies
> fewer bits for marking and prioritization.
>
>
>- Page 15: It is said that
> >Conversational service group consists of five service classes:
>
>But only three classes appear
>
>
>
>- Page 16, Real-time paragraph. Games do not use RTP, but only bare UDP =

>an TCP.
> >Real-Time Interactive, which includes both 'admitted' and
> 'unadmitted' Realtime-Interactive
> >service class, is for bidirectional variable rate inelastic
> applications that require low
> >jitter and loss and very low delay, such as interactive gaming
> applications that use
> >RTP/UDP streams for game control commands,
>
>
>- Page 20: You say that real-time interactive service has to work with=20
>very low loss rate. This may be true or not. In a reference I sent=20
>yesterday, it was shown that Microsoft's game "Halo" stops working with =

>4% packet loss, whereas Quake III works well with 30%, and players do=20
>not notice
>
>
>- Page 42, first paragraph:
> >Interactive gaming applications that do not have the ability to
> change encoding rates
> >or to mark packets with different importance indications is one
> good example of such an
> >application.
>
>I think that talking about encoding rates for games is a bit confusing. =

>The games I have studied do not modify encoding rates, since they do=20
>not transmit audio or video. They only transmit the game status or the=20
>actions of the player. The game is rendered in the local machine. One=20
>exception: games including a voice channel in order to coordinate the=20
>different players.
>


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From: "Jose Saldana" <jsaldana@unizar.es>
To: "'James M. Polk'" <jmpolk@cisco.com>
References: <002101cd122a$d28a5560$779f0020$@unizar.es> <201204042135.q34LZZW0022804@mtv-core-4.cisco.com>
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Subject: RE: Some comments about draft-polk-tsvwg-rfc4594-update-00
Date: Fri, 6 Apr 2012 13:27:36 +0200
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James,

Another question: Clarifying the number of services and clases. I have
summarized the service classification you present in the draft, at the end
of this message. Some suggestions:

1) The fact of grouping "Audio", "Hi-Res Video" and "Video" into a group,
makes it a bit confusing. In the figures, you consider them as different
services, but in page 15 you put them at a different level. This results a
bit confusing for the reader: are them a class or three classes? Since you
use them separately in the next classifications, I think that the paragraph
" Conversational service group consists of five service classes" could be a
comment, and then you could put the three services using "o" instead of "-"

2) Perhaps "Broadcast video" in page 18 could be "Broadcast". There is also
audio broadcast.

3) Perhaps the order of the different services could be the same in section
2.1, section 2.2 and the figures 2, 3 and 4. Perhaps it is not possible, but
the draft would read better.

4) In figure 2, best effort service is called "standard", but in figure 3 it
is called "best effort".

Regards,

Jose

--------------------------------
Summary of the different classifications:

Classification of draft-polk-tsvwg-rfc4594-update-00, section 2.1:

* Network control

o "Network Control" for routing and network control function.
o "OAM" (Operations, Administration, and Management) for network
configuration and management functions.


* The user/subscriber traffic group is broken down into ten (eleven) service
classes to

o Audio		\
o Hi-Res Video	- Conversational service includes three service classes
o Video		/
o Conversational Signaling (it appears twice in page 16)
o Real-Time Interactive
o Multimedia Conferencing
o Multimedia Streaming
o Broadcast
o Low-Latency Data
o High-Throughput Data
o Standard service class, commonly called best effort (BE)
o Low-Priority Data

So we have 2 + 12 = 14 service classes

Figure 2, 3, 4 include 14 service classes.



Classification of draft-polk-tsvwg-rfc4594-update-00, section 2.2 (only
user/subscriber traffic group): 12 application classes, divided into 4
categories

* Application Control Category

o Conversational Signaling 


* Media-Oriented Category

o Audio
o Video
o Hi-Res Video
o Realtime-Interactive
o Multimedia Conferencing 
o Broadcast video 
o Multimedia Streaming


* Data Category

o Low-Latency Data
o High-Throughput Data
o Low-Priority Data

* Best-Effort Category
o Standard


From jsaldana@unizar.es  Sat Apr  7 05:03:40 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
To: <Ruediger.Geib@telekom.de>, "James M. Polk" <jmpolk@cisco.com>
References: <007b01cd10b9$c1c80dc0$45582940$@unizar.es> <580BEA5E3B99744AB1F5BFF5E9A3C67D13DE980BDC@HE111648.emea1.cds.t-internal.com> <000b01cd130e$78c48f70$6a4dae50$@unizar.es> <580BEA5E3B99744AB1F5BFF5E9A3C67D13DE980E7A@HE111648.emea1.cds.t-internal.com>
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Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
Date: Sat, 7 Apr 2012 14:03:24 +0200
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I have been reading the two messages of Ruediger about Jame's draft. I =
will
try to make a "status of the question" and then a proposal. If something =
is
wrong, just tell me.

- We have the old 8 TOS bits of the IP header. Nowadays, two of them are
ECN, and six of them are the DSCP field. So the problem is how to use =
these
6 bits (64 possible values)
- Network control is 11xxxx, so this uses 16 values
	-11xx11 are defined in RFC 2474, and remain as only experimentally
assignable/useable (this uses 4 of these 16 values)
- This leaves 48 values
- Best effort is 000000
- The draft of James proposes 14 values for the DSCP field, which =
correspond
to service classes, mapped to application categories.

But Ruediger says that, in his opinion, more than 4 classes are not =
useful,
according to current implementations.

We can ask ourselves: which are  the key parameters that define how =
packets
have to be managed in a router? I think that they are:

1) "Real-time-ness": The tolerated delay of the service. The jitter can =
be
roughly included in this parameter.
2) loss tolerance: We can distinguish: TCP services are tolerant to =
loss,
since they retransmit packets. And some UDP services are also tolerant =
to
loss (VoIP, some games, etc)

In my opinion, the "real-time-ness" is the key: it distinguishes very
similar services: e.g. a video conference is different from a broadcast,
since the latter permits delays of some seconds, whereas the first has =
to be
interactive.

So one possible solution could be:

- Using the first two bits in order to define 4 classes, taking into =
account
the existing implementations. They can be the 4 classes of Fig. 1 of =
James'
draft, or the ones proposed by Ruediger.
	00xxxx	best effort
	01xxxx data
	10xxxx media oriented
	11xxxx control

- With the other 4 bits, something should be done. My proposal:
	- If the third bit is "1", then the other three bits include a value
defining the "real-time-ness" of the flow. So we have 8 levels of
real-time-ness. Some examples:
		- 10-1-111 would be a  media-oriented (10) and very
real-time flow, as VoIP or a game
		- 10-0-xxx would be a media-oriented flow for which the
real-time-ness is not defined
		- 10-1-010 could be a media-oriented flow with medium delay
requirements, as multimedia conferencing
		- 10-1-000 would be a media-oriented flow with no very hard
delay requirements, as multimedia streaming
		- 00-xxxx would be best-effort
		-01-1-111 would be very low-latency data
		-01-1-000 would be low-priority data

One advantage of this is that some routers may only take into account =
the
first two bits, and others would use all of them.

The new RFC could suggest or define different values of real-time-ness =
for
each application. So the applications should be ordered depending of the
maximum tolerated delay.

Ok. This is only an idea. Perhaps it does not make sense. Just tell me.
Regards,

Jose

-----Mensaje original-----
De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]=20
Enviado el: jueves, 05 de abril de 2012 12:23
Para: jsaldana@unizar.es
CC: tsvwg@ietf.org
Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

Jose,

regarding QoS, I'm happy to agree on 4 standard classes with properties =
as
mentioned in my last email. If there's consensus on some codepoints for
them, I'll support that.

If IETF recommends how and which applications to use these
classes/codepoints, that's again agreeable. I don't think, that by now
requiring a specific application to carry a specific DSCP makes sense, =
apart
from few exceptions (like VoIP payload, network management traffic or =
Best
Effort traffic).
IETF needs consensus to do that.
Note that the number of AF classes and codepoints are troublesome. Too =
many
with too little explanation how to use them, nothing is aligned with =
user
demand. EF/VoIP payload and network management are examples how to =
proceed:
just specify what's required, no extras.


Then regarding tunnels and header compression Yes, that's smart and =
there
may be more use cases than gaming. I'm a backbone engineer however and =
your
proposal is rather application specific. What I can do is ask some
colleagues on their view. But there's no guarantee, that they will =
respond.

Some initial thoughts:
To me a key point seems to be what can be reached at the bottleneck. =
There
the compression should save bandwidth with little cost and no negative
service impact. Any kind of shared access from which traffic is =
transmitted
to the same destination seems to be relevant. The application/tunnel
awareness of network elements may not be feasible, a network terminal at =
a
UNI may be able to handle that, or any service specific device in a =
network.

Regards,

Ruediger


-----Original Message-----
From: Jose Saldana [mailto:jsaldana@unizar.es]
Sent: Thursday, April 05, 2012 11:29 AM
To: Geib, R=FCdiger
Cc: tsvwg@ietf.org
Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

Ruediger,

I have read your message about draft-polk-tsvwg-rfc4594-update. As you =
may
know, I presented another draft in Paris (draft-saldana-tsvwg-tcmtf-02), =
and
after talking to James, I thought that the idea of standardizing the =
traffic
classification fits well with my proposal. I would like to know your =
opinion
about it, if possible.

The idea is to use tunnels in order to compress and multiplex real-time
traffic flows. Since these flows tend to use very small packets, they =
have
very bad efficiency. So perhaps tunnels could be used in order to group
these flows. The drawback is that they add small delays, i.e. the =
waiting
time in the multiplexer. So another use is to provide flexibility: if =
you
are a game provider, you may over provision your network in order to be =
able
to manage traffic peaks. But perhaps using this, you may be able to have =
a
smaller network, and use compression when necessary (rush hour, first =
days
after releasing a new game, etc). When traffic is small, you do not
multiplex, in order to reduce latency.

So if traffic classification is a standard, everyone in the way from the
origin to the destination will handle the traffic with the correct =
priority.

I attach a figure in order to clarify.

Thank you,

Jose

PS: In
http://community.virginmedia.com/t5/Fibre-optic-broadband-cable/World-of-=
War
craft-high-latency-issues-with-virgin-media-users/td-p/44498 you can =
find
the history of the problems of World of Warcraft for Virgin Media users. =
The
problem was caused by a modification in the traffic, and the provider =
did
not notice about it, so it treated the packets as best-effort.

-----Mensaje original-----
De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de] Enviado =
el:
jueves, 05 de abril de 2012 9:49
Para: jsaldana@unizar.es
CC: tsvwg@ietf.org
Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

Jose,

you raise an interesting point:

draft-polk-tsvwg-rfc4594-update suggests to standardise application to
codepoint mapping as well as codepoint to class properties. You observe =
that
some games are using TCP while others use UDP, which I think is a useful
level of QoS requirement abstraction. There's no need to have like 25+ =
QoS
(sub)classes, I think. But one class optimised to transport  "UDP" and =
one
for "TCP" are agreeable.

The feedback that I had on carrier class QoS usage is that there's a =
point
in standardising class properties, and certainly not to standardise
application mappings to classes (recommending them may however be =
useful).
My take would be to standardise four classes/properties (in brackets =
example
usage):

a) for traffic which doesn't support re-transmits, requiring lowest =
delay
and jitter, having limited volume and packet
   size, but being loss tolerant (example usage VoIP)
b) for any other traffic which doesn't support re-transmits (let's say
traffic with real time like properties, but any
   packet size and possibly high volume, UDP transport, like IPTV)
c) for any other traffic which supports re-transmits (QoS traffic using =
TCP,
like application signalling and network management traffic)
d) a default class, for any traffic supporting re-transmits (Best =
Effort) -
this class may not be regarded as QoS (or as lowest CoS)

What I can tell ist that many carriers and standards on QoS support 3-4
classes, having properties as described above. It is useful to recommend
codepoints matching to these classes. Standardising these codepoints
requires agreement between those having deplyed them. Straightforward =
for a)
EF and d) 000000, probably easy for c) being used to transport Network
Management (CS 6). I'm not aware of common ground for anything else. My
favourites are:

b) AF4 (GSMA supports this one, RFC4594 also)
c) AF3 (MEF and GSMA support this one)

Gaming from my point of view would then match to an AF3 or AF4 class, =
this
may depend on the implemented application transport. Further details may =
be
discussed.

Everything else could be reserved, but shouldn't be recommended. =
Carriers
have deployed differing schemes on codepoints and classes. They need =
coding
reserverves in 3 bit space (MPLS TC, IP Precedence and Ethernet P-Bits) =
as
well as in DSCP space if they should converge to a standardised
codepoint/class scheme.

I object to any approach standardising a match of classes and =
applications
with codepoints as a requirement and I also object to standards leaving =
no
spare codepoints in 3 bit or 6 bit QoS codepoint space (including 3 bit =
IP
precedence).

Bob Briscoe mentioned the difference between classes and queues and =
above
properties can be mapped to queues. If end to end QoS is desired, the =
same
class properties must be negotiated end to end. The lower the number of
classes and codepoints being deployed, the higher the probability of =
getting
end to end QoS. I don't support an approach requiring 8 three bit =
classes
and 25 or more 6 bit classes and codepoints.

Regards,

Ruediger

________________________________

From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf =
Of
Jose Saldana
Sent: Monday, April 02, 2012 12:17 PM
To: James Polk
Cc: tsvwg@ietf.org
Subject: About draft-polk-tsvwg-rfc4594-update-00.txt



James,



After returning home from IETF 83, I have been reading your draft. I =
find it
interesting to standardize the service classes. Otherwise, if each one
classifies each traffic according to different criteria, Diffserv =
becomes
less useful.



Regarding "Real-time interactive" service class, which refers to online
gaming (my main research field), I find it really interesting: online =
games
are at least as real-time as VoIP is. In most cases, the gamers are a =
very
difficult people to deal with. If the latency is high, they leave and =
never
return (this statement is not mine, it can be corroborated in the =
references
I put below). This has been studied by many research groups, with =
empirical
tests using real gamers and adding delay, jitter, packet loss.



Other question: in the draft it is said that online games use RTP/UDP
streams. This can be a little confusing. In fact, the vast majority of
online games do not use RTP, but bare UDP or even TCP. In
http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.pdf, =
the
authors proposed the use of an adaptation of RTP in order to manage =
gaming
traffic. But as far as I know, this approach is not used by commercial
games.





A typical classification of today's games is:



-          FPS (First Person Shooters) e.g. Counter Strike, Halo, Call =
of
Duty: They use UDP

-          MMORPG (Massively Multiplayer Online Role-playing Games), =
e.g.
World of Warcraft: They normally use TCP, even though some of them use =
UDP

-          RTS (Real-time strategy) e.g. Age of Empires: They may use =
TCP or
UDP

-          Sports games: Since there is a big variety, they have not =
been
deeply studied in the literature



Some research papers supporting this classification, and also the =
real-time
requirements of the games:



First Person Shooters



-          A survey of traffic models for 17 First Person Shooters can =
be
found here. All of them use UDP
http://www.discover.uottawa.ca/publications/files/MMOG-Survey_IEEE-IC.pdf=




-          In this paper from my research group, 8 FPS games were =
analyzed
in order to compress and multiplex their traffic. Figure 5 shows their
histograms (inter-packet time and packet size). It can be seen that they
generate real-time flows, with characteristics very similar to VoIP =
(packets
of 60-80 bytes every 15-40 ms):
http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana.pdf



-          Another paper about FPS using UDP. Section V presents some
interesting conclusions about routing infrastructure.
http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf



-          In this paper the network factors annoying the gamers are
analyzed. There are also some calculations of the MOS. Above 150 ms the
quality is bad (something similar to what happens with VoIP)
http://www.research.ibm.com/netgames2005/papers/dick.pdf



-          In this one, a MOS model for a FPS (Quake IV) is deployed. It =
is
only based on delay and jitter, since packet loss has vevy little =
influence
(for this concrete game). Delay above 140 ms produces MOS below 3
http://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf



-          In this paper, it is found that some FPS work properly even =
with
30% packet loss, whereas others stop working with 4%. RTT above 300 ms =
(150
OWD) makes MOS go above 3
http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pdf



MMORPGs:

-          A traffic model for World of Warcraft:
http://publik.tuwien.ac.at/files/pub-et_12119.pdf



-          This one presents a MOS for World of Warcraft. When added =
delay
is above 160ms, the MOS is roughly 3.:
http://publik.tuwien.ac.at/files/PubDat_170202.pdf



Best regards,



Jose Saldana



From jmpolk@cisco.com  Mon Apr  9 12:55:16 2012
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Date: Mon, 09 Apr 2012 14:55:12 -0500
To: "Jose Saldana" <jsaldana@unizar.es>, <Ruediger.Geib@telekom.de>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
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Jose

Thanks for giving this effort. SPs usually use=20
MPLS, which has 3 bits (8 values) of=20
differentiation. Thus, for SPs, this is the goal=20
-> to be 8 values or less. I don't expect too=20
many, if any, SPs to want to do 8 different=20
values per customer, so I understand where=20
Ruediger is coming from. At the same time, there=20
are many networks that want far more than 8=20
values for use, so SP desires to box everyone=20
into only their way of managing the world is not=20
satisfactory to these other networks.  Thus, the=20
draft discusses aggregating several service=20
classes into a since value when traveling over SP=20
networks. The draft says in many places that not=20
all service classes are needed, and in fact that=20
few likely would be used across SP networks. What=20
the draft *didn't* make clear is saying this idea=20
in a concise part (i.e., its own section) of the=20
draft (instead, it was spread over the whole=20
draft). I said in the meeting that was an error=20
on my part. I also said at the meeting the next=20
version of the draft will have this concept in a=20
nice, concise standalone section that (hopefully)=20
will make this idea more apparent.

I plan to get the next version written well before Vancouver, for WG review.

James

At 07:03 AM 4/7/2012, Jose Saldana wrote:
>I have been reading the two messages of Ruediger about Jame's draft. I will
>try to make a "status of the question" and then a proposal. If something is
>wrong, just tell me.
>
>- We have the old 8 TOS bits of the IP header. Nowadays, two of them are
>ECN, and six of them are the DSCP field. So the problem is how to use these
>6 bits (64 possible values)
>- Network control is 11xxxx, so this uses 16 values
>         -11xx11 are defined in RFC 2474, and remain as only experimentally
>assignable/useable (this uses 4 of these 16 values)
>- This leaves 48 values
>- Best effort is 000000
>- The draft of James proposes 14 values for the DSCP field, which=
 correspond
>to service classes, mapped to application categories.
>
>But Ruediger says that, in his opinion, more than 4 classes are not useful,
>according to current implementations.
>
>We can ask ourselves: which are  the key parameters that define how packets
>have to be managed in a router? I think that they are:
>
>1) "Real-time-ness": The tolerated delay of the service. The jitter can be
>roughly included in this parameter.
>2) loss tolerance: We can distinguish: TCP services are tolerant to loss,
>since they retransmit packets. And some UDP services are also tolerant to
>loss (VoIP, some games, etc)
>
>In my opinion, the "real-time-ness" is the key: it distinguishes very
>similar services: e.g. a video conference is different from a broadcast,
>since the latter permits delays of some seconds, whereas the first has to=
 be
>interactive.
>
>So one possible solution could be:
>
>- Using the first two bits in order to define 4 classes, taking into=
 account
>the existing implementations. They can be the 4 classes of Fig. 1 of James'
>draft, or the ones proposed by Ruediger.
>         00xxxx  best effort
>         01xxxx data
>         10xxxx media oriented
>         11xxxx control
>
>- With the other 4 bits, something should be done. My proposal:
>         - If the third bit is "1", then the other three bits include a=
 value
>defining the "real-time-ness" of the flow. So we have 8 levels of
>real-time-ness. Some examples:
>                 - 10-1-111 would be a  media-oriented (10) and very
>real-time flow, as VoIP or a game
>                 - 10-0-xxx would be a media-oriented flow for which the
>real-time-ness is not defined
>                 - 10-1-010 could be a media-oriented flow with medium=
 delay
>requirements, as multimedia conferencing
>                 - 10-1-000 would be a media-oriented flow with no very=
 hard
>delay requirements, as multimedia streaming
>                 - 00-xxxx would be best-effort
>                 -01-1-111 would be very low-latency data
>                 -01-1-000 would be low-priority data
>
>One advantage of this is that some routers may only take into account the
>first two bits, and others would use all of them.
>
>The new RFC could suggest or define different values of real-time-ness for
>each application. So the applications should be ordered depending of the
>maximum tolerated delay.
>
>Ok. This is only an idea. Perhaps it does not make sense. Just tell me.
>Regards,
>
>Jose
>
>-----Mensaje original-----
>De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]
>Enviado el: jueves, 05 de abril de 2012 12:23
>Para: jsaldana@unizar.es
>CC: tsvwg@ietf.org
>Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
>
>Jose,
>
>regarding QoS, I'm happy to agree on 4 standard classes with properties as
>mentioned in my last email. If there's consensus on some codepoints for
>them, I'll support that.
>
>If IETF recommends how and which applications to use these
>classes/codepoints, that's again agreeable. I don't think, that by now
>requiring a specific application to carry a specific DSCP makes sense,=
 apart
>from few exceptions (like VoIP payload, network management traffic or Best
>Effort traffic).
>IETF needs consensus to do that.
>Note that the number of AF classes and codepoints are troublesome. Too many
>with too little explanation how to use them, nothing is aligned with user
>demand. EF/VoIP payload and network management are examples how to proceed:
>just specify what's required, no extras.
>
>
>Then regarding tunnels and header compression Yes, that's smart and there
>may be more use cases than gaming. I'm a backbone engineer however and your
>proposal is rather application specific. What I can do is ask some
>colleagues on their view. But there's no guarantee, that they will respond.
>
>Some initial thoughts:
>To me a key point seems to be what can be reached at the bottleneck. There
>the compression should save bandwidth with little cost and no negative
>service impact. Any kind of shared access from which traffic is transmitted
>to the same destination seems to be relevant. The application/tunnel
>awareness of network elements may not be feasible, a network terminal at a
>UNI may be able to handle that, or any service specific device in a=
 network.
>
>Regards,
>
>Ruediger
>
>
>-----Original Message-----
>From: Jose Saldana [mailto:jsaldana@unizar.es]
>Sent: Thursday, April 05, 2012 11:29 AM
>To: Geib, R=FCdiger
>Cc: tsvwg@ietf.org
>Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
>
>Ruediger,
>
>I have read your message about draft-polk-tsvwg-rfc4594-update. As you may
>know, I presented another draft in Paris (draft-saldana-tsvwg-tcmtf-02),=
 and
>after talking to James, I thought that the idea of standardizing the=
 traffic
>classification fits well with my proposal. I would like to know your=
 opinion
>about it, if possible.
>
>The idea is to use tunnels in order to compress and multiplex real-time
>traffic flows. Since these flows tend to use very small packets, they have
>very bad efficiency. So perhaps tunnels could be used in order to group
>these flows. The drawback is that they add small delays, i.e. the waiting
>time in the multiplexer. So another use is to provide flexibility: if you
>are a game provider, you may over provision your network in order to be=
 able
>to manage traffic peaks. But perhaps using this, you may be able to have a
>smaller network, and use compression when necessary (rush hour, first days
>after releasing a new game, etc). When traffic is small, you do not
>multiplex, in order to reduce latency.
>
>So if traffic classification is a standard, everyone in the way from the
>origin to the destination will handle the traffic with the correct=
 priority.
>
>I attach a figure in order to clarify.
>
>Thank you,
>
>Jose
>
>PS: In
>http://community.virginmedia.com/t5/Fibre-optic-broadband-cable/World-of-Wa=
r
>craft-high-latency-issues-with-virgin-media-users/td-p/44498 you can find
>the history of the problems of World of Warcraft for Virgin Media users.=
 The
>problem was caused by a modification in the traffic, and the provider did
>not notice about it, so it treated the packets as best-effort.
>
>-----Mensaje original-----
>De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de] Enviado el:
>jueves, 05 de abril de 2012 9:49
>Para: jsaldana@unizar.es
>CC: tsvwg@ietf.org
>Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
>
>Jose,
>
>you raise an interesting point:
>
>draft-polk-tsvwg-rfc4594-update suggests to standardise application to
>codepoint mapping as well as codepoint to class properties. You observe=
 that
>some games are using TCP while others use UDP, which I think is a useful
>level of QoS requirement abstraction. There's no need to have like 25+ QoS
>(sub)classes, I think. But one class optimised to transport  "UDP" and one
>for "TCP" are agreeable.
>
>The feedback that I had on carrier class QoS usage is that there's a point
>in standardising class properties, and certainly not to standardise
>application mappings to classes (recommending them may however be useful).
>My take would be to standardise four classes/properties (in brackets=
 example
>usage):
>
>a) for traffic which doesn't support re-transmits, requiring lowest delay
>and jitter, having limited volume and packet
>    size, but being loss tolerant (example usage VoIP)
>b) for any other traffic which doesn't support re-transmits (let's say
>traffic with real time like properties, but any
>    packet size and possibly high volume, UDP transport, like IPTV)
>c) for any other traffic which supports re-transmits (QoS traffic using=
 TCP,
>like application signalling and network management traffic)
>d) a default class, for any traffic supporting re-transmits (Best Effort) -
>this class may not be regarded as QoS (or as lowest CoS)
>
>What I can tell ist that many carriers and standards on QoS support 3-4
>classes, having properties as described above. It is useful to recommend
>codepoints matching to these classes. Standardising these codepoints
>requires agreement between those having deplyed them. Straightforward for=
 a)
>EF and d) 000000, probably easy for c) being used to transport Network
>Management (CS 6). I'm not aware of common ground for anything else. My
>favourites are:
>
>b) AF4 (GSMA supports this one, RFC4594 also)
>c) AF3 (MEF and GSMA support this one)
>
>Gaming from my point of view would then match to an AF3 or AF4 class, this
>may depend on the implemented application transport. Further details may be
>discussed.
>
>Everything else could be reserved, but shouldn't be recommended. Carriers
>have deployed differing schemes on codepoints and classes. They need coding
>reserverves in 3 bit space (MPLS TC, IP Precedence and Ethernet P-Bits) as
>well as in DSCP space if they should converge to a standardised
>codepoint/class scheme.
>
>I object to any approach standardising a match of classes and applications
>with codepoints as a requirement and I also object to standards leaving no
>spare codepoints in 3 bit or 6 bit QoS codepoint space (including 3 bit IP
>precedence).
>
>Bob Briscoe mentioned the difference between classes and queues and above
>properties can be mapped to queues. If end to end QoS is desired, the same
>class properties must be negotiated end to end. The lower the number of
>classes and codepoints being deployed, the higher the probability of=
 getting
>end to end QoS. I don't support an approach requiring 8 three bit classes
>and 25 or more 6 bit classes and codepoints.
>
>Regards,
>
>Ruediger
>
>________________________________
>
>From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of
>Jose Saldana
>Sent: Monday, April 02, 2012 12:17 PM
>To: James Polk
>Cc: tsvwg@ietf.org
>Subject: About draft-polk-tsvwg-rfc4594-update-00.txt
>
>
>
>James,
>
>
>
>After returning home from IETF 83, I have been reading your draft. I find=
 it
>interesting to standardize the service classes. Otherwise, if each one
>classifies each traffic according to different criteria, Diffserv becomes
>less useful.
>
>
>
>Regarding "Real-time interactive" service class, which refers to online
>gaming (my main research field), I find it really interesting: online games
>are at least as real-time as VoIP is. In most cases, the gamers are a very
>difficult people to deal with. If the latency is high, they leave and never
>return (this statement is not mine, it can be corroborated in the=
 references
>I put below). This has been studied by many research groups, with empirical
>tests using real gamers and adding delay, jitter, packet loss.
>
>
>
>Other question: in the draft it is said that online games use RTP/UDP
>streams. This can be a little confusing. In fact, the vast majority of
>online games do not use RTP, but bare UDP or even TCP. In
>http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.pdf, the
>authors proposed the use of an adaptation of RTP in order to manage gaming
>traffic. But as far as I know, this approach is not used by commercial
>games.
>
>
>
>
>
>A typical classification of today's games is:
>
>
>
>-          FPS (First Person Shooters) e.g. Counter Strike, Halo, Call of
>Duty: They use UDP
>
>-          MMORPG (Massively Multiplayer Online Role-playing Games), e.g.
>World of Warcraft: They normally use TCP, even though some of them use UDP
>
>-          RTS (Real-time strategy) e.g. Age of Empires: They may use TCP=
 or
>UDP
>
>-          Sports games: Since there is a big variety, they have not been
>deeply studied in the literature
>
>
>
>Some research papers supporting this classification, and also the real-time
>requirements of the games:
>
>
>
>First Person Shooters
>
>
>
>-          A survey of traffic models for 17 First Person Shooters can be
>found here. All of them use UDP
>http://www.discover.uottawa.ca/publications/files/MMOG-Survey_IEEE-IC.pdf
>
>
>
>-          In this paper from my research group, 8 FPS games were analyzed
>in order to compress and multiplex their traffic. Figure 5 shows their
>histograms (inter-packet time and packet size). It can be seen that they
>generate real-time flows, with characteristics very similar to VoIP=
 (packets
>of 60-80 bytes every 15-40 ms):
>http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana.pdf
>
>
>
>-          Another paper about FPS using UDP. Section V presents some
>interesting conclusions about routing infrastructure.
>http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf
>
>
>
>-          In this paper the network factors annoying the gamers are
>analyzed. There are also some calculations of the MOS. Above 150 ms the
>quality is bad (something similar to what happens with VoIP)
>http://www.research.ibm.com/netgames2005/papers/dick.pdf
>
>
>
>-          In this one, a MOS model for a FPS (Quake IV) is deployed. It is
>only based on delay and jitter, since packet loss has vevy little influence
>(for this concrete game). Delay above 140 ms produces MOS below 3
>http://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf
>
>
>
>-          In this paper, it is found that some FPS work properly even with
>30% packet loss, whereas others stop working with 4%. RTT above 300 ms (150
>OWD) makes MOS go above 3
>http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pdf
>
>
>
>MMORPGs:
>
>-          A traffic model for World of Warcraft:
>http://publik.tuwien.ac.at/files/pub-et_12119.pdf
>
>
>
>-          This one presents a MOS for World of Warcraft. When added delay
>is above 160ms, the MOS is roughly 3.:
>http://publik.tuwien.ac.at/files/PubDat_170202.pdf
>
>
>
>Best regards,
>
>
>
>Jose Saldana


From david.black@emc.com  Tue Apr 10 15:35:03 2012
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To: <jsaldana@unizar.es>, <Ruediger.Geib@telekom.de>, <jmpolk@cisco.com>
Date: Tue, 10 Apr 2012 18:34:33 -0400
Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
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> - We have the old 8 TOS bits of the IP header. Nowadays, two of them are
> ECN, and six of them are the DSCP field. So the problem is how to use the=
se
> 6 bits (64 possible values)
> - Network control is 11xxxx, so this uses 16 values
>       -11xx11 are defined in RFC 2474, and remain as only experimentally
> assignable/useable (this uses 4 of these 16 values)
> - This leaves 48 values
> - Best effort is 000000
> - The draft of James proposes 14 values for the DSCP field, which corresp=
ond
> to service classes, mapped to application categories.
>
> But Ruediger says that, in his opinion, more than 4 classes are not usefu=
l,
> according to current implementations.

That's actually not appropriate for an RFC 4594 update - the proper place t=
o
address that requirement is an update to RFC 5127, which will need to be up=
dated
to be consistent with whatever is done to RFC 4594.

Thanks,
--David


> -----Original Message-----
> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of=
 Jose
> Saldana
> Sent: Saturday, April 07, 2012 8:03 AM
> To: Ruediger.Geib@telekom.de; James M. Polk
> Cc: tsvwg@ietf.org
> Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
>
> I have been reading the two messages of Ruediger about Jame's draft. I wi=
ll
> try to make a "status of the question" and then a proposal. If something =
is
> wrong, just tell me.
>
> - We have the old 8 TOS bits of the IP header. Nowadays, two of them are
> ECN, and six of them are the DSCP field. So the problem is how to use the=
se
> 6 bits (64 possible values)
> - Network control is 11xxxx, so this uses 16 values
>       -11xx11 are defined in RFC 2474, and remain as only experimentally
> assignable/useable (this uses 4 of these 16 values)
> - This leaves 48 values
> - Best effort is 000000
> - The draft of James proposes 14 values for the DSCP field, which corresp=
ond
> to service classes, mapped to application categories.
>
> But Ruediger says that, in his opinion, more than 4 classes are not usefu=
l,
> according to current implementations.
>
> We can ask ourselves: which are  the key parameters that define how packe=
ts
> have to be managed in a router? I think that they are:
>
> 1) "Real-time-ness": The tolerated delay of the service. The jitter can b=
e
> roughly included in this parameter.
> 2) loss tolerance: We can distinguish: TCP services are tolerant to loss,
> since they retransmit packets. And some UDP services are also tolerant to
> loss (VoIP, some games, etc)
>
> In my opinion, the "real-time-ness" is the key: it distinguishes very
> similar services: e.g. a video conference is different from a broadcast,
> since the latter permits delays of some seconds, whereas the first has to=
 be
> interactive.
>
> So one possible solution could be:
>
> - Using the first two bits in order to define 4 classes, taking into acco=
unt
> the existing implementations. They can be the 4 classes of Fig. 1 of Jame=
s'
> draft, or the ones proposed by Ruediger.
>       00xxxx  best effort
>       01xxxx data
>       10xxxx media oriented
>       11xxxx control
>
> - With the other 4 bits, something should be done. My proposal:
>       - If the third bit is "1", then the other three bits include a valu=
e
> defining the "real-time-ness" of the flow. So we have 8 levels of
> real-time-ness. Some examples:
>               - 10-1-111 would be a  media-oriented (10) and very
> real-time flow, as VoIP or a game
>               - 10-0-xxx would be a media-oriented flow for which the
> real-time-ness is not defined
>               - 10-1-010 could be a media-oriented flow with medium delay
> requirements, as multimedia conferencing
>               - 10-1-000 would be a media-oriented flow with no very hard
> delay requirements, as multimedia streaming
>               - 00-xxxx would be best-effort
>               -01-1-111 would be very low-latency data
>               -01-1-000 would be low-priority data
>
> One advantage of this is that some routers may only take into account the
> first two bits, and others would use all of them.
>
> The new RFC could suggest or define different values of real-time-ness fo=
r
> each application. So the applications should be ordered depending of the
> maximum tolerated delay.
>
> Ok. This is only an idea. Perhaps it does not make sense. Just tell me.
> Regards,
>
> Jose
>
> -----Mensaje original-----
> De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]
> Enviado el: jueves, 05 de abril de 2012 12:23
> Para: jsaldana@unizar.es
> CC: tsvwg@ietf.org
> Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
>
> Jose,
>
> regarding QoS, I'm happy to agree on 4 standard classes with properties a=
s
> mentioned in my last email. If there's consensus on some codepoints for
> them, I'll support that.
>
> If IETF recommends how and which applications to use these
> classes/codepoints, that's again agreeable. I don't think, that by now
> requiring a specific application to carry a specific DSCP makes sense, ap=
art
> from few exceptions (like VoIP payload, network management traffic or Bes=
t
> Effort traffic).
> IETF needs consensus to do that.
> Note that the number of AF classes and codepoints are troublesome. Too ma=
ny
> with too little explanation how to use them, nothing is aligned with user
> demand. EF/VoIP payload and network management are examples how to procee=
d:
> just specify what's required, no extras.
>
>
> Then regarding tunnels and header compression Yes, that's smart and there
> may be more use cases than gaming. I'm a backbone engineer however and yo=
ur
> proposal is rather application specific. What I can do is ask some
> colleagues on their view. But there's no guarantee, that they will respon=
d.
>
> Some initial thoughts:
> To me a key point seems to be what can be reached at the bottleneck. Ther=
e
> the compression should save bandwidth with little cost and no negative
> service impact. Any kind of shared access from which traffic is transmitt=
ed
> to the same destination seems to be relevant. The application/tunnel
> awareness of network elements may not be feasible, a network terminal at =
a
> UNI may be able to handle that, or any service specific device in a netwo=
rk.
>
> Regards,
>
> Ruediger
>
>
> -----Original Message-----
> From: Jose Saldana [mailto:jsaldana@unizar.es]
> Sent: Thursday, April 05, 2012 11:29 AM
> To: Geib, R=FCdiger
> Cc: tsvwg@ietf.org
> Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
>
> Ruediger,
>
> I have read your message about draft-polk-tsvwg-rfc4594-update. As you ma=
y
> know, I presented another draft in Paris (draft-saldana-tsvwg-tcmtf-02), =
and
> after talking to James, I thought that the idea of standardizing the traf=
fic
> classification fits well with my proposal. I would like to know your opin=
ion
> about it, if possible.
>
> The idea is to use tunnels in order to compress and multiplex real-time
> traffic flows. Since these flows tend to use very small packets, they hav=
e
> very bad efficiency. So perhaps tunnels could be used in order to group
> these flows. The drawback is that they add small delays, i.e. the waiting
> time in the multiplexer. So another use is to provide flexibility: if you
> are a game provider, you may over provision your network in order to be a=
ble
> to manage traffic peaks. But perhaps using this, you may be able to have =
a
> smaller network, and use compression when necessary (rush hour, first day=
s
> after releasing a new game, etc). When traffic is small, you do not
> multiplex, in order to reduce latency.
>
> So if traffic classification is a standard, everyone in the way from the
> origin to the destination will handle the traffic with the correct priori=
ty.
>
> I attach a figure in order to clarify.
>
> Thank you,
>
> Jose
>
> PS: In
> http://community.virginmedia.com/t5/Fibre-optic-broadband-cable/World-of-=
War
> craft-high-latency-issues-with-virgin-media-users/td-p/44498 you can find
> the history of the problems of World of Warcraft for Virgin Media users. =
The
> problem was caused by a modification in the traffic, and the provider did
> not notice about it, so it treated the packets as best-effort.
>
> -----Mensaje original-----
> De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de] Enviado el=
:
> jueves, 05 de abril de 2012 9:49
> Para: jsaldana@unizar.es
> CC: tsvwg@ietf.org
> Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
>
> Jose,
>
> you raise an interesting point:
>
> draft-polk-tsvwg-rfc4594-update suggests to standardise application to
> codepoint mapping as well as codepoint to class properties. You observe t=
hat
> some games are using TCP while others use UDP, which I think is a useful
> level of QoS requirement abstraction. There's no need to have like 25+ Qo=
S
> (sub)classes, I think. But one class optimised to transport  "UDP" and on=
e
> for "TCP" are agreeable.
>
> The feedback that I had on carrier class QoS usage is that there's a poin=
t
> in standardising class properties, and certainly not to standardise
> application mappings to classes (recommending them may however be useful)=
.
> My take would be to standardise four classes/properties (in brackets exam=
ple
> usage):
>
> a) for traffic which doesn't support re-transmits, requiring lowest delay
> and jitter, having limited volume and packet
>    size, but being loss tolerant (example usage VoIP)
> b) for any other traffic which doesn't support re-transmits (let's say
> traffic with real time like properties, but any
>    packet size and possibly high volume, UDP transport, like IPTV)
> c) for any other traffic which supports re-transmits (QoS traffic using T=
CP,
> like application signalling and network management traffic)
> d) a default class, for any traffic supporting re-transmits (Best Effort)=
 -
> this class may not be regarded as QoS (or as lowest CoS)
>
> What I can tell ist that many carriers and standards on QoS support 3-4
> classes, having properties as described above. It is useful to recommend
> codepoints matching to these classes. Standardising these codepoints
> requires agreement between those having deplyed them. Straightforward for=
 a)
> EF and d) 000000, probably easy for c) being used to transport Network
> Management (CS 6). I'm not aware of common ground for anything else. My
> favourites are:
>
> b) AF4 (GSMA supports this one, RFC4594 also)
> c) AF3 (MEF and GSMA support this one)
>
> Gaming from my point of view would then match to an AF3 or AF4 class, thi=
s
> may depend on the implemented application transport. Further details may =
be
> discussed.
>
> Everything else could be reserved, but shouldn't be recommended. Carriers
> have deployed differing schemes on codepoints and classes. They need codi=
ng
> reserverves in 3 bit space (MPLS TC, IP Precedence and Ethernet P-Bits) a=
s
> well as in DSCP space if they should converge to a standardised
> codepoint/class scheme.
>
> I object to any approach standardising a match of classes and application=
s
> with codepoints as a requirement and I also object to standards leaving n=
o
> spare codepoints in 3 bit or 6 bit QoS codepoint space (including 3 bit I=
P
> precedence).
>
> Bob Briscoe mentioned the difference between classes and queues and above
> properties can be mapped to queues. If end to end QoS is desired, the sam=
e
> class properties must be negotiated end to end. The lower the number of
> classes and codepoints being deployed, the higher the probability of gett=
ing
> end to end QoS. I don't support an approach requiring 8 three bit classes
> and 25 or more 6 bit classes and codepoints.
>
> Regards,
>
> Ruediger
>
> ________________________________
>
> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of
> Jose Saldana
> Sent: Monday, April 02, 2012 12:17 PM
> To: James Polk
> Cc: tsvwg@ietf.org
> Subject: About draft-polk-tsvwg-rfc4594-update-00.txt
>
>
>
> James,
>
>
>
> After returning home from IETF 83, I have been reading your draft. I find=
 it
> interesting to standardize the service classes. Otherwise, if each one
> classifies each traffic according to different criteria, Diffserv becomes
> less useful.
>
>
>
> Regarding "Real-time interactive" service class, which refers to online
> gaming (my main research field), I find it really interesting: online gam=
es
> are at least as real-time as VoIP is. In most cases, the gamers are a ver=
y
> difficult people to deal with. If the latency is high, they leave and nev=
er
> return (this statement is not mine, it can be corroborated in the referen=
ces
> I put below). This has been studied by many research groups, with empiric=
al
> tests using real gamers and adding delay, jitter, packet loss.
>
>
>
> Other question: in the draft it is said that online games use RTP/UDP
> streams. This can be a little confusing. In fact, the vast majority of
> online games do not use RTP, but bare UDP or even TCP. In
> http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.pdf, the
> authors proposed the use of an adaptation of RTP in order to manage gamin=
g
> traffic. But as far as I know, this approach is not used by commercial
> games.
>
>
>
>
>
> A typical classification of today's games is:
>
>
>
> -          FPS (First Person Shooters) e.g. Counter Strike, Halo, Call of
> Duty: They use UDP
>
> -          MMORPG (Massively Multiplayer Online Role-playing Games), e.g.
> World of Warcraft: They normally use TCP, even though some of them use UD=
P
>
> -          RTS (Real-time strategy) e.g. Age of Empires: They may use TCP=
 or
> UDP
>
> -          Sports games: Since there is a big variety, they have not been
> deeply studied in the literature
>
>
>
> Some research papers supporting this classification, and also the real-ti=
me
> requirements of the games:
>
>
>
> First Person Shooters
>
>
>
> -          A survey of traffic models for 17 First Person Shooters can be
> found here. All of them use UDP
> http://www.discover.uottawa.ca/publications/files/MMOG-Survey_IEEE-IC.pdf
>
>
>
> -          In this paper from my research group, 8 FPS games were analyze=
d
> in order to compress and multiplex their traffic. Figure 5 shows their
> histograms (inter-packet time and packet size). It can be seen that they
> generate real-time flows, with characteristics very similar to VoIP (pack=
ets
> of 60-80 bytes every 15-40 ms):
> http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana.pdf
>
>
>
> -          Another paper about FPS using UDP. Section V presents some
> interesting conclusions about routing infrastructure.
> http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf
>
>
>
> -          In this paper the network factors annoying the gamers are
> analyzed. There are also some calculations of the MOS. Above 150 ms the
> quality is bad (something similar to what happens with VoIP)
> http://www.research.ibm.com/netgames2005/papers/dick.pdf
>
>
>
> -          In this one, a MOS model for a FPS (Quake IV) is deployed. It =
is
> only based on delay and jitter, since packet loss has vevy little influen=
ce
> (for this concrete game). Delay above 140 ms produces MOS below 3
> http://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf
>
>
>
> -          In this paper, it is found that some FPS work properly even wi=
th
> 30% packet loss, whereas others stop working with 4%. RTT above 300 ms (1=
50
> OWD) makes MOS go above 3
> http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pdf
>
>
>
> MMORPGs:
>
> -          A traffic model for World of Warcraft:
> http://publik.tuwien.ac.at/files/pub-et_12119.pdf
>
>
>
> -          This one presents a MOS for World of Warcraft. When added dela=
y
> is above 160ms, the MOS is roughly 3.:
> http://publik.tuwien.ac.at/files/PubDat_170202.pdf
>
>
>
> Best regards,
>
>
>
> Jose Saldana
>
>


From jsaldana@unizar.es  Fri Apr 13 01:48:02 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
To: <tsvwg@ietf.org>
Subject: A suggestion for draft-polk-tsvwg-rfc4594-update
Date: Fri, 13 Apr 2012 10:47:58 +0200
Organization: Universidad de Zaragoza
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Although I am mainly interested on online games, in my university there =
are
more serious people working on Telemedicine services. I was wondering if =
it
could be interesting to include in the draft a class for these kind of
services. I have been talking with Jose Garcia, a Professor whose =
research
interests are in Telemedicine, and he has told me that there are two =
kinds
of telemedicine services:

=20

-          Non real-time: e.g. I send a radiography, and the doctor will
have it this afternoon.

-          Real-time: examples:

o    A doctor is in a hospital in a big town. A nurse is in a small =
village
with a patient. There is a service including video, voice and an
electrocardiogram signal established from the village to the hospital, =
that
allows a single doctor to attend a number of villages.

o   Another example: An ambulance carrying an injured man going to the
hospital, and connected to the Internet using the mobile cellular =
network to
send the electrocardiogram, voice, etc. to a doctor who is waiting in =
the
hospital.

=20

So we could think in the working group the possibility of including some
service class for this. If we are trying to reduce latency for VoIP or
online games, why not thinking about this?

=20

This would be similar to what we do in real-life: letting the ambulance
pass. Logically, we would have to be careful: some =93smarty-pants=94 =
person
would try to follow the ambulance and take advantage of it. In our case, =
a
=93smarty-pants=94 gamer could change his traffic in order to have a =
bigger
priority.

=20

What do you think?,

=20

Jose Saldana


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US>Although I am mainly interested on online games, in my =
university there are more serious people working on Telemedicine =
services. I was wondering if it could be interesting to include in the =
draft a class for these kind of services. I have been talking with Jose =
Garcia, a Professor whose research interests are in Telemedicine, and he =
has told me that there are two kinds of telemedicine =
services:<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>Non real-time: e.g. I =
send a radiography, and the doctor will have it this =
afternoon.<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>Real-time: =
examples:<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'margin-left:72.0pt;text-indent:-18.0pt;mso-list:l0 level2 =
lfo1'><![if !supportLists]><span lang=3DEN-US =
style=3D'font-family:"Courier New"'><span =
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp; </span></span></span><![endif]><span =
lang=3DEN-US>=A0A doctor is in a hospital in a big town. A nurse is in a =
small village with a patient. There is a service including video, voice =
and an electrocardiogram signal established from the village to the =
hospital, that allows a single doctor to attend a number of =
villages.<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'margin-left:72.0pt;text-indent:-18.0pt;mso-list:l0 level2 =
lfo1'><![if !supportLists]><span lang=3DEN-US =
style=3D'font-family:"Courier New"'><span =
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp; </span></span></span><![endif]><span =
lang=3DEN-US>Another example: An ambulance carrying an injured man going =
to the hospital, and connected to the Internet using the mobile cellular =
network to send the electrocardiogram, voice, etc. to a doctor who is =
waiting in the hospital.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>So we could think in the working group the possibility of =
including some service class for this. If we are trying to reduce =
latency for VoIP or online games, why not thinking about =
this?<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>This would be similar to what we do in real-life: letting =
the ambulance pass. Logically, we would have to be careful: some =
&#8220;smarty-pants&#8221; person would try to follow the ambulance and =
take advantage of it. In our case, a &#8220;smarty-pants&#8221; gamer =
could change his traffic in order to have a bigger =
priority.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>What do you think?,<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";mso-fareast-la=
nguage:ES'>Jose Saldana</span><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";mso-fareast-la=
nguage:ES'><o:p></o:p></span></p></div></body></html>
------=_NextPart_000_0022_01CD1962.E416C7C0--


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Date: Fri, 13 Apr 2012 07:17:06 -0400
To: jsaldana@unizar.es, <tsvwg@ietf.org>
From: Al Morton <acmorton@att.com>
Subject: Re: A suggestion for draft-polk-tsvwg-rfc4594-update
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At 04:47 AM 4/13/2012, Jose Saldana wrote:
>  ...
>
>This would be similar to what we do in real-life: letting the 
>ambulance pass. Logically, we would have to be careful: some 
>"smarty-pants" person would try to follow the ambulance and take 
>advantage of it. In our case, a "smarty-pants" gamer could change 
>his traffic in order to have a bigger priority.
>
>What do you think?

Hi Jose,

You are discussing two mechanisms in series:

  - classifying telemedicine traffic by itself
  - arranging that (some) telemedicine traffic has priority treatment

I don't think it is in scope to establish relative class priorities.

There *should* be other classes whose characteristics can satisfy
the needs of the telemedicine applications you describe
(there are many classes!), and priority can be established in many ways.
It is a common tendency to identify a new application and decide
it needs special treatment in a class by itself, that's partly how
we arrived were we are, IMO.

regards,
Al




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To: Al Morton <acmorton@att.com>, jsaldana@unizar.es, <tsvwg@ietf.org>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: A suggestion for draft-polk-tsvwg-rfc4594-update
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At 06:17 AM 4/13/2012, Al Morton wrote:
>At 04:47 AM 4/13/2012, Jose Saldana wrote:
>>  ...
>>
>>This would be similar to what we do in real-life: letting the 
>>ambulance pass. Logically, we would have to be careful: some 
>>"smarty-pants" person would try to follow the ambulance and take 
>>advantage of it. In our case, a "smarty-pants" gamer could change 
>>his traffic in order to have a bigger priority.
>>
>>What do you think?
>
>Hi Jose,
>
>You are discussing two mechanisms in series:
>
>  - classifying telemedicine traffic by itself
>  - arranging that (some) telemedicine traffic has priority treatment
>
>I don't think it is in scope to establish relative class priorities.
>
>There *should* be other classes whose characteristics can satisfy
>the needs of the telemedicine applications you describe
>(there are many classes!), and priority can be established in many ways.
>It is a common tendency to identify a new application and decide
>it needs special treatment in a class by itself, that's partly how
>we arrived were we are, IMO.


I agree with Al. We're trying to avoid application specific service 
classes, but rather determine the necessary traffic characteristics 
for applications, then slotting those applications into service 
classes - if, and only if - they are widely used. There are 
exceptions to this rule in both directions. But we want to keep them 
to a minimum.

James


>regards,
>Al
>
>


From jsaldana@unizar.es  Mon Apr 16 05:32:43 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
To: "'James M. Polk'" <jmpolk@cisco.com>, "'Al Morton'" <acmorton@att.com>, <tsvwg@ietf.org>
References: <002101cd1952$208c7120$61a55360$@unizar.es> <201204131116.q3DBFvB6024587@alpd052.aldc.att.com> <201204131905.q3DJ55aN022433@mtv-core-2.cisco.com>
In-Reply-To: <201204131905.q3DJ55aN022433@mtv-core-2.cisco.com>
Subject: RE: A suggestion for draft-polk-tsvwg-rfc4594-update
Date: Mon, 16 Apr 2012 14:32:32 +0200
Organization: Universidad de Zaragoza
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I think I get the point: if we have two VoIP conversations, one from an
ambulance carrying an injured man, and other one from a VoIP client, =
then
the two conversations have the same delay requirements, so they should =
be in
the same service class. The problem of prioritizing the first one does =
not
have to be solved by rfc4594-update. Perhaps the access provider may =
have a
special channel for ambulances, or something like that.

The different types of telemedicine can be found here:
http://en.wikipedia.org/wiki/Telemedicine#Types_of_telemedicine. Perhaps =
it
is correct that none of them can be seen as a new service. Perhaps
telesurgery (http://en.wikipedia.org/wiki/Telemedicine#Telesurgery), but =
I
don't think it is a good idea to use the Internet to do telesurgery (at
least if I am the patient). A dedicated link should be necessary in this
case.

Best regards,

Jose

-----Mensaje original-----
De: James M. Polk [mailto:jmpolk@cisco.com]=20
Enviado el: viernes, 13 de abril de 2012 21:05
Para: Al Morton; jsaldana@unizar.es; tsvwg@ietf.org
CC: Jos=E9 Garc=EDa Moros
Asunto: Re: A suggestion for draft-polk-tsvwg-rfc4594-update

At 06:17 AM 4/13/2012, Al Morton wrote:
>At 04:47 AM 4/13/2012, Jose Saldana wrote:
>>  ...
>>
>>This would be similar to what we do in real-life: letting the=20
>>ambulance pass. Logically, we would have to be careful: some=20
>>"smarty-pants" person would try to follow the ambulance and take=20
>>advantage of it. In our case, a "smarty-pants" gamer could change his=20
>>traffic in order to have a bigger priority.
>>
>>What do you think?
>
>Hi Jose,
>
>You are discussing two mechanisms in series:
>
>  - classifying telemedicine traffic by itself
>  - arranging that (some) telemedicine traffic has priority treatment
>
>I don't think it is in scope to establish relative class priorities.
>
>There *should* be other classes whose characteristics can satisfy the=20
>needs of the telemedicine applications you describe (there are many=20
>classes!), and priority can be established in many ways.
>It is a common tendency to identify a new application and decide it=20
>needs special treatment in a class by itself, that's partly how we=20
>arrived were we are, IMO.


I agree with Al. We're trying to avoid application specific service =
classes,
but rather determine the necessary traffic characteristics for =
applications,
then slotting those applications into service classes - if, and only if =
-
they are widely used. There are exceptions to this rule in both =
directions.
But we want to keep them to a minimum.

James


>regards,
>Al
>
>


From wwwrun@rfc-editor.org  Tue Apr 17 00:25:41 2012
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Subject: [Editorial Errata Reported] RFC5865 (3194)
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The following errata report has been submitted for RFC5865,
"A Differentiated Services Code Point (DSCP) for Capacity-Admitted Traffic".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=5865&eid=3194

--------------------------------------
Type: Editorial
Reported by: Sergey Antipov <sergey.antipov@nsn.com>

Section: 2.2

Original Text
-------------
   There are at least six major ways that capacity admission is done or
   has been proposed to be done for real-time applications.  Each will
   be described below, and Section 3 will judge which ones are likely to
   meet the requirements of the Admitted Telephony service class.  These
   include:


Corrected Text
--------------
   There are at least six major ways that capacity admission is done or
   has been proposed to be done for real-time applications.  Each will
   be described below, and Section 2.3 will judge which ones are likely to
   meet the requirements of the Admitted Telephony service class.  These
   include:


Notes
-----
Section 2.3 is the one, which recommends capacity admission procedures, while Section 3 summarizes proposed changes.

Instructions:
-------------
This errata is currently posted as "Reported". If necessary, please
use "Reply All" to discuss whether it should be verified or
rejected. When a decision is reached, the verifying party (IESG)
can log in to change the status and edit the report, if necessary. 

--------------------------------------
RFC5865 (draft-ietf-tsvwg-admitted-realtime-dscp-07)
--------------------------------------
Title               : A Differentiated Services Code Point (DSCP) for Capacity-Admitted Traffic
Publication Date    : May 2010
Author(s)           : F. Baker, J. Polk, M. Dolly
Category            : PROPOSED STANDARD
Source              : Transport Area Working Group
Area                : Transport
Stream              : IETF
Verifying Party     : IESG

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Subject: Re: [Editorial Errata Reported] RFC5865 (3194)
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From: Fred Baker <fred@cisco.com>
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Date: Tue, 17 Apr 2012 00:44:48 -0700
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I am not opposed to the erratum.

On Apr 17, 2012, at 12:24 AM, RFC Errata System wrote:

>=20
> The following errata report has been submitted for RFC5865,
> "A Differentiated Services Code Point (DSCP) for Capacity-Admitted =
Traffic".
>=20
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=3D5865&eid=3D3194
>=20
> --------------------------------------
> Type: Editorial
> Reported by: Sergey Antipov <sergey.antipov@nsn.com>
>=20
> Section: 2.2
>=20
> Original Text
> -------------
>   There are at least six major ways that capacity admission is done or
>=20
>   has been proposed to be done for real-time applications.  Each will
>=20
>   be described below, and Section 3 will judge which ones are likely =
to
>=20
>   meet the requirements of the Admitted Telephony service class.  =
These
>=20
>   include:
>=20
>=20
>=20
> Corrected Text
> --------------
>   There are at least six major ways that capacity admission is done or
>=20
>   has been proposed to be done for real-time applications.  Each will
>=20
>   be described below, and Section 2.3 will judge which ones are likely =
to
>=20
>   meet the requirements of the Admitted Telephony service class.  =
These
>=20
>   include:
>=20
>=20
>=20
> Notes
> -----
> Section 2.3 is the one, which recommends capacity admission =
procedures, while Section 3 summarizes proposed changes.
>=20
> Instructions:
> -------------
> This errata is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary.=20
>=20
> --------------------------------------
> RFC5865 (draft-ietf-tsvwg-admitted-realtime-dscp-07)
> --------------------------------------
> Title               : A Differentiated Services Code Point (DSCP) for =
Capacity-Admitted Traffic
> Publication Date    : May 2010
> Author(s)           : F. Baker, J. Polk, M. Dolly
> Category            : PROPOSED STANDARD
> Source              : Transport Area Working Group
> Area                : Transport
> Stream              : IETF
> Verifying Party     : IESG


From jmpolk@cisco.com  Tue Apr 17 11:35:14 2012
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To: Fred Baker <fred@cisco.com>, RFC Errata System <rfc-editor@rfc-editor.org>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Editorial Errata Reported] RFC5865 (3194)
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Nor am I.

At 02:44 AM 4/17/2012, Fred Baker wrote:
>I am not opposed to the erratum.
>
>On Apr 17, 2012, at 12:24 AM, RFC Errata System wrote:
>
> >
> > The following errata report has been submitted for RFC5865,
> > "A Differentiated Services Code Point (DSCP) for 
> Capacity-Admitted Traffic".
> >
> > --------------------------------------
> > You may review the report below and at:
> > http://www.rfc-editor.org/errata_search.php?rfc=5865&eid=3194
> >
> > --------------------------------------
> > Type: Editorial
> > Reported by: Sergey Antipov <sergey.antipov@nsn.com>
> >
> > Section: 2.2
> >
> > Original Text
> > -------------
> >   There are at least six major ways that capacity admission is done or
> >
> >   has been proposed to be done for real-time applications.  Each will
> >
> >   be described below, and Section 3 will judge which ones are likely to
> >
> >   meet the requirements of the Admitted Telephony service class.  These
> >
> >   include:
> >
> >
> >
> > Corrected Text
> > --------------
> >   There are at least six major ways that capacity admission is done or
> >
> >   has been proposed to be done for real-time applications.  Each will
> >
> >   be described below, and Section 2.3 will judge which ones are likely to
> >
> >   meet the requirements of the Admitted Telephony service class.  These
> >
> >   include:
> >
> >
> >
> > Notes
> > -----
> > Section 2.3 is the one, which recommends capacity admission 
> procedures, while Section 3 summarizes proposed changes.
> >
> > Instructions:
> > -------------
> > This errata is currently posted as "Reported". If necessary, please
> > use "Reply All" to discuss whether it should be verified or
> > rejected. When a decision is reached, the verifying party (IESG)
> > can log in to change the status and edit the report, if necessary.
> >
> > --------------------------------------
> > RFC5865 (draft-ietf-tsvwg-admitted-realtime-dscp-07)
> > --------------------------------------
> > Title               : A Differentiated Services Code Point (DSCP) 
> for Capacity-Admitted Traffic
> > Publication Date    : May 2010
> > Author(s)           : F. Baker, J. Polk, M. Dolly
> > Category            : PROPOSED STANDARD
> > Source              : Transport Area Working Group
> > Area                : Transport
> > Stream              : IETF
> > Verifying Party     : IESG


From Karen.Nielsen@tieto.com  Wed Apr 18 00:25:47 2012
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From: <Karen.Nielsen@tieto.com>
To: <mattmathis@google.com>
Date: Wed, 18 Apr 2012 10:25:26 +0300
Subject: RE: [tcpm] CWND increase after under-utilization in congestion avoidance
Thread-Topic: [tcpm] CWND increase after under-utilization in congestion avoidance
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Hi,

Thanks a lot!=20

My main concern is the slope of the increase of the CWND after a period of =
CWND underutilization in congestion avoidance phase.

My general understanding of the standards is that during slow start one may=
 raise the CWND per received (S)ACK whereas during congestion avoidance gen=
erally CWND may be increase by one MTU only per RTT. Then some of the more =
experimental CC algorithms allow for slow start slope increase after a pack=
et loss, but that may be a sidetrack here. At least my question really goes=
 to the base standard of SCTP as specified in RFC4960. But I assume that th=
e question is equally relevant for a TCP implementation that implements CWN=
D validation in some form.

The assumption here is that we are in congestion avoidance phase, but also =
that the accumulated partical_bytes_ack can allow for several increases of =
the CWND within the same RTT as SACKs will be returned. Provided that the a=
pplication continues to fill the increasing CWND, the question is whether S=
CTP (TCP) then is allowed to proactively increase the CWND several times pe=
r RTT (slow start slope) or whether the increase should be limited to be do=
ne one per RTT (congestion avoidance slope).

Thanks!

BR, Karen

-----Original Message-----
From: Matt Mathis [mailto:mattmathis@google.com]=20
Sent: 17. april 2012 22:37
To: Nielsen Karen
Cc: tcpm@ietf.org
Subject: Re: [tcpm] CWND increase after under-utilization in congestion avo=
idance

Well, my opinion would be that as long as  flightsize >=3D cwnd at least
once recently it should be ok to continue to raise cwnd.  Note that
raising cwnd probably causes the predicate to become not true.  Also
"recently" should be no shorter than one RTT, but may be a longer (the
details are not so important).   The point is that as long as FS
reaches cwnd at least once per RTT, it is fine to continue to grow the
window.  I believe I have seen it implemented "as long as FS comes
close to cwnd..."

In the big picture is it important not to keep raising cwnd if cwnd in
not controlling the connection, because then you have no way to know
if it is a valid measure of anything useful.  However, as long as cwnd
has been "tested" it is fine to keep pushing it up.

Thanks,
--MM--
The best way to predict the future is to create it. =A0- Alan Kay



On Tue, Apr 17, 2012 at 5:50 AM,  <Karen.Nielsen@tieto.com> wrote:
> Hi,
>
>
>
> SCTP as prescribed by RFC4960 operates with CWND validation in that CWND
> increase always
>
> is subject to its fully usage. That is, CWND only may increase when
> flightsize >=3D CWND.
>
> Opposed to TCP, as standardized by RFC5681, CWND is thus not necessarily
> increased even when partial_bytes_ack
>
> increases beyond the present CWND.
>
>
>
> An implementation which boldly augments the partial_bytes_ack during the
> time where a CWND is underutilized (but the connection is not idle) may
> observe that the partial_bytes_ack increases to a very high value, possib=
ly
> it increases to
>
> a value many times higher than the CWND.
>
>
>
> The question now arises what should happen once the CWND resumes fully
> utilization and thus may be increased. That is, is the intention of the
> standard of SCTP CC to have :
>
> 1.=A0=A0=A0=A0=A0 the CWND now be increased as during "normal" congestion=
 control
> avoidance, i.e., one MTU per acked CWND amount=A0 of data
>
> -=A0=A0=A0=A0=A0=A0=A0 Or to have -
>
> 2.=A0=A0=A0=A0=A0 the CWND now be allowed to recapture the growth that it=
 did not
> realize due to under-utilization but which may be recaptured from the
> partial_bytes_ack value. I.e., let it be increased one MTU per SACK until
> partial_bytes_ack has been decreased below CWND, one CWND at a time.
>
>
>
> From an implementation perspective the question likely turns into a quest=
ion
> about the maintaining of the partial_bytes_ack value during the period of
> time when the CWND is underutilized. E.g., should then partial_bytes_ack =
be
> decreased with CWND for each times it increases beyond the CWND or should=
 it
> be allowed to increase unlimited. =A0But first we need to find out what t=
he
> intention of the standard is..
>
>
>
> The above of course is subject to cumulative SACKs and other details like
> that. I hope that we can leave such details out of the discussion.
>
>
>
> I hope that you may help.
>
>
>
> Thanks in advance!
>
>
>
> BR, Karen
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From Karen.Nielsen@tieto.com  Wed Apr 18 00:53:07 2012
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From: <Karen.Nielsen@tieto.com>
To: <tuexen@fh-muenster.de>, <randall@lakerest.net>
Date: Wed, 18 Apr 2012 10:52:16 +0300
Subject: FW: [tcpm] CWND increase after under-utilization in congestion avoidance
Thread-Topic: [tcpm] CWND increase after under-utilization in congestion avoidance
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Hi,

I am sorry to bother you again.
My emails to tsvwg generally bounces back...

I do not know if you are interested in the following topic.

We have checked the BSD code and it seems that BSD SCTP allows for=20
slow start type increase of the CWND if the accumulated partial_bytes_ack a=
llows=20
after a situation with CWND underutilization in congestion avoidance phase.

Do you know if that is a conscious choice and/or if that is how you
see the RFC4960 precription ? (which then is perfectly mute on what to do w=
ith
the partial_bytes-ack value in this situation).

Actually I/we think that it should not be like this, namely that the partia=
l_bytes_ack should not be allowed to increase much above CWND - so that it =
will allow for only one +MTU increase of the CWND per RTT in congestion avo=
idance phase.

It is then an implementation decision as to how exactly to handle the parti=
al_bytes_ack in the under-utilization phase and which value to start with o=
nce the condition for CWND (flightsize >=3D CWND) resumes to be met.

BR, Karen

-----Original Message-----
From: Nielsen Karen=20
Sent: 18. april 2012 09:25
To: 'Matt Mathis'
Cc: tcpm@ietf.org; tsvwg@ietf.org
Subject: RE: [tcpm] CWND increase after under-utilization in congestion avo=
idance

Hi,

Thanks a lot!=20

My main concern is the slope of the increase of the CWND after a period of =
CWND underutilization in congestion avoidance phase.

My general understanding of the standards is that during slow start one may=
 raise the CWND per received (S)ACK whereas during congestion avoidance gen=
erally CWND may be increase by one MTU only per RTT. Then some of the more =
experimental CC algorithms allow for slow start slope increase after a pack=
et loss, but that may be a sidetrack here. At least my question really goes=
 to the base standard of SCTP as specified in RFC4960. But I assume that th=
e question is equally relevant for a TCP implementation that implements CWN=
D validation in some form.

The assumption here is that we are in congestion avoidance phase, but also =
that the accumulated partical_bytes_ack can allow for several increases of =
the CWND within the same RTT as SACKs will be returned. Provided that the a=
pplication continues to fill the increasing CWND, the question is whether S=
CTP (TCP) then is allowed to proactively increase the CWND several times pe=
r RTT (slow start slope) or whether the increase should be limited to be do=
ne one per RTT (congestion avoidance slope).

Thanks!

BR, Karen

-----Original Message-----
From: Matt Mathis [mailto:mattmathis@google.com]=20
Sent: 17. april 2012 22:37
To: Nielsen Karen
Cc: tcpm@ietf.org
Subject: Re: [tcpm] CWND increase after under-utilization in congestion avo=
idance

Well, my opinion would be that as long as  flightsize >=3D cwnd at least
once recently it should be ok to continue to raise cwnd.  Note that
raising cwnd probably causes the predicate to become not true.  Also
"recently" should be no shorter than one RTT, but may be a longer (the
details are not so important).   The point is that as long as FS
reaches cwnd at least once per RTT, it is fine to continue to grow the
window.  I believe I have seen it implemented "as long as FS comes
close to cwnd..."

In the big picture is it important not to keep raising cwnd if cwnd in
not controlling the connection, because then you have no way to know
if it is a valid measure of anything useful.  However, as long as cwnd
has been "tested" it is fine to keep pushing it up.

Thanks,
--MM--
The best way to predict the future is to create it. =A0- Alan Kay



On Tue, Apr 17, 2012 at 5:50 AM,  <Karen.Nielsen@tieto.com> wrote:
> Hi,
>
>
>
> SCTP as prescribed by RFC4960 operates with CWND validation in that CWND
> increase always
>
> is subject to its fully usage. That is, CWND only may increase when
> flightsize >=3D CWND.
>
> Opposed to TCP, as standardized by RFC5681, CWND is thus not necessarily
> increased even when partial_bytes_ack
>
> increases beyond the present CWND.
>
>
>
> An implementation which boldly augments the partial_bytes_ack during the
> time where a CWND is underutilized (but the connection is not idle) may
> observe that the partial_bytes_ack increases to a very high value, possib=
ly
> it increases to
>
> a value many times higher than the CWND.
>
>
>
> The question now arises what should happen once the CWND resumes fully
> utilization and thus may be increased. That is, is the intention of the
> standard of SCTP CC to have :
>
> 1.=A0=A0=A0=A0=A0 the CWND now be increased as during "normal" congestion=
 control
> avoidance, i.e., one MTU per acked CWND amount=A0 of data
>
> -=A0=A0=A0=A0=A0=A0=A0 Or to have -
>
> 2.=A0=A0=A0=A0=A0 the CWND now be allowed to recapture the growth that it=
 did not
> realize due to under-utilization but which may be recaptured from the
> partial_bytes_ack value. I.e., let it be increased one MTU per SACK until
> partial_bytes_ack has been decreased below CWND, one CWND at a time.
>
>
>
> From an implementation perspective the question likely turns into a quest=
ion
> about the maintaining of the partial_bytes_ack value during the period of
> time when the CWND is underutilized. E.g., should then partial_bytes_ack =
be
> decreased with CWND for each times it increases beyond the CWND or should=
 it
> be allowed to increase unlimited. =A0But first we need to find out what t=
he
> intention of the standard is..
>
>
>
> The above of course is subject to cumulative SACKs and other details like
> that. I hope that we can leave such details out of the discussion.
>
>
>
> I hope that you may help.
>
>
>
> Thanks in advance!
>
>
>
> BR, Karen
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From bob.briscoe@bt.com  Wed Apr 18 11:13:53 2012
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From: Bob Briscoe <bob.briscoe@bt.com>
Subject: Re: Note: WGLC Announcement for draft-ietf-tsvwg-byte-pkt-congest-07
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John,

inline...

At 16:50 30/03/2012, John Leslie wrote:
>Gorry Fairhurst <gorry@erg.abdn.ac.uk> wrote:
> >
> > Please respond to the WGLC that ends on Friday 30th March 2012.
> >
> > Notes to the list would be helpful to complete the WGCC.
>
>    I cannot support this draft as written.
>
>    Perhaps some background would help...
>
>    It will be no secret to anyone following the ConEx WG that Bob and I
>fundamentally disagree about whether to model congestion as per-packet
>or per-byte. I strongly believe that it's packets that are congested, and
>that we'll do better to accurately count congested packets than estimate
>congested bytes; Bob seems to strongly believe that we'll do better to
>estimate congested bytes than accurately count congested packets.

This doesn't need to be personal. There are two co-authors, and this 
is also the output of everyone else who has contributed (see acks). 
This is not just me, this is a WG document.

If you believe differently, we need to hear a technical argument. 
There are 5 technical motivating arguments in this doc (S.3) that 
back up the recommendations. One specifically relates to the 
recommendations you don't like about the transport layer (S3.4). Then 
there is a more equivocal discussion in S.5 about whether 
bit-congestible is more prevalent than pkt-congestible, and whether 
this might change in future.

I have been encouraging you to have this argument out, rather than 
just saying we shouldn't decide because 'some people' disagree. Let's hear it.

I believe the disagreement is rooted in the idea that vendors hit the 
design limits of equipment performance when they cannot process 
packets any faster, whereas raw line transmission rates could be a 
lot faster. The point referred to by this draft [RFC6077] is that 
vendors then limit the bit-rate of the ports to the packet-processing 
rates they can achieve with a workload that includes reasonable runs 
of small packets (what point would there be in faster line rates?). 
Therefore, at *run-time*, congestion signals from the queues in such 
equipment generally represent bit-congestion, not packet-congestion, 
even though the kit was mostly constrained by packet processing 
during the *design* phase.

>    When I previously skimmed this document, I believed it contained only
>common ground: that AQM should drop/mark packets without favoring small
>packets, and that to whatever extent packet size _is_ considered, that
>should be a transport-layer responsibility.

Good - this is the main thing we wanted to ensure was said.


>    Alas, when I read it carefully this week, I find that Bob is actually
>saying that transport-layer should _only_ consider "congested bytes", not
>"congested packets".

Assuming we are talking about the general recommendation in 2.3 here, 
it endorses protocols like TFRC-SP [RFC4828] that weights losses 
proportionate to packet size within a flow with variable size 
packets. The recommendation in 2.3 is largely unchanged since Sally 
Floyd (one of the authors of RFC4828) reviewed byte-pkt.

In fact, TFRC-SP wrestled with the possibility that the network, not 
the transport, could weight its congestion signals relative to packet 
size. But it didn't even entertain the possibility that congestion 
might be packet-based not bit-based, so packet size wouldn't be an 
issue at all.

byte-pkt-congest does at least entertain the possibility that 
prevalent bit-congestion may be a wrong assumption and describes a 
future where it might be wrong, but it also justifies fairly 
carefully why it is a safe working assumption and it will continue to 
be safe, based on advice from equipment designers. (Actually, in 
reading the latest draft, I notice some of this argument has 
disappeared in the editing and it now mainly justifies this position 
by reference to RFC6077. I would be happy to bring some of that 
removed text back in.)

But, I gather that wouldn't satisfy you, because you believe 
packet-congestion is prevalent, so let's please have that technical discussion.

>    In fact, Bob openly endorses replacing TCP congestion-control with an
>algorithm which calculates "fair-share bytes" and doesn't back off at all
>(even in the presence of 20% packet loss) unless you're already sending
>more than this "fair-share".

I assume you're referring to bullet 1. under "What this advice means 
for the case of TCP:" in S2.3.

If so, that's not how it should read at all.

It says
"  1.  If two TCP flows with different packet sizes are required to run
        at equal bit rates under the same path conditions, this should be
        done by altering TCP (Section 4.2.2), not network equipment [...]
"

This doesn't say TCP should do this, it just says TCP is the place to 
fiddle, *if* you want to equalise bit-rates.

If that's not clear, I would be happy (and I hope Jukka would too) to 
make it clearer that this recommendation is irrelevant if TCP flows 
are *not* required to run at equal bit-rates (Personally I would like 
to say that there should be no need for any equal-bit-rate 
requirement - however, that *would* be going beyond our charter).

How about this text after S2.3. bullet 1?

"  For the avoidance of doubt, this is not a recommendation that TCP
    should be changed so that all TCP flows run at equal bit-rates.  It is a
    recommendation that the transport protocol not the network is where any
    attempt should be made to equalise bit-rates of competing flows, *if*
    that is ever considered an important requirement.
"


>    I do not believe that is the consensus of this WG; and I believe if
>that were the consensus it would exceed our charter.
>
>    The draft contains a number of things I _do_ support; and I'd be happy
>to support a considerably shorter draft which concentrates on the AQM
>recommendation, omitting any suggestions of modifying TCP congestion
>control.

We could omit "What this means for TCP" altogether. However, given 
people of the standing of Sally Floyd originally tried to equalise 
bit-rates in the network, then moved to doing it in the transport, I 
don't think it is wise to leave a vacuum. It is our duty to record 
the advice from this collective experience. But if the wording isn't 
nuanced correctly, let's change it.


Bob


>--
>John Leslie <john@jlc.net>

________________________________________________________________
Bob Briscoe,                                BT Innovate & Design 


From john@jlc.net  Thu Apr 19 06:38:04 2012
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From: John Leslie <john@jlc.net>
To: Bob Briscoe <bob.briscoe@bt.com>
Subject: Re: Note: WGLC Announcement for draft-ietf-tsvwg-byte-pkt-congest-07
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Bob Briscoe <bob.briscoe@bt.com> wrote:
> 
> I believe the disagreement is rooted in the idea that vendors hit the 
> design limits of equipment performance when they cannot process 
> packets any faster, whereas raw line transmission rates could be a 
> lot faster.

   Actually, that's not it at all.

>> When I previously skimmed this document, I believed it contained only
>> common ground: that AQM should drop/mark packets without favoring small
>> packets, and that to whatever extent packet size _is_ considered, that
>> should be a transport-layer responsibility.
> 
> Good - this is the main thing we wanted to ensure was said.
> 
>> Alas, when I read it carefully this week, I find that Bob is actually
>> saying that transport-layer should _only_ consider "congested bytes",
>> not "congested packets".
> 
> In fact, TFRC-SP wrestled with the possibility that the network, not 
> the transport, could weight its congestion signals relative to packet 
> size. But it didn't even entertain the possibility that congestion 
> might be packet-based not bit-based, so packet size wouldn't be an 
> issue at all.

   Hopefully, we all can agree the network should not weigh congestion
signals by packet size. But that doesn't mean some other layer needs
to do so in any "ordinary" case.

> byte-pkt-congest does at least entertain the possibility that 
> prevalent bit-congestion may be a wrong assumption and describes a 
> future where it might be wrong, but it also justifies fairly 
> carefully why it is a safe working assumption and it will continue to 
> be safe, based on advice from equipment designers. (Actually, in 
> reading the latest draft, I notice some of this argument has 
> disappeared in the editing and it now mainly justifies this position 
> by reference to RFC6077. I would be happy to bring some of that 
> removed text back in.)

   I very much doubt that what we need here is to add such text.

> But, I gather that wouldn't satisfy you, because you believe 
> packet-congestion is prevalent, so let's please have that technical 
> discussion.

   Once again, that's not it at all.

   My technical argument is it's really-hard to accurately estimate
per-packet overhead when counting bits for bit-congestible node pairs;
and that counting packets (which we can do accurately) works well in
all cases where packet size within a flow is fairly constant.

   This is basically what we do today; and I'm very resistant to any
call to replace it with algorithms which tend to underestimate the
overhead when counting payload bytes.

>> In fact, Bob openly endorses replacing TCP congestion-control with an
>> algorithm which calculates "fair-share bytes" and doesn't back off at all
>> (even in the presence of 20% packet loss) unless you're already sending
>> more than this "fair-share".

   BTW, I was very surprised to find "fair-share" as a recommendation
in a Bob Briscoe document, given his previous "dismantling a religion"
documents.

> I assume you're referring to bullet 1. under "What this advice means 
> for the case of TCP:" in S2.3.

   Actually, that's not what I was referring to.

> It says
> "  1.  If two TCP flows with different packet sizes are required to run
>        at equal bit rates under the same path conditions, this should be
>        done by altering TCP (Section 4.2.2), not network equipment [...]
> "

   More likely, it was Section 4.2.2... I don't think this document
needs a 4.2.2 at all. This statement would be stronger if we didn't try
to describe possible modifications to TCP.

> This doesn't say TCP should do this, it just says TCP is the place to 
> fiddle, *if* you want to equalise bit-rates.

   And I didn't find it objectionble until I read the rest of the
document carefully.

> If that's not clear, I would be happy (and I hope Jukka would too) to 
> make it clearer that this recommendation is irrelevant if TCP flows 
> are *not* required to run at equal bit-rates (Personally I would like 
> to say that there should be no need for any equal-bit-rate 
> requirement - however, that *would* be going beyond our charter).

   Maybe we should explore exactly what about our charter excludes such
a statement?

> How about this text after S2.3. bullet 1?
> 
> "  For the avoidance of doubt, this is not a recommendation that TCP
>    should be changed so that all TCP flows run at equal bit-rates.  It is a
>    recommendation that the transport protocol not the network is where any
>    attempt should be made to equalise bit-rates of competing flows, *if*
>    that is ever considered an important requirement.
> "

   I preferred your short-form:
" 
" If you want to equalize bit-rates between TCP flows, this should be
" done within TCP, not by network equipment.

>> The draft contains a number of things I _do_ support; and I'd be happy
>> to support a considerably shorter draft which concentrates on the AQM
>> recommendation, omitting any suggestions of modifying TCP congestion
>> control.
> 
> We could omit "What this means for TCP" altogether.

   IMHO, the document would be stronger without it.

> However, given people of the standing of Sally Floyd originally tried
> to equalise bit-rates in the network, then moved to doing it in the
> transport, I don't think it is wise to leave a vacuum.

   I don't perceive any such vacuum. At most, we might need an informative
reference to a Sally Floyd document.

--
John Leslie <john@jlc.net>

From Karen.Nielsen@tieto.com  Fri Apr 20 00:52:41 2012
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From: <Karen.Nielsen@tieto.com>
To: <tsvwg@ietf.org>, <randall@lakerest.net>, <tuexen@fh-muenster.de>
Date: Fri, 20 Apr 2012 10:52:35 +0300
Subject: SCTP CWND increase after under-utilization in congestion avoidance
Thread-Topic: SCTP CWND increase after under-utilization in congestion avoidance
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Hi,

RFC4960 specifies for the partial_bytes_ack (pba) to be uncritically increa=
sed upon receipt of SACKs. Uncritically here means also if the CWND is unde=
r-utilized, that is, also if flightsize < CWND.
Further then once traffic increases (or resumes) and the CWND becomes fully=
 utilized and SACKs are received,
then the prescription is for the CWND to be increases one MTU provided the =
pba value >=3D CWND (and pba then set =3D pba - CWND)

The pba parameter first was introduced in TCP ABC RFC3465 and the parameter=
 is used in the implementation to mimic congestion avoidance CWND increase =
slope of "+MTU*MTU/CWND per SACK" as specified by RFC5681.

Following the prescriptions of RFC4960 uncritically one may have the pba in=
crease to multiple times the CWND during a
period of CWND underutilization and further, once traffic and CWND utilizat=
ion resumes, one may have a situation where the CWND in principle may be in=
creases by one MTU per received SACK (more or less) a number of times corre=
sponding to the value INT(pba/CWND) until the pba parameter iteratively has=
 been decreased below present CWND. This effectively results in slow start =
slope increase of the CWND during congestion avoidance.

Randy and Michael, as authors of the RFc4960, could I ask you to confirm th=
at while the explicit handling of the pba in this particular case may be an=
 implementation decision THEN THE INTENTION of SCTP CC as specified in RFC =
4960 is for the CWND to be increased following congestion avoidance slope i=
n this situation (traffic resume after CWND underutilization in congestion =
avoidance phase).

Many Thanks in advance !!

BR, Karen


-----Original Message-----
From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of K=
aren.Nielsen@tieto.com
Sent: 18. april 2012 09:25
To: mattmathis@google.com
Cc: tcpm@ietf.org; tsvwg@ietf.org
Subject: RE: [tcpm] CWND increase after under-utilization in congestion avo=
idance

Hi,

Thanks a lot!=20

My main concern is the slope of the increase of the CWND after a period of =
CWND underutilization in congestion avoidance phase.

My general understanding of the standards is that during slow start one may=
 raise the CWND per received (S)ACK whereas during congestion avoidance gen=
erally CWND may be increase by one MTU only per RTT. Then some of the more =
experimental CC algorithms allow for slow start slope increase after a pack=
et loss, but that may be a sidetrack here. At least my question really goes=
 to the base standard of SCTP as specified in RFC4960. But I assume that th=
e question is equally relevant for a TCP implementation that implements CWN=
D validation in some form.

The assumption here is that we are in congestion avoidance phase, but also =
that the accumulated partical_bytes_ack can allow for several increases of =
the CWND within the same RTT as SACKs will be returned. Provided that the a=
pplication continues to fill the increasing CWND, the question is whether S=
CTP (TCP) then is allowed to proactively increase the CWND several times pe=
r RTT (slow start slope) or whether the increase should be limited to be do=
ne one per RTT (congestion avoidance slope).

Thanks!

BR, Karen

-----Original Message-----
From: Matt Mathis [mailto:mattmathis@google.com]=20
Sent: 17. april 2012 22:37
To: Nielsen Karen
Cc: tcpm@ietf.org
Subject: Re: [tcpm] CWND increase after under-utilization in congestion avo=
idance

Well, my opinion would be that as long as  flightsize >=3D cwnd at least
once recently it should be ok to continue to raise cwnd.  Note that
raising cwnd probably causes the predicate to become not true.  Also
"recently" should be no shorter than one RTT, but may be a longer (the
details are not so important).   The point is that as long as FS
reaches cwnd at least once per RTT, it is fine to continue to grow the
window.  I believe I have seen it implemented "as long as FS comes
close to cwnd..."

In the big picture is it important not to keep raising cwnd if cwnd in
not controlling the connection, because then you have no way to know
if it is a valid measure of anything useful.  However, as long as cwnd
has been "tested" it is fine to keep pushing it up.

Thanks,
--MM--
The best way to predict the future is to create it. =A0- Alan Kay



On Tue, Apr 17, 2012 at 5:50 AM,  <Karen.Nielsen@tieto.com> wrote:
> Hi,
>
>
>
> SCTP as prescribed by RFC4960 operates with CWND validation in that CWND
> increase always
>
> is subject to its fully usage. That is, CWND only may increase when
> flightsize >=3D CWND.
>
> Opposed to TCP, as standardized by RFC5681, CWND is thus not necessarily
> increased even when partial_bytes_ack
>
> increases beyond the present CWND.
>
>
>
> An implementation which boldly augments the partial_bytes_ack during the
> time where a CWND is underutilized (but the connection is not idle) may
> observe that the partial_bytes_ack increases to a very high value, possib=
ly
> it increases to
>
> a value many times higher than the CWND.
>
>
>
> The question now arises what should happen once the CWND resumes fully
> utilization and thus may be increased. That is, is the intention of the
> standard of SCTP CC to have :
>
> 1.=A0=A0=A0=A0=A0 the CWND now be increased as during "normal" congestion=
 control
> avoidance, i.e., one MTU per acked CWND amount=A0 of data
>
> -=A0=A0=A0=A0=A0=A0=A0 Or to have -
>
> 2.=A0=A0=A0=A0=A0 the CWND now be allowed to recapture the growth that it=
 did not
> realize due to under-utilization but which may be recaptured from the
> partial_bytes_ack value. I.e., let it be increased one MTU per SACK until
> partial_bytes_ack has been decreased below CWND, one CWND at a time.
>
>
>
> From an implementation perspective the question likely turns into a quest=
ion
> about the maintaining of the partial_bytes_ack value during the period of
> time when the CWND is underutilized. E.g., should then partial_bytes_ack =
be
> decreased with CWND for each times it increases beyond the CWND or should=
 it
> be allowed to increase unlimited. =A0But first we need to find out what t=
he
> intention of the standard is..
>
>
>
> The above of course is subject to cumulative SACKs and other details like
> that. I hope that we can leave such details out of the discussion.
>
>
>
> I hope that you may help.
>
>
>
> Thanks in advance!
>
>
>
> BR, Karen
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www.ietf.org/mailman/listinfo/tcpm
>

From tuexen@fh-muenster.de  Sat Apr 21 00:38:15 2012
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Subject: Re: SCTP CWND increase after under-utilization in congestion avoidance
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To: <Karen.Nielsen@tieto.com> <Karen.Nielsen@tieto.com>
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On Apr 20, 2012, at 9:52 AM, <Karen.Nielsen@tieto.com> =
<Karen.Nielsen@tieto.com> wrote:

> Hi,
>=20
> RFC4960 specifies for the partial_bytes_ack (pba) to be uncritically =
increased upon receipt of SACKs. Uncritically here means also if the =
CWND is under-utilized, that is, also if flightsize < CWND.
> Further then once traffic increases (or resumes) and the CWND becomes =
fully utilized and SACKs are received,
> then the prescription is for the CWND to be increases one MTU provided =
the pba value >=3D CWND (and pba then set =3D pba - CWND)
>=20
> The pba parameter first was introduced in TCP ABC RFC3465 and the =
parameter is used in the implementation to mimic congestion avoidance =
CWND increase slope of "+MTU*MTU/CWND per SACK" as specified by RFC5681.
>=20
> Following the prescriptions of RFC4960 uncritically one may have the =
pba increase to multiple times the CWND during a
> period of CWND underutilization and further, once traffic and CWND =
utilization resumes, one may have a situation where the CWND in =
principle may be increases by one MTU per received SACK (more or less) a =
number of times corresponding to the value INT(pba/CWND) until the pba =
parameter iteratively has been decreased below present CWND. This =
effectively results in slow start slope increase of the CWND during =
congestion avoidance.
>=20
> Randy and Michael, as authors of the RFc4960, could I ask you to =
confirm that while the explicit handling of the pba in this particular =
case may be an implementation decision THEN THE INTENTION of SCTP CC as =
specified in RFC 4960 is for the CWND to be increased following =
congestion avoidance slope in this situation (traffic resume after CWND =
underutilization in congestion avoidance phase).
Hi Karen,

the text in RFC 4960 is already in RFC 2960 and it was not intended to =
do anything different
than TCP CC. The first sentence of section 7.2.2 states the goal:
  When cwnd is greater than ssthresh, cwnd should be incremented by
  1*MTU per RTT if the sender has cwnd or more bytes of data
  outstanding for the corresponding transport address.
So I guess the algorithm described after that paragraph doesn't realize =
this goal
in the scenario you mention.

I FreeBSD we try to follow the RFC.

Best regards
Michael

>=20
> Many Thanks in advance !!
>=20
> BR, Karen
>=20
>=20
> -----Original Message-----
> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf =
Of Karen.Nielsen@tieto.com
> Sent: 18. april 2012 09:25
> To: mattmathis@google.com
> Cc: tcpm@ietf.org; tsvwg@ietf.org
> Subject: RE: [tcpm] CWND increase after under-utilization in =
congestion avoidance
>=20
> Hi,
>=20
> Thanks a lot!=20
>=20
> My main concern is the slope of the increase of the CWND after a =
period of CWND underutilization in congestion avoidance phase.
>=20
> My general understanding of the standards is that during slow start =
one may raise the CWND per received (S)ACK whereas during congestion =
avoidance generally CWND may be increase by one MTU only per RTT. Then =
some of the more experimental CC algorithms allow for slow start slope =
increase after a packet loss, but that may be a sidetrack here. At least =
my question really goes to the base standard of SCTP as specified in =
RFC4960. But I assume that the question is equally relevant for a TCP =
implementation that implements CWND validation in some form.
>=20
> The assumption here is that we are in congestion avoidance phase, but =
also that the accumulated partical_bytes_ack can allow for several =
increases of the CWND within the same RTT as SACKs will be returned. =
Provided that the application continues to fill the increasing CWND, the =
question is whether SCTP (TCP) then is allowed to proactively increase =
the CWND several times per RTT (slow start slope) or whether the =
increase should be limited to be done one per RTT (congestion avoidance =
slope).
>=20
> Thanks!
>=20
> BR, Karen
>=20
> -----Original Message-----
> From: Matt Mathis [mailto:mattmathis@google.com]=20
> Sent: 17. april 2012 22:37
> To: Nielsen Karen
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] CWND increase after under-utilization in =
congestion avoidance
>=20
> Well, my opinion would be that as long as  flightsize >=3D cwnd at =
least
> once recently it should be ok to continue to raise cwnd.  Note that
> raising cwnd probably causes the predicate to become not true.  Also
> "recently" should be no shorter than one RTT, but may be a longer (the
> details are not so important).   The point is that as long as FS
> reaches cwnd at least once per RTT, it is fine to continue to grow the
> window.  I believe I have seen it implemented "as long as FS comes
> close to cwnd..."
>=20
> In the big picture is it important not to keep raising cwnd if cwnd in
> not controlling the connection, because then you have no way to know
> if it is a valid measure of anything useful.  However, as long as cwnd
> has been "tested" it is fine to keep pushing it up.
>=20
> Thanks,
> --MM--
> The best way to predict the future is to create it.  - Alan Kay
>=20
>=20
>=20
> On Tue, Apr 17, 2012 at 5:50 AM,  <Karen.Nielsen@tieto.com> wrote:
>> Hi,
>>=20
>>=20
>>=20
>> SCTP as prescribed by RFC4960 operates with CWND validation in that =
CWND
>> increase always
>>=20
>> is subject to its fully usage. That is, CWND only may increase when
>> flightsize >=3D CWND.
>>=20
>> Opposed to TCP, as standardized by RFC5681, CWND is thus not =
necessarily
>> increased even when partial_bytes_ack
>>=20
>> increases beyond the present CWND.
>>=20
>>=20
>>=20
>> An implementation which boldly augments the partial_bytes_ack during =
the
>> time where a CWND is underutilized (but the connection is not idle) =
may
>> observe that the partial_bytes_ack increases to a very high value, =
possibly
>> it increases to
>>=20
>> a value many times higher than the CWND.
>>=20
>>=20
>>=20
>> The question now arises what should happen once the CWND resumes =
fully
>> utilization and thus may be increased. That is, is the intention of =
the
>> standard of SCTP CC to have :
>>=20
>> 1.      the CWND now be increased as during "normal" congestion =
control
>> avoidance, i.e., one MTU per acked CWND amount  of data
>>=20
>> -        Or to have -
>>=20
>> 2.      the CWND now be allowed to recapture the growth that it did =
not
>> realize due to under-utilization but which may be recaptured from the
>> partial_bytes_ack value. I.e., let it be increased one MTU per SACK =
until
>> partial_bytes_ack has been decreased below CWND, one CWND at a time.
>>=20
>>=20
>>=20
>> =46rom an implementation perspective the question likely turns into a =
question
>> about the maintaining of the partial_bytes_ack value during the =
period of
>> time when the CWND is underutilized. E.g., should then =
partial_bytes_ack be
>> decreased with CWND for each times it increases beyond the CWND or =
should it
>> be allowed to increase unlimited.  But first we need to find out what =
the
>> intention of the standard is..
>>=20
>>=20
>>=20
>> The above of course is subject to cumulative SACKs and other details =
like
>> that. I hope that we can leave such details out of the discussion.
>>=20
>>=20
>>=20
>> I hope that you may help.
>>=20
>>=20
>>=20
>> Thanks in advance!
>>=20
>>=20
>>=20
>> BR, Karen
>>=20
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>=20
>=20


From randall@lakerest.net  Sat Apr 21 05:34:49 2012
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Subject: Re: SCTP CWND increase after under-utilization in congestion avoidance
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Karen:

As Michael stated, this algorithm was written out in RFC2960 and was not
touched in 4960 to my knowledge. I believe Lixia texted the original =
wording
(if my memory serves me correct). I guess the thing that is lacking is =
reseting
pba to 0 when you increase <or> would have increased the cwnd (but were =
blocked
due to other issues.. i.e. you were not using the full cwnd).

And as Michael stated, we try to follow 100% the RFC in the BSD =
implementation ;-)

R
On Apr 20, 2012, at 3:52 AM, <Karen.Nielsen@tieto.com> =
<Karen.Nielsen@tieto.com> wrote:

> Hi,
>=20
> RFC4960 specifies for the partial_bytes_ack (pba) to be uncritically =
increased upon receipt of SACKs. Uncritically here means also if the =
CWND is under-utilized, that is, also if flightsize < CWND.
> Further then once traffic increases (or resumes) and the CWND becomes =
fully utilized and SACKs are received,
> then the prescription is for the CWND to be increases one MTU provided =
the pba value >=3D CWND (and pba then set =3D pba - CWND)
>=20
> The pba parameter first was introduced in TCP ABC RFC3465 and the =
parameter is used in the implementation to mimic congestion avoidance =
CWND increase slope of "+MTU*MTU/CWND per SACK" as specified by RFC5681.
>=20
> Following the prescriptions of RFC4960 uncritically one may have the =
pba increase to multiple times the CWND during a
> period of CWND underutilization and further, once traffic and CWND =
utilization resumes, one may have a situation where the CWND in =
principle may be increases by one MTU per received SACK (more or less) a =
number of times corresponding to the value INT(pba/CWND) until the pba =
parameter iteratively has been decreased below present CWND. This =
effectively results in slow start slope increase of the CWND during =
congestion avoidance.
>=20
> Randy and Michael, as authors of the RFc4960, could I ask you to =
confirm that while the explicit handling of the pba in this particular =
case may be an implementation decision THEN THE INTENTION of SCTP CC as =
specified in RFC 4960 is for the CWND to be increased following =
congestion avoidance slope in this situation (traffic resume after CWND =
underutilization in congestion avoidance phase).
>=20
> Many Thanks in advance !!
>=20
> BR, Karen
>=20
>=20
> -----Original Message-----
> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf =
Of Karen.Nielsen@tieto.com
> Sent: 18. april 2012 09:25
> To: mattmathis@google.com
> Cc: tcpm@ietf.org; tsvwg@ietf.org
> Subject: RE: [tcpm] CWND increase after under-utilization in =
congestion avoidance
>=20
> Hi,
>=20
> Thanks a lot!=20
>=20
> My main concern is the slope of the increase of the CWND after a =
period of CWND underutilization in congestion avoidance phase.
>=20
> My general understanding of the standards is that during slow start =
one may raise the CWND per received (S)ACK whereas during congestion =
avoidance generally CWND may be increase by one MTU only per RTT. Then =
some of the more experimental CC algorithms allow for slow start slope =
increase after a packet loss, but that may be a sidetrack here. At least =
my question really goes to the base standard of SCTP as specified in =
RFC4960. But I assume that the question is equally relevant for a TCP =
implementation that implements CWND validation in some form.
>=20
> The assumption here is that we are in congestion avoidance phase, but =
also that the accumulated partical_bytes_ack can allow for several =
increases of the CWND within the same RTT as SACKs will be returned. =
Provided that the application continues to fill the increasing CWND, the =
question is whether SCTP (TCP) then is allowed to proactively increase =
the CWND several times per RTT (slow start slope) or whether the =
increase should be limited to be done one per RTT (congestion avoidance =
slope).
>=20
> Thanks!
>=20
> BR, Karen
>=20
> -----Original Message-----
> From: Matt Mathis [mailto:mattmathis@google.com]=20
> Sent: 17. april 2012 22:37
> To: Nielsen Karen
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] CWND increase after under-utilization in =
congestion avoidance
>=20
> Well, my opinion would be that as long as  flightsize >=3D cwnd at =
least
> once recently it should be ok to continue to raise cwnd.  Note that
> raising cwnd probably causes the predicate to become not true.  Also
> "recently" should be no shorter than one RTT, but may be a longer (the
> details are not so important).   The point is that as long as FS
> reaches cwnd at least once per RTT, it is fine to continue to grow the
> window.  I believe I have seen it implemented "as long as FS comes
> close to cwnd..."
>=20
> In the big picture is it important not to keep raising cwnd if cwnd in
> not controlling the connection, because then you have no way to know
> if it is a valid measure of anything useful.  However, as long as cwnd
> has been "tested" it is fine to keep pushing it up.
>=20
> Thanks,
> --MM--
> The best way to predict the future is to create it.  - Alan Kay
>=20
>=20
>=20
> On Tue, Apr 17, 2012 at 5:50 AM,  <Karen.Nielsen@tieto.com> wrote:
>> Hi,
>>=20
>>=20
>>=20
>> SCTP as prescribed by RFC4960 operates with CWND validation in that =
CWND
>> increase always
>>=20
>> is subject to its fully usage. That is, CWND only may increase when
>> flightsize >=3D CWND.
>>=20
>> Opposed to TCP, as standardized by RFC5681, CWND is thus not =
necessarily
>> increased even when partial_bytes_ack
>>=20
>> increases beyond the present CWND.
>>=20
>>=20
>>=20
>> An implementation which boldly augments the partial_bytes_ack during =
the
>> time where a CWND is underutilized (but the connection is not idle) =
may
>> observe that the partial_bytes_ack increases to a very high value, =
possibly
>> it increases to
>>=20
>> a value many times higher than the CWND.
>>=20
>>=20
>>=20
>> The question now arises what should happen once the CWND resumes =
fully
>> utilization and thus may be increased. That is, is the intention of =
the
>> standard of SCTP CC to have :
>>=20
>> 1.      the CWND now be increased as during "normal" congestion =
control
>> avoidance, i.e., one MTU per acked CWND amount  of data
>>=20
>> -        Or to have -
>>=20
>> 2.      the CWND now be allowed to recapture the growth that it did =
not
>> realize due to under-utilization but which may be recaptured from the
>> partial_bytes_ack value. I.e., let it be increased one MTU per SACK =
until
>> partial_bytes_ack has been decreased below CWND, one CWND at a time.
>>=20
>>=20
>>=20
>> =46rom an implementation perspective the question likely turns into a =
question
>> about the maintaining of the partial_bytes_ack value during the =
period of
>> time when the CWND is underutilized. E.g., should then =
partial_bytes_ack be
>> decreased with CWND for each times it increases beyond the CWND or =
should it
>> be allowed to increase unlimited.  But first we need to find out what =
the
>> intention of the standard is..
>>=20
>>=20
>>=20
>> The above of course is subject to cumulative SACKs and other details =
like
>> that. I hope that we can leave such details out of the discussion.
>>=20
>>=20
>>=20
>> I hope that you may help.
>>=20
>>=20
>>=20
>> Thanks in advance!
>>=20
>>=20
>>=20
>> BR, Karen
>>=20
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>=20
>=20

-----
Randall Stewart
randall@lakerest.net





From Karen.Nielsen@tieto.com  Sun Apr 22 23:30:28 2012
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From: <Karen.Nielsen@tieto.com>
To: <randall@lakerest.net>
Date: Mon, 23 Apr 2012 09:30:19 +0300
Subject: RE: SCTP CWND increase after under-utilization in congestion avoidance
Thread-Topic: SCTP CWND increase after under-utilization in congestion avoidance
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Hi,

Thanks a lot for the comments and for the confirmation.

I think, as you say, that the conclusion is that the algorithm as described=
 in the RFC
is not complete and that it need, by implementation choice, to be adjusted=
=20
in order to ensure that the CWND does not increase more than 1MTU per RTT i=
n congestion avoidance.
The exact way this is achieved, either by adjusting how the pba is increase=
d when the CWBD is underutilized or
by adjusting the CWND increase or the pba value when CWWND usage resumes (o=
r something completely different) would then, for now at least, remain an i=
mplementation decision.

Right ?!

BR, Karen

BSD implementation can then discuss with itself how to find peace with the =
conflicting statements of the RFC in=20
this respect. Difficult to be 100% compliant with all statements, I should =
think ;-)=20

-----Original Message-----
From: Randy Stewart [mailto:randall@lakerest.net]=20
Sent: 21. april 2012 14:33
To: Nielsen Karen
Cc: tsvwg@ietf.org; tuexen@fh-muenster.de
Subject: Re: SCTP CWND increase after under-utilization in congestion avoid=
ance

Karen:

As Michael stated, this algorithm was written out in RFC2960 and was not
touched in 4960 to my knowledge. I believe Lixia texted the original wordin=
g
(if my memory serves me correct). I guess the thing that is lacking is rese=
ting
pba to 0 when you increase <or> would have increased the cwnd (but were blo=
cked
due to other issues.. i.e. you were not using the full cwnd).

And as Michael stated, we try to follow 100% the RFC in the BSD implementat=
ion ;-)

R
On Apr 20, 2012, at 3:52 AM, <Karen.Nielsen@tieto.com> <Karen.Nielsen@tieto=
.com> wrote:

> Hi,
>=20
> RFC4960 specifies for the partial_bytes_ack (pba) to be uncritically incr=
eased upon receipt of SACKs. Uncritically here means also if the CWND is un=
der-utilized, that is, also if flightsize < CWND.
> Further then once traffic increases (or resumes) and the CWND becomes ful=
ly utilized and SACKs are received,
> then the prescription is for the CWND to be increases one MTU provided th=
e pba value >=3D CWND (and pba then set =3D pba - CWND)
>=20
> The pba parameter first was introduced in TCP ABC RFC3465 and the paramet=
er is used in the implementation to mimic congestion avoidance CWND increas=
e slope of "+MTU*MTU/CWND per SACK" as specified by RFC5681.
>=20
> Following the prescriptions of RFC4960 uncritically one may have the pba =
increase to multiple times the CWND during a
> period of CWND underutilization and further, once traffic and CWND utiliz=
ation resumes, one may have a situation where the CWND in principle may be =
increases by one MTU per received SACK (more or less) a number of times cor=
responding to the value INT(pba/CWND) until the pba parameter iteratively h=
as been decreased below present CWND. This effectively results in slow star=
t slope increase of the CWND during congestion avoidance.
>=20
> Randy and Michael, as authors of the RFc4960, could I ask you to confirm =
that while the explicit handling of the pba in this particular case may be =
an implementation decision THEN THE INTENTION of SCTP CC as specified in RF=
C 4960 is for the CWND to be increased following congestion avoidance slope=
 in this situation (traffic resume after CWND underutilization in congestio=
n avoidance phase).
>=20
> Many Thanks in advance !!
>=20
> BR, Karen
>=20
>=20
> -----Original Message-----
> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of=
 Karen.Nielsen@tieto.com
> Sent: 18. april 2012 09:25
> To: mattmathis@google.com
> Cc: tcpm@ietf.org; tsvwg@ietf.org
> Subject: RE: [tcpm] CWND increase after under-utilization in congestion a=
voidance
>=20
> Hi,
>=20
> Thanks a lot!=20
>=20
> My main concern is the slope of the increase of the CWND after a period o=
f CWND underutilization in congestion avoidance phase.
>=20
> My general understanding of the standards is that during slow start one m=
ay raise the CWND per received (S)ACK whereas during congestion avoidance g=
enerally CWND may be increase by one MTU only per RTT. Then some of the mor=
e experimental CC algorithms allow for slow start slope increase after a pa=
cket loss, but that may be a sidetrack here. At least my question really go=
es to the base standard of SCTP as specified in RFC4960. But I assume that =
the question is equally relevant for a TCP implementation that implements C=
WND validation in some form.
>=20
> The assumption here is that we are in congestion avoidance phase, but als=
o that the accumulated partical_bytes_ack can allow for several increases o=
f the CWND within the same RTT as SACKs will be returned. Provided that the=
 application continues to fill the increasing CWND, the question is whether=
 SCTP (TCP) then is allowed to proactively increase the CWND several times =
per RTT (slow start slope) or whether the increase should be limited to be =
done one per RTT (congestion avoidance slope).
>=20
> Thanks!
>=20
> BR, Karen
>=20
> -----Original Message-----
> From: Matt Mathis [mailto:mattmathis@google.com]=20
> Sent: 17. april 2012 22:37
> To: Nielsen Karen
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] CWND increase after under-utilization in congestion a=
voidance
>=20
> Well, my opinion would be that as long as  flightsize >=3D cwnd at least
> once recently it should be ok to continue to raise cwnd.  Note that
> raising cwnd probably causes the predicate to become not true.  Also
> "recently" should be no shorter than one RTT, but may be a longer (the
> details are not so important).   The point is that as long as FS
> reaches cwnd at least once per RTT, it is fine to continue to grow the
> window.  I believe I have seen it implemented "as long as FS comes
> close to cwnd..."
>=20
> In the big picture is it important not to keep raising cwnd if cwnd in
> not controlling the connection, because then you have no way to know
> if it is a valid measure of anything useful.  However, as long as cwnd
> has been "tested" it is fine to keep pushing it up.
>=20
> Thanks,
> --MM--
> The best way to predict the future is to create it.  - Alan Kay
>=20
>=20
>=20
> On Tue, Apr 17, 2012 at 5:50 AM,  <Karen.Nielsen@tieto.com> wrote:
>> Hi,
>>=20
>>=20
>>=20
>> SCTP as prescribed by RFC4960 operates with CWND validation in that CWND
>> increase always
>>=20
>> is subject to its fully usage. That is, CWND only may increase when
>> flightsize >=3D CWND.
>>=20
>> Opposed to TCP, as standardized by RFC5681, CWND is thus not necessarily
>> increased even when partial_bytes_ack
>>=20
>> increases beyond the present CWND.
>>=20
>>=20
>>=20
>> An implementation which boldly augments the partial_bytes_ack during the
>> time where a CWND is underutilized (but the connection is not idle) may
>> observe that the partial_bytes_ack increases to a very high value, possi=
bly
>> it increases to
>>=20
>> a value many times higher than the CWND.
>>=20
>>=20
>>=20
>> The question now arises what should happen once the CWND resumes fully
>> utilization and thus may be increased. That is, is the intention of the
>> standard of SCTP CC to have :
>>=20
>> 1.      the CWND now be increased as during "normal" congestion control
>> avoidance, i.e., one MTU per acked CWND amount  of data
>>=20
>> -        Or to have -
>>=20
>> 2.      the CWND now be allowed to recapture the growth that it did not
>> realize due to under-utilization but which may be recaptured from the
>> partial_bytes_ack value. I.e., let it be increased one MTU per SACK unti=
l
>> partial_bytes_ack has been decreased below CWND, one CWND at a time.
>>=20
>>=20
>>=20
>> From an implementation perspective the question likely turns into a ques=
tion
>> about the maintaining of the partial_bytes_ack value during the period o=
f
>> time when the CWND is underutilized. E.g., should then partial_bytes_ack=
 be
>> decreased with CWND for each times it increases beyond the CWND or shoul=
d it
>> be allowed to increase unlimited.  But first we need to find out what th=
e
>> intention of the standard is..
>>=20
>>=20
>>=20
>> The above of course is subject to cumulative SACKs and other details lik=
e
>> that. I hope that we can leave such details out of the discussion.
>>=20
>>=20
>>=20
>> I hope that you may help.
>>=20
>>=20
>>=20
>> Thanks in advance!
>>=20
>>=20
>>=20
>> BR, Karen
>>=20
>>=20
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www.ietf.org/mailman/listinfo/tcpm
>>=20
>=20

-----
Randall Stewart
randall@lakerest.net





From seggelmann@fh-muenster.de  Mon Apr 23 06:35:31 2012
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Subject: Re: WGLC for draft-ietf-tsvwg-sctp-udp-encaps-03, ending Friday 20th April 2012
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I just had a look at the draft again, and I'm wondering how the Path MTU =
Discovery (section 4.6) is handled when the Don't Fragment bit cannot be =
set. If a userland implementation is used, this is the case on Mac OS X, =
for example.

Best regards
Robin





On Mar 28, 2012, at 3:25 PM, Gorry Fairhurst wrote:

> This email announces the start of a working group last call for
> draft-ietf-tsvwg-sctp-udp-encaps, "UDP Encapsulation of SCTP Packets".
>=20
> Following the meeting this week, this document is now thought to be =
ready to proceed to be published as a BCP. Please send any comments to =
the TSVWG list.
>=20
> The draft is available at:
> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-udp-encaps-03
>=20
> The last call will run for THREE weeks, ending Friday 20th April 2012.
>=20
> Emails saying "I support" or "I don't support" publication are also =
most helpful in judging the WG consensus. Please let us know if this =
draft is useful and seems to provide the correct advice.
>=20
> James and Gorry
> (TSVWG Chairs)
> tsvwg-chairs@ietf.org


From martin.becke@uni-due.de  Mon Apr 23 07:36:51 2012
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From: "Becke, Martin" <martin.becke@uni-due.de>
To: Robin Seggelmann <seggelmann@fh-muenster.de>
Subject: Re: WGLC for draft-ietf-tsvwg-sctp-udp-encaps-03,	ending Friday 20th April 2012
Thread-Topic: WGLC for draft-ietf-tsvwg-sctp-udp-encaps-03,	ending Friday 20th April 2012
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--_000_D15F3562C85C4307A1A22E5EC877A20Duniduede_
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AFAIK, on MAC OS X you could do a work around by set the DF bit on a raw so=
cket, but I do not think that this will work for a SCTP userland implementa=
tion.


Am 23.04.2012 um 15:35 schrieb Robin Seggelmann:

I just had a look at the draft again, and I'm wondering how the Path MTU Di=
scovery (section 4.6) is handled when the Don't Fragment bit cannot be set.=
 If a userland implementation is used, this is the case on Mac OS X, for ex=
ample.

Best regards
Robin





On Mar 28, 2012, at 3:25 PM, Gorry Fairhurst wrote:

This email announces the start of a working group last call for
draft-ietf-tsvwg-sctp-udp-encaps, "UDP Encapsulation of SCTP Packets".

Following the meeting this week, this document is now thought to be ready t=
o proceed to be published as a BCP. Please send any comments to the TSVWG l=
ist.

The draft is available at:
http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-udp-encaps-03

The last call will run for THREE weeks, ending Friday 20th April 2012.

Emails saying "I support" or "I don't support" publication are also most he=
lpful in judging the WG consensus. Please let us know if this draft is usef=
ul and seems to provide the correct advice.

James and Gorry
(TSVWG Chairs)
tsvwg-chairs@ietf.org<mailto:tsvwg-chairs@ietf.org>


--_000_D15F3562C85C4307A1A22E5EC877A20Duniduede_
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Content-ID: <054B9AF511E99449A28DA4D31F0DCB65@win.uni-due.de>
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<div>AFAIK, on MAC OS X you could do a work around by set the DF bit on a r=
aw socket, but I do not think that this will work for a SCTP userland imple=
mentation.&nbsp;</div>
<div><br>
</div>
<br>
<div>
<div>Am 23.04.2012 um 15:35 schrieb Robin Seggelmann:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite"><span class=3D"Apple-style-span" style=3D"border-=
collapse: separate; font-family: Helvetica; font-style: normal; font-varian=
t: normal; font-weight: normal; letter-spacing: normal; line-height: normal=
; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: n=
one; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-hori=
zontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-dec=
orations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stro=
ke-width: 0px; font-size: medium; ">I
 just had a look at the draft again, and I'm wondering how the Path MTU Dis=
covery (section 4.6) is handled when the Don't Fragment bit cannot be set. =
If a userland implementation is used, this is the case on Mac OS X, for exa=
mple.<br>
<br>
Best regards<br>
Robin<br>
<br>
<br>
<br>
<br>
<br>
On Mar 28, 2012, at 3:25 PM, Gorry Fairhurst wrote:<br>
<br>
<blockquote type=3D"cite">This email announces the start of a working group=
 last call for<br>
</blockquote>
<blockquote type=3D"cite">draft-ietf-tsvwg-sctp-udp-encaps, &quot;UDP Encap=
sulation of SCTP Packets&quot;.<br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">Following the meeting this week, this document is=
 now thought to be ready to proceed to be published as a BCP. Please send a=
ny comments to the TSVWG list.<br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">The draft is available at:<br>
</blockquote>
<blockquote type=3D"cite"><a href=3D"http://tools.ietf.org/html/draft-ietf-=
tsvwg-sctp-udp-encaps-03">http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-=
udp-encaps-03</a><br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">The last call will run for THREE weeks, ending Fr=
iday 20th April 2012.<br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">Emails saying &quot;I support&quot; or &quot;I do=
n't support&quot; publication are also most helpful in judging the WG conse=
nsus. Please let us know if this draft is useful and seems to provide the c=
orrect advice.<br>
</blockquote>
<blockquote type=3D"cite"><br>
</blockquote>
<blockquote type=3D"cite">James and Gorry<br>
</blockquote>
<blockquote type=3D"cite">(TSVWG Chairs)<br>
</blockquote>
<blockquote type=3D"cite"><a href=3D"mailto:tsvwg-chairs@ietf.org">tsvwg-ch=
airs@ietf.org</a></blockquote>
</span></blockquote>
</div>
<br>
</body>
</html>

--_000_D15F3562C85C4307A1A22E5EC877A20Duniduede_--

From Michael.Tuexen@lurchi.franken.de  Mon Apr 23 07:49:50 2012
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Subject: Re: WGLC for draft-ietf-tsvwg-sctp-udp-encaps-03, ending Friday 20th April 2012
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On Apr 23, 2012, at 4:36 PM, Becke, Martin wrote:

> AFAIK, on MAC OS X you could do a work around by set the DF bit on a =
raw socket, but I do not think that this will work for a SCTP userland =
implementation.=20
I guess Robins points is the you SCTP/UDP because you don't have the =
privileges to open a raw socket.
Some OSes support the controlling of the DF bit for a normal user, some =
don't.

I'll update the text accordingly. I guess if you can't control the DF =
bit, you can't do PMTU discovery.
So you need to choose a small MTU value... Or use IPv6, which doesn't =
have this problem.

Best regards
Michael
>=20
>=20
> Am 23.04.2012 um 15:35 schrieb Robin Seggelmann:
>=20
>> I just had a look at the draft again, and I'm wondering how the Path =
MTU Discovery (section 4.6) is handled when the Don't Fragment bit =
cannot be set. If a userland implementation is used, this is the case on =
Mac OS X, for example.
>>=20
>> Best regards
>> Robin
>>=20
>>=20
>>=20
>>=20
>>=20
>> On Mar 28, 2012, at 3:25 PM, Gorry Fairhurst wrote:
>>=20
>>> This email announces the start of a working group last call for
>>> draft-ietf-tsvwg-sctp-udp-encaps, "UDP Encapsulation of SCTP =
Packets".
>>>=20
>>> Following the meeting this week, this document is now thought to be =
ready to proceed to be published as a BCP. Please send any comments to =
the TSVWG list.
>>>=20
>>> The draft is available at:
>>> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-udp-encaps-03
>>>=20
>>> The last call will run for THREE weeks, ending Friday 20th April =
2012.
>>>=20
>>> Emails saying "I support" or "I don't support" publication are also =
most helpful in judging the WG consensus. Please let us know if this =
draft is useful and seems to provide the correct advice.
>>>=20
>>> James and Gorry
>>> (TSVWG Chairs)
>>> tsvwg-chairs@ietf.org
>=20


From martin.becke@uni-due.de  Mon Apr 23 09:06:56 2012
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From: "Becke, Martin" <martin.becke@uni-due.de>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Subject: Re: WGLC for draft-ietf-tsvwg-sctp-udp-encaps-03,	ending Friday 20th April 2012
Thread-Topic: WGLC for draft-ietf-tsvwg-sctp-udp-encaps-03,	ending Friday 20th April 2012
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Am 23.04.2012 um 16:49 schrieb Michael Tuexen:

> On Apr 23, 2012, at 4:36 PM, Becke, Martin wrote:
>=20
>> AFAIK, on MAC OS X you could do a work around by set the DF bit on a raw=
 socket, but I do not think that this will work for a SCTP userland impleme=
ntation.=20
> I guess Robins points is the you SCTP/UDP because you don't have the priv=
ileges to open a raw socket.
> Some OSes support the controlling of the DF bit for a normal user, some d=
on't.

Yes, this results in a privilege issue.=20

>=20
> I'll update the text accordingly. I guess if you can't control the DF bit=
, you can't do PMTU discovery.
> So you need to choose a small MTU value=85

I agree.=20

Is "you" for you the protocol developer or the user of the protocol. =20
If I, as a user, have enough knowledge about the connection (even without c=
ontrol and without PMTUD) I would like to change the value to increase my p=
erformance.=20

Martin


> Or use IPv6, which doesn't have this problem.
>=20
> Best regards
> Michael
>>=20
>>=20
>> Am 23.04.2012 um 15:35 schrieb Robin Seggelmann:
>>=20
>>> I just had a look at the draft again, and I'm wondering how the Path MT=
U Discovery (section 4.6) is handled when the Don't Fragment bit cannot be =
set. If a userland implementation is used, this is the case on Mac OS X, fo=
r example.
>>>=20
>>> Best regards
>>> Robin
>>>=20
>>>=20
>>>=20
>>>=20
>>>=20
>>> On Mar 28, 2012, at 3:25 PM, Gorry Fairhurst wrote:
>>>=20
>>>> This email announces the start of a working group last call for
>>>> draft-ietf-tsvwg-sctp-udp-encaps, "UDP Encapsulation of SCTP Packets".
>>>>=20
>>>> Following the meeting this week, this document is now thought to be re=
ady to proceed to be published as a BCP. Please send any comments to the TS=
VWG list.
>>>>=20
>>>> The draft is available at:
>>>> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-udp-encaps-03
>>>>=20
>>>> The last call will run for THREE weeks, ending Friday 20th April 2012.
>>>>=20
>>>> Emails saying "I support" or "I don't support" publication are also mo=
st helpful in judging the WG consensus. Please let us know if this draft is=
 useful and seems to provide the correct advice.
>>>>=20
>>>> James and Gorry
>>>> (TSVWG Chairs)
>>>> tsvwg-chairs@ietf.org
>>=20
>=20


From Michael.Tuexen@lurchi.franken.de  Mon Apr 23 10:29:49 2012
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On Apr 23, 2012, at 6:06 PM, Becke, Martin wrote:

>=20
> Am 23.04.2012 um 16:49 schrieb Michael Tuexen:
>=20
>> On Apr 23, 2012, at 4:36 PM, Becke, Martin wrote:
>>=20
>>> AFAIK, on MAC OS X you could do a work around by set the DF bit on a =
raw socket, but I do not think that this will work for a SCTP userland =
implementation.=20
>> I guess Robins points is the you SCTP/UDP because you don't have the =
privileges to open a raw socket.
>> Some OSes support the controlling of the DF bit for a normal user, =
some don't.
>=20
> Yes, this results in a privilege issue.=20
>=20
>>=20
>> I'll update the text accordingly. I guess if you can't control the DF =
bit, you can't do PMTU discovery.
>> So you need to choose a small MTU value=85
>=20
> I agree.=20
>=20
> Is "you" for you the protocol developer or the user of the protocol. =20=

I meant the developer of the SCTP stack...
> If I, as a user, have enough knowledge about the connection (even =
without control and without PMTUD) I would like to change the value to =
increase my performance.=20
... then you as a user of an SCTP stack providing a socket API will =
allow you set
the MTU and disable PMTU discovery by using the SCTP_PEER_ADDR_PARAMS =
socket option
as specified in
http://tools.ietf.org/html/rfc6458#section-8.1.12

Best regards
Michael
>=20
> Martin
>=20
>=20
>> Or use IPv6, which doesn't have this problem.
>>=20
>> Best regards
>> Michael
>>>=20
>>>=20
>>> Am 23.04.2012 um 15:35 schrieb Robin Seggelmann:
>>>=20
>>>> I just had a look at the draft again, and I'm wondering how the =
Path MTU Discovery (section 4.6) is handled when the Don't Fragment bit =
cannot be set. If a userland implementation is used, this is the case on =
Mac OS X, for example.
>>>>=20
>>>> Best regards
>>>> Robin
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>> On Mar 28, 2012, at 3:25 PM, Gorry Fairhurst wrote:
>>>>=20
>>>>> This email announces the start of a working group last call for
>>>>> draft-ietf-tsvwg-sctp-udp-encaps, "UDP Encapsulation of SCTP =
Packets".
>>>>>=20
>>>>> Following the meeting this week, this document is now thought to =
be ready to proceed to be published as a BCP. Please send any comments =
to the TSVWG list.
>>>>>=20
>>>>> The draft is available at:
>>>>> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-udp-encaps-03
>>>>>=20
>>>>> The last call will run for THREE weeks, ending Friday 20th April =
2012.
>>>>>=20
>>>>> Emails saying "I support" or "I don't support" publication are =
also most helpful in judging the WG consensus. Please let us know if =
this draft is useful and seems to provide the correct advice.
>>>>>=20
>>>>> James and Gorry
>>>>> (TSVWG Chairs)
>>>>> tsvwg-chairs@ietf.org
>>>=20
>>=20
>=20
>=20


From Ruediger.Geib@telekom.de  Tue Apr 24 01:28:12 2012
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From: <Ruediger.Geib@telekom.de>
To: <jsaldana@unizar.es>
Date: Tue, 24 Apr 2012 10:28:05 +0200
Subject: RE: About the tunnels: packets per second savings
Thread-Topic: About the tunnels: packets per second savings
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Jose,

sorry for being late with my reply. Two of my colleagues replied. Simple ba=
ndwidth
savings at the access itself would be interesting, especially in the case o=
f shared
networks.

Your proposal is related to the "core" part of the network and as you
describe below, high end routers can deal with bigger numbers of packets pe=
r second.

That's the reason why my colleagues are not interested in working on the su=
bject.

Regards,

Ruediger


-----Original Message-----
From: Jose Saldana [mailto:jsaldana@unizar.es]
Sent: Thursday, April 05, 2012 2:14 PM
To: Geib, R=FCdiger
Cc: tsvwg@ietf.org
Subject: About the tunnels: packets per second savings

Ruediger,

Thank you for your fast response.  I will separately answer the two parts o=
f
your email. First, this message about the tunnels. This afternoon or
tomorrow I will send to the list an idea about QoS classes, but I have to
think more before sending it.

I'm glad to hear that you consider the "tunnels" proposal interesting. Ther=
e
is another issue that I discussed with Gorry in Paris: the saving in terms
of packets per second. I attach a graph of the saving in terms of this
parameter using Counter Strike. If you multiplex all the packets you have
received e.g. every 20ms, then you will send  50 pps (the inverse of the
multiplexing period). So you can save a lot of pps.

It is clear that bandwidth saving is interesting, but I have read some
papers in which a "packets per second" upper bound in the routers was
reported:

Section V.A of
W. Feng, F. Chang, W. Feng, J. Walpole: "A Traffic Characterization of
Popular On-Line Games". IEEE/ACM Trans. Netw., pp. 488-500, 2005.
http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf

Section IV.3 of
J. Yu, I. Al-Ajarmeh. "Call Admission Control and Traffic Engineering of
VoIP", In Proc. Second Intenational Conference on Digital
Telecommunications, IEEE ICDT 2007.
http://140.192.40.4:8001/~imad/voip/YuJ-CallAdmin.pdf

They not only talk about the bandwidth limitation of routers, but also the
packets per second limit. Gorry told me that high-end routers may not have
this problem. But low-end routers are not able to back-to-back send small
packets, due to the processing limitation.

So, do you think this saving is interesting? Since you work in the backbone=
,
your opinion is important for me, taking into account that I am an Academia
researcher, and sometimes we are a little far from real things. Perhaps you
may ask some of your colleagues.

Thanks,

Jose

-----Mensaje original-----
De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]
Enviado el: jueves, 05 de abril de 2012 12:23
Para: jsaldana@unizar.es
CC: tsvwg@ietf.org
Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

Jose,

(...) I have removed the first part of the message for clarity.

Then regarding tunnels and header compression Yes, that's smart and there
may be more use cases than gaming. I'm a backbone engineer however and your
proposal is rather application specific. What I can do is ask some
colleagues on their view. But there's no guarantee, that they will respond.

Some initial thoughts:
To me a key point seems to be what can be reached at the bottleneck. There
the compression should save bandwidth with little cost and no negative
service impact. Any kind of shared access from which traffic is transmitted
to the same destination seems to be relevant. The application/tunnel
awareness of network elements may not be feasible, a network terminal at a
UNI may be able to handle that, or any service specific device in a network=
.

Regards,

Ruediger


-----Original Message-----
From: Jose Saldana [mailto:jsaldana@unizar.es]
Sent: Thursday, April 05, 2012 11:29 AM
To: Geib, R=FCdiger
Cc: tsvwg@ietf.org
Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

Ruediger,

I have read your message about draft-polk-tsvwg-rfc4594-update. As you may
know, I presented another draft in Paris (draft-saldana-tsvwg-tcmtf-02), an=
d
after talking to James, I thought that the idea of standardizing the traffi=
c
classification fits well with my proposal. I would like to know your opinio=
n
about it, if possible.

The idea is to use tunnels in order to compress and multiplex real-time
traffic flows. Since these flows tend to use very small packets, they have
very bad efficiency. So perhaps tunnels could be used in order to group
these flows. The drawback is that they add small delays, i.e. the waiting
time in the multiplexer. So another use is to provide flexibility: if you
are a game provider, you may over provision your network in order to be abl=
e
to manage traffic peaks. But perhaps using this, you may be able to have a
smaller network, and use compression when necessary (rush hour, first days
after releasing a new game, etc). When traffic is small, you do not
multiplex, in order to reduce latency.

So if traffic classification is a standard, everyone in the way from the
origin to the destination will handle the traffic with the correct priority=
.

I attach a figure in order to clarify.

Thank you,

Jose

PS: In
http://community.virginmedia.com/t5/Fibre-optic-broadband-cable/World-of-Wa=
r
craft-high-latency-issues-with-virgin-media-users/td-p/44498 you can find
the history of the problems of World of Warcraft for Virgin Media users. Th=
e
problem was caused by a modification in the traffic, and the provider did
not notice about it, so it treated the packets as best-effort.

-----Mensaje original-----
De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de] Enviado el:
jueves, 05 de abril de 2012 9:49
Para: jsaldana@unizar.es
CC: tsvwg@ietf.org
Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

Jose,

you raise an interesting point:

draft-polk-tsvwg-rfc4594-update suggests to standardise application to
codepoint mapping as well as codepoint to class properties. You observe tha=
t
some games are using TCP while others use UDP, which I think is a useful
level of QoS requirement abstraction. There's no need to have like 25+ QoS
(sub)classes, I think. But one class optimised to transport  "UDP" and one
for "TCP" are agreeable.

The feedback that I had on carrier class QoS usage is that there's a point
in standardising class properties, and certainly not to standardise
application mappings to classes (recommending them may however be useful).
My take would be to standardise four classes/properties (in brackets exampl=
e
usage):

a) for traffic which doesn't support re-transmits, requiring lowest delay
and jitter, having limited volume and packet
   size, but being loss tolerant (example usage VoIP)
b) for any other traffic which doesn't support re-transmits (let's say
traffic with real time like properties, but any
   packet size and possibly high volume, UDP transport, like IPTV)
c) for any other traffic which supports re-transmits (QoS traffic using TCP=
,
like application signalling and network management traffic)
d) a default class, for any traffic supporting re-transmits (Best Effort) -
this class may not be regarded as QoS (or as lowest CoS)

What I can tell ist that many carriers and standards on QoS support 3-4
classes, having properties as described above. It is useful to recommend
codepoints matching to these classes. Standardising these codepoints
requires agreement between those having deplyed them. Straightforward for a=
)
EF and d) 000000, probably easy for c) being used to transport Network
Management (CS 6). I'm not aware of common ground for anything else. My
favourites are:

b) AF4 (GSMA supports this one, RFC4594 also)
c) AF3 (MEF and GSMA support this one)

Gaming from my point of view would then match to an AF3 or AF4 class, this
may depend on the implemented application transport. Further details may be
discussed.

Everything else could be reserved, but shouldn't be recommended. Carriers
have deployed differing schemes on codepoints and classes. They need coding
reserverves in 3 bit space (MPLS TC, IP Precedence and Ethernet P-Bits) as
well as in DSCP space if they should converge to a standardised
codepoint/class scheme.

I object to any approach standardising a match of classes and applications
with codepoints as a requirement and I also object to standards leaving no
spare codepoints in 3 bit or 6 bit QoS codepoint space (including 3 bit IP
precedence).

Bob Briscoe mentioned the difference between classes and queues and above
properties can be mapped to queues. If end to end QoS is desired, the same
class properties must be negotiated end to end. The lower the number of
classes and codepoints being deployed, the higher the probability of gettin=
g
end to end QoS. I don't support an approach requiring 8 three bit classes
and 25 or more 6 bit classes and codepoints.

Regards,

Ruediger

________________________________

From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of
Jose Saldana
Sent: Monday, April 02, 2012 12:17 PM
To: James Polk
Cc: tsvwg@ietf.org
Subject: About draft-polk-tsvwg-rfc4594-update-00.txt



James,



After returning home from IETF 83, I have been reading your draft. I find i=
t
interesting to standardize the service classes. Otherwise, if each one
classifies each traffic according to different criteria, Diffserv becomes
less useful.



Regarding "Real-time interactive" service class, which refers to online
gaming (my main research field), I find it really interesting: online games
are at least as real-time as VoIP is. In most cases, the gamers are a very
difficult people to deal with. If the latency is high, they leave and never
return (this statement is not mine, it can be corroborated in the reference=
s
I put below). This has been studied by many research groups, with empirical
tests using real gamers and adding delay, jitter, packet loss.



Other question: in the draft it is said that online games use RTP/UDP
streams. This can be a little confusing. In fact, the vast majority of
online games do not use RTP, but bare UDP or even TCP. In
http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.pdf, the
authors proposed the use of an adaptation of RTP in order to manage gaming
traffic. But as far as I know, this approach is not used by commercial
games.





A typical classification of today's games is:



-          FPS (First Person Shooters) e.g. Counter Strike, Halo, Call of
Duty: They use UDP

-          MMORPG (Massively Multiplayer Online Role-playing Games), e.g.
World of Warcraft: They normally use TCP, even though some of them use UDP

-          RTS (Real-time strategy) e.g. Age of Empires: They may use TCP o=
r
UDP

-          Sports games: Since there is a big variety, they have not been
deeply studied in the literature



Some research papers supporting this classification, and also the real-time
requirements of the games:



First Person Shooters



-          A survey of traffic models for 17 First Person Shooters can be
found here. All of them use UDP
http://www.discover.uottawa.ca/publications/files/MMOG-Survey_IEEE-IC.pdf



-          In this paper from my research group, 8 FPS games were analyzed
in order to compress and multiplex their traffic. Figure 5 shows their
histograms (inter-packet time and packet size). It can be seen that they
generate real-time flows, with characteristics very similar to VoIP (packet=
s
of 60-80 bytes every 15-40 ms):
http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana.pdf



-          Another paper about FPS using UDP. Section V presents some
interesting conclusions about routing infrastructure.
http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf



-          In this paper the network factors annoying the gamers are
analyzed. There are also some calculations of the MOS. Above 150 ms the
quality is bad (something similar to what happens with VoIP)
http://www.research.ibm.com/netgames2005/papers/dick.pdf



-          In this one, a MOS model for a FPS (Quake IV) is deployed. It is
only based on delay and jitter, since packet loss has vevy little influence
(for this concrete game). Delay above 140 ms produces MOS below 3
http://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf



-          In this paper, it is found that some FPS work properly even with
30% packet loss, whereas others stop working with 4%. RTT above 300 ms (150
OWD) makes MOS go above 3
http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pdf



MMORPGs:

-          A traffic model for World of Warcraft:
http://publik.tuwien.ac.at/files/pub-et_12119.pdf



-          This one presents a MOS for World of Warcraft. When added delay
is above 160ms, the MOS is roughly 3.:
http://publik.tuwien.ac.at/files/PubDat_170202.pdf



Best regards,



Jose Saldana


From jsaldana@unizar.es  Tue Apr 24 02:14:59 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
To: <Ruediger.Geib@telekom.de>
References: <000601cd1325$9a1afcb0$ce50f610$@unizar.es> <580BEA5E3B99744AB1F5BFF5E9A3C67D13DEF36B2D@HE111648.emea1.cds.t-internal.com>
In-Reply-To: <580BEA5E3B99744AB1F5BFF5E9A3C67D13DEF36B2D@HE111648.emea1.cds.t-internal.com>
Subject: RE: About the tunnels: packets per second savings
Date: Tue, 24 Apr 2012 11:14:54 +0200
Organization: Universidad de Zaragoza
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Ruediger,

Thanks for your response. This is good for understanding which are the =
best
scenarios for TCMTF.

Best regards,

Jose

-----Mensaje original-----
De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]=20
Enviado el: martes, 24 de abril de 2012 10:28
Para: jsaldana@unizar.es
CC: tsvwg@ietf.org
Asunto: RE: About the tunnels: packets per second savings

Jose,

sorry for being late with my reply. Two of my colleagues replied. Simple
bandwidth savings at the access itself would be interesting, especially =
in
the case of shared networks.

Your proposal is related to the "core" part of the network and as you
describe below, high end routers can deal with bigger numbers of packets =
per
second.

That's the reason why my colleagues are not interested in working on the
subject.

Regards,

Ruediger


-----Original Message-----
From: Jose Saldana [mailto:jsaldana@unizar.es]
Sent: Thursday, April 05, 2012 2:14 PM
To: Geib, R=FCdiger
Cc: tsvwg@ietf.org
Subject: About the tunnels: packets per second savings

Ruediger,

Thank you for your fast response.  I will separately answer the two =
parts of
your email. First, this message about the tunnels. This afternoon or
tomorrow I will send to the list an idea about QoS classes, but I have =
to
think more before sending it.

I'm glad to hear that you consider the "tunnels" proposal interesting. =
There
is another issue that I discussed with Gorry in Paris: the saving in =
terms
of packets per second. I attach a graph of the saving in terms of this
parameter using Counter Strike. If you multiplex all the packets you =
have
received e.g. every 20ms, then you will send  50 pps (the inverse of the
multiplexing period). So you can save a lot of pps.

It is clear that bandwidth saving is interesting, but I have read some
papers in which a "packets per second" upper bound in the routers was
reported:

Section V.A of
W. Feng, F. Chang, W. Feng, J. Walpole: "A Traffic Characterization of
Popular On-Line Games". IEEE/ACM Trans. Netw., pp. 488-500, 2005.
http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf

Section IV.3 of
J. Yu, I. Al-Ajarmeh. "Call Admission Control and Traffic Engineering of
VoIP", In Proc. Second Intenational Conference on Digital
Telecommunications, IEEE ICDT 2007.
http://140.192.40.4:8001/~imad/voip/YuJ-CallAdmin.pdf

They not only talk about the bandwidth limitation of routers, but also =
the
packets per second limit. Gorry told me that high-end routers may not =
have
this problem. But low-end routers are not able to back-to-back send =
small
packets, due to the processing limitation.

So, do you think this saving is interesting? Since you work in the =
backbone,
your opinion is important for me, taking into account that I am an =
Academia
researcher, and sometimes we are a little far from real things. Perhaps =
you
may ask some of your colleagues.

Thanks,

Jose

-----Mensaje original-----
De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de] Enviado =
el:
jueves, 05 de abril de 2012 12:23
Para: jsaldana@unizar.es
CC: tsvwg@ietf.org
Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

Jose,

(...) I have removed the first part of the message for clarity.

Then regarding tunnels and header compression Yes, that's smart and =
there
may be more use cases than gaming. I'm a backbone engineer however and =
your
proposal is rather application specific. What I can do is ask some
colleagues on their view. But there's no guarantee, that they will =
respond.

Some initial thoughts:
To me a key point seems to be what can be reached at the bottleneck. =
There
the compression should save bandwidth with little cost and no negative
service impact. Any kind of shared access from which traffic is =
transmitted
to the same destination seems to be relevant. The application/tunnel
awareness of network elements may not be feasible, a network terminal at =
a
UNI may be able to handle that, or any service specific device in a =
network.

Regards,

Ruediger


-----Original Message-----
From: Jose Saldana [mailto:jsaldana@unizar.es]
Sent: Thursday, April 05, 2012 11:29 AM
To: Geib, R=FCdiger
Cc: tsvwg@ietf.org
Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

Ruediger,

I have read your message about draft-polk-tsvwg-rfc4594-update. As you =
may
know, I presented another draft in Paris (draft-saldana-tsvwg-tcmtf-02), =
and
after talking to James, I thought that the idea of standardizing the =
traffic
classification fits well with my proposal. I would like to know your =
opinion
about it, if possible.

The idea is to use tunnels in order to compress and multiplex real-time
traffic flows. Since these flows tend to use very small packets, they =
have
very bad efficiency. So perhaps tunnels could be used in order to group
these flows. The drawback is that they add small delays, i.e. the =
waiting
time in the multiplexer. So another use is to provide flexibility: if =
you
are a game provider, you may over provision your network in order to be =
able
to manage traffic peaks. But perhaps using this, you may be able to have =
a
smaller network, and use compression when necessary (rush hour, first =
days
after releasing a new game, etc). When traffic is small, you do not
multiplex, in order to reduce latency.

So if traffic classification is a standard, everyone in the way from the
origin to the destination will handle the traffic with the correct =
priority.

I attach a figure in order to clarify.

Thank you,

Jose

PS: In
http://community.virginmedia.com/t5/Fibre-optic-broadband-cable/World-of-=
War
craft-high-latency-issues-with-virgin-media-users/td-p/44498 you can =
find
the history of the problems of World of Warcraft for Virgin Media users. =
The
problem was caused by a modification in the traffic, and the provider =
did
not notice about it, so it treated the packets as best-effort.

-----Mensaje original-----
De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de] Enviado =
el:
jueves, 05 de abril de 2012 9:49
Para: jsaldana@unizar.es
CC: tsvwg@ietf.org
Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

Jose,

you raise an interesting point:

draft-polk-tsvwg-rfc4594-update suggests to standardise application to
codepoint mapping as well as codepoint to class properties. You observe =
that
some games are using TCP while others use UDP, which I think is a useful
level of QoS requirement abstraction. There's no need to have like 25+ =
QoS
(sub)classes, I think. But one class optimised to transport  "UDP" and =
one
for "TCP" are agreeable.

The feedback that I had on carrier class QoS usage is that there's a =
point
in standardising class properties, and certainly not to standardise
application mappings to classes (recommending them may however be =
useful).
My take would be to standardise four classes/properties (in brackets =
example
usage):

a) for traffic which doesn't support re-transmits, requiring lowest =
delay
and jitter, having limited volume and packet
   size, but being loss tolerant (example usage VoIP)
b) for any other traffic which doesn't support re-transmits (let's say
traffic with real time like properties, but any
   packet size and possibly high volume, UDP transport, like IPTV)
c) for any other traffic which supports re-transmits (QoS traffic using =
TCP,
like application signalling and network management traffic)
d) a default class, for any traffic supporting re-transmits (Best =
Effort) -
this class may not be regarded as QoS (or as lowest CoS)

What I can tell ist that many carriers and standards on QoS support 3-4
classes, having properties as described above. It is useful to recommend
codepoints matching to these classes. Standardising these codepoints
requires agreement between those having deplyed them. Straightforward =
for a)
EF and d) 000000, probably easy for c) being used to transport Network
Management (CS 6). I'm not aware of common ground for anything else. My
favourites are:

b) AF4 (GSMA supports this one, RFC4594 also)
c) AF3 (MEF and GSMA support this one)

Gaming from my point of view would then match to an AF3 or AF4 class, =
this
may depend on the implemented application transport. Further details may =
be
discussed.

Everything else could be reserved, but shouldn't be recommended. =
Carriers
have deployed differing schemes on codepoints and classes. They need =
coding
reserverves in 3 bit space (MPLS TC, IP Precedence and Ethernet P-Bits) =
as
well as in DSCP space if they should converge to a standardised
codepoint/class scheme.

I object to any approach standardising a match of classes and =
applications
with codepoints as a requirement and I also object to standards leaving =
no
spare codepoints in 3 bit or 6 bit QoS codepoint space (including 3 bit =
IP
precedence).

Bob Briscoe mentioned the difference between classes and queues and =
above
properties can be mapped to queues. If end to end QoS is desired, the =
same
class properties must be negotiated end to end. The lower the number of
classes and codepoints being deployed, the higher the probability of =
getting
end to end QoS. I don't support an approach requiring 8 three bit =
classes
and 25 or more 6 bit classes and codepoints.

Regards,

Ruediger

________________________________

From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf =
Of
Jose Saldana
Sent: Monday, April 02, 2012 12:17 PM
To: James Polk
Cc: tsvwg@ietf.org
Subject: About draft-polk-tsvwg-rfc4594-update-00.txt



James,



After returning home from IETF 83, I have been reading your draft. I =
find it
interesting to standardize the service classes. Otherwise, if each one
classifies each traffic according to different criteria, Diffserv =
becomes
less useful.



Regarding "Real-time interactive" service class, which refers to online
gaming (my main research field), I find it really interesting: online =
games
are at least as real-time as VoIP is. In most cases, the gamers are a =
very
difficult people to deal with. If the latency is high, they leave and =
never
return (this statement is not mine, it can be corroborated in the =
references
I put below). This has been studied by many research groups, with =
empirical
tests using real gamers and adding delay, jitter, packet loss.



Other question: in the draft it is said that online games use RTP/UDP
streams. This can be a little confusing. In fact, the vast majority of
online games do not use RTP, but bare UDP or even TCP. In
http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.pdf, =
the
authors proposed the use of an adaptation of RTP in order to manage =
gaming
traffic. But as far as I know, this approach is not used by commercial
games.





A typical classification of today's games is:



-          FPS (First Person Shooters) e.g. Counter Strike, Halo, Call =
of
Duty: They use UDP

-          MMORPG (Massively Multiplayer Online Role-playing Games), =
e.g.
World of Warcraft: They normally use TCP, even though some of them use =
UDP

-          RTS (Real-time strategy) e.g. Age of Empires: They may use =
TCP or
UDP

-          Sports games: Since there is a big variety, they have not =
been
deeply studied in the literature



Some research papers supporting this classification, and also the =
real-time
requirements of the games:



First Person Shooters



-          A survey of traffic models for 17 First Person Shooters can =
be
found here. All of them use UDP
http://www.discover.uottawa.ca/publications/files/MMOG-Survey_IEEE-IC.pdf=




-          In this paper from my research group, 8 FPS games were =
analyzed
in order to compress and multiplex their traffic. Figure 5 shows their
histograms (inter-packet time and packet size). It can be seen that they
generate real-time flows, with characteristics very similar to VoIP =
(packets
of 60-80 bytes every 15-40 ms):
http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana.pdf



-          Another paper about FPS using UDP. Section V presents some
interesting conclusions about routing infrastructure.
http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf



-          In this paper the network factors annoying the gamers are
analyzed. There are also some calculations of the MOS. Above 150 ms the
quality is bad (something similar to what happens with VoIP)
http://www.research.ibm.com/netgames2005/papers/dick.pdf



-          In this one, a MOS model for a FPS (Quake IV) is deployed. It =
is
only based on delay and jitter, since packet loss has vevy little =
influence
(for this concrete game). Delay above 140 ms produces MOS below 3
http://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf



-          In this paper, it is found that some FPS work properly even =
with
30% packet loss, whereas others stop working with 4%. RTT above 300 ms =
(150
OWD) makes MOS go above 3
http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pdf



MMORPGs:

-          A traffic model for World of Warcraft:
http://publik.tuwien.ac.at/files/pub-et_12119.pdf



-          This one presents a MOS for World of Warcraft. When added =
delay
is above 160ms, the MOS is roughly 3.:
http://publik.tuwien.ac.at/files/PubDat_170202.pdf



Best regards,



Jose Saldana



From Ruediger.Geib@telekom.de  Tue Apr 24 02:32:13 2012
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From: <Ruediger.Geib@telekom.de>
To: <jsaldana@unizar.es>
Date: Tue, 24 Apr 2012 11:32:07 +0200
Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
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Jose,

thanks, part of your proposal isn't that far from the current deployment.

IP offers 6 bits to encode QoS classes, while MPLS and Ethernet only offer =
three.
The commodity solution mapping IP QoS classes to MPLS or Ethernet is to cop=
y the
three most significant bits of the DSCP into the three bit QoS codepoint of
Ethernet/MPLS (you proposed to use 2 bits).
These three most significant bits are the so called IP-Precedence (RFC791).

Making this copy mechanism a standard behaviour is a feature I'd support. I=
t's
simple, straightforward an widely deployed. Also DiffServ respects the prec=
edence,
EF has IP Prec 5, AF4 has IP Prec 4 and so on.

All in all now, there are 8 possible codepoints. I suggest to standardise 4
classes. Some may use sub-classes with own codepoints. Further, a reserve
of codepoints should be left for experiments and for provider internal use.

So on standards track I support at most 4 classes and 4 IP precedence value=
s.

The number of IP DSCPs to be standardised should follow the usual IETF way =
of
dealing with things: If there's widespread deployment and more than one
interoperable solution, put things on standards track. My take of
interoperable solution here is not IETF like, as I look at this meaning
two carriers or more having deployed a class in the same way.

Starting from there you soon divide the problem into two issues:

- carrier QoS classes having similar properties.
- codepoints identifying these classes.

Classes and codepoints I like:

Class                   IP Prec, TC, P-Bits     Example Use
UDP short packets       5                               Telephony
UDP any         4                               Streaming, IPTV, Gaming
TCP any         3                               Data, Signaling, 3. party n=
etmgmt
Default         0                               Best Effort

IP prec 6 should be reserved for network management traffic. I don't look
at this as a public class (meaning it will be mapped to another class/DSCP
at interconnection).

Sub-classes on IP level are possible, the TCP any class is a candidate (AF3=
).

The former should see standardisation by now. The latter is not in a good
shape and requires goodwill and allow for migration plan, if something easy=
 to
deploy should result. End to end QoS based with well known class properties
and codepoints only results that way.

A clear cause of the problem is the high number of DSCPs and the
low degree of usage guidance of the AF classes. There's a need for two
standard AF classes and up to 5 codepoints, that's my point of view.

Compare EF: it uses a single DSCP and IP precedence and it has well
defined properties. It saw widespread deployment. EF can be produced end
to end.

I oppose any concept to standardise over 25 DSCPs or any standard
leaving no IP precedence codepoints for reserve. It doesn't matter whether
that's in a VPN or public. QoS classes and codepoints to be put on standard=
s
track should have seen deployment and get support by more than one carrier.

Entertaining documentation on classes and codepoints without a clear demand
for usage impeded end to end QoS deployment so far.

Regards,

Ruediger


-----Original Message-----
From: Jose Saldana [mailto:jsaldana@unizar.es]
Sent: Saturday, April 07, 2012 2:03 PM
To: Geib, R=FCdiger; James M. Polk
Cc: tsvwg@ietf.org
Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt

I have been reading the two messages of Ruediger about Jame's draft. I will
try to make a "status of the question" and then a proposal. If something is
wrong, just tell me.

- We have the old 8 TOS bits of the IP header. Nowadays, two of them are
ECN, and six of them are the DSCP field. So the problem is how to use these
6 bits (64 possible values)
- Network control is 11xxxx, so this uses 16 values
        -11xx11 are defined in RFC 2474, and remain as only experimentally
assignable/useable (this uses 4 of these 16 values)
- This leaves 48 values
- Best effort is 000000
- The draft of James proposes 14 values for the DSCP field, which correspon=
d
to service classes, mapped to application categories.

But Ruediger says that, in his opinion, more than 4 classes are not useful,
according to current implementations.

We can ask ourselves: which are  the key parameters that define how packets
have to be managed in a router? I think that they are:

1) "Real-time-ness": The tolerated delay of the service. The jitter can be
roughly included in this parameter.
2) loss tolerance: We can distinguish: TCP services are tolerant to loss,
since they retransmit packets. And some UDP services are also tolerant to
loss (VoIP, some games, etc)

In my opinion, the "real-time-ness" is the key: it distinguishes very
similar services: e.g. a video conference is different from a broadcast,
since the latter permits delays of some seconds, whereas the first has to b=
e
interactive.

So one possible solution could be:

- Using the first two bits in order to define 4 classes, taking into accoun=
t
the existing implementations. They can be the 4 classes of Fig. 1 of James'
draft, or the ones proposed by Ruediger.
        00xxxx  best effort
        01xxxx data
        10xxxx media oriented
        11xxxx control

- With the other 4 bits, something should be done. My proposal:
        - If the third bit is "1", then the other three bits include a valu=
e
defining the "real-time-ness" of the flow. So we have 8 levels of
real-time-ness. Some examples:
                - 10-1-111 would be a  media-oriented (10) and very
real-time flow, as VoIP or a game
                - 10-0-xxx would be a media-oriented flow for which the
real-time-ness is not defined
                - 10-1-010 could be a media-oriented flow with medium delay
requirements, as multimedia conferencing
                - 10-1-000 would be a media-oriented flow with no very hard
delay requirements, as multimedia streaming
                - 00-xxxx would be best-effort
                -01-1-111 would be very low-latency data
                -01-1-000 would be low-priority data

One advantage of this is that some routers may only take into account the
first two bits, and others would use all of them.

The new RFC could suggest or define different values of real-time-ness for
each application. So the applications should be ordered depending of the
maximum tolerated delay.

Ok. This is only an idea. Perhaps it does not make sense. Just tell me.
Regards,

Jose


From Ruediger.Geib@telekom.de  Tue Apr 24 02:56:12 2012
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From: <Ruediger.Geib@telekom.de>
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Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
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David,

taking a formal point of view, you are probably right. These isues are
intertwined however.

End to end QoS will certainly not result from continuing to act like
almost all standards bodies did so far: as there's a lot flexibility
and little non-ambiguous guidance, anything is customised.

I don't regard the update of RFC 4594 as being helpful if the
expectation of the author is that even supporters of that standard
will only deploy a subset of the defined classes and sub classes. Those
standards close to deployment I know offer 3-4 classes and require no
more than 6 codepoints (the latter is personal judgement).

Any standards track QoS document should only determine things seeing
widespread deployment.

Some QoS classes have seen deployment. It's trickier with codepoints.

Anything which might be useful in future or has seen an informational
proposal in past  should be critically reviewed and remain
informational or experimental.

Any QoS class should be well defined any easy to be differentiated
against others.

QoS codepoints should not identify applications.

I also don't think that one needs to differentiate "admitted" from
"non-admitted" classes. In a traffic engineered DiffServ network,
customers get an SLA describing the worst case QoS class performance.
To me it doesn't matter how a carrier ensures that the SLAs are kept.
It only matters that they are kept.

Regards,

Ruediger


> - We have the old 8 TOS bits of the IP header. Nowadays, two of them are
> ECN, and six of them are the DSCP field. So the problem is how to use the=
se
> 6 bits (64 possible values)
> - Network control is 11xxxx, so this uses 16 values
>       -11xx11 are defined in RFC 2474, and remain as only experimentally
> assignable/useable (this uses 4 of these 16 values)
> - This leaves 48 values
> - Best effort is 000000
> - The draft of James proposes 14 values for the DSCP field, which corresp=
ond
> to service classes, mapped to application categories.
>
> But Ruediger says that, in his opinion, more than 4 classes are not usefu=
l,
> according to current implementations.

That's actually not appropriate for an RFC 4594 update - the proper place t=
o
address that requirement is an update to RFC 5127, which will need to be up=
dated
to be consistent with whatever is done to RFC 4594.

Thanks,
--David


> -----Original Message-----
> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of=
 Jose
> Saldana
> Sent: Saturday, April 07, 2012 8:03 AM
> To: Ruediger.Geib@telekom.de; James M. Polk
> Cc: tsvwg@ietf.org
> Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
>
> I have been reading the two messages of Ruediger about Jame's draft. I wi=
ll
> try to make a "status of the question" and then a proposal. If something =
is
> wrong, just tell me.
>
> - We have the old 8 TOS bits of the IP header. Nowadays, two of them are
> ECN, and six of them are the DSCP field. So the problem is how to use the=
se
> 6 bits (64 possible values)
> - Network control is 11xxxx, so this uses 16 values
>       -11xx11 are defined in RFC 2474, and remain as only experimentally
> assignable/useable (this uses 4 of these 16 values)
> - This leaves 48 values
> - Best effort is 000000
> - The draft of James proposes 14 values for the DSCP field, which corresp=
ond
> to service classes, mapped to application categories.
>
> But Ruediger says that, in his opinion, more than 4 classes are not usefu=
l,
> according to current implementations.
>
> We can ask ourselves: which are  the key parameters that define how packe=
ts
> have to be managed in a router? I think that they are:
>
> 1) "Real-time-ness": The tolerated delay of the service. The jitter can b=
e
> roughly included in this parameter.
> 2) loss tolerance: We can distinguish: TCP services are tolerant to loss,
> since they retransmit packets. And some UDP services are also tolerant to
> loss (VoIP, some games, etc)
>
> In my opinion, the "real-time-ness" is the key: it distinguishes very
> similar services: e.g. a video conference is different from a broadcast,
> since the latter permits delays of some seconds, whereas the first has to=
 be
> interactive.
>
> So one possible solution could be:
>
> - Using the first two bits in order to define 4 classes, taking into acco=
unt
> the existing implementations. They can be the 4 classes of Fig. 1 of Jame=
s'
> draft, or the ones proposed by Ruediger.
>       00xxxx  best effort
>       01xxxx data
>       10xxxx media oriented
>       11xxxx control
>
> - With the other 4 bits, something should be done. My proposal:
>       - If the third bit is "1", then the other three bits include a valu=
e
> defining the "real-time-ness" of the flow. So we have 8 levels of
> real-time-ness. Some examples:
>               - 10-1-111 would be a  media-oriented (10) and very
> real-time flow, as VoIP or a game
>               - 10-0-xxx would be a media-oriented flow for which the
> real-time-ness is not defined
>               - 10-1-010 could be a media-oriented flow with medium delay
> requirements, as multimedia conferencing
>               - 10-1-000 would be a media-oriented flow with no very hard
> delay requirements, as multimedia streaming
>               - 00-xxxx would be best-effort
>               -01-1-111 would be very low-latency data
>               -01-1-000 would be low-priority data
>
> One advantage of this is that some routers may only take into account the
> first two bits, and others would use all of them.
>
> The new RFC could suggest or define different values of real-time-ness fo=
r
> each application. So the applications should be ordered depending of the
> maximum tolerated delay.
>
> Ok. This is only an idea. Perhaps it does not make sense. Just tell me.
> Regards,
>
> Jose
>
> -----Mensaje original-----
> De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]
> Enviado el: jueves, 05 de abril de 2012 12:23
> Para: jsaldana@unizar.es
> CC: tsvwg@ietf.org
> Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
>
> Jose,
>
> regarding QoS, I'm happy to agree on 4 standard classes with properties a=
s
> mentioned in my last email. If there's consensus on some codepoints for
> them, I'll support that.
>
> If IETF recommends how and which applications to use these
> classes/codepoints, that's again agreeable. I don't think, that by now
> requiring a specific application to carry a specific DSCP makes sense, ap=
art
> from few exceptions (like VoIP payload, network management traffic or Bes=
t
> Effort traffic).
> IETF needs consensus to do that.
> Note that the number of AF classes and codepoints are troublesome. Too ma=
ny
> with too little explanation how to use them, nothing is aligned with user
> demand. EF/VoIP payload and network management are examples how to procee=
d:
> just specify what's required, no extras.
>
>
> Then regarding tunnels and header compression Yes, that's smart and there
> may be more use cases than gaming. I'm a backbone engineer however and yo=
ur
> proposal is rather application specific. What I can do is ask some
> colleagues on their view. But there's no guarantee, that they will respon=
d.
>
> Some initial thoughts:
> To me a key point seems to be what can be reached at the bottleneck. Ther=
e
> the compression should save bandwidth with little cost and no negative
> service impact. Any kind of shared access from which traffic is transmitt=
ed
> to the same destination seems to be relevant. The application/tunnel
> awareness of network elements may not be feasible, a network terminal at =
a
> UNI may be able to handle that, or any service specific device in a netwo=
rk.
>
> Regards,
>
> Ruediger
>
>
> -----Original Message-----
> From: Jose Saldana [mailto:jsaldana@unizar.es]
> Sent: Thursday, April 05, 2012 11:29 AM
> To: Geib, R=FCdiger
> Cc: tsvwg@ietf.org
> Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
>
> Ruediger,
>
> I have read your message about draft-polk-tsvwg-rfc4594-update. As you ma=
y
> know, I presented another draft in Paris (draft-saldana-tsvwg-tcmtf-02), =
and
> after talking to James, I thought that the idea of standardizing the traf=
fic
> classification fits well with my proposal. I would like to know your opin=
ion
> about it, if possible.
>
> The idea is to use tunnels in order to compress and multiplex real-time
> traffic flows. Since these flows tend to use very small packets, they hav=
e
> very bad efficiency. So perhaps tunnels could be used in order to group
> these flows. The drawback is that they add small delays, i.e. the waiting
> time in the multiplexer. So another use is to provide flexibility: if you
> are a game provider, you may over provision your network in order to be a=
ble
> to manage traffic peaks. But perhaps using this, you may be able to have =
a
> smaller network, and use compression when necessary (rush hour, first day=
s
> after releasing a new game, etc). When traffic is small, you do not
> multiplex, in order to reduce latency.
>
> So if traffic classification is a standard, everyone in the way from the
> origin to the destination will handle the traffic with the correct priori=
ty.
>
> I attach a figure in order to clarify.
>
> Thank you,
>
> Jose
>
> PS: In
> http://community.virginmedia.com/t5/Fibre-optic-broadband-cable/World-of-=
War
> craft-high-latency-issues-with-virgin-media-users/td-p/44498 you can find
> the history of the problems of World of Warcraft for Virgin Media users. =
The
> problem was caused by a modification in the traffic, and the provider did
> not notice about it, so it treated the packets as best-effort.
>
> -----Mensaje original-----
> De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de] Enviado el=
:
> jueves, 05 de abril de 2012 9:49
> Para: jsaldana@unizar.es
> CC: tsvwg@ietf.org
> Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
>
> Jose,
>
> you raise an interesting point:
>
> draft-polk-tsvwg-rfc4594-update suggests to standardise application to
> codepoint mapping as well as codepoint to class properties. You observe t=
hat
> some games are using TCP while others use UDP, which I think is a useful
> level of QoS requirement abstraction. There's no need to have like 25+ Qo=
S
> (sub)classes, I think. But one class optimised to transport  "UDP" and on=
e
> for "TCP" are agreeable.
>
> The feedback that I had on carrier class QoS usage is that there's a poin=
t
> in standardising class properties, and certainly not to standardise
> application mappings to classes (recommending them may however be useful)=
.
> My take would be to standardise four classes/properties (in brackets exam=
ple
> usage):
>
> a) for traffic which doesn't support re-transmits, requiring lowest delay
> and jitter, having limited volume and packet
>    size, but being loss tolerant (example usage VoIP)
> b) for any other traffic which doesn't support re-transmits (let's say
> traffic with real time like properties, but any
>    packet size and possibly high volume, UDP transport, like IPTV)
> c) for any other traffic which supports re-transmits (QoS traffic using T=
CP,
> like application signalling and network management traffic)
> d) a default class, for any traffic supporting re-transmits (Best Effort)=
 -
> this class may not be regarded as QoS (or as lowest CoS)
>
> What I can tell ist that many carriers and standards on QoS support 3-4
> classes, having properties as described above. It is useful to recommend
> codepoints matching to these classes. Standardising these codepoints
> requires agreement between those having deplyed them. Straightforward for=
 a)
> EF and d) 000000, probably easy for c) being used to transport Network
> Management (CS 6). I'm not aware of common ground for anything else. My
> favourites are:
>
> b) AF4 (GSMA supports this one, RFC4594 also)
> c) AF3 (MEF and GSMA support this one)
>
> Gaming from my point of view would then match to an AF3 or AF4 class, thi=
s
> may depend on the implemented application transport. Further details may =
be
> discussed.
>
> Everything else could be reserved, but shouldn't be recommended. Carriers
> have deployed differing schemes on codepoints and classes. They need codi=
ng
> reserverves in 3 bit space (MPLS TC, IP Precedence and Ethernet P-Bits) a=
s
> well as in DSCP space if they should converge to a standardised
> codepoint/class scheme.
>
> I object to any approach standardising a match of classes and application=
s
> with codepoints as a requirement and I also object to standards leaving n=
o
> spare codepoints in 3 bit or 6 bit QoS codepoint space (including 3 bit I=
P
> precedence).
>
> Bob Briscoe mentioned the difference between classes and queues and above
> properties can be mapped to queues. If end to end QoS is desired, the sam=
e
> class properties must be negotiated end to end. The lower the number of
> classes and codepoints being deployed, the higher the probability of gett=
ing
> end to end QoS. I don't support an approach requiring 8 three bit classes
> and 25 or more 6 bit classes and codepoints.
>
> Regards,
>
> Ruediger
>
> ________________________________
>
> From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of
> Jose Saldana
> Sent: Monday, April 02, 2012 12:17 PM
> To: James Polk
> Cc: tsvwg@ietf.org
> Subject: About draft-polk-tsvwg-rfc4594-update-00.txt
>
>
>
> James,
>
>
>
> After returning home from IETF 83, I have been reading your draft. I find=
 it
> interesting to standardize the service classes. Otherwise, if each one
> classifies each traffic according to different criteria, Diffserv becomes
> less useful.
>
>
>
> Regarding "Real-time interactive" service class, which refers to online
> gaming (my main research field), I find it really interesting: online gam=
es
> are at least as real-time as VoIP is. In most cases, the gamers are a ver=
y
> difficult people to deal with. If the latency is high, they leave and nev=
er
> return (this statement is not mine, it can be corroborated in the referen=
ces
> I put below). This has been studied by many research groups, with empiric=
al
> tests using real gamers and adding delay, jitter, packet loss.
>
>
>
> Other question: in the draft it is said that online games use RTP/UDP
> streams. This can be a little confusing. In fact, the vast majority of
> online games do not use RTP, but bare UDP or even TCP. In
> http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.pdf, the
> authors proposed the use of an adaptation of RTP in order to manage gamin=
g
> traffic. But as far as I know, this approach is not used by commercial
> games.
>
>
>
>
>
> A typical classification of today's games is:
>
>
>
> -          FPS (First Person Shooters) e.g. Counter Strike, Halo, Call of
> Duty: They use UDP
>
> -          MMORPG (Massively Multiplayer Online Role-playing Games), e.g.
> World of Warcraft: They normally use TCP, even though some of them use UD=
P
>
> -          RTS (Real-time strategy) e.g. Age of Empires: They may use TCP=
 or
> UDP
>
> -          Sports games: Since there is a big variety, they have not been
> deeply studied in the literature
>
>
>
> Some research papers supporting this classification, and also the real-ti=
me
> requirements of the games:
>
>
>
> First Person Shooters
>
>
>
> -          A survey of traffic models for 17 First Person Shooters can be
> found here. All of them use UDP
> http://www.discover.uottawa.ca/publications/files/MMOG-Survey_IEEE-IC.pdf
>
>
>
> -          In this paper from my research group, 8 FPS games were analyze=
d
> in order to compress and multiplex their traffic. Figure 5 shows their
> histograms (inter-packet time and packet size). It can be seen that they
> generate real-time flows, with characteristics very similar to VoIP (pack=
ets
> of 60-80 bytes every 15-40 ms):
> http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana.pdf
>
>
>
> -          Another paper about FPS using UDP. Section V presents some
> interesting conclusions about routing infrastructure.
> http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf
>
>
>
> -          In this paper the network factors annoying the gamers are
> analyzed. There are also some calculations of the MOS. Above 150 ms the
> quality is bad (something similar to what happens with VoIP)
> http://www.research.ibm.com/netgames2005/papers/dick.pdf
>
>
>
> -          In this one, a MOS model for a FPS (Quake IV) is deployed. It =
is
> only based on delay and jitter, since packet loss has vevy little influen=
ce
> (for this concrete game). Delay above 140 ms produces MOS below 3
> http://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf
>
>
>
> -          In this paper, it is found that some FPS work properly even wi=
th
> 30% packet loss, whereas others stop working with 4%. RTT above 300 ms (1=
50
> OWD) makes MOS go above 3
> http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pdf
>
>
>
> MMORPGs:
>
> -          A traffic model for World of Warcraft:
> http://publik.tuwien.ac.at/files/pub-et_12119.pdf
>
>
>
> -          This one presents a MOS for World of Warcraft. When added dela=
y
> is above 160ms, the MOS is roughly 3.:
> http://publik.tuwien.ac.at/files/PubDat_170202.pdf
>
>
>
> Best regards,
>
>
>
> Jose Saldana
>
>


From trammell@tik.ee.ethz.ch  Tue Apr 24 04:35:21 2012
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Greetings, all,

Apologies for missing the WGLC end on this draft; however as it appears =
it may be rev'd anyway on the MTU DF question, I hope the following =
comments prove useful.

I've reviewed the document. It fills a need and the approach makes =
sense, so I'm in favor of its publication. I do have a couple of =
suggestions for clarification, though none of these change the normative =
content of the document. I'll state that my perspective here is as an =
SCTP application developer, and as someone interested both in this draft =
and draft-tuexen-tsvwg-sctp-dtls-encaps within the context of IPFIX over =
SCTP. (I'll also state that I didn't follow the WG discussion on this =
draft to date, as I only sporadically pay attention to tsvwg, so =
apologies if some of these comments are old news.)

I'm still a little confused about how exactly UDP port mappings to SCTP =
associations are meant to work in practice. Clearly, there is one UDP =
port per "stack": one per application with application-bound =
SCTP-over-UDP and one per host, defaulting to 9989, with SCTP-over-UDP =
in the kernel or via a (probably privileged) userspace SCTP-over-UDP =
daemon. This point might be made more clearly in section 4.1 -- that =
there are really two envisioned deployment scenarios (or three, =
depending on how you count), and explicitly spelling out the recommended =
port numbering for each.

(This opens an issue at the user interface level, though I am reasonably =
certain this is out of scope for this document. Now you potentially have =
one port that is exposed to the user or inherent to the application (the =
SCTP port), plus one port that may have to be exposed to the client or =
not based on the deployment scenario in use at the server. There are =
ways for dealing with this with a single port in user- or =
administrator-exposed names (URIs, DNS SRV...) but I don't know how well =
this works for two ports.=20
Again, I'm not sure it fits in this document at all, but some discussion =
of this issue (and suggested solutions) somewhere would keep each =
application developer or specifier of application protocols from having =
to solve it themselves. This issue would appear to apply to =
draft-tuexen-tsvwg-sctp-dtls-encaps as well.)

There are a couple of other bits that were terse to the point of being a =
little unclear on a first reading. In section 4.2 it may be helpful to =
point out that the point of UDP checksumming is to reduce the chance of =
UDP header corruption. (I had to think about this for a couple of =
seconds; SCTP has its own reliability mechanisms, after all...) =
Likewise, section 4.8 might benefit from a note that you're talking =
about ECN bits in the IP header, and that ECN over SCTP works otherwise =
as specified in draft-stewart-tsvwg-sctpecn (with an informative =
reference).

Best regards,

Brian=

From Michael.Tuexen@lurchi.franken.de  Tue Apr 24 12:33:10 2012
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Hi Brian,

see my comments in-line.

Best regards
Michael

On Apr 24, 2012, at 1:35 PM, Brian Trammell wrote:

> Greetings, all,
>=20
> Apologies for missing the WGLC end on this draft; however as it =
appears it may be rev'd anyway on the MTU DF question, I hope the =
following comments prove useful.
>=20
> I've reviewed the document. It fills a need and the approach makes =
sense, so I'm in favor of its publication. I do have a couple of =
suggestions for clarification, though none of these change the normative =
content of the document. I'll state that my perspective here is as an =
SCTP application developer, and as someone interested both in this draft =
and draft-tuexen-tsvwg-sctp-dtls-encaps within the context of IPFIX over =
SCTP. (I'll also state that I didn't follow the WG discussion on this =
draft to date, as I only sporadically pay attention to tsvwg, so =
apologies if some of these comments are old news.)
Why do you prefer IPFIX over SCTP over DTLS over UDP compared to IPFIX =
over DTLS over SCTP over UDP?
>=20
> I'm still a little confused about how exactly UDP port mappings to =
SCTP associations are meant to work in practice. Clearly, there is one =
UDP port per "stack": one per application with application-bound =
SCTP-over-UDP and one per host, defaulting to 9989, with SCTP-over-UDP =
in the kernel or via a (probably privileged) userspace SCTP-over-UDP =
daemon. This point might be made more clearly in section 4.1 -- that =
there are really two envisioned deployment scenarios (or three, =
depending on how you count), and explicitly spelling out the recommended =
port numbering for each.
Each SCTP stack running in userland will use its own UDP port number for =
receiving the packets.
Each stack will then use the SCTP port number for =
multiplexing/demultiplexing the packets
belonging to different associations.
>=20
> (This opens an issue at the user interface level, though I am =
reasonably certain this is out of scope for this document. Now you =
potentially have one port that is exposed to the user or inherent to the =
application (the SCTP port), plus one port that may have to be exposed =
to the client or not based on the deployment scenario in use at the =
server. There are ways for dealing with this with a single port in user- =
or administrator-exposed names (URIs, DNS SRV...) but I don't know how =
well this works for two ports.=20
> Again, I'm not sure it fits in this document at all, but some =
discussion of this issue (and suggested solutions) somewhere would keep =
each application developer or specifier of application protocols from =
having to solve it themselves. This issue would appear to apply to =
draft-tuexen-tsvwg-sctp-dtls-encaps as well.)
I guess this is not within the scope of the document.
>=20
> There are a couple of other bits that were terse to the point of being =
a little unclear on a first reading. In section 4.2 it may be helpful to =
point out that the point of UDP checksumming is to reduce the chance of =
UDP header corruption. (I had
And UDP checksum is required for IPv6... However, the MUST will be =
changed to a SHOULD. On FreeBSD we
will just use the system wide setting for UDP/IPv4.
> to think about this for a couple of seconds; SCTP has its own =
reliability mechanisms, after all...) Likewise, section 4.8 might =
benefit from a note that you're talking about ECN bits in the IP header, =
and that ECN over SCTP works otherwise as specified in =
draft-stewart-tsvwg-sctpecn (with an informative reference).
I didn't wanted to add another reference. Just wanted to make sure that =
encaps/decaps should try to
transfer the ECN stuff. However, I have to double check if that is =
possible with the socket API on
plattforms like FreeBSD, Linux, Mac OS X using the socket API and =
Windows using whatever they provide.

Thanks a lot for the comments!

Best regards
Michael
>=20
> Best regards,
>=20
> Brian


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From: <david.black@emc.com>
To: <Ruediger.Geib@telekom.de>
Date: Wed, 25 Apr 2012 20:17:21 -0400
Thread-Topic: About draft-polk-tsvwg-rfc4594-update-00.txt
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Subject: Re: [tsvwg] About draft-polk-tsvwg-rfc4594-update-00.txt
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Ruediger,

We may be disagreeing more about means than ends.

RFC 5127's intent was to aggregate all of the classes defined in RFC 4594
into a much smaller number of classes that would be suitable for a core
network,  e.g., "3-4 classes and require no more than 6 codepoints".  The
net would be that a choice of considerably more traffic classes from RFC
4594 would be deployable via aggregation as a suitably small number of
classes on the wire.

I believe (and I've said so to James), that RFC 5127 needs to be revised
in concert with any work on RFC 4594 (in particular, I strongly oppose any
revision to 4594 that ignores 5127).  I also think that RFC 5127 is a
better starting point to address the small number of classes use case
that appears to be of interest and concern to you.

Thanks,
--David

> -----Original Message-----
> From: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]
> Sent: Tuesday, April 24, 2012 5:56 AM
> To: Black, David
> Cc: tsvwg@ietf.org
> Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
>
> David,
>
> taking a formal point of view, you are probably right. These isues are
> intertwined however.
>
> End to end QoS will certainly not result from continuing to act like
> almost all standards bodies did so far: as there's a lot flexibility
> and little non-ambiguous guidance, anything is customised.
>
> I don't regard the update of RFC 4594 as being helpful if the
> expectation of the author is that even supporters of that standard
> will only deploy a subset of the defined classes and sub classes. Those
> standards close to deployment I know offer 3-4 classes and require no
> more than 6 codepoints (the latter is personal judgement).
>
> Any standards track QoS document should only determine things seeing
> widespread deployment.
>
> Some QoS classes have seen deployment. It's trickier with codepoints.
>
> Anything which might be useful in future or has seen an informational
> proposal in past  should be critically reviewed and remain
> informational or experimental.
>
> Any QoS class should be well defined any easy to be differentiated
> against others.
>
> QoS codepoints should not identify applications.
>
> I also don't think that one needs to differentiate "admitted" from
> "non-admitted" classes. In a traffic engineered DiffServ network,
> customers get an SLA describing the worst case QoS class performance.
> To me it doesn't matter how a carrier ensures that the SLAs are kept.
> It only matters that they are kept.
>
> Regards,
>
> Ruediger
>
>
> > - We have the old 8 TOS bits of the IP header. Nowadays, two of them ar=
e
> > ECN, and six of them are the DSCP field. So the problem is how to use t=
hese
> > 6 bits (64 possible values)
> > - Network control is 11xxxx, so this uses 16 values
> >       -11xx11 are defined in RFC 2474, and remain as only experimentall=
y
> > assignable/useable (this uses 4 of these 16 values)
> > - This leaves 48 values
> > - Best effort is 000000
> > - The draft of James proposes 14 values for the DSCP field, which corre=
spond
> > to service classes, mapped to application categories.
> >
> > But Ruediger says that, in his opinion, more than 4 classes are not use=
ful,
> > according to current implementations.
>
> That's actually not appropriate for an RFC 4594 update - the proper place=
 to
> address that requirement is an update to RFC 5127, which will need to be
> updated
> to be consistent with whatever is done to RFC 4594.
>
> Thanks,
> --David
>
>
> > -----Original Message-----
> > From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf =
Of
> Jose
> > Saldana
> > Sent: Saturday, April 07, 2012 8:03 AM
> > To: Ruediger.Geib@telekom.de; James M. Polk
> > Cc: tsvwg@ietf.org
> > Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> >
> > I have been reading the two messages of Ruediger about Jame's draft. I =
will
> > try to make a "status of the question" and then a proposal. If somethin=
g is
> > wrong, just tell me.
> >
> > - We have the old 8 TOS bits of the IP header. Nowadays, two of them ar=
e
> > ECN, and six of them are the DSCP field. So the problem is how to use t=
hese
> > 6 bits (64 possible values)
> > - Network control is 11xxxx, so this uses 16 values
> >       -11xx11 are defined in RFC 2474, and remain as only experimentall=
y
> > assignable/useable (this uses 4 of these 16 values)
> > - This leaves 48 values
> > - Best effort is 000000
> > - The draft of James proposes 14 values for the DSCP field, which corre=
spond
> > to service classes, mapped to application categories.
> >
> > But Ruediger says that, in his opinion, more than 4 classes are not use=
ful,
> > according to current implementations.
> >
> > We can ask ourselves: which are  the key parameters that define how pac=
kets
> > have to be managed in a router? I think that they are:
> >
> > 1) "Real-time-ness": The tolerated delay of the service. The jitter can=
 be
> > roughly included in this parameter.
> > 2) loss tolerance: We can distinguish: TCP services are tolerant to los=
s,
> > since they retransmit packets. And some UDP services are also tolerant =
to
> > loss (VoIP, some games, etc)
> >
> > In my opinion, the "real-time-ness" is the key: it distinguishes very
> > similar services: e.g. a video conference is different from a broadcast=
,
> > since the latter permits delays of some seconds, whereas the first has =
to be
> > interactive.
> >
> > So one possible solution could be:
> >
> > - Using the first two bits in order to define 4 classes, taking into ac=
count
> > the existing implementations. They can be the 4 classes of Fig. 1 of Ja=
mes'
> > draft, or the ones proposed by Ruediger.
> >       00xxxx  best effort
> >       01xxxx data
> >       10xxxx media oriented
> >       11xxxx control
> >
> > - With the other 4 bits, something should be done. My proposal:
> >       - If the third bit is "1", then the other three bits include a va=
lue
> > defining the "real-time-ness" of the flow. So we have 8 levels of
> > real-time-ness. Some examples:
> >               - 10-1-111 would be a  media-oriented (10) and very
> > real-time flow, as VoIP or a game
> >               - 10-0-xxx would be a media-oriented flow for which the
> > real-time-ness is not defined
> >               - 10-1-010 could be a media-oriented flow with medium del=
ay
> > requirements, as multimedia conferencing
> >               - 10-1-000 would be a media-oriented flow with no very ha=
rd
> > delay requirements, as multimedia streaming
> >               - 00-xxxx would be best-effort
> >               -01-1-111 would be very low-latency data
> >               -01-1-000 would be low-priority data
> >
> > One advantage of this is that some routers may only take into account t=
he
> > first two bits, and others would use all of them.
> >
> > The new RFC could suggest or define different values of real-time-ness =
for
> > each application. So the applications should be ordered depending of th=
e
> > maximum tolerated delay.
> >
> > Ok. This is only an idea. Perhaps it does not make sense. Just tell me.
> > Regards,
> >
> > Jose
> >
> > -----Mensaje original-----
> > De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]
> > Enviado el: jueves, 05 de abril de 2012 12:23
> > Para: jsaldana@unizar.es
> > CC: tsvwg@ietf.org
> > Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> >
> > Jose,
> >
> > regarding QoS, I'm happy to agree on 4 standard classes with properties=
 as
> > mentioned in my last email. If there's consensus on some codepoints for
> > them, I'll support that.
> >
> > If IETF recommends how and which applications to use these
> > classes/codepoints, that's again agreeable. I don't think, that by now
> > requiring a specific application to carry a specific DSCP makes sense, =
apart
> > from few exceptions (like VoIP payload, network management traffic or B=
est
> > Effort traffic).
> > IETF needs consensus to do that.
> > Note that the number of AF classes and codepoints are troublesome. Too =
many
> > with too little explanation how to use them, nothing is aligned with us=
er
> > demand. EF/VoIP payload and network management are examples how to proc=
eed:
> > just specify what's required, no extras.
> >
> >
> > Then regarding tunnels and header compression Yes, that's smart and the=
re
> > may be more use cases than gaming. I'm a backbone engineer however and =
your
> > proposal is rather application specific. What I can do is ask some
> > colleagues on their view. But there's no guarantee, that they will resp=
ond.
> >
> > Some initial thoughts:
> > To me a key point seems to be what can be reached at the bottleneck. Th=
ere
> > the compression should save bandwidth with little cost and no negative
> > service impact. Any kind of shared access from which traffic is transmi=
tted
> > to the same destination seems to be relevant. The application/tunnel
> > awareness of network elements may not be feasible, a network terminal a=
t a
> > UNI may be able to handle that, or any service specific device in a net=
work.
> >
> > Regards,
> >
> > Ruediger
> >
> >
> > -----Original Message-----
> > From: Jose Saldana [mailto:jsaldana@unizar.es]
> > Sent: Thursday, April 05, 2012 11:29 AM
> > To: Geib, R=FCdiger
> > Cc: tsvwg@ietf.org
> > Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> >
> > Ruediger,
> >
> > I have read your message about draft-polk-tsvwg-rfc4594-update. As you =
may
> > know, I presented another draft in Paris (draft-saldana-tsvwg-tcmtf-02)=
, and
> > after talking to James, I thought that the idea of standardizing the tr=
affic
> > classification fits well with my proposal. I would like to know your op=
inion
> > about it, if possible.
> >
> > The idea is to use tunnels in order to compress and multiplex real-time
> > traffic flows. Since these flows tend to use very small packets, they h=
ave
> > very bad efficiency. So perhaps tunnels could be used in order to group
> > these flows. The drawback is that they add small delays, i.e. the waiti=
ng
> > time in the multiplexer. So another use is to provide flexibility: if y=
ou
> > are a game provider, you may over provision your network in order to be=
 able
> > to manage traffic peaks. But perhaps using this, you may be able to hav=
e a
> > smaller network, and use compression when necessary (rush hour, first d=
ays
> > after releasing a new game, etc). When traffic is small, you do not
> > multiplex, in order to reduce latency.
> >
> > So if traffic classification is a standard, everyone in the way from th=
e
> > origin to the destination will handle the traffic with the correct prio=
rity.
> >
> > I attach a figure in order to clarify.
> >
> > Thank you,
> >
> > Jose
> >
> > PS: In
> > http://community.virginmedia.com/t5/Fibre-optic-broadband-cable/World-o=
f-War
> > craft-high-latency-issues-with-virgin-media-users/td-p/44498 you can fi=
nd
> > the history of the problems of World of Warcraft for Virgin Media users=
. The
> > problem was caused by a modification in the traffic, and the provider d=
id
> > not notice about it, so it treated the packets as best-effort.
> >
> > -----Mensaje original-----
> > De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de] Enviado =
el:
> > jueves, 05 de abril de 2012 9:49
> > Para: jsaldana@unizar.es
> > CC: tsvwg@ietf.org
> > Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> >
> > Jose,
> >
> > you raise an interesting point:
> >
> > draft-polk-tsvwg-rfc4594-update suggests to standardise application to
> > codepoint mapping as well as codepoint to class properties. You observe=
 that
> > some games are using TCP while others use UDP, which I think is a usefu=
l
> > level of QoS requirement abstraction. There's no need to have like 25+ =
QoS
> > (sub)classes, I think. But one class optimised to transport  "UDP" and =
one
> > for "TCP" are agreeable.
> >
> > The feedback that I had on carrier class QoS usage is that there's a po=
int
> > in standardising class properties, and certainly not to standardise
> > application mappings to classes (recommending them may however be usefu=
l).
> > My take would be to standardise four classes/properties (in brackets ex=
ample
> > usage):
> >
> > a) for traffic which doesn't support re-transmits, requiring lowest del=
ay
> > and jitter, having limited volume and packet
> >    size, but being loss tolerant (example usage VoIP)
> > b) for any other traffic which doesn't support re-transmits (let's say
> > traffic with real time like properties, but any
> >    packet size and possibly high volume, UDP transport, like IPTV)
> > c) for any other traffic which supports re-transmits (QoS traffic using=
 TCP,
> > like application signalling and network management traffic)
> > d) a default class, for any traffic supporting re-transmits (Best Effor=
t) -
> > this class may not be regarded as QoS (or as lowest CoS)
> >
> > What I can tell ist that many carriers and standards on QoS support 3-4
> > classes, having properties as described above. It is useful to recommen=
d
> > codepoints matching to these classes. Standardising these codepoints
> > requires agreement between those having deplyed them. Straightforward f=
or a)
> > EF and d) 000000, probably easy for c) being used to transport Network
> > Management (CS 6). I'm not aware of common ground for anything else. My
> > favourites are:
> >
> > b) AF4 (GSMA supports this one, RFC4594 also)
> > c) AF3 (MEF and GSMA support this one)
> >
> > Gaming from my point of view would then match to an AF3 or AF4 class, t=
his
> > may depend on the implemented application transport. Further details ma=
y be
> > discussed.
> >
> > Everything else could be reserved, but shouldn't be recommended. Carrie=
rs
> > have deployed differing schemes on codepoints and classes. They need co=
ding
> > reserverves in 3 bit space (MPLS TC, IP Precedence and Ethernet P-Bits)=
 as
> > well as in DSCP space if they should converge to a standardised
> > codepoint/class scheme.
> >
> > I object to any approach standardising a match of classes and applicati=
ons
> > with codepoints as a requirement and I also object to standards leaving=
 no
> > spare codepoints in 3 bit or 6 bit QoS codepoint space (including 3 bit=
 IP
> > precedence).
> >
> > Bob Briscoe mentioned the difference between classes and queues and abo=
ve
> > properties can be mapped to queues. If end to end QoS is desired, the s=
ame
> > class properties must be negotiated end to end. The lower the number of
> > classes and codepoints being deployed, the higher the probability of ge=
tting
> > end to end QoS. I don't support an approach requiring 8 three bit class=
es
> > and 25 or more 6 bit classes and codepoints.
> >
> > Regards,
> >
> > Ruediger
> >
> > ________________________________
> >
> > From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf =
Of
> > Jose Saldana
> > Sent: Monday, April 02, 2012 12:17 PM
> > To: James Polk
> > Cc: tsvwg@ietf.org
> > Subject: About draft-polk-tsvwg-rfc4594-update-00.txt
> >
> >
> >
> > James,
> >
> >
> >
> > After returning home from IETF 83, I have been reading your draft. I fi=
nd it
> > interesting to standardize the service classes. Otherwise, if each one
> > classifies each traffic according to different criteria, Diffserv becom=
es
> > less useful.
> >
> >
> >
> > Regarding "Real-time interactive" service class, which refers to online
> > gaming (my main research field), I find it really interesting: online g=
ames
> > are at least as real-time as VoIP is. In most cases, the gamers are a v=
ery
> > difficult people to deal with. If the latency is high, they leave and n=
ever
> > return (this statement is not mine, it can be corroborated in the refer=
ences
> > I put below). This has been studied by many research groups, with empir=
ical
> > tests using real gamers and adding delay, jitter, packet loss.
> >
> >
> >
> > Other question: in the draft it is said that online games use RTP/UDP
> > streams. This can be a little confusing. In fact, the vast majority of
> > online games do not use RTP, but bare UDP or even TCP. In
> > http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.pdf, t=
he
> > authors proposed the use of an adaptation of RTP in order to manage gam=
ing
> > traffic. But as far as I know, this approach is not used by commercial
> > games.
> >
> >
> >
> >
> >
> > A typical classification of today's games is:
> >
> >
> >
> > -          FPS (First Person Shooters) e.g. Counter Strike, Halo, Call =
of
> > Duty: They use UDP
> >
> > -          MMORPG (Massively Multiplayer Online Role-playing Games), e.=
g.
> > World of Warcraft: They normally use TCP, even though some of them use =
UDP
> >
> > -          RTS (Real-time strategy) e.g. Age of Empires: They may use T=
CP or
> > UDP
> >
> > -          Sports games: Since there is a big variety, they have not be=
en
> > deeply studied in the literature
> >
> >
> >
> > Some research papers supporting this classification, and also the real-=
time
> > requirements of the games:
> >
> >
> >
> > First Person Shooters
> >
> >
> >
> > -          A survey of traffic models for 17 First Person Shooters can =
be
> > found here. All of them use UDP
> > http://www.discover.uottawa.ca/publications/files/MMOG-Survey_IEEE-IC.p=
df
> >
> >
> >
> > -          In this paper from my research group, 8 FPS games were analy=
zed
> > in order to compress and multiplex their traffic. Figure 5 shows their
> > histograms (inter-packet time and packet size). It can be seen that the=
y
> > generate real-time flows, with characteristics very similar to VoIP (pa=
ckets
> > of 60-80 bytes every 15-40 ms):
> > http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana.pdf
> >
> >
> >
> > -          Another paper about FPS using UDP. Section V presents some
> > interesting conclusions about routing infrastructure.
> > http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf
> >
> >
> >
> > -          In this paper the network factors annoying the gamers are
> > analyzed. There are also some calculations of the MOS. Above 150 ms the
> > quality is bad (something similar to what happens with VoIP)
> > http://www.research.ibm.com/netgames2005/papers/dick.pdf
> >
> >
> >
> > -          In this one, a MOS model for a FPS (Quake IV) is deployed. I=
t is
> > only based on delay and jitter, since packet loss has vevy little influ=
ence
> > (for this concrete game). Delay above 140 ms produces MOS below 3
> > http://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf
> >
> >
> >
> > -          In this paper, it is found that some FPS work properly even =
with
> > 30% packet loss, whereas others stop working with 4%. RTT above 300 ms =
(150
> > OWD) makes MOS go above 3
> > http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pdf
> >
> >
> >
> > MMORPGs:
> >
> > -          A traffic model for World of Warcraft:
> > http://publik.tuwien.ac.at/files/pub-et_12119.pdf
> >
> >
> >
> > -          This one presents a MOS for World of Warcraft. When added de=
lay
> > is above 160ms, the MOS is roughly 3.:
> > http://publik.tuwien.ac.at/files/PubDat_170202.pdf
> >
> >
> >
> > Best regards,
> >
> >
> >
> > Jose Saldana
> >
> >
>


From gorry@erg.abdn.ac.uk  Wed Apr 25 23:53:57 2012
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Subject: [tsvwg] 6man WGLC of draft-ietf-6man-udpchecksums-02.txt to end May 3, 2012
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TSVWG... please note the following WGLC in the 6man WG to update the UDP 
spec when used with IPv6. If you have transport comments or think this 
update is OK, please note this on the 6man or tsvwg list.

Gorry
<tsvwg co-chair>

----

This message starts a two week 6MAN Working Group on advancing:

	Title           : UDP Checksums for Tunneled Packets
	Author(s)       : Marshall Eubanks
                           P.F. Chimento
	Filename        : draft-ietf-6man-udpchecksums-02.txt
	Pages           : 12
	Date            : 2012-03-12

         http://tools.ietf.org/html/draft-ietf-6man-udpchecksums-02

as Proposed Standard.  Substantive comments and statements of support 
for advancing this document should be directed to the mailing list. 
Editorial suggestions can be sent to the authors.  This last call will 
end on May 3, 2012.

Regards,
Ole Troan & Bob Hinden

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Subject: [tsvwg] 6man WGLC of draft-ietf-6man-udpzero-05.txt to end May 3, 2012
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TSVWG... please note the following WGLC in the 6man WG this I-D presents 
transport-layer issues for the proposed update the UDP spec with IPv6.

If you have transport comments or think this update is OK, please note 
this on the 6man or tsvwg list.

Gorry
<tsvwg co-chair>

----

This message starts a two week 6MAN Working Group on advancing:

         Title           : IPv6 UDP Checksum Considerations
	Author(s)       : Godred Fairhurst
                           Magnus Westerlund
	Filename        : draft-ietf-6man-udpzero-05.txt
	Pages           : 34
	Date            : 2011-12-23

         http://tools.ietf.org/html/draft-ietf-6man-udpzero-05

as an Informational RFC.  Substantive comments and statements of support 
for advancing this document should be directed to the mailing list. 
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end on May 3, 2012.

Regards,
Ole Troan & Bob Hinden

From jukka.manner@aalto.fi  Thu Apr 26 01:50:43 2012
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From: Manner Jukka <jukka.manner@aalto.fi>
To: John Leslie <john@jlc.net>
Thread-Topic: Note: WGLC Announcement for draft-ietf-tsvwg-byte-pkt-congest-07
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Subject: Re: [tsvwg] Note: WGLC Announcement for draft-ietf-tsvwg-byte-pkt-congest-07
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Hi John,

Could you give a concrete clear list of things, you would like to be change=
d?

Jukka

On Apr 19, 2012, at 4:38 PM, John Leslie wrote:

> Bob Briscoe <bob.briscoe@bt.com> wrote:
>>=20
>> I believe the disagreement is rooted in the idea that vendors hit the=20
>> design limits of equipment performance when they cannot process=20
>> packets any faster, whereas raw line transmission rates could be a=20
>> lot faster.
>=20
>   Actually, that's not it at all.
>=20
>>> When I previously skimmed this document, I believed it contained only
>>> common ground: that AQM should drop/mark packets without favoring small
>>> packets, and that to whatever extent packet size _is_ considered, that
>>> should be a transport-layer responsibility.
>>=20
>> Good - this is the main thing we wanted to ensure was said.
>>=20
>>> Alas, when I read it carefully this week, I find that Bob is actually
>>> saying that transport-layer should _only_ consider "congested bytes",
>>> not "congested packets".
>>=20
>> In fact, TFRC-SP wrestled with the possibility that the network, not=20
>> the transport, could weight its congestion signals relative to packet=20
>> size. But it didn't even entertain the possibility that congestion=20
>> might be packet-based not bit-based, so packet size wouldn't be an=20
>> issue at all.
>=20
>   Hopefully, we all can agree the network should not weigh congestion
> signals by packet size. But that doesn't mean some other layer needs
> to do so in any "ordinary" case.
>=20
>> byte-pkt-congest does at least entertain the possibility that=20
>> prevalent bit-congestion may be a wrong assumption and describes a=20
>> future where it might be wrong, but it also justifies fairly=20
>> carefully why it is a safe working assumption and it will continue to=20
>> be safe, based on advice from equipment designers. (Actually, in=20
>> reading the latest draft, I notice some of this argument has=20
>> disappeared in the editing and it now mainly justifies this position=20
>> by reference to RFC6077. I would be happy to bring some of that=20
>> removed text back in.)
>=20
>   I very much doubt that what we need here is to add such text.
>=20
>> But, I gather that wouldn't satisfy you, because you believe=20
>> packet-congestion is prevalent, so let's please have that technical=20
>> discussion.
>=20
>   Once again, that's not it at all.
>=20
>   My technical argument is it's really-hard to accurately estimate
> per-packet overhead when counting bits for bit-congestible node pairs;
> and that counting packets (which we can do accurately) works well in
> all cases where packet size within a flow is fairly constant.
>=20
>   This is basically what we do today; and I'm very resistant to any
> call to replace it with algorithms which tend to underestimate the
> overhead when counting payload bytes.
>=20
>>> In fact, Bob openly endorses replacing TCP congestion-control with an
>>> algorithm which calculates "fair-share bytes" and doesn't back off at a=
ll
>>> (even in the presence of 20% packet loss) unless you're already sending
>>> more than this "fair-share".
>=20
>   BTW, I was very surprised to find "fair-share" as a recommendation
> in a Bob Briscoe document, given his previous "dismantling a religion"
> documents.
>=20
>> I assume you're referring to bullet 1. under "What this advice means=20
>> for the case of TCP:" in S2.3.
>=20
>   Actually, that's not what I was referring to.
>=20
>> It says
>> "  1.  If two TCP flows with different packet sizes are required to run
>>       at equal bit rates under the same path conditions, this should be
>>       done by altering TCP (Section 4.2.2), not network equipment [...]
>> "
>=20
>   More likely, it was Section 4.2.2... I don't think this document
> needs a 4.2.2 at all. This statement would be stronger if we didn't try
> to describe possible modifications to TCP.
>=20
>> This doesn't say TCP should do this, it just says TCP is the place to=20
>> fiddle, *if* you want to equalise bit-rates.
>=20
>   And I didn't find it objectionble until I read the rest of the
> document carefully.
>=20
>> If that's not clear, I would be happy (and I hope Jukka would too) to=20
>> make it clearer that this recommendation is irrelevant if TCP flows=20
>> are *not* required to run at equal bit-rates (Personally I would like=20
>> to say that there should be no need for any equal-bit-rate=20
>> requirement - however, that *would* be going beyond our charter).
>=20
>   Maybe we should explore exactly what about our charter excludes such
> a statement?
>=20
>> How about this text after S2.3. bullet 1?
>>=20
>> "  For the avoidance of doubt, this is not a recommendation that TCP
>>   should be changed so that all TCP flows run at equal bit-rates.  It is=
 a
>>   recommendation that the transport protocol not the network is where an=
y
>>   attempt should be made to equalise bit-rates of competing flows, *if*
>>   that is ever considered an important requirement.
>> "
>=20
>   I preferred your short-form:
> "=20
> " If you want to equalize bit-rates between TCP flows, this should be
> " done within TCP, not by network equipment.
>=20
>>> The draft contains a number of things I _do_ support; and I'd be happy
>>> to support a considerably shorter draft which concentrates on the AQM
>>> recommendation, omitting any suggestions of modifying TCP congestion
>>> control.
>>=20
>> We could omit "What this means for TCP" altogether.
>=20
>   IMHO, the document would be stronger without it.
>=20
>> However, given people of the standing of Sally Floyd originally tried
>> to equalise bit-rates in the network, then moved to doing it in the
>> transport, I don't think it is wise to leave a vacuum.
>=20
>   I don't perceive any such vacuum. At most, we might need an informative
> reference to a Sally Floyd document.
>=20
> --
> John Leslie <john@jlc.net>


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Date: Thu, 26 Apr 2012 16:16:58 -0500
To: <david.black@emc.com>
From: "James M. Polk" <jmpolk@cisco.com>
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Cc: tsvwg@ietf.org
Subject: Re: [tsvwg] About draft-polk-tsvwg-rfc4594-update-00.txt
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David

I absolutely agree with you that these two=20
efforts (the current rewrite of RFC4594, and the=20
coming rewrite of RFC5127) need to be=20
synchronized to have either make proper sense. I=20
also agree that the rewrite of RFC5127 is where a=20
smaller, aggregated set of classes for core=20
networks. Ruediger mentioned his worry about=20
"folks viewing tables and just configuring based=20
on the tables". Clearly his concerns on that need to be addressed.

I'll work to get my initial version of the=20
rewrite of RFC5127 in before Vancouver.

James

At 07:17 PM 4/25/2012, david.black@emc.com wrote:
>Ruediger,
>
>We may be disagreeing more about means than ends.
>
>RFC 5127's intent was to aggregate all of the classes defined in RFC 4594
>into a much smaller number of classes that would be suitable for a core
>network,  e.g., "3-4 classes and require no more than 6 codepoints".  The
>net would be that a choice of considerably more traffic classes from RFC
>4594 would be deployable via aggregation as a suitably small number of
>classes on the wire.
>
>I believe (and I've said so to James), that RFC 5127 needs to be revised
>in concert with any work on RFC 4594 (in particular, I strongly oppose any
>revision to 4594 that ignores 5127).  I also think that RFC 5127 is a
>better starting point to address the small number of classes use case
>that appears to be of interest and concern to you.
>
>Thanks,
>--David
>
> > -----Original Message-----
> > From: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]
> > Sent: Tuesday, April 24, 2012 5:56 AM
> > To: Black, David
> > Cc: tsvwg@ietf.org
> > Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> >
> > David,
> >
> > taking a formal point of view, you are probably right. These isues are
> > intertwined however.
> >
> > End to end QoS will certainly not result from continuing to act like
> > almost all standards bodies did so far: as there's a lot flexibility
> > and little non-ambiguous guidance, anything is customised.
> >
> > I don't regard the update of RFC 4594 as being helpful if the
> > expectation of the author is that even supporters of that standard
> > will only deploy a subset of the defined classes and sub classes. Those
> > standards close to deployment I know offer 3-4 classes and require no
> > more than 6 codepoints (the latter is personal judgement).
> >
> > Any standards track QoS document should only determine things seeing
> > widespread deployment.
> >
> > Some QoS classes have seen deployment. It's trickier with codepoints.
> >
> > Anything which might be useful in future or has seen an informational
> > proposal in past  should be critically reviewed and remain
> > informational or experimental.
> >
> > Any QoS class should be well defined any easy to be differentiated
> > against others.
> >
> > QoS codepoints should not identify applications.
> >
> > I also don't think that one needs to differentiate "admitted" from
> > "non-admitted" classes. In a traffic engineered DiffServ network,
> > customers get an SLA describing the worst case QoS class performance.
> > To me it doesn't matter how a carrier ensures that the SLAs are kept.
> > It only matters that they are kept.
> >
> > Regards,
> >
> > Ruediger
> >
> >
> > > - We have the old 8 TOS bits of the IP header. Nowadays, two of them=
 are
> > > ECN, and six of them are the DSCP field. So=20
> the problem is how to use these
> > > 6 bits (64 possible values)
> > > - Network control is 11xxxx, so this uses 16 values
> > >       -11xx11 are defined in RFC 2474, and remain as only=
 experimentally
> > > assignable/useable (this uses 4 of these 16 values)
> > > - This leaves 48 values
> > > - Best effort is 000000
> > > - The draft of James proposes 14 values for=20
> the DSCP field, which correspond
> > > to service classes, mapped to application categories.
> > >
> > > But Ruediger says that, in his opinion,=20
> more than 4 classes are not useful,
> > > according to current implementations.
> >
> > That's actually not appropriate for an RFC=20
> 4594 update - the proper place to
> > address that requirement is an update to RFC 5127, which will need to be
> > updated
> > to be consistent with whatever is done to RFC 4594.
> >
> > Thanks,
> > --David
> >
> >
> > > -----Original Message-----
> > > From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf=
 Of
> > Jose
> > > Saldana
> > > Sent: Saturday, April 07, 2012 8:03 AM
> > > To: Ruediger.Geib@telekom.de; James M. Polk
> > > Cc: tsvwg@ietf.org
> > > Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> > >
> > > I have been reading the two messages of=20
> Ruediger about Jame's draft. I will
> > > try to make a "status of the question" and=20
> then a proposal. If something is
> > > wrong, just tell me.
> > >
> > > - We have the old 8 TOS bits of the IP header. Nowadays, two of them=
 are
> > > ECN, and six of them are the DSCP field. So=20
> the problem is how to use these
> > > 6 bits (64 possible values)
> > > - Network control is 11xxxx, so this uses 16 values
> > >       -11xx11 are defined in RFC 2474, and remain as only=
 experimentally
> > > assignable/useable (this uses 4 of these 16 values)
> > > - This leaves 48 values
> > > - Best effort is 000000
> > > - The draft of James proposes 14 values for=20
> the DSCP field, which correspond
> > > to service classes, mapped to application categories.
> > >
> > > But Ruediger says that, in his opinion,=20
> more than 4 classes are not useful,
> > > according to current implementations.
> > >
> > > We can ask ourselves: which are  the key=20
> parameters that define how packets
> > > have to be managed in a router? I think that they are:
> > >
> > > 1) "Real-time-ness": The tolerated delay of=20
> the service. The jitter can be
> > > roughly included in this parameter.
> > > 2) loss tolerance: We can distinguish: TCP services are tolerant to=
 loss,
> > > since they retransmit packets. And some UDP services are also tolerant=
 to
> > > loss (VoIP, some games, etc)
> > >
> > > In my opinion, the "real-time-ness" is the key: it distinguishes very
> > > similar services: e.g. a video conference is different from a=
 broadcast,
> > > since the latter permits delays of some=20
> seconds, whereas the first has to be
> > > interactive.
> > >
> > > So one possible solution could be:
> > >
> > > - Using the first two bits in order to=20
> define 4 classes, taking into account
> > > the existing implementations. They can be=20
> the 4 classes of Fig. 1 of James'
> > > draft, or the ones proposed by Ruediger.
> > >       00xxxx  best effort
> > >       01xxxx data
> > >       10xxxx media oriented
> > >       11xxxx control
> > >
> > > - With the other 4 bits, something should be done. My proposal:
> > >       - If the third bit is "1", then the=20
> other three bits include a value
> > > defining the "real-time-ness" of the flow. So we have 8 levels of
> > > real-time-ness. Some examples:
> > >               - 10-1-111 would be a  media-oriented (10) and very
> > > real-time flow, as VoIP or a game
> > >               - 10-0-xxx would be a media-oriented flow for which the
> > > real-time-ness is not defined
> > >               - 10-1-010 could be a media-oriented flow with medium=
 delay
> > > requirements, as multimedia conferencing
> > >               - 10-1-000 would be a media-oriented flow with no very=
 hard
> > > delay requirements, as multimedia streaming
> > >               - 00-xxxx would be best-effort
> > >               -01-1-111 would be very low-latency data
> > >               -01-1-000 would be low-priority data
> > >
> > > One advantage of this is that some routers may only take into account=
 the
> > > first two bits, and others would use all of them.
> > >
> > > The new RFC could suggest or define=20
> different values of real-time-ness for
> > > each application. So the applications should be ordered depending of=
 the
> > > maximum tolerated delay.
> > >
> > > Ok. This is only an idea. Perhaps it does not make sense. Just tell=
 me.
> > > Regards,
> > >
> > > Jose
> > >
> > > -----Mensaje original-----
> > > De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]
> > > Enviado el: jueves, 05 de abril de 2012 12:23
> > > Para: jsaldana@unizar.es
> > > CC: tsvwg@ietf.org
> > > Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> > >
> > > Jose,
> > >
> > > regarding QoS, I'm happy to agree on 4=20
> standard classes with properties as
> > > mentioned in my last email. If there's consensus on some codepoints=
 for
> > > them, I'll support that.
> > >
> > > If IETF recommends how and which applications to use these
> > > classes/codepoints, that's again agreeable. I don't think, that by now
> > > requiring a specific application to carry a=20
> specific DSCP makes sense, apart
> > > from few exceptions (like VoIP payload,=20
> network management traffic or Best
> > > Effort traffic).
> > > IETF needs consensus to do that.
> > > Note that the number of AF classes and=20
> codepoints are troublesome. Too many
> > > with too little explanation how to use them, nothing is aligned with=
 user
> > > demand. EF/VoIP payload and network=20
> management are examples how to proceed:
> > > just specify what's required, no extras.
> > >
> > >
> > > Then regarding tunnels and header compression Yes, that's smart and=
 there
> > > may be more use cases than gaming. I'm a=20
> backbone engineer however and your
> > > proposal is rather application specific. What I can do is ask some
> > > colleagues on their view. But there's no=20
> guarantee, that they will respond.
> > >
> > > Some initial thoughts:
> > > To me a key point seems to be what can be=20
> reached at the bottleneck. There
> > > the compression should save bandwidth with little cost and no negative
> > > service impact. Any kind of shared access=20
> from which traffic is transmitted
> > > to the same destination seems to be relevant. The application/tunnel
> > > awareness of network elements may not be=20
> feasible, a network terminal at a
> > > UNI may be able to handle that, or any=20
> service specific device in a network.
> > >
> > > Regards,
> > >
> > > Ruediger
> > >
> > >
> > > -----Original Message-----
> > > From: Jose Saldana [mailto:jsaldana@unizar.es]
> > > Sent: Thursday, April 05, 2012 11:29 AM
> > > To: Geib, R=FCdiger
> > > Cc: tsvwg@ietf.org
> > > Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> > >
> > > Ruediger,
> > >
> > > I have read your message about=20
> draft-polk-tsvwg-rfc4594-update. As you may
> > > know, I presented another draft in Paris=20
> (draft-saldana-tsvwg-tcmtf-02), and
> > > after talking to James, I thought that the=20
> idea of standardizing the traffic
> > > classification fits well with my proposal.=20
> I would like to know your opinion
> > > about it, if possible.
> > >
> > > The idea is to use tunnels in order to compress and multiplex=
 real-time
> > > traffic flows. Since these flows tend to=20
> use very small packets, they have
> > > very bad efficiency. So perhaps tunnels could be used in order to=
 group
> > > these flows. The drawback is that they add small delays, i.e. the=
 waiting
> > > time in the multiplexer. So another use is to provide flexibility: if=
 you
> > > are a game provider, you may over provision=20
> your network in order to be able
> > > to manage traffic peaks. But perhaps using=20
> this, you may be able to have a
> > > smaller network, and use compression when=20
> necessary (rush hour, first days
> > > after releasing a new game, etc). When traffic is small, you do not
> > > multiplex, in order to reduce latency.
> > >
> > > So if traffic classification is a standard, everyone in the way from=
 the
> > > origin to the destination will handle the=20
> traffic with the correct priority.
> > >
> > > I attach a figure in order to clarify.
> > >
> > > Thank you,
> > >
> > > Jose
> > >
> > > PS: In
> > >=20
>=
 http://community.virginmedia.com/t5/Fibre-optic-broadband-cable/World-of-Wa=
r
> > > craft-high-latency-issues-with-virgin-media-users/td-p/44498 you can=
 find
> > > the history of the problems of World of=20
> Warcraft for Virgin Media users. The
> > > problem was caused by a modification in the traffic, and the provider=
 did
> > > not notice about it, so it treated the packets as best-effort.
> > >
> > > -----Mensaje original-----
> > > De: Ruediger.Geib@telekom.de=20
> [mailto:Ruediger.Geib@telekom.de] Enviado el:
> > > jueves, 05 de abril de 2012 9:49
> > > Para: jsaldana@unizar.es
> > > CC: tsvwg@ietf.org
> > > Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> > >
> > > Jose,
> > >
> > > you raise an interesting point:
> > >
> > > draft-polk-tsvwg-rfc4594-update suggests to standardise application to
> > > codepoint mapping as well as codepoint to=20
> class properties. You observe that
> > > some games are using TCP while others use UDP, which I think is a=
 useful
> > > level of QoS requirement abstraction.=20
> There's no need to have like 25+ QoS
> > > (sub)classes, I think. But one class=20
> optimised to transport  "UDP" and one
> > > for "TCP" are agreeable.
> > >
> > > The feedback that I had on carrier class=20
> QoS usage is that there's a point
> > > in standardising class properties, and certainly not to standardise
> > > application mappings to classes=20
> (recommending them may however be useful).
> > > My take would be to standardise four=20
> classes/properties (in brackets example
> > > usage):
> > >
> > > a) for traffic which doesn't support re-transmits, requiring lowest=
 delay
> > > and jitter, having limited volume and packet
> > >    size, but being loss tolerant (example usage VoIP)
> > > b) for any other traffic which doesn't support re-transmits (let's say
> > > traffic with real time like properties, but any
> > >    packet size and possibly high volume, UDP transport, like IPTV)
> > > c) for any other traffic which supports=20
> re-transmits (QoS traffic using TCP,
> > > like application signalling and network management traffic)
> > > d) a default class, for any traffic=20
> supporting re-transmits (Best Effort) -
> > > this class may not be regarded as QoS (or as lowest CoS)
> > >
> > > What I can tell ist that many carriers and standards on QoS support=
 3-4
> > > classes, having properties as described above. It is useful to=
 recommend
> > > codepoints matching to these classes. Standardising these codepoints
> > > requires agreement between those having=20
> deplyed them. Straightforward for a)
> > > EF and d) 000000, probably easy for c) being used to transport Network
> > > Management (CS 6). I'm not aware of common ground for anything else.=
 My
> > > favourites are:
> > >
> > > b) AF4 (GSMA supports this one, RFC4594 also)
> > > c) AF3 (MEF and GSMA support this one)
> > >
> > > Gaming from my point of view would then=20
> match to an AF3 or AF4 class, this
> > > may depend on the implemented application=20
> transport. Further details may be
> > > discussed.
> > >
> > > Everything else could be reserved, but shouldn't be recommended.=
 Carriers
> > > have deployed differing schemes on=20
> codepoints and classes. They need coding
> > > reserverves in 3 bit space (MPLS TC, IP=20
> Precedence and Ethernet P-Bits) as
> > > well as in DSCP space if they should converge to a standardised
> > > codepoint/class scheme.
> > >
> > > I object to any approach standardising a=20
> match of classes and applications
> > > with codepoints as a requirement and I also=20
> object to standards leaving no
> > > spare codepoints in 3 bit or 6 bit QoS=20
> codepoint space (including 3 bit IP
> > > precedence).
> > >
> > > Bob Briscoe mentioned the difference between classes and queues and=
 above
> > > properties can be mapped to queues. If end=20
> to end QoS is desired, the same
> > > class properties must be negotiated end to end. The lower the number=
 of
> > > classes and codepoints being deployed, the=20
> higher the probability of getting
> > > end to end QoS. I don't support an approach requiring 8 three bit=
 classes
> > > and 25 or more 6 bit classes and codepoints.
> > >
> > > Regards,
> > >
> > > Ruediger
> > >
> > > ________________________________
> > >
> > > From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf=
 Of
> > > Jose Saldana
> > > Sent: Monday, April 02, 2012 12:17 PM
> > > To: James Polk
> > > Cc: tsvwg@ietf.org
> > > Subject: About draft-polk-tsvwg-rfc4594-update-00.txt
> > >
> > >
> > >
> > > James,
> > >
> > >
> > >
> > > After returning home from IETF 83, I have=20
> been reading your draft. I find it
> > > interesting to standardize the service classes. Otherwise, if each one
> > > classifies each traffic according to different criteria, Diffserv=
 becomes
> > > less useful.
> > >
> > >
> > >
> > > Regarding "Real-time interactive" service class, which refers to=
 online
> > > gaming (my main research field), I find it=20
> really interesting: online games
> > > are at least as real-time as VoIP is. In=20
> most cases, the gamers are a very
> > > difficult people to deal with. If the=20
> latency is high, they leave and never
> > > return (this statement is not mine, it can=20
> be corroborated in the references
> > > I put below). This has been studied by many=20
> research groups, with empirical
> > > tests using real gamers and adding delay, jitter, packet loss.
> > >
> > >
> > >
> > > Other question: in the draft it is said that online games use RTP/UDP
> > > streams. This can be a little confusing. In fact, the vast majority of
> > > online games do not use RTP, but bare UDP or even TCP. In
> > > http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.pdf,=
 the
> > > authors proposed the use of an adaptation=20
> of RTP in order to manage gaming
> > > traffic. But as far as I know, this approach is not used by commercial
> > > games.
> > >
> > >
> > >
> > >
> > >
> > > A typical classification of today's games is:
> > >
> > >
> > >
> > > -          FPS (First Person Shooters) e.g. Counter Strike, Halo, Call=
 of
> > > Duty: They use UDP
> > >
> > > -          MMORPG (Massively Multiplayer Online Role-playing Games),=
 e.g.
> > > World of Warcraft: They normally use TCP,=20
> even though some of them use UDP
> > >
> > > -          RTS (Real-time strategy) e.g.=20
> Age of Empires: They may use TCP or
> > > UDP
> > >
> > > -          Sports games: Since there is a big variety, they have not=
 been
> > > deeply studied in the literature
> > >
> > >
> > >
> > > Some research papers supporting this=20
> classification, and also the real-time
> > > requirements of the games:
> > >
> > >
> > >
> > > First Person Shooters
> > >
> > >
> > >
> > > -          A survey of traffic models for 17 First Person Shooters can=
 be
> > > found here. All of them use UDP
> > >=
 http://www.discover.uottawa.ca/publications/files/MMOG-Survey_IEEE-IC.pdf
> > >
> > >
> > >
> > > -          In this paper from my research=20
> group, 8 FPS games were analyzed
> > > in order to compress and multiplex their traffic. Figure 5 shows their
> > > histograms (inter-packet time and packet size). It can be seen that=
 they
> > > generate real-time flows, with=20
> characteristics very similar to VoIP (packets
> > > of 60-80 bytes every 15-40 ms):
> > > http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana.pdf
> > >
> > >
> > >
> > > -          Another paper about FPS using UDP. Section V presents some
> > > interesting conclusions about routing infrastructure.
> > > http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf
> > >
> > >
> > >
> > > -          In this paper the network factors annoying the gamers are
> > > analyzed. There are also some calculations of the MOS. Above 150 ms=
 the
> > > quality is bad (something similar to what happens with VoIP)
> > > http://www.research.ibm.com/netgames2005/papers/dick.pdf
> > >
> > >
> > >
> > > -          In this one, a MOS model for a=20
> FPS (Quake IV) is deployed. It is
> > > only based on delay and jitter, since=20
> packet loss has vevy little influence
> > > (for this concrete game). Delay above 140 ms produces MOS below 3
> > > http://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf
> > >
> > >
> > >
> > > -          In this paper, it is found that=20
> some FPS work properly even with
> > > 30% packet loss, whereas others stop=20
> working with 4%. RTT above 300 ms (150
> > > OWD) makes MOS go above 3
> > > http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pdf
> > >
> > >
> > >
> > > MMORPGs:
> > >
> > > -          A traffic model for World of Warcraft:
> > > http://publik.tuwien.ac.at/files/pub-et_12119.pdf
> > >
> > >
> > >
> > > -          This one presents a MOS for=20
> World of Warcraft. When added delay
> > > is above 160ms, the MOS is roughly 3.:
> > > http://publik.tuwien.ac.at/files/PubDat_170202.pdf
> > >
> > >
> > >
> > > Best regards,
> > >
> > >
> > >
> > > Jose Saldana
> > >
> > >
> >


From jsaldana@unizar.es  Fri Apr 27 01:06:14 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
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Subject: Re: [tsvwg] About draft-polk-tsvwg-rfc4594-update-00.txt
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James,

I think it is clear that people agree with the need of rewriting RFC5127 =
in
conjunction with RFC4594. I also agree.

If you want, I could help you with that task. Just tell me. Next two =
months
I think I will have a little more available time.

Jose

-----Mensaje original-----
De: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] En nombre de
James M. Polk
Enviado el: jueves, 26 de abril de 2012 23:17
Para: david.black@emc.com
CC: tsvwg@ietf.org
Asunto: Re: [tsvwg] About draft-polk-tsvwg-rfc4594-update-00.txt

David

I absolutely agree with you that these two efforts (the current rewrite =
of
RFC4594, and the coming rewrite of RFC5127) need to be synchronized to =
have
either make proper sense. I also agree that the rewrite of RFC5127 is =
where
a smaller, aggregated set of classes for core networks. Ruediger =
mentioned
his worry about "folks viewing tables and just configuring based on the
tables". Clearly his concerns on that need to be addressed.

I'll work to get my initial version of the rewrite of RFC5127 in before
Vancouver.

James

At 07:17 PM 4/25/2012, david.black@emc.com wrote:
>Ruediger,
>
>We may be disagreeing more about means than ends.
>
>RFC 5127's intent was to aggregate all of the classes defined in RFC=20
>4594 into a much smaller number of classes that would be suitable for a =

>core network,  e.g., "3-4 classes and require no more than 6=20
>codepoints".  The net would be that a choice of considerably more=20
>traffic classes from RFC
>4594 would be deployable via aggregation as a suitably small number of=20
>classes on the wire.
>
>I believe (and I've said so to James), that RFC 5127 needs to be=20
>revised in concert with any work on RFC 4594 (in particular, I strongly =

>oppose any revision to 4594 that ignores 5127).  I also think that RFC=20
>5127 is a better starting point to address the small number of classes=20
>use case that appears to be of interest and concern to you.
>
>Thanks,
>--David
>
> > -----Original Message-----
> > From: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]
> > Sent: Tuesday, April 24, 2012 5:56 AM
> > To: Black, David
> > Cc: tsvwg@ietf.org
> > Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> >
> > David,
> >
> > taking a formal point of view, you are probably right. These isues=20
> > are intertwined however.
> >
> > End to end QoS will certainly not result from continuing to act like =

> > almost all standards bodies did so far: as there's a lot flexibility =

> > and little non-ambiguous guidance, anything is customised.
> >
> > I don't regard the update of RFC 4594 as being helpful if the=20
> > expectation of the author is that even supporters of that standard=20
> > will only deploy a subset of the defined classes and sub classes.=20
> > Those standards close to deployment I know offer 3-4 classes and=20
> > require no more than 6 codepoints (the latter is personal =
judgement).
> >
> > Any standards track QoS document should only determine things seeing =

> > widespread deployment.
> >
> > Some QoS classes have seen deployment. It's trickier with =
codepoints.
> >
> > Anything which might be useful in future or has seen an=20
> > informational proposal in past  should be critically reviewed and=20
> > remain informational or experimental.
> >
> > Any QoS class should be well defined any easy to be differentiated=20
> > against others.
> >
> > QoS codepoints should not identify applications.
> >
> > I also don't think that one needs to differentiate "admitted" from=20
> > "non-admitted" classes. In a traffic engineered DiffServ network,=20
> > customers get an SLA describing the worst case QoS class =
performance.
> > To me it doesn't matter how a carrier ensures that the SLAs are =
kept.
> > It only matters that they are kept.
> >
> > Regards,
> >
> > Ruediger
> >
> >
> > > - We have the old 8 TOS bits of the IP header. Nowadays, two of=20
> > > them are ECN, and six of them are the DSCP field. So
> the problem is how to use these
> > > 6 bits (64 possible values)
> > > - Network control is 11xxxx, so this uses 16 values
> > >       -11xx11 are defined in RFC 2474, and remain as only=20
> > > experimentally assignable/useable (this uses 4 of these 16 values)
> > > - This leaves 48 values
> > > - Best effort is 000000
> > > - The draft of James proposes 14 values for
> the DSCP field, which correspond
> > > to service classes, mapped to application categories.
> > >
> > > But Ruediger says that, in his opinion,
> more than 4 classes are not useful,
> > > according to current implementations.
> >
> > That's actually not appropriate for an RFC
> 4594 update - the proper place to
> > address that requirement is an update to RFC 5127, which will need=20
> > to be updated to be consistent with whatever is done to RFC 4594.
> >
> > Thanks,
> > --David
> >
> >
> > > -----Original Message-----
> > > From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On=20
> > > Behalf Of
> > Jose
> > > Saldana
> > > Sent: Saturday, April 07, 2012 8:03 AM
> > > To: Ruediger.Geib@telekom.de; James M. Polk
> > > Cc: tsvwg@ietf.org
> > > Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> > >
> > > I have been reading the two messages of
> Ruediger about Jame's draft. I will
> > > try to make a "status of the question" and
> then a proposal. If something is
> > > wrong, just tell me.
> > >
> > > - We have the old 8 TOS bits of the IP header. Nowadays, two of=20
> > > them are ECN, and six of them are the DSCP field. So
> the problem is how to use these
> > > 6 bits (64 possible values)
> > > - Network control is 11xxxx, so this uses 16 values
> > >       -11xx11 are defined in RFC 2474, and remain as only=20
> > > experimentally assignable/useable (this uses 4 of these 16 values)
> > > - This leaves 48 values
> > > - Best effort is 000000
> > > - The draft of James proposes 14 values for
> the DSCP field, which correspond
> > > to service classes, mapped to application categories.
> > >
> > > But Ruediger says that, in his opinion,
> more than 4 classes are not useful,
> > > according to current implementations.
> > >
> > > We can ask ourselves: which are  the key
> parameters that define how packets
> > > have to be managed in a router? I think that they are:
> > >
> > > 1) "Real-time-ness": The tolerated delay of
> the service. The jitter can be
> > > roughly included in this parameter.
> > > 2) loss tolerance: We can distinguish: TCP services are tolerant=20
> > > to loss, since they retransmit packets. And some UDP services are=20
> > > also tolerant to loss (VoIP, some games, etc)
> > >
> > > In my opinion, the "real-time-ness" is the key: it distinguishes=20
> > > very similar services: e.g. a video conference is different from a =

> > > broadcast, since the latter permits delays of some
> seconds, whereas the first has to be
> > > interactive.
> > >
> > > So one possible solution could be:
> > >
> > > - Using the first two bits in order to
> define 4 classes, taking into account
> > > the existing implementations. They can be
> the 4 classes of Fig. 1 of James'
> > > draft, or the ones proposed by Ruediger.
> > >       00xxxx  best effort
> > >       01xxxx data
> > >       10xxxx media oriented
> > >       11xxxx control
> > >
> > > - With the other 4 bits, something should be done. My proposal:
> > >       - If the third bit is "1", then the
> other three bits include a value
> > > defining the "real-time-ness" of the flow. So we have 8 levels of=20
> > > real-time-ness. Some examples:
> > >               - 10-1-111 would be a  media-oriented (10) and very=20
> > > real-time flow, as VoIP or a game
> > >               - 10-0-xxx would be a media-oriented flow for which=20
> > > the real-time-ness is not defined
> > >               - 10-1-010 could be a media-oriented flow with=20
> > > medium delay requirements, as multimedia conferencing
> > >               - 10-1-000 would be a media-oriented flow with no=20
> > > very hard delay requirements, as multimedia streaming
> > >               - 00-xxxx would be best-effort
> > >               -01-1-111 would be very low-latency data
> > >               -01-1-000 would be low-priority data
> > >
> > > One advantage of this is that some routers may only take into=20
> > > account the first two bits, and others would use all of them.
> > >
> > > The new RFC could suggest or define
> different values of real-time-ness for
> > > each application. So the applications should be ordered depending=20
> > > of the maximum tolerated delay.
> > >
> > > Ok. This is only an idea. Perhaps it does not make sense. Just =
tell
me.
> > > Regards,
> > >
> > > Jose
> > >
> > > -----Mensaje original-----
> > > De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]=20
> > > Enviado el: jueves, 05 de abril de 2012 12:23
> > > Para: jsaldana@unizar.es
> > > CC: tsvwg@ietf.org
> > > Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> > >
> > > Jose,
> > >
> > > regarding QoS, I'm happy to agree on 4
> standard classes with properties as
> > > mentioned in my last email. If there's consensus on some=20
> > > codepoints for them, I'll support that.
> > >
> > > If IETF recommends how and which applications to use these=20
> > > classes/codepoints, that's again agreeable. I don't think, that by =

> > > now requiring a specific application to carry a
> specific DSCP makes sense, apart
> > > from few exceptions (like VoIP payload,
> network management traffic or Best
> > > Effort traffic).
> > > IETF needs consensus to do that.
> > > Note that the number of AF classes and
> codepoints are troublesome. Too many
> > > with too little explanation how to use them, nothing is aligned=20
> > > with user demand. EF/VoIP payload and network
> management are examples how to proceed:
> > > just specify what's required, no extras.
> > >
> > >
> > > Then regarding tunnels and header compression Yes, that's smart=20
> > > and there may be more use cases than gaming. I'm a
> backbone engineer however and your
> > > proposal is rather application specific. What I can do is ask some =

> > > colleagues on their view. But there's no
> guarantee, that they will respond.
> > >
> > > Some initial thoughts:
> > > To me a key point seems to be what can be
> reached at the bottleneck. There
> > > the compression should save bandwidth with little cost and no=20
> > > negative service impact. Any kind of shared access
> from which traffic is transmitted
> > > to the same destination seems to be relevant. The=20
> > > application/tunnel awareness of network elements may not be
> feasible, a network terminal at a
> > > UNI may be able to handle that, or any
> service specific device in a network.
> > >
> > > Regards,
> > >
> > > Ruediger
> > >
> > >
> > > -----Original Message-----
> > > From: Jose Saldana [mailto:jsaldana@unizar.es]
> > > Sent: Thursday, April 05, 2012 11:29 AM
> > > To: Geib, R=FCdiger
> > > Cc: tsvwg@ietf.org
> > > Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> > >
> > > Ruediger,
> > >
> > > I have read your message about
> draft-polk-tsvwg-rfc4594-update. As you may
> > > know, I presented another draft in Paris
> (draft-saldana-tsvwg-tcmtf-02), and
> > > after talking to James, I thought that the
> idea of standardizing the traffic
> > > classification fits well with my proposal.=20
> I would like to know your opinion
> > > about it, if possible.
> > >
> > > The idea is to use tunnels in order to compress and multiplex=20
> > > real-time traffic flows. Since these flows tend to
> use very small packets, they have
> > > very bad efficiency. So perhaps tunnels could be used in order to=20
> > > group these flows. The drawback is that they add small delays,=20
> > > i.e. the waiting time in the multiplexer. So another use is to=20
> > > provide flexibility: if you are a game provider, you may over=20
> > > provision
> your network in order to be able
> > > to manage traffic peaks. But perhaps using
> this, you may be able to have a
> > > smaller network, and use compression when
> necessary (rush hour, first days
> > > after releasing a new game, etc). When traffic is small, you do=20
> > > not multiplex, in order to reduce latency.
> > >
> > > So if traffic classification is a standard, everyone in the way=20
> > > from the origin to the destination will handle the
> traffic with the correct priority.
> > >
> > > I attach a figure in order to clarify.
> > >
> > > Thank you,
> > >
> > > Jose
> > >
> > > PS: In
> > >=20
> http://community.virginmedia.com/t5/Fibre-optic-broadband-cable/World-
> of-War
> > > craft-high-latency-issues-with-virgin-media-users/td-p/44498 you=20
> > > can find the history of the problems of World of
> Warcraft for Virgin Media users. The
> > > problem was caused by a modification in the traffic, and the=20
> > > provider did not notice about it, so it treated the packets as
best-effort.
> > >
> > > -----Mensaje original-----
> > > De: Ruediger.Geib@telekom.de
> [mailto:Ruediger.Geib@telekom.de] Enviado el:
> > > jueves, 05 de abril de 2012 9:49
> > > Para: jsaldana@unizar.es
> > > CC: tsvwg@ietf.org
> > > Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> > >
> > > Jose,
> > >
> > > you raise an interesting point:
> > >
> > > draft-polk-tsvwg-rfc4594-update suggests to standardise=20
> > > application to codepoint mapping as well as codepoint to
> class properties. You observe that
> > > some games are using TCP while others use UDP, which I think is a=20
> > > useful level of QoS requirement abstraction.
> There's no need to have like 25+ QoS
> > > (sub)classes, I think. But one class
> optimised to transport  "UDP" and one
> > > for "TCP" are agreeable.
> > >
> > > The feedback that I had on carrier class
> QoS usage is that there's a point
> > > in standardising class properties, and certainly not to=20
> > > standardise application mappings to classes
> (recommending them may however be useful).
> > > My take would be to standardise four
> classes/properties (in brackets example
> > > usage):
> > >
> > > a) for traffic which doesn't support re-transmits, requiring=20
> > > lowest delay and jitter, having limited volume and packet
> > >    size, but being loss tolerant (example usage VoIP)
> > > b) for any other traffic which doesn't support re-transmits (let's =

> > > say traffic with real time like properties, but any
> > >    packet size and possibly high volume, UDP transport, like IPTV)
> > > c) for any other traffic which supports
> re-transmits (QoS traffic using TCP,
> > > like application signalling and network management traffic)
> > > d) a default class, for any traffic
> supporting re-transmits (Best Effort) -
> > > this class may not be regarded as QoS (or as lowest CoS)
> > >
> > > What I can tell ist that many carriers and standards on QoS=20
> > > support 3-4 classes, having properties as described above. It is=20
> > > useful to recommend codepoints matching to these classes.=20
> > > Standardising these codepoints requires agreement between those=20
> > > having
> deplyed them. Straightforward for a)
> > > EF and d) 000000, probably easy for c) being used to transport=20
> > > Network Management (CS 6). I'm not aware of common ground for=20
> > > anything else. My favourites are:
> > >
> > > b) AF4 (GSMA supports this one, RFC4594 also)
> > > c) AF3 (MEF and GSMA support this one)
> > >
> > > Gaming from my point of view would then
> match to an AF3 or AF4 class, this
> > > may depend on the implemented application
> transport. Further details may be
> > > discussed.
> > >
> > > Everything else could be reserved, but shouldn't be recommended.=20
> > > Carriers have deployed differing schemes on
> codepoints and classes. They need coding
> > > reserverves in 3 bit space (MPLS TC, IP
> Precedence and Ethernet P-Bits) as
> > > well as in DSCP space if they should converge to a standardised=20
> > > codepoint/class scheme.
> > >
> > > I object to any approach standardising a
> match of classes and applications
> > > with codepoints as a requirement and I also
> object to standards leaving no
> > > spare codepoints in 3 bit or 6 bit QoS
> codepoint space (including 3 bit IP
> > > precedence).
> > >
> > > Bob Briscoe mentioned the difference between classes and queues=20
> > > and above properties can be mapped to queues. If end
> to end QoS is desired, the same
> > > class properties must be negotiated end to end. The lower the=20
> > > number of classes and codepoints being deployed, the
> higher the probability of getting
> > > end to end QoS. I don't support an approach requiring 8 three bit=20
> > > classes and 25 or more 6 bit classes and codepoints.
> > >
> > > Regards,
> > >
> > > Ruediger
> > >
> > > ________________________________
> > >
> > > From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On=20
> > > Behalf Of Jose Saldana
> > > Sent: Monday, April 02, 2012 12:17 PM
> > > To: James Polk
> > > Cc: tsvwg@ietf.org
> > > Subject: About draft-polk-tsvwg-rfc4594-update-00.txt
> > >
> > >
> > >
> > > James,
> > >
> > >
> > >
> > > After returning home from IETF 83, I have
> been reading your draft. I find it
> > > interesting to standardize the service classes. Otherwise, if each =

> > > one classifies each traffic according to different criteria,=20
> > > Diffserv becomes less useful.
> > >
> > >
> > >
> > > Regarding "Real-time interactive" service class, which refers to=20
> > > online gaming (my main research field), I find it
> really interesting: online games
> > > are at least as real-time as VoIP is. In
> most cases, the gamers are a very
> > > difficult people to deal with. If the
> latency is high, they leave and never
> > > return (this statement is not mine, it can
> be corroborated in the references
> > > I put below). This has been studied by many
> research groups, with empirical
> > > tests using real gamers and adding delay, jitter, packet loss.
> > >
> > >
> > >
> > > Other question: in the draft it is said that online games use=20
> > > RTP/UDP streams. This can be a little confusing. In fact, the vast =

> > > majority of online games do not use RTP, but bare UDP or even TCP. =

> > > In=20
> > > http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.p
> > > df, the authors proposed the use of an adaptation
> of RTP in order to manage gaming
> > > traffic. But as far as I know, this approach is not used by=20
> > > commercial games.
> > >
> > >
> > >
> > >
> > >
> > > A typical classification of today's games is:
> > >
> > >
> > >
> > > -          FPS (First Person Shooters) e.g. Counter Strike, Halo, =
Call
of
> > > Duty: They use UDP
> > >
> > > -          MMORPG (Massively Multiplayer Online Role-playing =
Games),
e.g.
> > > World of Warcraft: They normally use TCP,
> even though some of them use UDP
> > >
> > > -          RTS (Real-time strategy) e.g.=20
> Age of Empires: They may use TCP or
> > > UDP
> > >
> > > -          Sports games: Since there is a big variety, they have =
not
been
> > > deeply studied in the literature
> > >
> > >
> > >
> > > Some research papers supporting this
> classification, and also the real-time
> > > requirements of the games:
> > >
> > >
> > >
> > > First Person Shooters
> > >
> > >
> > >
> > > -          A survey of traffic models for 17 First Person Shooters =
can
be
> > > found here. All of them use UDP
> > > http://www.discover.uottawa.ca/publications/files/MMOG-Survey_IEEE
> > > -IC.pdf
> > >
> > >
> > >
> > > -          In this paper from my research=20
> group, 8 FPS games were analyzed
> > > in order to compress and multiplex their traffic. Figure 5 shows=20
> > > their histograms (inter-packet time and packet size). It can be=20
> > > seen that they generate real-time flows, with
> characteristics very similar to VoIP (packets
> > > of 60-80 bytes every 15-40 ms):
> > > http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana.
> > > pdf
> > >
> > >
> > >
> > > -          Another paper about FPS using UDP. Section V presents =
some
> > > interesting conclusions about routing infrastructure.
> > > http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf
> > >
> > >
> > >
> > > -          In this paper the network factors annoying the gamers =
are
> > > analyzed. There are also some calculations of the MOS. Above 150=20
> > > ms the quality is bad (something similar to what happens with=20
> > > VoIP) http://www.research.ibm.com/netgames2005/papers/dick.pdf
> > >
> > >
> > >
> > > -          In this one, a MOS model for a=20
> FPS (Quake IV) is deployed. It is
> > > only based on delay and jitter, since
> packet loss has vevy little influence
> > > (for this concrete game). Delay above 140 ms produces MOS below 3=20
> > > http://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf
> > >
> > >
> > >
> > > -          In this paper, it is found that=20
> some FPS work properly even with
> > > 30% packet loss, whereas others stop
> working with 4%. RTT above 300 ms (150
> > > OWD) makes MOS go above 3
> > > http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pdf
> > >
> > >
> > >
> > > MMORPGs:
> > >
> > > -          A traffic model for World of Warcraft:
> > > http://publik.tuwien.ac.at/files/pub-et_12119.pdf
> > >
> > >
> > >
> > > -          This one presents a MOS for=20
> World of Warcraft. When added delay
> > > is above 160ms, the MOS is roughly 3.:
> > > http://publik.tuwien.ac.at/files/PubDat_170202.pdf
> > >
> > >
> > >
> > > Best regards,
> > >
> > >
> > >
> > > Jose Saldana
> > >
> > >
> >



From Ruediger.Geib@telekom.de  Fri Apr 27 01:31:46 2012
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Thread-Topic: [tsvwg] About draft-polk-tsvwg-rfc4594-update-00.txt
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Subject: Re: [tsvwg] About draft-polk-tsvwg-rfc4594-update-00.txt
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Jose,

I'd like to join in and also volunteer to co-author, but that
for me depends on the contents we find consensus upon. My time is
administered in Best Effort mode however...

Regards, Ruediger

-----Original Message-----
From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf Of J=
ose Saldana
Sent: Friday, April 27, 2012 10:06 AM
To: 'James M. Polk'
Cc: tsvwg@ietf.org
Subject: Re: [tsvwg] About draft-polk-tsvwg-rfc4594-update-00.txt

James,

I think it is clear that people agree with the need of rewriting RFC5127 in
conjunction with RFC4594. I also agree.

If you want, I could help you with that task. Just tell me. Next two months
I think I will have a little more available time.

Jose

-----Mensaje original-----
De: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] En nombre de
James M. Polk
Enviado el: jueves, 26 de abril de 2012 23:17
Para: david.black@emc.com
CC: tsvwg@ietf.org
Asunto: Re: [tsvwg] About draft-polk-tsvwg-rfc4594-update-00.txt

David

I absolutely agree with you that these two efforts (the current rewrite of
RFC4594, and the coming rewrite of RFC5127) need to be synchronized to have
either make proper sense. I also agree that the rewrite of RFC5127 is where
a smaller, aggregated set of classes for core networks. Ruediger mentioned
his worry about "folks viewing tables and just configuring based on the
tables". Clearly his concerns on that need to be addressed.

I'll work to get my initial version of the rewrite of RFC5127 in before
Vancouver.

James

At 07:17 PM 4/25/2012, david.black@emc.com wrote:
>Ruediger,
>
>We may be disagreeing more about means than ends.
>
>RFC 5127's intent was to aggregate all of the classes defined in RFC
>4594 into a much smaller number of classes that would be suitable for a
>core network,  e.g., "3-4 classes and require no more than 6
>codepoints".  The net would be that a choice of considerably more
>traffic classes from RFC
>4594 would be deployable via aggregation as a suitably small number of
>classes on the wire.
>
>I believe (and I've said so to James), that RFC 5127 needs to be
>revised in concert with any work on RFC 4594 (in particular, I strongly
>oppose any revision to 4594 that ignores 5127).  I also think that RFC
>5127 is a better starting point to address the small number of classes
>use case that appears to be of interest and concern to you.
>
>Thanks,
>--David
>
> > -----Original Message-----
> > From: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]
> > Sent: Tuesday, April 24, 2012 5:56 AM
> > To: Black, David
> > Cc: tsvwg@ietf.org
> > Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> >
> > David,
> >
> > taking a formal point of view, you are probably right. These isues
> > are intertwined however.
> >
> > End to end QoS will certainly not result from continuing to act like
> > almost all standards bodies did so far: as there's a lot flexibility
> > and little non-ambiguous guidance, anything is customised.
> >
> > I don't regard the update of RFC 4594 as being helpful if the
> > expectation of the author is that even supporters of that standard
> > will only deploy a subset of the defined classes and sub classes.
> > Those standards close to deployment I know offer 3-4 classes and
> > require no more than 6 codepoints (the latter is personal judgement).
> >
> > Any standards track QoS document should only determine things seeing
> > widespread deployment.
> >
> > Some QoS classes have seen deployment. It's trickier with codepoints.
> >
> > Anything which might be useful in future or has seen an
> > informational proposal in past  should be critically reviewed and
> > remain informational or experimental.
> >
> > Any QoS class should be well defined any easy to be differentiated
> > against others.
> >
> > QoS codepoints should not identify applications.
> >
> > I also don't think that one needs to differentiate "admitted" from
> > "non-admitted" classes. In a traffic engineered DiffServ network,
> > customers get an SLA describing the worst case QoS class performance.
> > To me it doesn't matter how a carrier ensures that the SLAs are kept.
> > It only matters that they are kept.
> >
> > Regards,
> >
> > Ruediger
> >
> >
> > > - We have the old 8 TOS bits of the IP header. Nowadays, two of
> > > them are ECN, and six of them are the DSCP field. So
> the problem is how to use these
> > > 6 bits (64 possible values)
> > > - Network control is 11xxxx, so this uses 16 values
> > >       -11xx11 are defined in RFC 2474, and remain as only
> > > experimentally assignable/useable (this uses 4 of these 16 values)
> > > - This leaves 48 values
> > > - Best effort is 000000
> > > - The draft of James proposes 14 values for
> the DSCP field, which correspond
> > > to service classes, mapped to application categories.
> > >
> > > But Ruediger says that, in his opinion,
> more than 4 classes are not useful,
> > > according to current implementations.
> >
> > That's actually not appropriate for an RFC
> 4594 update - the proper place to
> > address that requirement is an update to RFC 5127, which will need
> > to be updated to be consistent with whatever is done to RFC 4594.
> >
> > Thanks,
> > --David
> >
> >
> > > -----Original Message-----
> > > From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On
> > > Behalf Of
> > Jose
> > > Saldana
> > > Sent: Saturday, April 07, 2012 8:03 AM
> > > To: Ruediger.Geib@telekom.de; James M. Polk
> > > Cc: tsvwg@ietf.org
> > > Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> > >
> > > I have been reading the two messages of
> Ruediger about Jame's draft. I will
> > > try to make a "status of the question" and
> then a proposal. If something is
> > > wrong, just tell me.
> > >
> > > - We have the old 8 TOS bits of the IP header. Nowadays, two of
> > > them are ECN, and six of them are the DSCP field. So
> the problem is how to use these
> > > 6 bits (64 possible values)
> > > - Network control is 11xxxx, so this uses 16 values
> > >       -11xx11 are defined in RFC 2474, and remain as only
> > > experimentally assignable/useable (this uses 4 of these 16 values)
> > > - This leaves 48 values
> > > - Best effort is 000000
> > > - The draft of James proposes 14 values for
> the DSCP field, which correspond
> > > to service classes, mapped to application categories.
> > >
> > > But Ruediger says that, in his opinion,
> more than 4 classes are not useful,
> > > according to current implementations.
> > >
> > > We can ask ourselves: which are  the key
> parameters that define how packets
> > > have to be managed in a router? I think that they are:
> > >
> > > 1) "Real-time-ness": The tolerated delay of
> the service. The jitter can be
> > > roughly included in this parameter.
> > > 2) loss tolerance: We can distinguish: TCP services are tolerant
> > > to loss, since they retransmit packets. And some UDP services are
> > > also tolerant to loss (VoIP, some games, etc)
> > >
> > > In my opinion, the "real-time-ness" is the key: it distinguishes
> > > very similar services: e.g. a video conference is different from a
> > > broadcast, since the latter permits delays of some
> seconds, whereas the first has to be
> > > interactive.
> > >
> > > So one possible solution could be:
> > >
> > > - Using the first two bits in order to
> define 4 classes, taking into account
> > > the existing implementations. They can be
> the 4 classes of Fig. 1 of James'
> > > draft, or the ones proposed by Ruediger.
> > >       00xxxx  best effort
> > >       01xxxx data
> > >       10xxxx media oriented
> > >       11xxxx control
> > >
> > > - With the other 4 bits, something should be done. My proposal:
> > >       - If the third bit is "1", then the
> other three bits include a value
> > > defining the "real-time-ness" of the flow. So we have 8 levels of
> > > real-time-ness. Some examples:
> > >               - 10-1-111 would be a  media-oriented (10) and very
> > > real-time flow, as VoIP or a game
> > >               - 10-0-xxx would be a media-oriented flow for which
> > > the real-time-ness is not defined
> > >               - 10-1-010 could be a media-oriented flow with
> > > medium delay requirements, as multimedia conferencing
> > >               - 10-1-000 would be a media-oriented flow with no
> > > very hard delay requirements, as multimedia streaming
> > >               - 00-xxxx would be best-effort
> > >               -01-1-111 would be very low-latency data
> > >               -01-1-000 would be low-priority data
> > >
> > > One advantage of this is that some routers may only take into
> > > account the first two bits, and others would use all of them.
> > >
> > > The new RFC could suggest or define
> different values of real-time-ness for
> > > each application. So the applications should be ordered depending
> > > of the maximum tolerated delay.
> > >
> > > Ok. This is only an idea. Perhaps it does not make sense. Just tell
me.
> > > Regards,
> > >
> > > Jose
> > >
> > > -----Mensaje original-----
> > > De: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]
> > > Enviado el: jueves, 05 de abril de 2012 12:23
> > > Para: jsaldana@unizar.es
> > > CC: tsvwg@ietf.org
> > > Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> > >
> > > Jose,
> > >
> > > regarding QoS, I'm happy to agree on 4
> standard classes with properties as
> > > mentioned in my last email. If there's consensus on some
> > > codepoints for them, I'll support that.
> > >
> > > If IETF recommends how and which applications to use these
> > > classes/codepoints, that's again agreeable. I don't think, that by
> > > now requiring a specific application to carry a
> specific DSCP makes sense, apart
> > > from few exceptions (like VoIP payload,
> network management traffic or Best
> > > Effort traffic).
> > > IETF needs consensus to do that.
> > > Note that the number of AF classes and
> codepoints are troublesome. Too many
> > > with too little explanation how to use them, nothing is aligned
> > > with user demand. EF/VoIP payload and network
> management are examples how to proceed:
> > > just specify what's required, no extras.
> > >
> > >
> > > Then regarding tunnels and header compression Yes, that's smart
> > > and there may be more use cases than gaming. I'm a
> backbone engineer however and your
> > > proposal is rather application specific. What I can do is ask some
> > > colleagues on their view. But there's no
> guarantee, that they will respond.
> > >
> > > Some initial thoughts:
> > > To me a key point seems to be what can be
> reached at the bottleneck. There
> > > the compression should save bandwidth with little cost and no
> > > negative service impact. Any kind of shared access
> from which traffic is transmitted
> > > to the same destination seems to be relevant. The
> > > application/tunnel awareness of network elements may not be
> feasible, a network terminal at a
> > > UNI may be able to handle that, or any
> service specific device in a network.
> > >
> > > Regards,
> > >
> > > Ruediger
> > >
> > >
> > > -----Original Message-----
> > > From: Jose Saldana [mailto:jsaldana@unizar.es]
> > > Sent: Thursday, April 05, 2012 11:29 AM
> > > To: Geib, R=FCdiger
> > > Cc: tsvwg@ietf.org
> > > Subject: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> > >
> > > Ruediger,
> > >
> > > I have read your message about
> draft-polk-tsvwg-rfc4594-update. As you may
> > > know, I presented another draft in Paris
> (draft-saldana-tsvwg-tcmtf-02), and
> > > after talking to James, I thought that the
> idea of standardizing the traffic
> > > classification fits well with my proposal.
> I would like to know your opinion
> > > about it, if possible.
> > >
> > > The idea is to use tunnels in order to compress and multiplex
> > > real-time traffic flows. Since these flows tend to
> use very small packets, they have
> > > very bad efficiency. So perhaps tunnels could be used in order to
> > > group these flows. The drawback is that they add small delays,
> > > i.e. the waiting time in the multiplexer. So another use is to
> > > provide flexibility: if you are a game provider, you may over
> > > provision
> your network in order to be able
> > > to manage traffic peaks. But perhaps using
> this, you may be able to have a
> > > smaller network, and use compression when
> necessary (rush hour, first days
> > > after releasing a new game, etc). When traffic is small, you do
> > > not multiplex, in order to reduce latency.
> > >
> > > So if traffic classification is a standard, everyone in the way
> > > from the origin to the destination will handle the
> traffic with the correct priority.
> > >
> > > I attach a figure in order to clarify.
> > >
> > > Thank you,
> > >
> > > Jose
> > >
> > > PS: In
> > >
> http://community.virginmedia.com/t5/Fibre-optic-broadband-cable/World-
> of-War
> > > craft-high-latency-issues-with-virgin-media-users/td-p/44498 you
> > > can find the history of the problems of World of
> Warcraft for Virgin Media users. The
> > > problem was caused by a modification in the traffic, and the
> > > provider did not notice about it, so it treated the packets as
best-effort.
> > >
> > > -----Mensaje original-----
> > > De: Ruediger.Geib@telekom.de
> [mailto:Ruediger.Geib@telekom.de] Enviado el:
> > > jueves, 05 de abril de 2012 9:49
> > > Para: jsaldana@unizar.es
> > > CC: tsvwg@ietf.org
> > > Asunto: RE: About draft-polk-tsvwg-rfc4594-update-00.txt
> > >
> > > Jose,
> > >
> > > you raise an interesting point:
> > >
> > > draft-polk-tsvwg-rfc4594-update suggests to standardise
> > > application to codepoint mapping as well as codepoint to
> class properties. You observe that
> > > some games are using TCP while others use UDP, which I think is a
> > > useful level of QoS requirement abstraction.
> There's no need to have like 25+ QoS
> > > (sub)classes, I think. But one class
> optimised to transport  "UDP" and one
> > > for "TCP" are agreeable.
> > >
> > > The feedback that I had on carrier class
> QoS usage is that there's a point
> > > in standardising class properties, and certainly not to
> > > standardise application mappings to classes
> (recommending them may however be useful).
> > > My take would be to standardise four
> classes/properties (in brackets example
> > > usage):
> > >
> > > a) for traffic which doesn't support re-transmits, requiring
> > > lowest delay and jitter, having limited volume and packet
> > >    size, but being loss tolerant (example usage VoIP)
> > > b) for any other traffic which doesn't support re-transmits (let's
> > > say traffic with real time like properties, but any
> > >    packet size and possibly high volume, UDP transport, like IPTV)
> > > c) for any other traffic which supports
> re-transmits (QoS traffic using TCP,
> > > like application signalling and network management traffic)
> > > d) a default class, for any traffic
> supporting re-transmits (Best Effort) -
> > > this class may not be regarded as QoS (or as lowest CoS)
> > >
> > > What I can tell ist that many carriers and standards on QoS
> > > support 3-4 classes, having properties as described above. It is
> > > useful to recommend codepoints matching to these classes.
> > > Standardising these codepoints requires agreement between those
> > > having
> deplyed them. Straightforward for a)
> > > EF and d) 000000, probably easy for c) being used to transport
> > > Network Management (CS 6). I'm not aware of common ground for
> > > anything else. My favourites are:
> > >
> > > b) AF4 (GSMA supports this one, RFC4594 also)
> > > c) AF3 (MEF and GSMA support this one)
> > >
> > > Gaming from my point of view would then
> match to an AF3 or AF4 class, this
> > > may depend on the implemented application
> transport. Further details may be
> > > discussed.
> > >
> > > Everything else could be reserved, but shouldn't be recommended.
> > > Carriers have deployed differing schemes on
> codepoints and classes. They need coding
> > > reserverves in 3 bit space (MPLS TC, IP
> Precedence and Ethernet P-Bits) as
> > > well as in DSCP space if they should converge to a standardised
> > > codepoint/class scheme.
> > >
> > > I object to any approach standardising a
> match of classes and applications
> > > with codepoints as a requirement and I also
> object to standards leaving no
> > > spare codepoints in 3 bit or 6 bit QoS
> codepoint space (including 3 bit IP
> > > precedence).
> > >
> > > Bob Briscoe mentioned the difference between classes and queues
> > > and above properties can be mapped to queues. If end
> to end QoS is desired, the same
> > > class properties must be negotiated end to end. The lower the
> > > number of classes and codepoints being deployed, the
> higher the probability of getting
> > > end to end QoS. I don't support an approach requiring 8 three bit
> > > classes and 25 or more 6 bit classes and codepoints.
> > >
> > > Regards,
> > >
> > > Ruediger
> > >
> > > ________________________________
> > >
> > > From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On
> > > Behalf Of Jose Saldana
> > > Sent: Monday, April 02, 2012 12:17 PM
> > > To: James Polk
> > > Cc: tsvwg@ietf.org
> > > Subject: About draft-polk-tsvwg-rfc4594-update-00.txt
> > >
> > >
> > >
> > > James,
> > >
> > >
> > >
> > > After returning home from IETF 83, I have
> been reading your draft. I find it
> > > interesting to standardize the service classes. Otherwise, if each
> > > one classifies each traffic according to different criteria,
> > > Diffserv becomes less useful.
> > >
> > >
> > >
> > > Regarding "Real-time interactive" service class, which refers to
> > > online gaming (my main research field), I find it
> really interesting: online games
> > > are at least as real-time as VoIP is. In
> most cases, the gamers are a very
> > > difficult people to deal with. If the
> latency is high, they leave and never
> > > return (this statement is not mine, it can
> be corroborated in the references
> > > I put below). This has been studied by many
> research groups, with empirical
> > > tests using real gamers and adding delay, jitter, packet loss.
> > >
> > >
> > >
> > > Other question: in the draft it is said that online games use
> > > RTP/UDP streams. This can be a little confusing. In fact, the vast
> > > majority of online games do not use RTP, but bare UDP or even TCP.
> > > In
> > > http://www.cn.uni-duesseldorf.de/publications/library/Mauve2001a.p
> > > df, the authors proposed the use of an adaptation
> of RTP in order to manage gaming
> > > traffic. But as far as I know, this approach is not used by
> > > commercial games.
> > >
> > >
> > >
> > >
> > >
> > > A typical classification of today's games is:
> > >
> > >
> > >
> > > -          FPS (First Person Shooters) e.g. Counter Strike, Halo, Cal=
l
of
> > > Duty: They use UDP
> > >
> > > -          MMORPG (Massively Multiplayer Online Role-playing Games),
e.g.
> > > World of Warcraft: They normally use TCP,
> even though some of them use UDP
> > >
> > > -          RTS (Real-time strategy) e.g.
> Age of Empires: They may use TCP or
> > > UDP
> > >
> > > -          Sports games: Since there is a big variety, they have not
been
> > > deeply studied in the literature
> > >
> > >
> > >
> > > Some research papers supporting this
> classification, and also the real-time
> > > requirements of the games:
> > >
> > >
> > >
> > > First Person Shooters
> > >
> > >
> > >
> > > -          A survey of traffic models for 17 First Person Shooters ca=
n
be
> > > found here. All of them use UDP
> > > http://www.discover.uottawa.ca/publications/files/MMOG-Survey_IEEE
> > > -IC.pdf
> > >
> > >
> > >
> > > -          In this paper from my research
> group, 8 FPS games were analyzed
> > > in order to compress and multiplex their traffic. Figure 5 shows
> > > their histograms (inter-packet time and packet size). It can be
> > > seen that they generate real-time flows, with
> characteristics very similar to VoIP (packets
> > > of 60-80 bytes every 15-40 ms):
> > > http://diec.unizar.es/~jsaldana/personal/commag_nov_2011_jsaldana.
> > > pdf
> > >
> > >
> > >
> > > -          Another paper about FPS using UDP. Section V presents some
> > > interesting conclusions about routing infrastructure.
> > > http://www.thefengs.com/wuchang/work/cstrike/ton05_traffic.pdf
> > >
> > >
> > >
> > > -          In this paper the network factors annoying the gamers are
> > > analyzed. There are also some calculations of the MOS. Above 150
> > > ms the quality is bad (something similar to what happens with
> > > VoIP) http://www.research.ibm.com/netgames2005/papers/dick.pdf
> > >
> > >
> > >
> > > -          In this one, a MOS model for a
> FPS (Quake IV) is deployed. It is
> > > only based on delay and jitter, since
> packet loss has vevy little influence
> > > (for this concrete game). Delay above 140 ms produces MOS below 3
> > > http://www.nas.ewi.tudelft.nl/publications/2006/netgames.pdf
> > >
> > >
> > >
> > > -          In this paper, it is found that
> some FPS work properly even with
> > > 30% packet loss, whereas others stop
> working with 4%. RTT above 300 ms (150
> > > OWD) makes MOS go above 3
> > > http://caia.swin.edu.au/pubs/ATNAC04/zander-armitage-ATNAC2004.pdf
> > >
> > >
> > >
> > > MMORPGs:
> > >
> > > -          A traffic model for World of Warcraft:
> > > http://publik.tuwien.ac.at/files/pub-et_12119.pdf
> > >
> > >
> > >
> > > -          This one presents a MOS for
> World of Warcraft. When added delay
> > > is above 160ms, the MOS is roughly 3.:
> > > http://publik.tuwien.ac.at/files/PubDat_170202.pdf
> > >
> > >
> > >
> > > Best regards,
> > >
> > >
> > >
> > > Jose Saldana
> > >
> > >
> >


