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Hi all,
 
reading the experiments on SCTP performance 
 
LKSCTP 
http://datatag.web.cern.ch/datatag/WP3/sctp/tests.htm

OPENSS7
http://www.openss7.org/sctp_perf.html
 
If we ignore the gap underline between TCP and SCTP in LKSCTP, becose due to implementation problem ( and I think now the situation is better, isn't it? does someone have up to date results about?)
and use just the result underline on OPENSSS7
 
I've some questions/doubts and I'd like know you opinion about:
 
1) the test SCTP with 1 interface shows a substanzial equivalent to TCP.
Is it possible SCTP with 1 interface is TCP, or if you want if there is   not HoL block there are not difference between two protocols due to the same congestion algorithm.
 
2) SCTP with 2 interface has a better performance just for message size greater than 64-byte.
But SCTP was born for Signaling transport, and signaling messages are yes lager than 64-bye but not very large.
 
when in the experiment is used SOCK_SEQPACKET
3) the real start up with 1 or 2 interfaces are both worse than the Theoretical one. The strange is the reale one with 2 interfaces is worse than the reale one with just one interface.
4) similar considerations ( done above ) could be done also for the other phases . So SOCK_SEQPACKET seems not so good.
 
 
regards
sal

 
 

		
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<DIV>Hi all,</DIV>
<DIV>&nbsp;</DIV>
<DIV>reading the experiments on SCTP performance </DIV>
<DIV>&nbsp;</DIV>
<DIV>LKSCTP </DIV>
<DIV><A href="http://datatag.web.cern.ch/datatag/WP3/sctp/tests.htm" target=_blank>http://datatag.web.cern.ch/datatag/WP3/sctp/tests.htm</A></DIV>
<DIV><BR>OPENSS7<BR><A href="http://www.openss7.org/sctp_perf.html" target=_blank><FONT color=#003399>http://www.openss7.org/sctp_perf.html</FONT></A></DIV>
<DIV>&nbsp;</DIV>
<DIV>If we ignore the gap underline&nbsp;between TCP and SCTP in LKSCTP, becose&nbsp;due to implementation problem (&nbsp;and I think now the situation&nbsp;is better, isn't it? does someone have up to date results about?)</DIV>
<DIV>and use just the result underline on OPENSSS7</DIV>
<DIV>&nbsp;</DIV>
<DIV>I've some questions/doubts and I'd like know you opinion about:</DIV>
<DIV>&nbsp;</DIV>
<DIV>1) the test SCTP with 1 interface shows a substanzial equivalent to TCP.</DIV>
<DIV>Is it possible SCTP with&nbsp;1 interface is TCP, or if you want if there is&nbsp;&nbsp; not HoL block there are not difference between two protocols due to the same congestion algorithm.</DIV>
<DIV>&nbsp;</DIV>
<DIV>2) SCTP with 2 interface has a better performance just for message size greater than 64-byte.</DIV>
<DIV>But SCTP was born for Signaling transport, and signaling messages are yes lager than 64-bye but not very large.</DIV>
<DIV>&nbsp;</DIV>
<DIV>when in the experiment is&nbsp;used SOCK_SEQPACKET</DIV>
<DIV>3) the real start up with 1 or 2 interfaces are both worse than the Theoretical one. The strange is the reale one with 2 interfaces is worse than the reale one with just one interface.</DIV>
<DIV>4) similar considerations (&nbsp;done above&nbsp;) could be done also for the other phases . So SOCK_SEQPACKET seems not so good.</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>regards</DIV>
<DIV>sal</DIV>
<DIV><BR>&nbsp;</DIV>
<DIV>&nbsp;</DIV><p>
		<hr size=1><font face="Arial" size="2"><a href="http://it.rd.yahoo.com/mail/taglines/*http://it.mail.yahoo.com">Yahoo! Mail</a> ti protegge gratis dalla posta indesiderata e dai virus!</font>
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From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Salvatore Loreto <loretosal@yahoo.it>
Subject: Re: [Tsvwg] sctp performance
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Salvatore,

Performance tests on a clean network do not exercise SCTP's strong
points.  SCTP was not really intended to outperform TCP on a clean
network with a single interface: it was intended to outperform TCP on a
real network with noise, failures and reconfiguration.  If you look at
even single stream tests under noise SCTP rapidly outperforms TCP in
dirty networks.

The SOCK_SEQPACKET test deviation from theoretical is probably more of a
software glitch in bundling, Nagle'ing and Minshall'ing in the OpenSS7
stack more than a problem with the protocol (that was noted in the
results).

The SOCK_STREAM tests on a single interface did outperform TCP, however,
note that the tests saturated the network and both are running very
close to theoretical maximums.

The interesting thing about the OpenSS7 tests were the multiple
interface tests on SOCK_STREAM, that showed that max throughput was a
multiple of the number of available interfaces.  Once TCP saturates
a single interface it has nowhere else to go.  SCTP can load multiple
interfaces to saturation.

These were throughput tests.  Latency tests show SCTP has comparable
results to TCP (with slight improvements) on clean networks, and
SCTP exhibits significant reduced latencies to TCP under even as little
as 5% noise.  On distrupted networks (50% interface failure on a 1 or
2 second duty cycle) TCP stops and SCTP continues.

There are a lot of good papers out there testing more real environments
as well as NS-2 simulations that demonstrate the benefits of SCTP over
TCP for its original intended application, telephony signalling.  Search
on "SCTP Performance" and you can read for the better part of a day.

--brian

On Thu, 07 Oct 2004, Salvatore Loreto wrote:

>=20
>    Hi all,
>=20
>=20
>=20
>    reading the experiments on SCTP performance
>=20
>=20
>=20
>    LKSCTP
>=20
>    [1]http://datatag.web.cern.ch/datatag/WP3/sctp/tests.htm
>=20
>    OPENSS7
>    [2]http://www.openss7.org/sctp_perf.html
>=20
>=20
>=20
>    If  we  ignore  the  gap  underline between  TCP  and  SCTP in LKSCT=
P,
>    becose due   to   implementation   problem   ( and  I  think  now  t=
he
>    situation is  better,  isn't  it? does someone have up to date resul=
ts
>    about?)
>=20
>    and use just the result underline on OPENSSS7
>=20
>=20
>=20
>    I've some questions/doubts and I'd like know you opinion about:
>=20
>=20
>=20
>    1)  the  test  SCTP with 1 interface shows a substanzial equivalent =
to
>    TCP.
>=20
>    Is  it  possible SCTP with 1 interface is TCP, or if you want if the=
re
>    is    not HoL block there are not difference between two protocols d=
ue
>    to the same congestion algorithm.
>=20
>=20
>=20
>    2)  SCTP  with  2  interface has a better performance just for messa=
ge
>    size greater than 64-byte.
>=20
>    But  SCTP was born for Signaling transport, and signaling messages a=
re
>    yes lager than 64-bye but not very large.
>=20
>=20
>=20
>    when in the experiment is used SOCK_SEQPACKET
>=20
>    3)  the  real  start up with 1 or 2 interfaces are both worse than t=
he
>    Theoretical  one.  The  strange  is the reale one with 2 interfaces =
is
>    worse than the reale one with just one interface.
>=20
>    4)  similar  considerations  ( done above ) could be done also for t=
he
>    other phases . So SOCK_SEQPACKET seems not so good.
>=20
>=20
>=20
>=20
>=20
>    regards
>=20
>    sal
>=20
>=20
>=20
>=20
>      _________________________________________________________________
>=20
>    [3]Yahoo!  Mail  ti  protegge  gratis  dalla  posta indesiderata e d=
ai
>    virus!
>=20
> References
>=20
>    1. http://datatag.web.cern.ch/datatag/WP3/sctp/tests.htm
>    2. http://www.openss7.org/sctp_perf.html
>    3. http://it.rd.yahoo.com/mail/taglines/*http://it.mail.yahoo.com

> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg


--=20
Brian F. G. Bidulock    =A6 The reasonable man adapts himself to the =A6
bidulock@openss7.org    =A6 world; the unreasonable one persists in  =A6
http://www.openss7.org/ =A6 trying  to adapt the  world  to himself. =A6
                        =A6 Therefore  all  progress  depends on the =A6
                        =A6 unreasonable man. -- George Bernard Shaw =A6

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Brian/Salvatore:

Just a couple of comments...


Brian F. G. Bidulock wrote:
> Salvatore,
> 
> Performance tests on a clean network do not exercise SCTP's strong
> points.  SCTP was not really intended to outperform TCP on a clean
> network with a single interface: it was intended to outperform TCP on a
> real network with noise, failures and reconfiguration.  If you look at
> even single stream tests under noise SCTP rapidly outperforms TCP in
> dirty networks.
> 
> The SOCK_SEQPACKET test deviation from theoretical is probably more of a
> software glitch in bundling, Nagle'ing and Minshall'ing in the OpenSS7
> stack more than a problem with the protocol (that was noted in the
> results).

I would tend to agree with Brian here.. SOCK_SEQPACKET is really
an implementation shade.. In KAME other than controlling a
few minor things all the same code runs... so it sould make
very little difference....

> 
> The SOCK_STREAM tests on a single interface did outperform TCP, however,
> note that the tests saturated the network and both are running very
> close to theoretical maximums.
> 
> The interesting thing about the OpenSS7 tests were the multiple
> interface tests on SOCK_STREAM, that showed that max throughput was a
> multiple of the number of available interfaces.  Once TCP saturates
> a single interface it has nowhere else to go.  SCTP can load multiple
> interfaces to saturation.
> 
> These were throughput tests.  Latency tests show SCTP has comparable
> results to TCP (with slight improvements) on clean networks, and
> SCTP exhibits significant reduced latencies to TCP under even as little
> as 5% noise.  On distrupted networks (50% interface failure on a 1 or
> 2 second duty cycle) TCP stops and SCTP continues.
> 
> There are a lot of good papers out there testing more real environments
> as well as NS-2 simulations that demonstrate the benefits of SCTP over
> TCP for its original intended application, telephony signalling.  Search
> on "SCTP Performance" and you can read for the better part of a day.
> 

A couple of other things:

a) The CERN tests were done on an OLD OLD OLD version of lk-sctp which
    was NOT performance tuned at all.. I am not even sure if they have
    completed a pass yet of tuning.. I know its on their to-do list.

b) IPERF ... which is used on both sites.. to me.. is NOT a very good
    benchmark. SCTP sends "messages" from one end to the other. TCP sends
    a byte stream from one end to the other. You then take and use IPERF
    to test the number of bytes streaming from one end to the other.
    This puts SCTP at a dis-advantage. Especially considering that MOST
    ALL applications send messages NOT streams of bytes. So if you added
    some additional overhead to TCP.. say framing messages at the app
    layer.. this would slow TCP down I would wager.

    I have always wanted to develop a variable message sender that could
    send a sequence of variable messages and get a reading of messages
    per second for various sizes....

c) CRC32c is extremely expensive (in overall terms of CPU use) on the
    TCP side they generally have an off-load engine they can use to
    not even do the c-sum... so in such a case when you begin to
    get CPU limited SCTP will lag behind TCP... just on the checksum
    basis alone. Bottom line is once you get CRC32c offload SCTP will
    be able to do a bit better :-D

Regards

R


> --brian
> 
> On Thu, 07 Oct 2004, Salvatore Loreto wrote:
> 
> 
>>   Hi all,
>>
>>
>>
>>   reading the experiments on SCTP performance
>>
>>
>>
>>   LKSCTP
>>
>>   [1]http://datatag.web.cern.ch/datatag/WP3/sctp/tests.htm
>>
>>   OPENSS7
>>   [2]http://www.openss7.org/sctp_perf.html
>>
>>
>>
>>   If  we  ignore  the  gap  underline between  TCP  and  SCTP in LKSCTP,
>>   becose due   to   implementation   problem   ( and  I  think  now  the
>>   situation is  better,  isn't  it? does someone have up to date results
>>   about?)
>>
>>   and use just the result underline on OPENSSS7
>>
>>
>>
>>   I've some questions/doubts and I'd like know you opinion about:
>>
>>
>>
>>   1)  the  test  SCTP with 1 interface shows a substanzial equivalent to
>>   TCP.
>>
>>   Is  it  possible SCTP with 1 interface is TCP, or if you want if there
>>   is    not HoL block there are not difference between two protocols due
>>   to the same congestion algorithm.
>>
>>
>>
>>   2)  SCTP  with  2  interface has a better performance just for message
>>   size greater than 64-byte.
>>
>>   But  SCTP was born for Signaling transport, and signaling messages are
>>   yes lager than 64-bye but not very large.
>>
>>
>>
>>   when in the experiment is used SOCK_SEQPACKET
>>
>>   3)  the  real  start up with 1 or 2 interfaces are both worse than the
>>   Theoretical  one.  The  strange  is the reale one with 2 interfaces is
>>   worse than the reale one with just one interface.
>>
>>   4)  similar  considerations  ( done above ) could be done also for the
>>   other phases . So SOCK_SEQPACKET seems not so good.
>>
>>
>>
>>
>>
>>   regards
>>
>>   sal
>>
>>
>>
>>
>>     _________________________________________________________________
>>
>>   [3]Yahoo!  Mail  ti  protegge  gratis  dalla  posta indesiderata e dai
>>   virus!
>>
>>References
>>
>>   1. http://datatag.web.cern.ch/datatag/WP3/sctp/tests.htm
>>   2. http://www.openss7.org/sctp_perf.html
>>   3. http://it.rd.yahoo.com/mail/taglines/*http://it.mail.yahoo.com
> 
> 
>>_______________________________________________
>>tsvwg mailing list
>>tsvwg@ietf.org
>>https://www1.ietf.org/mailman/listinfo/tsvwg
> 
> 
> 


-- 
Randall Stewart
ITD
803-345-0369 <or> 815-342-5222

_______________________________________________
tsvwg mailing list
tsvwg@ietf.org
https://www1.ietf.org/mailman/listinfo/tsvwg


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	Thu, 07 Oct 2004 17:25:11 CEST
Date: Thu, 7 Oct 2004 17:25:11 +0200 (CEST)
From: Salvatore Loreto <loretosal@yahoo.it>
Subject: Re: [Tsvwg] sctp performance
To: Randall Stewart <rrs@cisco.com>,
        "Brian F. G. Bidulock" <bidulock@openss7.org>
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Hi Randall
=20
do you have some report or paper where I can find information
about SCTP in  KAME  ...=20
I read it includes HighSpeed SCTP, is it true?
Can I have more information about?
=20
thanks in advance
/Sal
=20
=20
=20


Randall Stewart <rrs@cisco.com> wrote:
Brian/Salvatore:

Just a couple of comments...


Brian F. G. Bidulock wrote:
> Salvatore,
>=20
> Performance tests on a clean network do not exercise SCTP's strong
> points. SCTP was not really intended to outperform TCP on a clean
> network with a single interface: it was intended to outperform TCP on a
> real network with noise, failures and reconfiguration. If you look at
> even single stream tests under noise SCTP rapidly outperforms TCP in
> dirty networks.
>=20
> The SOCK_SEQPACKET test deviation from theoretical is probably more of =
a
> software glitch in bundling, Nagle'ing and Minshall'ing in the OpenSS7
> stack more than a problem with the protocol (that was noted in the
> results).

I would tend to agree with Brian here.. SOCK_SEQPACKET is really
an implementation shade.. In KAME other than controlling a
few minor things all the same code runs... so it sould make
very little difference....

>=20
> The SOCK_STREAM tests on a single interface did outperform TCP, however=
,
> note that the tests saturated the network and both are running very
> close to theoretical maximums.
>=20
> The interesting thing about the OpenSS7 tests were the multiple
> interface tests on SOCK_STREAM, that showed that max throughput was a
> multiple of the number of available interfaces. Once TCP saturates
> a single interface it has nowhere else to go. SCTP can load multiple
> interfaces to saturation.
>=20
> These were throughput tests. Latency tests show SCTP has comparable
> results to TCP (with slight improvements) on clean networks, and
> SCTP exhibits significant reduced latencies to TCP under even as little
> as 5% noise. On distrupted networks (50% interface failure on a 1 or
> 2 second duty cycle) TCP stops and SCTP continues.
>=20
> There are a lot of good papers out there testing more real environments
> as well as NS-2 simulations that demonstrate the benefits of SCTP over
> TCP for its original intended application, telephony signalling. Search
> on "SCTP Performance" and you can read for the better part of a day.
>=20

A couple of other things:

a) The CERN tests were done on an OLD OLD OLD version of lk-sctp which
was NOT performance tuned at all.. I am not even sure if they have
completed a pass yet of tuning.. I know its on their to-do list.

b) IPERF ... which is used on both sites.. to me.. is NOT a very good
benchmark. SCTP sends "messages" from one end to the other. TCP sends
a byte stream from one end to the other. You then take and use IPERF
to test the number of bytes streaming from one end to the other.
This puts SCTP at a dis-advantage. Especially considering that MOST
ALL applications send messages NOT streams of bytes. So if you added
some additional overhead to TCP.. say framing messages at the app
layer.. this would slow TCP down I would wager.

I have always wanted to develop a variable message sender that could
send a sequence of variable messages and get a reading of messages
per second for various sizes....

c) CRC32c is extremely expensive (in overall terms of CPU use) on the
TCP side they generally have an off-load engine they can use to
not even do the c-sum... so in such a case when you begin to
get CPU limited SCTP will lag behind TCP... just on the checksum
basis alone. Bottom line is once you get CRC32c offload SCTP will
be able to do a bit better :-D

Regards

R


> --brian
>=20
> On Thu, 07 Oct 2004, Salvatore Loreto wrote:
>=20
>=20
>> Hi all,
>>
>>
>>
>> reading the experiments on SCTP performance
>>
>>
>>
>> LKSCTP
>>
>> [1]http://datatag.web.cern.ch/datatag/WP3/sctp/tests.htm
>>
>> OPENSS7
>> [2]http://www.openss7.org/sctp_perf.html
>>
>>
>>
>> If we ignore the gap underline between TCP and SCTP in LKSCTP,
>> becose due to implementation problem ( and I think now the
>> situation is better, isn't it? does someone have up to date results
>> about?)
>>
>> and use just the result underline on OPENSSS7
>>
>>
>>
>> I've some questions/doubts and I'd like know you opinion about:
>>
>>
>>
>> 1) the test SCTP with 1 interface shows a substanzial equivalent to
>> TCP.
>>
>> Is it possible SCTP with 1 interface is TCP, or if you want if there
>> is not HoL block there are not difference between two protocols due
>> to the same congestion algorithm.
>>
>>
>>
>> 2) SCTP with 2 interface has a better performance just for message
>> size greater than 64-byte.
>>
>> But SCTP was born for Signaling transport, and signaling messages are
>> yes lager than 64-bye but not very large.
>>
>>
>>
>> when in the experiment is used SOCK_SEQPACKET
>>
>> 3) the real start up with 1 or 2 interfaces are both worse than the
>> Theoretical one. The strange is the reale one with 2 interfaces is
>> worse than the reale one with just one interface.
>>
>> 4) similar considerations ( done above ) could be done also for the
>> other phases . So SOCK_SEQPACKET seems not so good.
>>
>>
>>
>>
>>
>> regards
>>
>> sal
>>
>>
>>
>>
>> _________________________________________________________________
>>
>> [3]Yahoo! Mail ti protegge gratis dalla posta indesiderata e dai
>> virus!
>>
>>References
>>
>> 1. http://datatag.web.cern.ch/datatag/WP3/sctp/tests.htm
>> 2. http://www.openss7.org/sctp_perf.html
>> 3. http://it.rd.yahoo.com/mail/taglines/*http://it.mail.yahoo.com
>=20
>=20
>>_______________________________________________
>>tsvwg mailing list
>>tsvwg@ietf.org
>>https://www1.ietf.org/mailman/listinfo/tsvwg
>=20
>=20
>=20


--=20
Randall Stewart
ITD
803-345-0369 815-342-5222

			=09
---------------------------------
Nuovo Yahoo! Messenger E' molto pi=F9 divertente: Audibles, Avatar, Webca=
m, Giochi, Rubrica=85 Scaricalo ora!=20
--0-1021742537-1097162711=:90499
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<DIV>Hi Randall</DIV>
<DIV>&nbsp;</DIV>
<DIV>do you have some report or paper where I can find information</DIV>
<DIV>about SCTP in&nbsp; KAME&nbsp; ... </DIV>
<DIV>I read it includes HighSpeed SCTP, is it true?</DIV>
<DIV>Can I have more information about?</DIV>
<DIV>&nbsp;</DIV>
<DIV>thanks in advance</DIV>
<DIV>/Sal</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR><BR><B><I>Randall Stewart &lt;rrs@cisco.com&gt;</I></B> wrote:</=
DIV>
<BLOCKQUOTE class=3Dreplbq style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #1010ff 2px solid">Brian/Salvatore:<BR><BR>Just a couple of =
comments...<BR><BR><BR>Brian F. G. Bidulock wrote:<BR>&gt; Salvatore,<BR>=
&gt; <BR>&gt; Performance tests on a clean network do not exercise SCTP's=
 strong<BR>&gt; points. SCTP was not really intended to outperform TCP on=
 a clean<BR>&gt; network with a single interface: it was intended to outp=
erform TCP on a<BR>&gt; real network with noise, failures and reconfigura=
tion. If you look at<BR>&gt; even single stream tests under noise SCTP ra=
pidly outperforms TCP in<BR>&gt; dirty networks.<BR>&gt; <BR>&gt; The SOC=
K_SEQPACKET test deviation from theoretical is probably more of a<BR>&gt;=
 software glitch in bundling, Nagle'ing and Minshall'ing in the OpenSS7<B=
R>&gt; stack more than a problem with the protocol (that was noted in the=
<BR>&gt; results).<BR><BR>I would tend to agree with Brian here.. SOCK_SE=
QPACKET is really<BR>an implementation shade.. In
 KAME other than controlling a<BR>few minor things all the same code runs=
... so it sould make<BR>very little difference....<BR><BR>&gt; <BR>&gt; T=
he SOCK_STREAM tests on a single interface did outperform TCP, however,<B=
R>&gt; note that the tests saturated the network and both are running ver=
y<BR>&gt; close to theoretical maximums.<BR>&gt; <BR>&gt; The interesting=
 thing about the OpenSS7 tests were the multiple<BR>&gt; interface tests =
on SOCK_STREAM, that showed that max throughput was a<BR>&gt; multiple of=
 the number of available interfaces. Once TCP saturates<BR>&gt; a single =
interface it has nowhere else to go. SCTP can load multiple<BR>&gt; inter=
faces to saturation.<BR>&gt; <BR>&gt; These were throughput tests. Latenc=
y tests show SCTP has comparable<BR>&gt; results to TCP (with slight impr=
ovements) on clean networks, and<BR>&gt; SCTP exhibits significant reduce=
d latencies to TCP under even as little<BR>&gt; as 5% noise. On distrupte=
d networks (50% interface failure on a 1
 or<BR>&gt; 2 second duty cycle) TCP stops and SCTP continues.<BR>&gt; <B=
R>&gt; There are a lot of good papers out there testing more real environ=
ments<BR>&gt; as well as NS-2 simulations that demonstrate the benefits o=
f SCTP over<BR>&gt; TCP for its original intended application, telephony =
signalling. Search<BR>&gt; on "SCTP Performance" and you can read for the=
 better part of a day.<BR>&gt; <BR><BR>A couple of other things:<BR><BR>a=
) The CERN tests were done on an OLD OLD OLD version of lk-sctp which<BR>=
was NOT performance tuned at all.. I am not even sure if they have<BR>com=
pleted a pass yet of tuning.. I know its on their to-do list.<BR><BR>b) I=
PERF ... which is used on both sites.. to me.. is NOT a very good<BR>benc=
hmark. SCTP sends "messages" from one end to the other. TCP sends<BR>a by=
te stream from one end to the other. You then take and use IPERF<BR>to te=
st the number of bytes streaming from one end to the other.<BR>This puts =
SCTP at a dis-advantage. Especially considering
 that MOST<BR>ALL applications send messages NOT streams of bytes. So if =
you added<BR>some additional overhead to TCP.. say framing messages at th=
e app<BR>layer.. this would slow TCP down I would wager.<BR><BR>I have al=
ways wanted to develop a variable message sender that could<BR>send a seq=
uence of variable messages and get a reading of messages<BR>per second fo=
r various sizes....<BR><BR>c) CRC32c is extremely expensive (in overall t=
erms of CPU use) on the<BR>TCP side they generally have an off-load engin=
e they can use to<BR>not even do the c-sum... so in such a case when you =
begin to<BR>get CPU limited SCTP will lag behind TCP... just on the check=
sum<BR>basis alone. Bottom line is once you get CRC32c offload SCTP will<=
BR>be able to do a bit better :-D<BR><BR>Regards<BR><BR>R<BR><BR><BR>&gt;=
 --brian<BR>&gt; <BR>&gt; On Thu, 07 Oct 2004, Salvatore Loreto wrote:<BR=
>&gt; <BR>&gt; <BR>&gt;&gt; Hi all,<BR>&gt;&gt;<BR>&gtt;&gt;<BR>&gt;&gt;<=
BR>&gt;&gt; reading the experiments on SCTP
 performance<BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&gt; LKSCTP<BR>&g=
t;&gt;<BR>&gt;&gt; [1]http://datatag.web.cern.ch/datatag/WP3/sctp/tests.h=
tm<BR>&gt;&gt;<BR>&gt;&gt; OPENSS7<BR>&gt;&gt; [2]http://www.openss7.org/=
sctp_perf.html<BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&gt; If we igno=
re the gap underline between TCP and SCTP in LKSCTP,<BR>&gt;&gt; becose d=
ue to implementation problem ( and I think now the<BR>&gt;&gt; situation =
is better, isn't it? does someone have up to date results<BR>&gt;&gt; abo=
ut?)<BR>&gt;&gt;<BR>&gt;&gt; and use just the result underline on OPENSSS=
7<BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&gt; I've some questions/dou=
bts and I'd like know you opinion about:<BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&=
gt;<BR>&gt;&gt; 1) the test SCTP with 1 interface shows a substanzial equ=
ivalent to<BR>&gt;&gt; TCP.<BR>&gt;&gt;<BR>&gt;&gt; Is it possible SCTP w=
ith 1 interface is TCP, or if you want if there<BR>&gt;&gt; is not HoL bl=
ock there are not difference between two
 protocols due<BR>&gt;&gt; to the same congestion algorithm.<BR>&gt;&gt;<=
BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&gt; 2) SCTP with 2 interface has a better=
 performance just for message<BR>&gt;&gt; size greater than 64-byte.<BR>&=
gt;&gt;<BR>&gt;&gt; But SCTP was born for Signaling transport, and signal=
ing messages are<BR>&gt;&gt; yes lager than 64-bye but not very large.<BR=
>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&gt; when in the experiment is u=
sed SOCK_SEQPACKET<BR>&gt;&gt;<BR>&gt;&gt; 3) the real start up with 1 or=
 2 interfaces are both worse than the<BR>&gt;&gt; Theoretical one. The st=
range is the reale one with 2 interfaces is<BR>&gt;&gt; worse than the re=
ale one with just one interface.<BR>&gt;&gt;<BR>&gt;&gt; 4) similar consi=
derations ( done above ) could be done also for the<BR>&gt;&gt; other pha=
ses . So SOCK_SEQPACKET seems not so good.<BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt=
;&gt;<BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&gt; regards<BR>&gt;&gt;<BR>&gt;&gt;
 sal<BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&gt; ________=
_________________________________________________________<BR>&gt;&gt;<BR>=
&gt;&gt; [3]Yahoo! Mail ti protegge gratis dalla posta indesiderata e dai=
<BR>&gt;&gt; virus!<BR>&gt;&gt;<BR>&gt;&gt;References<BR>&gt;&gt;<BR>&gt;=
&gt; 1. http://datatag.web.cern.ch/datatag/WP3/sctp/tests.htm<BR>&gt;&gt;=
 2. http://www.openss7.org/sctp_perf.html<BR>&gt;&gt; 3. http://it.rd.yah=
oo.com/mail/taglines/*http://it.mail.yahoo.com<BR>&gt; <BR>&gt; <BR>&gt;&=
gt;_______________________________________________<BR>&gt;&gt;tsvwg maili=
ng list<BR>&gt;&gt;tsvwg@ietf.org<BR>&gt;&gt;https://www1.ietf.org/mailma=
n/listinfo/tsvwg<BR>&gt; <BR>&gt; <BR>&gt; <BR><BR><BR>-- <BR>Randall Ste=
wart<BR>ITD<BR>803-345-0369 <OR>815-342-5222<BR></BLOCKQUOTE><p>
=09

=09
		<hr size=3D1><font face=3D"Arial" size=3D"2"><a href=3D"http://it.rd.ya=
hoo.com/mail/taglines/*http://it.messenger.yahoo.com"><b>Nuovo Yahoo! Mes=
senger</b></a> E' molto pi=F9 divertente: Audibles, Avatar, Webcam, Gioch=
i, Rubrica=85 Scaricalo ora!=20
</font>
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From: Janardhan Iyengar <iyengar@mail.eecis.udel.edu>
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Hi,

Just adding another pointer -

Sourabh Ladha had some (throughput) results on the KAME stack on FreeBSD
4.6 (TCP NewReno):
http://www.cis.udel.edu/~amer/PEL/poc/pdf/IPCCC2004-ladha.FTP.over.SCTP.pdf

regards,
jana

On Thu, 7 Oct 2004, Pawel Hadam wrote:

> (I am sorry for my previous, signed e-mail)
>
> Hi
>
>  > a) The CERN tests were done on an OLD OLD OLD version of lk-sctp which
>  >    was NOT performance tuned at all.. I am not even sure if they have
>  >    completed a pass yet of tuning.. I know its on their to-do list.
>
>
> I was working with linux kernel 2.6.7 with lkSCTP 1.0.1 and I obtained
> much better results.
>
>  >    I have always wanted to develop a variable message sender that could
>  >    send a sequence of variable messages and get a reading of messages
>  >    per second for various sizes....
>
>
> Sounds like my own testing apps. I wrote a sender and receiver that
> accept some customization parameters (message size, buffers).
>
> I wrote a raport about my measurements and experiments with lkSCTP on
> 1Gb network, but:
>
> 1) It is now being read by my supervisors.
> 2) It is in french :) - I really am sorry, it was written for a french
> project.
>
> Once the raport finished, I will, I hope, try to publish something in
> this subject.
>
> If anybody wants the french version, I may send him the PDF (there are a
> lot of pictures :))). Any comments are always welcome.
>
> Regards
> Pawel
>

---------------------------------------------------------------
Janardhan R. Iyengar           http://www.cis.udel.edu/~iyengar
Protocol Engineering Lab  --   CIS   --  University Of Delaware
---------------------------------------------------------------


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From tsvwg-bounces@ietf.org  Thu Oct  7 12:27:39 2004
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From: "Brian F. G. Bidulock" <bidulock@openss7.org>
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Subject: Re: [Tsvwg] sctp performance
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Janardhan,

Interesting: at 0% probability of loss, single homed KAME SCTP performed
poorer than TCP.  It looks like SCTP implementors still have something to
learn from TCP implementors in some cases.  SCTP wasn't really aimed
exclusively at greater than 3% message loss networks...  Signalling netwo=
rks
normally have far better than 1ppm BER.

--brian

On Thu, 07 Oct 2004, Janardhan Iyengar wrote:

> Hi,
>=20
> Just adding another pointer -
>=20
> Sourabh Ladha had some (throughput) results on the KAME stack on FreeBS=
D
> 4.6 (TCP NewReno):
> http://www.cis.udel.edu/~amer/PEL/poc/pdf/IPCCC2004-ladha.FTP.over.SCTP=
.pdf
>=20
> regards,
> jana
>=20
> On Thu, 7 Oct 2004, Pawel Hadam wrote:
>=20
> > (I am sorry for my previous, signed e-mail)
> >
> > Hi
> >
> >  > a) The CERN tests were done on an OLD OLD OLD version of lk-sctp w=
hich
> >  >    was NOT performance tuned at all.. I am not even sure if they h=
ave
> >  >    completed a pass yet of tuning.. I know its on their to-do list.
> >
> >
> > I was working with linux kernel 2.6.7 with lkSCTP 1.0.1 and I obtaine=
d
> > much better results.
> >
> >  >    I have always wanted to develop a variable message sender that =
could
> >  >    send a sequence of variable messages and get a reading of messa=
ges
> >  >    per second for various sizes....
> >
> >
> > Sounds like my own testing apps. I wrote a sender and receiver that
> > accept some customization parameters (message size, buffers).
> >
> > I wrote a raport about my measurements and experiments with lkSCTP on
> > 1Gb network, but:
> >
> > 1) It is now being read by my supervisors.
> > 2) It is in french :) - I really am sorry, it was written for a frenc=
h
> > project.
> >
> > Once the raport finished, I will, I hope, try to publish something in
> > this subject.
> >
> > If anybody wants the french version, I may send him the PDF (there ar=
e a
> > lot of pictures :))). Any comments are always welcome.
> >
> > Regards
> > Pawel
> >
>=20
> ---------------------------------------------------------------
> Janardhan R. Iyengar           http://www.cis.udel.edu/~iyengar
> Protocol Engineering Lab  --   CIS   --  University Of Delaware
> ---------------------------------------------------------------

--=20
Brian F. G. Bidulock    =A6 The reasonable man adapts himself to the =A6
bidulock@openss7.org    =A6 world; the unreasonable one persists in  =A6
http://www.openss7.org/ =A6 trying  to adapt the  world  to himself. =A6
                        =A6 Therefore  all  progress  depends on the =A6
                        =A6 unreasonable man. -- George Bernard Shaw =A6

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Brian F. G. Bidulock wrote:
> Janardhan,
> 
> Interesting: at 0% probability of loss, single homed KAME SCTP performed
> poorer than TCP.  It looks like SCTP implementors still have something to
> learn from TCP implementors in some cases.  SCTP wasn't really aimed
> exclusively at greater than 3% message loss networks...  Signalling networks
> normally have far better than 1ppm BER.


Hmm..

The results are interesting.. but I think your 3% figure is a
bit off..

Thats only one of the many graphs present...

Some considerations of these results:

a) Netbsd does NOT apply maxburst in TCP. So this will
    definetly influence small transfers... and if you
    note the larger the transfer the less the difference
    between the graphs.. in fact the 1Meg transfer in the
    later pages looks like possibly SCTP in all cases did
    better instead of in 2 out of 3 cases as it did in all others.

b) Note that at ALL times if you use all of sctp .. multi-streaming
    etc. it does better.. it is only one of the 3 graphs that is slower
    in small transfers.

c) 2002 was about the time I was performance optimizing KAME.. which
    actually needs to have a second pass... so I am not really sure
    which version Sourabh was actually using...


d) Considering the very very little difference between TCP and SCTP
    even in these cases .. thats pretty darn good.. especially when
    you consider that the netbsd stack as been tweaked and tuned for
    at least 10 years... and KAME was about 1 - 1.5 years when Sourabh
    did this work... all in all I think the results quite positive :-D


Note to Salvatore: Yes KAME does have H-S TCP in it.. its a configurable
option when you compile it. And no I have no papers on any results since
my focus as been in other places then that :-0



R

 >
> 
> --brian
> 
> On Thu, 07 Oct 2004, Janardhan Iyengar wrote:
> 
> 
>>Hi,
>>
>>Just adding another pointer -
>>
>>Sourabh Ladha had some (throughput) results on the KAME stack on FreeBSD
>>4.6 (TCP NewReno):
>>http://www.cis.udel.edu/~amer/PEL/poc/pdf/IPCCC2004-ladha.FTP.over.SCTP.pdf
>>
>>regards,
>>jana
>>
>>On Thu, 7 Oct 2004, Pawel Hadam wrote:
>>
>>
>>>(I am sorry for my previous, signed e-mail)
>>>
>>>Hi
>>>
>>> > a) The CERN tests were done on an OLD OLD OLD version of lk-sctp which
>>> >    was NOT performance tuned at all.. I am not even sure if they have
>>> >    completed a pass yet of tuning.. I know its on their to-do list.
>>>
>>>
>>>I was working with linux kernel 2.6.7 with lkSCTP 1.0.1 and I obtained
>>>much better results.
>>>
>>> >    I have always wanted to develop a variable message sender that could
>>> >    send a sequence of variable messages and get a reading of messages
>>> >    per second for various sizes....
>>>
>>>
>>>Sounds like my own testing apps. I wrote a sender and receiver that
>>>accept some customization parameters (message size, buffers).
>>>
>>>I wrote a raport about my measurements and experiments with lkSCTP on
>>>1Gb network, but:
>>>
>>>1) It is now being read by my supervisors.
>>>2) It is in french :) - I really am sorry, it was written for a french
>>>project.
>>>
>>>Once the raport finished, I will, I hope, try to publish something in
>>>this subject.
>>>
>>>If anybody wants the french version, I may send him the PDF (there are a
>>>lot of pictures :))). Any comments are always welcome.
>>>
>>>Regards
>>>Pawel
>>>
>>
>>---------------------------------------------------------------
>>Janardhan R. Iyengar           http://www.cis.udel.edu/~iyengar
>>Protocol Engineering Lab  --   CIS   --  University Of Delaware
>>---------------------------------------------------------------
> 
> 


-- 
Randall Stewart
ITD
803-345-0369 <or> 815-342-5222

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To: Randall Stewart <rrs@cisco.com>
Subject: Re: [Tsvwg] sctp performance
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(I am sorry for my previous, signed e-mail)

Hi

 > a) The CERN tests were done on an OLD OLD OLD version of lk-sctp which
 >    was NOT performance tuned at all.. I am not even sure if they have
 >    completed a pass yet of tuning.. I know its on their to-do list.


I was working with linux kernel 2.6.7 with lkSCTP 1.0.1 and I obtained 
much better results.

 >    I have always wanted to develop a variable message sender that could
 >    send a sequence of variable messages and get a reading of messages
 >    per second for various sizes....


Sounds like my own testing apps. I wrote a sender and receiver that 
accept some customization parameters (message size, buffers).

I wrote a raport about my measurements and experiments with lkSCTP on 
1Gb network, but:

1) It is now being read by my supervisors.
2) It is in french :) - I really am sorry, it was written for a french 
project.

Once the raport finished, I will, I hope, try to publish something in 
this subject.

If anybody wants the french version, I may send him the PDF (there are a 
lot of pictures :))). Any comments are always welcome.

Regards
Pawel

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Hi

> a) The CERN tests were done on an OLD OLD OLD version of lk-sctp which
>    was NOT performance tuned at all.. I am not even sure if they have
>    completed a pass yet of tuning.. I know its on their to-do list.

I was working with linux kernel 2.6.7 with lkSCTP 1.0.1 and I obtained 
much better results.

>    I have always wanted to develop a variable message sender that could
>    send a sequence of variable messages and get a reading of messages
>    per second for various sizes....

Sounds like my own testing apps. I wrote a sender and receiver that 
accept some customization parameters (message size, buffers).

I wrote a raport about my measurements and experiments with lkSCTP on 
1Gb network, but:

1) It is now being read by my supervisors.
2) It is in french :) - I really am sorry, it was written for a french 
project.

Once the raport finished, I will, I hope, try to publish something in 
this subject.

If anybody wants the french version, I may send him the PDF (there are a 
lot of pictures :))). Any comments are always welcome.

Regards
Pawel

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Hi

> OPENSS7
> http://www.openss7.org/sctp_perf.html

I am interested in a case with 2 interfaces.
How did they do the load-balancing between 2 interfaces?
It was on SCTP or application (iperf ???) level ???

Does anybody know?

Pawel


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From: The IESG <iesg-secretary@ietf.org>
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        tsvwg chair <mankin@psg.com>, RFC Editor <rfc-editor@rfc-editor.org>
Subject: [Tsvwg] Protocol Action: 'The Eifel Response Algorithm for TCP' to 
 Proposed Standard 
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The IESG has approved the following document:

- 'The Eifel Response Algorithm for TCP '
   <draft-ietf-tsvwg-tcp-eifel-response-06.txt> as a Proposed Standard

This document is the product of the Transport Area Working Group Working 
Group.

The IESG contact persons are Jon Peterson and Allison Mankin.

Technical Summary
 
This document recommends a way for a sender to respond to a spurious TCP 
timeout, provided that one has been detected (such detection mechanisms are ou
of scope of this document). Furthermore, it adapts the retransmission timer to 
avoid additional spurious timeouts, and restores congestion state in such a wa
that packet bursts are avoided.
 
Working Group Summary
 
The TSVWG working group supports the advancement of this document. It describe
a building block for a complete system that avoids spurious TCP 
retransmissions. Although there are a number of different philosophies for 
overall approaches to this problem, this document provides a modular component 
that could be integrated into several such approaches.
 
Protocol Quality
 
This document was reviewed for the IESG by Jon Peterson.


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Date: Thu, 7 Oct 2004 20:06:09 -0600
From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Pawel Hadam <Pawel.Hadam@imag.fr>
Subject: Re: [Tsvwg] Re: sctp performance
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Pawel,

The load balancing is performed in the SCTP implementation.  It's the sam=
e
algorithm that allowed us to acheive 20ms nominal fail-over times between
interfaces in lmco's testing.

--brian

On Thu, 07 Oct 2004, Pawel Hadam wrote:

> Hi
>=20
> > OPENSS7
> > http://www.openss7.org/sctp_perf.html
>=20
> I am interested in a case with 2 interfaces.
> How did they do the load-balancing between 2 interfaces?
> It was on SCTP or application (iperf ???) level ???
>=20
> Does anybody know?
>=20
> Pawel
>=20



> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg


--=20
Brian F. G. Bidulock    =A6 The reasonable man adapts himself to the =A6
bidulock@openss7.org    =A6 world; the unreasonable one persists in  =A6
http://www.openss7.org/ =A6 trying  to adapt the  world  to himself. =A6
                        =A6 Therefore  all  progress  depends on the =A6
                        =A6 unreasonable man. -- George Bernard Shaw =A6

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Subject: [Tsvwg] [sctp performance] english version presentation
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Hi all

Together with my report I did a presentation. It is in english, and it 
presents the same work and the same results as described in the report.

http://www-lsr.imag.fr/Les.Personnes/Pawel.Hadam/work/index.html
->
"Multihoming & SCTP"

Available as PPT and PDF

Any comments are welcome.

Regards
Pawel


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At the San Diego meeting last August, Allison closed the meeting by
talking very broadly about hosts having multiple addresses, and how this
affects the transport layer [1].  One of the potential uses is for
handling host mobility at the transport layer.  This has been done
experimentally several times in several different transport protocols,
but not within any specific unified architecture.

For example, over the years there have been mobile SCTP, TCP, and DCCP
proposals that all have used different code for supporting movement
detection (finding out when the local host has moved) and doing location
management (providing some way for the mobile node to be reached for new
connections).  These are two common problems for all transport layer
mobility protocols, which happen to be independent of the particular
transport protocol used.  It makes sense to define a common framework to
provide this functionality, so that individual transports don't have to
worry about it.

As a first step, we've begun exploring a simple architecture to provide
these two services using existing standards (eg neighbor discovery,
dynamic DNS, etc), and described our architecture in:

http://www.ietf.org/internet-drafts/draft-eddy-tlmarch-00.txt

We would appreciate any feedback from the community on this, and would
welcome the opportunity to speak breifly in Washington if the group is
interested.  We feel that while there is little new here, providing the
common framework for transport protocols to easily develop mobility
support with little replication of effort, is an important step.
Additionally, this draft attempts to provide a detailed comparison of
the architectural differences between transport layer mobility and
Mobile IP, the predominant mobility protocol which operates at a
completely different layer and has different deployment and stack-design
considerations.

-Wes

[1] See meeting notes at http://www.ietf.org/proceedings/04aug/261.htm

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This work is being explored in the Multi6 WG; it might be useful to 
check with their current state.

In particular, one of their early conclusions was that there were 
multiple layers at which this problem _could_ be fixed (transport, net, 
shim under net, etc.), but that some were more appropriate than others 
(I think they concluded that transport solutions to mobility weren't as 
useful in the general sense since they need to be recapitulated for each 
transport design).

Joe

Wesley Eddy wrote:

> At the San Diego meeting last August, Allison closed the meeting by
> talking very broadly about hosts having multiple addresses, and how this
> affects the transport layer [1].  One of the potential uses is for
> handling host mobility at the transport layer.  This has been done
> experimentally several times in several different transport protocols,
> but not within any specific unified architecture.
> 
> For example, over the years there have been mobile SCTP, TCP, and DCCP
> proposals that all have used different code for supporting movement
> detection (finding out when the local host has moved) and doing location
> management (providing some way for the mobile node to be reached for new
> connections).  These are two common problems for all transport layer
> mobility protocols, which happen to be independent of the particular
> transport protocol used.  It makes sense to define a common framework to
> provide this functionality, so that individual transports don't have to
> worry about it.
> 
> As a first step, we've begun exploring a simple architecture to provide
> these two services using existing standards (eg neighbor discovery,
> dynamic DNS, etc), and described our architecture in:
> 
> http://www.ietf.org/internet-drafts/draft-eddy-tlmarch-00.txt
> 
> We would appreciate any feedback from the community on this, and would
> welcome the opportunity to speak breifly in Washington if the group is
> interested.  We feel that while there is little new here, providing the
> common framework for transport protocols to easily develop mobility
> support with little replication of effort, is an important step.
> Additionally, this draft attempts to provide a detailed comparison of
> the architectural differences between transport layer mobility and
> Mobile IP, the predominant mobility protocol which operates at a
> completely different layer and has different deployment and stack-design
> considerations.
> 
> -Wes
> 
> [1] See meeting notes at http://www.ietf.org/proceedings/04aug/261.htm
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

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From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Pawel Hadam <Pawel.Hadam@imag.fr>
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        sctp-impl@external.cisco.com, alefelici@libero.it,
        lksctp-developers@lists.sourceforge.net,
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Subject: [Tsvwg] Re: [sctp performance] english version presentation
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Pawel,

Take a look at

  http://www.omg.org/news/meetings/workshops/RT_2003_Manual/Presentations=
/5-4_Thaker_etal.pdf

which presents failover results nominal 10ms in test situations similar
to your full mesh experiment on slide 35 of your presentation and also
show much better 1% packet loss latencies.

--brian

On Fri, 08 Oct 2004, Pawel Hadam wrote:

> Hi all
>=20
> Together with my report I did a presentation. It is in english, and it=20
> presents the same work and the same results as described in the report.
>=20
> http://www-lsr.imag.fr/Les.Personnes/Pawel.Hadam/work/index.html
> ->
> "Multihoming & SCTP"
>=20
> Available as PPT and PDF
>=20
> Any comments are welcome.
>=20
> Regards
> Pawel

--=20
Brian F. G. Bidulock    =A6 The reasonable man adapts himself to the =A6
bidulock@openss7.org    =A6 world; the unreasonable one persists in  =A6
http://www.openss7.org/ =A6 trying  to adapt the  world  to himself. =A6
                        =A6 Therefore  all  progress  depends on the =A6
                        =A6 unreasonable man. -- George Bernard Shaw =A6

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From: Brad Penoff <penoff@cs.ubc.ca>
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Subject: [Tsvwg] SCTP one-to-one style feature clarification
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Greetings,

We have noticed differing descriptions of the feature set for the 
one-to-one style in SCTP, and were hoping to receive some clarification. 
In the 3rd ed. of Stephen's "Unix Network Programming, Vol. 1", p.288 
mentions that using the SOCK_STREAM "would not take advantage of any of 
the additional features provided by SCTP except multihoming." It doesn't 
specify exactly what features are NOT taken advantage.  Are there any?

More recently, in the latest release of 
draft-ietf-tsvwg-sctpsocket-09.txt , it says on p.5 that "all new SCTP 
features can be used with both styles of socket".  This is seemingly 
contradictory to the Stephen's book.

Understanding that this is a draft, we are just curious as to what the 
current exact features are that are supported in the one-to-one style of 
LKSCTP.

Any assistance would be appreciated.  Thanks,

brad

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Subject: RE: [Tsvwg] architecture for transport layer mobility
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> -----Original Message-----
> From: Joe Touch [mailto:touch@ISI.EDU]=20
> Sent: Friday, October 08, 2004 1:11 PM
> To: weddy@grc.nasa.gov
> Cc: tsvwg@ietf.org
> Subject: Re: [Tsvwg] architecture for transport layer mobility
>=20
>=20
> This work is being explored in the Multi6 WG; it might be useful to=20
> check with their current state.

See also the CELP proposal from earlier this year, which was proposing
a framework that also included transport layer (and layer 3.5 "wedge"
layer):
http://www.brandenburg.com/specifications/draft-crocker-celp-00.txt

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From tsvwg-bounces@ietf.org  Sat Oct  9 14:04:04 2004
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Date: Sat, 9 Oct 2004 19:56:41 +0200
To: Brad Penoff <penoff@cs.ubc.ca>
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Subject: [Tsvwg] Re: [Lksctp-developers] SCTP one-to-one style feature
	clarification
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Brad,
see my comments below.
Best regards
Michael

On Oct 9, 2004, at 1:51 Uhr, Brad Penoff wrote:

> Greetings,
>
> We have noticed differing descriptions of the feature set for the 
> one-to-one style in SCTP, and were hoping to receive some 
> clarification. In the 3rd ed. of Stephen's "Unix Network Programming, 
> Vol. 1", p.288 mentions that using the SOCK_STREAM "would not take 
> advantage of any of the additional features provided by SCTP except 
> multihoming." It doesn't specify exactly what features are NOT taken 
> advantage.  Are there any?
This sentence is in the context of an existing TCP application which 
you 'port' to
SCTP by simply changing the socket() call. By not using any additional 
function you
do not use for example
- multiple streams
- unordered delivery
- special parameter settings available by SCTP socket options
>
> More recently, in the latest release of 
> draft-ietf-tsvwg-sctpsocket-09.txt , it says on p.5 that "all new SCTP 
> features can be used with both styles of socket".  This is seemingly 
> contradictory to the Stephen's book.
No it is not. The point is that you need additional code (sctp_* 
functions, SCTP socket options)
to make use of the additional functionality.
>
> Understanding that this is a draft, we are just curious as to what the 
> current exact features are that are supported in the one-to-one style 
> of LKSCTP.
>
> Any assistance would be appreciated.  Thanks,
>
> brad
>
>
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Hello Brian:

> Interesting: at 0% probability of loss, single homed KAME SCTP performed
> poorer than TCP.  It looks like SCTP implementors still have something to
> learn from TCP implementors in some cases.  SCTP wasn't really aimed
> exclusively at greater than 3% message loss networks...  Signalling netwo=
> rks
> normally have far better than 1ppm BER.

A more important result is that by exploiting SCTP's multistreaming
mechanism (a feature unavailable with TCP), FTP over KAME SCTP
was able to perform the same or slightly better than TCP at 0%
loss, and increasingly much better as loss increases.

Thanks,
Paul

Paul D. Amer, Professor
CISC Graduate Admissions Committee, Chair
101C Smith Hall
University of Delaware
Newark, DE  19716-2586
(302) 831-1944 (office)
(302) 831-8458 (fax)
http://www.cis.udel.edu/~amer/


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From tsvwg-bounces@ietf.org  Sun Oct 10 20:50:35 2004
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Subject: Re: [Tsvwg] architecture for transport layer mobility
Date: Mon, 11 Oct 2004 09:47:58 +0900
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Hello Wesley,

I think this kind of work will be very useful so to foster the further refinements
of the existing several approaches for transport and network layer mobility.
Basically, this will help service providers offer a lot of mobility solutions to users.

Two minor comments on your draft:
1) We may consider SIP (session initiation protocol) as a candidate location 
management scheme in the application layer

2) As such, a transport layer mobility may be used tegether with the other network 
or application layer mobility schemes such as Mobile IP and SIP, if necessary.
On use of MIP and mSCTP, you might refer to this I-D.
http://www.ietf.org/internet-drafts/draft-sjkoh-mobile-sctp-mobileip-04.txt


Seok J. Koh
http://protocol.knu.ac.kr/


----- Original Message ----- 
From: "Wesley Eddy" <weddy@grc.nasa.gov>
To: <tsvwg@ietf.org>
Sent: Saturday, October 09, 2004 3:37 AM
Subject: [Tsvwg] architecture for transport layer mobility

At the San Diego meeting last August, Allison closed the meeting by
talking very broadly about hosts having multiple addresses, and how this
affects the transport layer [1].  One of the potential uses is for
handling host mobility at the transport layer.  This has been done
experimentally several times in several different transport protocols,
but not within any specific unified architecture.

For example, over the years there have been mobile SCTP, TCP, and DCCP
proposals that all have used different code for supporting movement
detection (finding out when the local host has moved) and doing location
management (providing some way for the mobile node to be reached for new
connections).  These are two common problems for all transport layer
mobility protocols, which happen to be independent of the particular
transport protocol used.  It makes sense to define a common framework to
provide this functionality, so that individual transports don't have to
worry about it.

As a first step, we've begun exploring a simple architecture to provide
these two services using existing standards (eg neighbor discovery,
dynamic DNS, etc), and described our architecture in:

http://www.ietf.org/internet-drafts/draft-eddy-tlmarch-00.txt

We would appreciate any feedback from the community on this, and would
welcome the opportunity to speak breifly in Washington if the group is
interested.  We feel that while there is little new here, providing the
common framework for transport protocols to easily develop mobility
support with little replication of effort, is an important step.
Additionally, this draft attempts to provide a detailed comparison of
the architectural differences between transport layer mobility and
Mobile IP, the predominant mobility protocol which operates at a
completely different layer and has different deployment and stack-design
considerations.

-Wes

[1] See meeting notes at http://www.ietf.org/proceedings/04aug/261.htm


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To: "Joe Touch" <touch@ISI.EDU>, <weddy@grc.nasa.gov>
References: <20041008183702.GJ27317@grc.nasa.gov> <4166F459.7080907@isi.edu>
Subject: Re: [Tsvwg] architecture for transport layer mobility
Date: Mon, 11 Oct 2004 10:00:15 +0900
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Hi Joe,

> (I think they concluded that transport solutions to mobility weren't as 
> useful in the general sense since they need to be recapitulated for each 
> transport design).

The above opinion may come from the network-layer point of view.
We note that the Transport Layer Mobility (TLM) does not try to add 
a new transport protocol just for mobility. Instead, we consider the case 
in which the user's host (and its peering server) are already equipped with 
the newly developed transport protocol such as SCTP or DCCP.

The point is that in this case, the SCTP, DCCP or any new scheme could be
used for supporting the mobility, possibly with the other existing schemes 
(such as DDNS, MIP, SIP, etc)

Regards,

Seok

----- Original Message ----- 
From: "Joe Touch" <touch@ISI.EDU>
To: <weddy@grc.nasa.gov>
Cc: <tsvwg@ietf.org>
Sent: Saturday, October 09, 2004 5:11 AM
Subject: Re: [Tsvwg] architecture for transport layer mobility

This work is being explored in the Multi6 WG; it might be useful to 
check with their current state.

In particular, one of their early conclusions was that there were 
multiple layers at which this problem _could_ be fixed (transport, net, 
shim under net, etc.), but that some were more appropriate than others 
(I think they concluded that transport solutions to mobility weren't as 
useful in the general sense since they need to be recapitulated for each 
transport design).

Joe

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To: "Seok J. Koh" <sjkoh@cs.knu.ac.kr>
Subject: Re: [Tsvwg] architecture for transport layer mobility
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On Mon, Oct 11, 2004 at 10:00:15AM +0900, Seok J. Koh wrote:
> Hi Joe,
>=20
> > (I think they concluded that transport solutions to mobility weren't as=
=20
> > useful in the general sense since they need to be recapitulated for eac=
h=20
> > transport design).
>=20
> The above opinion may come from the network-layer point of view.
> We note that the Transport Layer Mobility (TLM) does not try to add=20
> a new transport protocol just for mobility. Instead, we consider the case=
=20
> in which the user's host (and its peering server) are already equipped wi=
th=20
> the newly developed transport protocol such as SCTP or DCCP.
>

Right.  The main idea behind the draft was that we boil "mobility
support" down into three components:

1) movement detection - using mechanisms like link triggers, neighbor/router
discovery, etc to determine when a new network is available, and using some
form of autoconfiguration to obtain an address and routing configuration.

2) location managment - maintaining some globally reachable "address".  In
Mobile IP, it's just the IP address.  In this draft, we use the hostname in
DNS, which may be preferable for a number of reasons.  Other proposals use
SIP, or a "host identity", etc.

3) binding updates - transport protocols are bound to some address, and
if the location management is such that the address changes with
location, then bindings need to be updated to maintain existing
connections.  Some congestion control state should be reset too for new
paths, etc.

We view #1 and #2 as independent of individual transport protocols.  These
are services that an operating system or daemon can handle in one place.  T=
he
only modifications that transports have to support in this architecture are
some form of #3, which are mechanisms like ADDIP and Change Primary in SCTP.

In short summary, our goal was to combat precisely the problem that Joe
brought up of every transport having to be modified, by making the
required modifications as light as possible, while avoiding burdening the
infrastructure with warts like HAs, FAs, and handover-smoothing goop.

-Wes

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Wesley Eddy wrote:
> On Mon, Oct 11, 2004 at 10:00:15AM +0900, Seok J. Koh wrote:
> 
>>Hi Joe,
>>
>>
>>>(I think they concluded that transport solutions to mobility weren't as 
>>>useful in the general sense since they need to be recapitulated for each 
>>>transport design).
>>
>>The above opinion may come from the network-layer point of view.
>>We note that the Transport Layer Mobility (TLM) does not try to add 
>>a new transport protocol just for mobility. Instead, we consider the case 
>>in which the user's host (and its peering server) are already equipped with 
>>the newly developed transport protocol such as SCTP or DCCP.
>>
> 
> 
> Right.  The main idea behind the draft was that we boil "mobility
> support" down into three components:
> 
> 1) movement detection - using mechanisms like link triggers, neighbor/router
> discovery, etc to determine when a new network is available, and using some
> form of autoconfiguration to obtain an address and routing configuration.
> 
> 2) location managment - maintaining some globally reachable "address".  In
> Mobile IP, it's just the IP address.  In this draft, we use the hostname in
> DNS, which may be preferable for a number of reasons.  Other proposals use
> SIP, or a "host identity", etc.
> 
> 3) binding updates - transport protocols are bound to some address, and
> if the location management is such that the address changes with
> location, then bindings need to be updated to maintain existing
> connections.  Some congestion control state should be reset too for new
> paths, etc.
> 
> We view #1 and #2 as independent of individual transport protocols.

My point is that Multi-6 views #3 as independent of transport protocols 
as well, and that they have a number of documents discussing 
alternatives for #1, #2, and #3.

> These
> are services that an operating system or daemon can handle in one place.  The
> only modifications that transports have to support in this architecture are
> some form of #3, which are mechanisms like ADDIP and Change Primary in SCTP.
> 
> In short summary, our goal was to combat precisely the problem that Joe
> brought up of every transport having to be modified, by making the
> required modifications as light as possible, while avoiding burdening the
> infrastructure with warts like HAs, FAs, and handover-smoothing goop.
> 
> -Wes

One issue is whether to burden the infrastructure with mobility support 
or to burden each transport protocol implementation. I like the idea of 
minimizing the impact on transport protocols, and certainly 
orthogonalization is a step in the right direction, but is this _really_ 
something that needs to happen at the transport layer?

Joe

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Seok J. Koh wrote:

> Hi Joe,
> 
> 
>>(I think they concluded that transport solutions to mobility weren't as 
>>useful in the general sense since they need to be recapitulated for each 
>>transport design).
> 
> The above opinion may come from the network-layer point of view.

It came from a discussion about the desire to have a single solution,
rather than to proscribe requirements for each new protocol being 
designed, as Eddy is doing. It's simpler to fix a problem once than to 
provide a set of 'rules for current and future protocol mods'.

Yes, it does come from the view that the network provides one set of 
services, and the transport provides a different set of services. And 
that the complexity of a network layer solution isn't different from 
that of a transport solution, except that a single network solution 
could be used by multiple transport protocols, and that the same is not 
true for transport solutions.

> We note that the Transport Layer Mobility (TLM) does not try to add 
> a new transport protocol just for mobility. Instead, we consider the case 
> in which the user's host (and its peering server) are already equipped with 
> the newly developed transport protocol such as SCTP or DCCP.

That works only where _each_ new protocol has support for mobility. Any
new protocol would need to incorporate mobility as well.

> The point is that in this case, the SCTP, DCCP or any new scheme could be
> used for supporting the mobility, possibly with the other existing schemes 
> (such as DDNS, MIP, SIP, etc)
> 
> Regards,
> 
> Seok

There can be interactions between schemes implemented at multiple layers
as well.

Joe



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From: Wesley Eddy <weddy@grc.nasa.gov>
To: Joe Touch <touch@isi.edu>
Subject: Re: [Tsvwg] architecture for transport layer mobility
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On Mon, Oct 11, 2004 at 10:24:03AM -0700, Joe Touch wrote:
>=20
> >We note that the Transport Layer Mobility (TLM) does not try to add=20
> >a new transport protocol just for mobility. Instead, we consider the cas=
e=20
> >in which the user's host (and its peering server) are already equipped=
=20
> >with the newly developed transport protocol such as SCTP or DCCP.
>=20
> That works only where _each_ new protocol has support for mobility. Any
> new protocol would need to incorporate mobility as well.
>=20

Not really.  Transports are build to suit classes of applications.  View
"robust to mobility" as we would other transport features like "ordered
byte stream", "congestion-controlled", "reliable datagram service",
"multi-streaming", etc.  We don't put features into transports for
applications that don't want them (at least we shouldn't).  Just because
SCTP does multi-streaming doesn't mean we now have to build it into TCP
=2E..  TCP applications figure out their own way to do that if they have
to.  Mobility could work similarly, either use a mobility-robust transport,
or do it yourself at a higher layer (if your application wants it).  In
this case, we'd only be building the binding updates into transports for
long-lived types of applications that we expect mobile hosts to use.
That may or may not be a large group of protocols.

Building mobility support too low, as in the network layer, is a
*terrible* idea, as it has unforseen consequences at higher layers that
haven't been built with mobility specifically in mind.  As evidence, TCP
congestion control can be observed to exhibit quite inappropriate
behavior when Mobile IP moves a host between two widely differing
networks.

-Wes

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From tsvwg-bounces@ietf.org  Mon Oct 11 14:01:16 2004
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From: Michael Vittrup Larsen <michael.vittrup.larsen@ericsson.com>
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Subject: [Tsvwg] New ID on efficient port randomisation
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Hi all,

A new ID on efficient port randomisation is now available at:

http://www.ietf.org/internet-drafts/draft-larsen-tsvwg-port-randomisation-00.txt

The draft describes and algorithm for port randomisation with a reasonable 
compromise between:

* Obfuscation (port randomisation is obfuscation)

* Fast convergence (i.e. port selection) in cases where many
  active sockets exist (i.e. probability of port collision is
  high).

* Ease of implementation, i.e. little code and no additional data
  structures (e.g. port free-lists) to describe current port usage.

The algorithm somewhat follows the idea of RFC1948, which describe TCP ISN 
randomisation.

Feedback appreciated.

(Sorry for the cross-post to TCPM. Port randomisation was discussed during the 
summer on TCPM, but IMHO belongs to TSVWG.)

Michael Vittrup Larsen

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From: Wesley Eddy <weddy@grc.nasa.gov>
To: Joe Touch <touch@isi.edu>
Subject: Re: [Tsvwg] architecture for transport layer mobility
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On Mon, Oct 11, 2004 at 10:21:17AM -0700, Joe Touch wrote:
>=20
> My point is that Multi-6 views #3 as independent of transport protocols=
=20
> as well, and that they have a number of documents discussing=20
> alternatives for #1, #2, and #3.
>

Yep, I think there is room for a lot of views here though, given the number
of layers and slots between layers where people have been cramming mobility
schemes.  Certainly some are more palletable than others ...
=20
> One issue is whether to burden the infrastructure with mobility support=
=20
> or to burden each transport protocol implementation. I like the idea of=
=20
> minimizing the impact on transport protocols, and certainly=20
> orthogonalization is a step in the right direction, but is this _really_=
=20
> something that needs to happen at the transport layer?
>=20

A question I think is better: Given the Internet architecture, where is
the best place to build mobility support?  Or: How can we build mobility
support into the architecture without screwing it up?

-Wes

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Wesley Eddy wrote:

> On Mon, Oct 11, 2004 at 10:24:03AM -0700, Joe Touch wrote:
> 
....
> Building mobility support too low, as in the network layer, is a
> *terrible* idea, as it has unforseen consequences at higher layers that
> haven't been built with mobility specifically in mind.  As evidence, TCP
> congestion control can be observed to exhibit quite inappropriate
> behavior when Mobile IP moves a host between two widely differing
> networks.
> 
> -Wes

That can be said of any transport layer service, given particular link 
conditions; e.g., TCP works horribly over paths that have wireless hops. 
Does that mean we need TCP to be aware of wireless hops? Or does that 
mean that we should have a link that 'plays nice' (ala the recent PILC BCP).

TCP and other transport protocols need to know addresses, and thus 
address mobility, only as an artifact of using the address as a 
connection identifer. IMO, mobility support should happen at the network 
layer; if the transport layer needs to detect and adjust to changing 
path properties, then make it more agile or deploy sub-path enhancements 
to 'play nice'.

Much of the rest of this overal direction seems to be pushing transport 
protocols to provide telco-like services (rapid handover, etc.); IMO, 
being different from telco-service is the defining useful property of 
the Internet, not something to be 'patched over'.

Joe




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Date: Tue, 12 Oct 2004 08:58:42 +0200 (CEST)
From: Salvatore Loreto <loretosal@yahoo.it>
Subject: Re: [Tsvwg] architecture for transport layer mobility
To: Joe Touch <touch@ISI.EDU>, weddy@grc.nasa.gov
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Hello
=20
the mobility support is becoming a hot issue.
The mobility is always more request from people,
but when we think mobility in Internet=20
=20
IMO we can't forget
1) the telco approach: continuity in communication or if you want in sess=
ion, especially for real time application
2) the congestion problem... if we change network, maybe we change the th=
roughput too, the link capacity and other...
3) the security problems... there are some good papers about all the poss=
ible intrusion made take advantages from mobility
=20
=20
/Sal


Joe Touch <touch@ISI.EDU> wrote:


Wesley Eddy wrote:

> On Mon, Oct 11, 2004 at 10:24:03AM -0700, Joe Touch wrote:
>=20
....
> Building mobility support too low, as in the network layer, is a
> *terrible* idea, as it has unforseen consequences at higher layers that
> haven't been built with mobility specifically in mind. As evidence, TCP
> congestion control can be observed to exhibit quite inappropriate
> behavior when Mobile IP moves a host between two widely differing
> networks.
>=20
> -Wes

That can be said of any transport layer service, given particular link=20
conditions; e.g., TCP works horribly over paths that have wireless hops.=20
Does that mean we need TCP to be aware of wireless hops? Or does that=20
mean that we should have a link that 'plays nice' (ala the recent PILC BC=
P).

TCP and other transport protocols need to know addresses, and thus=20
address mobility, only as an artifact of using the address as a=20
connection identifer. IMO, mobility support should happen at the network=20
layer; if the transport layer needs to detect and adjust to changing=20
path properties, then make it more agile or deploy sub-path enhancements=20
to 'play nice'.

Much of the rest of this overal direction seems to be pushing transport=20
protocols to provide telco-like services (rapid handover, etc.); IMO,=20
being different from telco-service is the defining useful property of=20
the Internet, not something to be 'patched over'.

Joe





> ATTACHMENT part 1.2 application/pgp-signature name=3Dsignature.asc
_______________________________________________
tsvwg mailing list
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			=09
---------------------------------
Nuovo Yahoo! Messenger E' molto pi=F9 divertente: Audibles, Avatar, Webca=
m, Giochi, Rubrica=85 Scaricalo ora!=20
--0-1766325685-1097564322=:7867
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<DIV>
<DIV>Hello</DIV>
<DIV>&nbsp;</DIV>
<DIV>the mobility support is becoming a hot issue.</DIV>
<DIV>The mobility is always more request from people,</DIV>
<DIV>but when we think mobility in Internet </DIV>
<DIV>&nbsp;</DIV>
<DIV>IMO we can't forget</DIV>
<DIV>1)&nbsp;the telco approach: continuity in communication or if you wa=
nt in session, especially for real time application</DIV>
<DIV>2) the congestion problem... if we change network, maybe we change t=
he throughput too, the link capacity and other...</DIV>
<DIV>3) the security problems... there are some good papers about all the=
 possible intrusion made take advantages from mobility</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>/Sal<BR><BR><BR><B><I>Joe Touch &lt;touch@ISI.EDU&gt;</I></B> wrote:=
</DIV>
<BLOCKQUOTE class=3Dreplbq style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #1010ff 2px solid"><BR><BR>Wesley Eddy wrote:<BR><BR>&gt; On=
 Mon, Oct 11, 2004 at 10:24:03AM -0700, Joe Touch wrote:<BR>&gt; <BR>....=
<BR>&gt; Building mobility support too low, as in the network layer, is a=
<BR>&gt; *terrible* idea, as it has unforseen consequences at higher laye=
rs that<BR>&gt; haven't been built with mobility specifically in mind. As=
 evidence, TCP<BR>&gt; congestion control can be observed to exhibit quit=
e inappropriate<BR>&gt; behavior when Mobile IP moves a host between two =
widely differing<BR>&gt; networks.<BR>&gt; <BR>&gt; -Wes<BR><BR>That can =
be said of any transport layer service, given particular link <BR>conditi=
ons; e.g., TCP works horribly over paths that have wireless hops. <BR>Doe=
s that mean we need TCP to be aware of wireless hops? Or does that <BR>me=
an that we should have a link that 'plays nice' (ala the recent PILC BCP)=
.<BR><BR>TCP and other transport protocols need to
 know addresses, and thus <BR>address mobility, only as an artifact of us=
ing the address as a <BR>connection identifer. IMO, mobility support shou=
ld happen at the network <BR>layer; if the transport layer needs to detec=
t and adjust to changing <BR>path properties, then make it more agile or =
deploy sub-path enhancements <BR>to 'play nice'.<BR><BR>Much of the rest =
of this overal direction seems to be pushing transport <BR>protocols to p=
rovide telco-like services (rapid handover, etc.); IMO, <BR>being differe=
nt from telco-service is the defining useful property of <BR>the Internet=
, not something to be 'patched over'.<BR><BR>Joe<BR><BR><BR><BR><BR><BR>&=
gt; ATTACHMENT part 1.2 application/pgp-signature name=3Dsignature.asc<BR=
>_______________________________________________<BR>tsvwg mailing list<BR=
>tsvwg@ietf.org<BR>https://www1.ietf.org/mailman/listinfo/tsvwg<BR></BLOC=
KQUOTE></DIV><p>
=09

=09
		<hr size=3D1><font face=3D"Arial" size=3D"2"><a href=3D"http://it.rd.ya=
hoo.com/mail/taglines/*http://it.messenger.yahoo.com"><b>Nuovo Yahoo! Mes=
senger</b></a> E' molto pi=F9 divertente: Audibles, Avatar, Webcam, Gioch=
i, Rubrica=85 Scaricalo ora!=20
</font>
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Hi Brian/all,

Sorry about the late reply - I was on a paper deadline.

My thoughts have been voiced by Randy and Prof. Amer. The experiments were
performed about 2 years ago, and performance tuning has happened since
then in the KAME stack.

> It looks like SCTP implementors still have something to learn from TCP
> implementors in some cases.

I'm not sure much else can replace performance tuning that happens with
time and usage. That a new SCTP could match up to an aged TCP stack is
quite good, I think.

regards,
jana

---------------------------------------------------------------
Janardhan R. Iyengar           http://www.cis.udel.edu/~iyengar
Protocol Engineering Lab  --   CIS   --  University Of Delaware
---------------------------------------------------------------


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Sorry for reviving an old thread, but I'm just getting caught up on this
list.

On Thu, 7 Oct 2004, Brian F. G. Bidulock wrote:

> The interesting thing about the OpenSS7 tests were the multiple
> interface tests on SOCK_STREAM, that showed that max throughput was a
> multiple of the number of available interfaces.  Once TCP saturates
> a single interface it has nowhere else to go.  SCTP can load multiple
> interfaces to saturation.

I am curious about SCTP is able to increase throughput as the number of
interfaces increase. Jana has showed that simply using all paths with SCTP
is counter-productive. The transport has to be "smart" about getting
around different issues when doing Concurrent Multipath Transfer (CMT).
Jana has found several issues with "naive" CMT, but has done a nice
job of addressing them one by one. He's still in the process of tackling
more of these issues.

http://www.cis.udel.edu/~iyengar/publications/2004.spects.iyengar.pdf
http://www.cis.udel.edu/~iyengar/publications/2004.icon.iyengar.pdf

~armando

0--                                              --0
| Armando L. Caro Jr.  |  Protocol Engineering Lab |
| www.armandocaro.net  |    University of Delaware |
0--                                              --0

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On Fri, 8 Oct 2004, Brian F. G. Bidulock wrote:

> Take a look at
>
>   http://www.omg.org/news/meetings/workshops/RT_2003_Manual/Presentations/5-4_Thaker_etal.pdf
>
> which presents failover results nominal 10ms in test situations similar
> to your full mesh experiment on slide 35 of your presentation and also
> show much better 1% packet loss latencies.

10ms failover?! Clearly this is a special case network and not operating
over a WAN.

~armando

0--                                              --0
| Armando L. Caro Jr.  |  Protocol Engineering Lab |
| www.armandocaro.net  |    University of Delaware |
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Subject: [Tsvwg] Re: [sctp performance] english version presentation
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My guess is RTO.Min must be set down to about
1ms and RTO.Max to some other low value...

Of course this is fine for special case networks.. not the
big-I :-D

R


Armando L. Caro Jr. wrote:
> On Fri, 8 Oct 2004, Brian F. G. Bidulock wrote:
> 
> 
>>Take a look at
>>
>>  http://www.omg.org/news/meetings/workshops/RT_2003_Manual/Presentations/5-4_Thaker_etal.pdf
>>
>>which presents failover results nominal 10ms in test situations similar
>>to your full mesh experiment on slide 35 of your presentation and also
>>show much better 1% packet loss latencies.
> 
> 
> 10ms failover?! Clearly this is a special case network and not operating
> over a WAN.
> 
> ~armando
> 
> 0--                                              --0
> | Armando L. Caro Jr.  |  Protocol Engineering Lab |
> | www.armandocaro.net  |    University of Delaware |
> 0--                                              --0
> 


-- 
Randall Stewart
ITD
803-345-0369 <or> 815-342-5222

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Subject: [Tsvwg] Re: [sctp performance] english version presentation
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Armando,

On Wed, 13 Oct 2004, Armando L. Caro Jr. wrote:

> On Fri, 8 Oct 2004, Brian F. G. Bidulock wrote:
>=20
> > Take a look at
> >
> >   http://www.omg.org/news/meetings/workshops/RT_2003_Manual/Presentat=
ions/5-4_Thaker_etal.pdf
> >
> > which presents failover results nominal 10ms in test situations simil=
ar
> > to your full mesh experiment on slide 35 of your presentation and als=
o
> > show much better 1% packet loss latencies.
>=20
> 10ms failover?! Clearly this is a special case network and not operatin=
g
> over a WAN.

The presentation details the network and test harness.  I believe it was =
ethernet
connected directly through the harness.  RTT for the network was substati=
ally below
10ms.  The RTO.Min was set to zero (which with Linux 2.4 timer granularit=
y is 10ms).
I would expect 1ms failovers for 1000Hz kernel clocks.

For a WAN, the experiment would lead one to believe that failover times o=
f one RTT
can be acheived.  10ms should get you across a continent.  100ms should g=
et you around
the planet.  1s, part way to the moon.

I suppose the point is that, when implemented with performance in mind, S=
CTP at the
transport layer can perform better than many Ethernet bonding solutions a=
t the link
layer.

--brian


>=20
> ~armando
>=20
> 0--                                              --0
> | Armando L. Caro Jr.  |  Protocol Engineering Lab |
> | www.armandocaro.net  |    University of Delaware |
> 0--                                              --0

--=20
Brian F. G. Bidulock    =A6 The reasonable man adapts himself to the =A6
bidulock@openss7.org    =A6 world; the unreasonable one persists in  =A6
http://www.openss7.org/ =A6 trying  to adapt the  world  to himself. =A6
                        =A6 Therefore  all  progress  depends on the =A6
                        =A6 unreasonable man. -- George Bernard Shaw =A6

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From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: "Armando L. Caro Jr." <me@armandocaro.net>
Subject: Re: [Tsvwg] sctp performance
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Armando,

As I'm sure you know, the OpenSS7 SCTP implementation has been using a
weighted version of the algorithms from the second paper since 2001.  It
also uses a fast restoration technique for restoring suspect
destinations since 2001.  The implementation also uses a (patentable) FR
algorithm that is far simpler than the split-FR presented in the first
paper.  These allow the implementation to exhibit multiplicative
throughput and single RTO fail-over simultaneously.  Neither algorithm
violates AIMD, FR or max burst rules.  They are in fact more
conservative.

I believe I sent you this weighted algorithm some years ago and said it
was worthy of attention. ;)

Perhaps more interesting than just the pure multiplicative increase is
the way that the implementation smoothly adapts and moves out of the way
of a single-interface TCP load resulting in maximization of both
throughputs.  Unfortunately, those results haven't been presented yet.

--brian

On Wed, 13 Oct 2004, Armando L. Caro Jr. wrote:

> Sorry for reviving an old thread, but I'm just getting caught up on thi=
s
> list.
>=20
> On Thu, 7 Oct 2004, Brian F. G. Bidulock wrote:
>=20
> > The interesting thing about the OpenSS7 tests were the multiple
> > interface tests on SOCK_STREAM, that showed that max throughput was a
> > multiple of the number of available interfaces.  Once TCP saturates
> > a single interface it has nowhere else to go.  SCTP can load multiple
> > interfaces to saturation.
>=20
> I am curious about SCTP is able to increase throughput as the number of
> interfaces increase. Jana has showed that simply using all paths with S=
CTP
> is counter-productive. The transport has to be "smart" about getting
> around different issues when doing Concurrent Multipath Transfer (CMT).
> Jana has found several issues with "naive" CMT, but has done a nice
> job of addressing them one by one. He's still in the process of tacklin=
g
> more of these issues.
>=20
> http://www.cis.udel.edu/~iyengar/publications/2004.spects.iyengar.pdf
> http://www.cis.udel.edu/~iyengar/publications/2004.icon.iyengar.pdf
>=20
> ~armando
>=20
> 0--                                              --0
> | Armando L. Caro Jr.  |  Protocol Engineering Lab |
> | www.armandocaro.net  |    University of Delaware |
> 0--                                              --0

--=20
Brian F. G. Bidulock    =A6 The reasonable man adapts himself to the =A6
bidulock@openss7.org    =A6 world; the unreasonable one persists in  =A6
http://www.openss7.org/ =A6 trying  to adapt the  world  to himself. =A6
                        =A6 Therefore  all  progress  depends on the =A6
                        =A6 unreasonable man. -- George Bernard Shaw =A6

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From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Randall Stewart <randall@stewart.chicago.il.us>
Subject: Re: [Tsvwg] sctp performance
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Randall,

Armando listed two papers:

1st paper:  http://www.cis.udel.edu/~iyengar/publications/2004.spects.iye=
ngar.pdf
2nd paper:  http://www.cis.udel.edu/~iyengar/publications/2004.icon.iyeng=
ar.pdf

--brian

On Thu, 14 Oct 2004, Randall Stewart wrote:

> Brian F. G. Bidulock wrote:
> > Armando,
> >=20
> > As I'm sure you know, the OpenSS7 SCTP implementation has been using =
a
> > weighted version of the algorithms from the second paper since 2001.=20
>=20
> Question....
>=20
> What is the "second paper" ... are you refering to some of the
> work Armando and Jana have been doing at the U-D or some other
> paper?
>=20
> R
> --
> Randall Stewart

--=20
Brian F. G. Bidulock    =A6 The reasonable man adapts himself to the =A6
bidulock@openss7.org    =A6 world; the unreasonable one persists in  =A6
http://www.openss7.org/ =A6 trying  to adapt the  world  to himself. =A6
                        =A6 Therefore  all  progress  depends on the =A6
                        =A6 unreasonable man. -- George Bernard Shaw =A6

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On Wed, 13 Oct 2004, Brian F. G. Bidulock wrote:

> As I'm sure you know, the OpenSS7 SCTP implementation has been using a
> weighted version of the algorithms from the second paper since 2001.  It
> also uses a fast restoration technique for restoring suspect
> destinations since 2001.  The implementation also uses a (patentable) FR
> algorithm that is far simpler than the split-FR presented in the first
> paper.  These allow the implementation to exhibit multiplicative
> throughput and single RTO fail-over simultaneously.  Neither algorithm
> violates AIMD, FR or max burst rules.  They are in fact more
> conservative.
>
> I believe I sent you this weighted algorithm some years ago and said it
> was worthy of attention. ;)

To be honest, I didn't remember you sending me an algorithm. I dug through
my old mails and found a private discussion we had in Nov 2001. I found
the weighted algorithm that you're talking about, but the algorithm you
sent me was for selecting a destination to retransmit to and failover to.
Since the email was sent privately, I don't want to post the algorithm in
this public forum without your permission. But in any case, the
algorithm/selection process looks reasonable, but I don't see how it
applies to Concurrent Multipath Transfer (CMT).

Maybe I am misunerstanding you, and you are not doing CMT? Are you saying
that additional interfaces multiplies the throughput performance because
your failover algorithm is so agressive? You are advocating using PMR=0,
RTO.Min=0-10ms, etc. I have also done work that comes to the conclusion
that PMR=0 provides the best performance
(http://www.armandocaro.net/papers/2004.TR2005-04.acaro.pdf), but an
RTO.Min=0-10ms would only work on special case networks (eg, SS7)  and not
the Internet.

~armando

0--                                              --0
| Armando L. Caro Jr.  |  Protocol Engineering Lab |
| www.armandocaro.net  |    University of Delaware |
0--                                              --0

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Hi Brian/all,

(I trimmed the Cc list. And I apologize for posting on several lists. But
this discussion started on all these lists. Maybe it could move to
one/more lists as per interest?)

The papers that are referred to in this thread are:
http://www.cis.udel.edu/~iyengar/publications/2004.spects.iyengar.pdf
http://www.cis.udel.edu/~iyengar/publications/2004.icon.iyengar.pdf

I add another paper here, which is currently under submission and is a
continuation of CMT work:
http://www.cis.udel.edu/~iyengar/publications/2005.icdcs.iyengar.pdf

> As I'm sure you know, the OpenSS7 SCTP implementation has been using a
> weighted version of the algorithms from the second paper since 2001.

[...]

> The implementation also uses a (patentable) FR algorithm that is far
> simpler than the split-FR presented in the first paper.

I certainly did not know that you already implemented these algorithms. I
don't think any of the co-authors on the papers knew either. Could you
post the algos, or send a reference that we could look up?

If your FR algo is far simpler, I would certainly like to look at it,
because I could not get the SFR-CACC algo to be simpler than it is. Well,
it is simpler than it _was_, but I do not know how to make it any simpler.
Also, the first paper presents two other algos - the CUC and DAC algos. At
least the CUC algorithm is, IMHO, necessary to see expected throughput, at
least if the paths in question have different end-to-end delays. Did  you
implement something like CUC too?

You say the algo is "patentable" - I do not understand what that means.
Did you mean to say the algo is "patentED"? So, can you post the algo?

Note: The ID (draft-iyengar-sctp-cacc-xx.txt) which was first published in
Feb 2002 has expired now, but I'm planning to resubmit it in a new form
and with significant additions/modifications. I wanted to spend more time
looking at changeover/CMT concerns with SCTP before putting something out
as an ID.

regards,
jana

---------------------------------------------------------------
Janardhan R. Iyengar           http://www.cis.udel.edu/~iyengar
Protocol Engineering Lab  --   CIS   --  University Of Delaware
---------------------------------------------------------------



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Brian F. G. Bidulock wrote:
> Armando,
> 
> As I'm sure you know, the OpenSS7 SCTP implementation has been using a
> weighted version of the algorithms from the second paper since 2001. 

Question....

What is the "second paper" ... are you refering to some of the
work Armando and Jana have been doing at the U-D or some other
paper?

R
--
Randall Stewart

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From: giuseppe de marco <gdemarco@unisa.it>
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Anyone know if the problem of source addressing in SCTP has been ever
solved?
Giuseppe

-- 
Giuseppe De Marco, Ph.D
Departement of Information and Electrical Engineering (DIIIE)
University of Salerno - Italy
via ponte don Melillo Fisciano (SA)
tel: +39 089 964012
mail: gdemarco@unisa.it



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giuseppe de marco wrote:
> Anyone know if the problem of source addressing in SCTP has been ever
> solved?
> Giuseppe
> 
And what is the 'source addressing problem' in SCTP to be solved?

R

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--NextPart

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

	Title		: Stream Control Transmission Protocol (SCTP) Implementer's Guide
	Author(s)	: R. Stewart, et al.
	Filename	: draft-ietf-tsvwg-sctpimpguide-12.txt
	Pages		: 96
	Date		: 2004-10-18
	
This document contains a compilation of all defects found up until
   the publishing of this document for the Stream Control Transmission
   Protocol (SCTP) RFC2960 [6].  These defects may be of an editorial or
   technical nature.  This document may be thought of as a companion
   document to be used in the implementation of SCTP to clarify errors
   in the original SCTP document.

   This document updates RFC2960 [6] and text within this document
   supersedes the text found in RFC2960 [6].

A URL for this Internet-Draft is:
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From tsvwg-bounces@ietf.org  Wed Oct 20 01:06:10 2004
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From: Brad Penoff <penoff@cs.ubc.ca>
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Greetings,

We have been reading about the ability to perform chunk bundling within 
SCTP, specifically the coalescing of multiple DATA chunks into a single 
SCTP packet.  It is said that this is an option by the application 
programmer; is the only control simply the ability to turn off/on Nagle's 
algorithm (SCTP_NODELAY), or is there a finer degree of control (i.e. can 
we pack a packet with specific DATA chunks) ?  If there are other ways, 
what are they? If SCTP_NODELAY is the only option, is this saying that 
chunk bundling is applicable to an entire association rather than per 
stream?

Thanks,
brad

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This is a request to give a talk on our resurrected and
significantly-revised quickstart draft at tsvwg in DC.

(I will have to leave IETF this time on Thursday afternoon,
so I will see if someone else can give the presentation if
tsvwg is being held after that...)

The draft:
http://www.ietf.org/internet-drafts/draft-amit-quick-start-03.txt
Quick-Start for TCP and IP. 
Amit Jain, Sally Floyd, Mark Allman, and Pasi Sarolahti. 

We will also be asking if QuickStart can be added to tsvwg
as a work item.

Many thanks.
- Sally




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A revised version of the subject document has been placed on the IETF site.

For those of you who haven't been following this topic, MLEF allows the 
application of different handling to different priority of Expedited Forwarding 
traffic so that higher priority traffic is less affected by congestion which 
results is packet discard.

It is not intended as a replacement for congestion management via some sort 
of call admission or call manager but complements this option. It must be used 
in conjuction with these other techniques to provide a complete QoS solution.

MLEF operates at the packet level, not the call level using various Diffserv 
codepoints to distinguish among the various priority of traffic and provides 
for different discard probabalities, as Assured Forwarding does.

It is currently the authors' intention to request that this be published as 
an individual submission as an Experimental RFC.

Comments specific to the described mechanism would be appreciated.

Mike Pierce
Artel



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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>A revised ver=
sion of the subject document has been placed on the IETF site.
<BR>
<BR>For those of you who haven't been following this topic, MLEF allows the=20=
application of different handling to different priority of Expedited Forward=
ing traffic so that higher priority traffic is less affected by congestion w=
hich results is packet discard.
<BR>
<BR>It is not intended as a replacement for congestion management via some s=
ort of call admission or call manager but complements this option. It must b=
e used in conjuction with these other techniques to provide a complete QoS s=
olution.
<BR>
<BR>MLEF operates at the packet level, not the call level using various Diff=
serv codepoints to distinguish among the various priority of traffic and pro=
vides for different discard probabalities, as Assured Forwarding does.
<BR>
<BR>It is currently the authors' intention to request that this be published=
 as an individual submission as an Experimental RFC.
<BR>
<BR>Comments specific to the described mechanism would be appreciated.
<BR>
<BR>Mike Pierce
<BR>Artel
<BR>
<BR></FONT></HTML>

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From tsvwg-bounces@ietf.org  Thu Oct 21 07:40:35 2004
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Brad,

To put it in a simple way, SCTP DATA chunk bundling is an opportunistic procedure - where 
the SCTP sender (not the application above) always tries to bundle as many ready-to-go DATA 
chunks as the current congestion window allows. There are only very few rules, such as that 
chunks marked for retrans will go first, etc. Turning SCTP_NODELAY off/on by the app simply 
increases/decreases the __opportunity__ for the SCTP sender to bundle, and that's all.

The idea for this is to maximize the bandwidth efficiency. I am afraid if we create finer 
controls to this bundling procedure (such as dis-allowing bundling across streams), we will 
be less successful in maximizing the bandwidth efficiency, which will mean poorer achievable 
throughput.

regards,
-Qiaobing

Brad Penoff wrote:

> Greetings,
> 
> We have been reading about the ability to perform chunk bundling within 
> SCTP, specifically the coalescing of multiple DATA chunks into a single 
> SCTP packet.  It is said that this is an option by the application 
> programmer; is the only control simply the ability to turn off/on 
> Nagle's algorithm (SCTP_NODELAY), or is there a finer degree of control 
> (i.e. can we pack a packet with specific DATA chunks) ?  If there are 
> other ways, what are they? If SCTP_NODELAY is the only option, is this 
> saying that chunk bundling is applicable to an entire association rather 
> than per stream?
> 
> Thanks,
> brad
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 


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From tsvwg-bounces@ietf.org  Thu Oct 21 08:23:21 2004
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Hi, Sally,

This slot request is fine: there's been significant past
discussion and interest on this draft.

I assume this would be a proposal for an Experimental work
item?  

Folks, please review.


Allison

> This is a request to give a talk on our resurrected and
> significantly-revised quickstart draft at tsvwg in DC.
> 
> (I will have to leave IETF this time on Thursday afternoon,
> so I will see if someone else can give the presentation if
> tsvwg is being held after that...)
> 
> The draft:
> http://www.ietf.org/internet-drafts/draft-amit-quick-start-03.txt
> Quick-Start for TCP and IP. 
> Amit Jain, Sally Floyd, Mark Allman, and Pasi Sarolahti. 
> 
> We will also be asking if QuickStart can be added to tsvwg
> as a work item.
> 

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From tsvwg-bounces@ietf.org  Thu Oct 21 08:25:43 2004
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>I assume this would be a proposal for an Experimental work
>item?  

We have done a lot of work on it, and what we would *really*
like would be for QuickStart to end up as Proposed Standard,
and to in fact really happen.  But it would be a long path from
here to there, including developing rough consensus in the working
group, the IESG, and the IETF.  One possibility would be a
step through Experimental on the way to Proposed, as happened
with ECN, NewReno, and the like.

- Sally

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Subject: [Tsvwg] New version of RSVP Bandwidth Reduction ID posted
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All

Yesterday draft-polk-tsvwg-rsvp-bw-reduction-00.txt was posted.

This is a revision to our original ID given last meeting about bandwidth 
reduction of RSVP flows. Previously, this ID focused on reducing aggregate 
flows. This version opens the scope up to individual and encrypted 
reservation flows as well.

This ID can be found at:

http://www.ietf.org/internet-drafts/draft-polk-tsvwg-rsvp-bw-reduction-00.txt

Comments are welcome

cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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Subject: [Tsvwg] agenda item for tsvwg in DC? 
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I would also like to request a slot to discuss the new version of 
"Aggregation of RSVP Reservations over MPLS TE/DS-TE Tunnels" 
(draft-lefaucheur-rsvp-dste-01.txt). We've revised the draft in response to 
feedback received during and after the last IETF.

 The significant changes from the previous (-00) version of this draft
   are:
      - added discussion of the relationship to RFC 2746 [RSVP-TUN]
      - added discussion of mapping policy at aggregator
      - added discussion of "RSVP proxy" behavior in conjunction with
        the aggregation scheme described here
      - added discussion on TTL processing on Deaggregator


Bruce


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	21 Oct 2004 09:10:38 EDT."
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Bruce,

Thanks for the email describing the updates of rsvp-dste.  With 
this new material, in response to the previous discussion, a tsvwg
slot makes a lot of sense.

> 
> I would also like to request a slot to discuss the new version of 
> "Aggregation of RSVP Reservations over MPLS TE/DS-TE Tunnels" 
> (draft-lefaucheur-rsvp-dste-01.txt). We've revised the draft in response to 
> feedback received during and after the last IETF.
> 
>  The significant changes from the previous (-00) version of this draft
>    are:
>       - added discussion of the relationship to RFC 2746 [RSVP-TUN]
>       - added discussion of mapping policy at aggregator
>       - added discussion of "RSVP proxy" behavior in conjunction with
>         the aggregation scheme described here
>       - added discussion on TTL processing on Deaggregator
> 
> 



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All:

A point and a question..

  We probably need to think about last calling this and closing
  up the document. I think, based on the last inter-op, we have
  things pretty well complete as far as clearifications go..


The question..

One thing that has off and on nagged at me is the
following wording from RFC2960 ...

 From 8.4:

    The receiver of an OOTB packet MUST do the following:

    1) If the OOTB packet is to or from a non-unicast address, silently
       discard the packet.  Otherwise,
*******************************************************************
and From 11.2.4.1 Flooding

   -  not responding to unexpected packets sent to non-unicast
       addresses.

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

Now I realize that these strictures were put there to keep us
from setting up an association with a smurf like attack
i.e. src from X (the victim) and send an INIT
broadcast  255.255.255.255... or even multi-cast...

But I wonder, is the wording too strict here??

In the future if you are dealing with "Anycast" .. this is
NON-Unicast... and so you have to discard it.. but I could
see sometime down the road when we would WANT to respond
when sent to an ANYCAST address.... right now V6 limits that to
routers... some day anycast may go beyond routers...
And SCTP won't be able to use it due to the above
wording...


I could see...

From:IP-A ---INIT(IP-A) -->To:Anycast Address

<----To:IP-A ---INIT-ACK(IP-Z,AnyCast)from IP-Z
--From:IP-A-COOKIE-ECHO------->To IP-Z
  From:IP-A<------COOKIE-ACK+ASCONF(Delete Anycast)----From:IP-Z

Working just great...and making it so you could
establish an SCTP association with an ANYCAST address.... neat
idea.. but right now NOT allowed...

Soo.. should we just cross that bridge when we come
to it... or should we attempt to loosen up the wording
for anycast??

R


Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> This draft is a work item of the Transport Area Working Group Working Group of the IETF.
> 
> 	Title		: Stream Control Transmission Protocol (SCTP) Implementer's Guide
> 	Author(s)	: R. Stewart, et al.
> 	Filename	: draft-ietf-tsvwg-sctpimpguide-12.txt
> 	Pages		: 96
> 	Date		: 2004-10-18
> 	
> This document contains a compilation of all defects found up until
>    the publishing of this document for the Stream Control Transmission
>    Protocol (SCTP) RFC2960 [6].  These defects may be of an editorial or
>    technical nature.  This document may be thought of as a companion
>    document to be used in the implementation of SCTP to clarify errors
>    in the original SCTP document.
> 
>    This document updates RFC2960 [6] and text within this document
>    supersedes the text found in RFC2960 [6].
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-sctpimpguide-12.txt
> 
> To remove yourself from the I-D Announcement list, send a message to 
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> 
> 
> Internet-Drafts are also available by anonymous FTP. Login with the username
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> type "cd internet-drafts" and then
> 	"get draft-ietf-tsvwg-sctpimpguide-12.txt".
> 
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html 
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> 
> Internet-Drafts can also be obtained by e-mail.
> 
> Send a message to:
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> 	
> NOTE:	The mail server at ietf.org can return the document in
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> 	feature, insert the command "ENCODING mime" before the "FILE"
> 	command.  To decode the response(s), you will need "munpack" or
> 	a MIME-compliant mail reader.  Different MIME-compliant mail readers
> 	exhibit different behavior, especially when dealing with
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-- 
Randall Stewart
803-345-0369 <or> 815-342-5222(cell)

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From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Randall Stewart <randall@stewart.chicago.il.us>
Subject: Re: [Tsvwg] I-D ACTION:draft-ietf-tsvwg-sctpimpguide-12.txt
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Randall,

Loosen 1) to SHOULD.

--brian

On Fri, 22 Oct 2004, Randall Stewart wrote:

> All:
>=20
> A point and a question..
>=20
>   We probably need to think about last calling this and closing
>   up the document. I think, based on the last inter-op, we have
>   things pretty well complete as far as clearifications go..
>=20
>=20
> The question..
>=20
> One thing that has off and on nagged at me is the
> following wording from RFC2960 ...
>=20
>  From 8.4:
>=20
>     The receiver of an OOTB packet MUST do the following:
>=20
>     1) If the OOTB packet is to or from a non-unicast address, silently
>        discard the packet.  Otherwise,
> *******************************************************************
> and From 11.2.4.1 Flooding
>=20
>    -  not responding to unexpected packets sent to non-unicast
>        addresses.
>=20
> *******************************************************************
>=20
> Now I realize that these strictures were put there to keep us
> from setting up an association with a smurf like attack
> i.e. src from X (the victim) and send an INIT
> broadcast  255.255.255.255... or even multi-cast...
>=20
> But I wonder, is the wording too strict here??
>=20
> In the future if you are dealing with "Anycast" .. this is
> NON-Unicast... and so you have to discard it.. but I could
> see sometime down the road when we would WANT to respond
> when sent to an ANYCAST address.... right now V6 limits that to
> routers... some day anycast may go beyond routers...
> And SCTP won't be able to use it due to the above
> wording...
>=20
>=20
> I could see...
>=20
> From:IP-A ---INIT(IP-A) -->To:Anycast Address
>=20
> <----To:IP-A ---INIT-ACK(IP-Z,AnyCast)from IP-Z
> --From:IP-A-COOKIE-ECHO------->To IP-Z
>   From:IP-A<------COOKIE-ACK+ASCONF(Delete Anycast)----From:IP-Z
>=20
> Working just great...and making it so you could
> establish an SCTP association with an ANYCAST address.... neat
> idea.. but right now NOT allowed...
>=20
> Soo.. should we just cross that bridge when we come
> to it... or should we attempt to loosen up the wording
> for anycast??
>=20
> R
>=20
>=20
> Internet-Drafts@ietf.org wrote:
> > A New Internet-Draft is available from the on-line Internet-Drafts di=
rectories.
> > This draft is a work item of the Transport Area Working Group Working=
 Group of the IETF.
> >=20
> > 	Title		: Stream Control Transmission Protocol (SCTP) Implementer's G=
uide
> > 	Author(s)	: R. Stewart, et al.
> > 	Filename	: draft-ietf-tsvwg-sctpimpguide-12.txt
> > 	Pages		: 96
> > 	Date		: 2004-10-18
> > =09
> > This document contains a compilation of all defects found up until
> >    the publishing of this document for the Stream Control Transmissio=
n
> >    Protocol (SCTP) RFC2960 [6].  These defects may be of an editorial=
 or
> >    technical nature.  This document may be thought of as a companion
> >    document to be used in the implementation of SCTP to clarify error=
s
> >    in the original SCTP document.
> >=20
> >    This document updates RFC2960 [6] and text within this document
> >    supersedes the text found in RFC2960 [6].
> >=20
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-sctpimpguide-12.=
txt
> >=20
> > To remove yourself from the I-D Announcement list, send a message to=20
> > i-d-announce-request@ietf.org with the word unsubscribe in the body o=
f the message. =20
> > You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announc=
e=20
> > to change your subscription settings.
> >=20
> >=20
> > Internet-Drafts are also available by anonymous FTP. Login with the u=
sername
> > "anonymous" and a password of your e-mail address. After logging in,
> > type "cd internet-drafts" and then
> > 	"get draft-ietf-tsvwg-sctpimpguide-12.txt".
> >=20
> > A list of Internet-Drafts directories can be found in
> > http://www.ietf.org/shadow.html=20
> > or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >=20
> >=20
> > Internet-Drafts can also be obtained by e-mail.
> >=20
> > Send a message to:
> > 	mailserv@ietf.org.
> > In the body type:
> > 	"FILE /internet-drafts/draft-ietf-tsvwg-sctpimpguide-12.txt".
> > =09
> > NOTE:	The mail server at ietf.org can return the document in
> > 	MIME-encoded form by using the "mpack" utility.  To use this
> > 	feature, insert the command "ENCODING mime" before the "FILE"
> > 	command.  To decode the response(s), you will need "munpack" or
> > 	a MIME-compliant mail reader.  Different MIME-compliant mail readers
> > 	exhibit different behavior, especially when dealing with
> > 	"multipart" MIME messages (i.e. documents which have been split
> > 	up into multiple messages), so check your local documentation on
> > 	how to manipulate these messages.
> > 	=09
> > 	=09
> > Below is the data which will enable a MIME compliant mail reader
> > implementation to automatically retrieve the ASCII version of the
> > Internet-Draft.
> >=20
> >=20
> > ---------------------------------------------------------------------=
---
> >=20
> > _______________________________________________
> > tsvwg mailing list
> > tsvwg@ietf.org
> > https://www1.ietf.org/mailman/listinfo/tsvwg
>=20
>=20
> --=20
> Randall Stewart
> 803-345-0369 <or> 815-342-5222(cell)
>=20
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

--=20
Brian F. G. Bidulock    =A6 The reasonable man adapts himself to the =A6
bidulock@openss7.org    =A6 world; the unreasonable one persists in  =A6
http://www.openss7.org/ =A6 trying  to adapt the  world  to himself. =A6
                        =A6 Therefore  all  progress  depends on the =A6
                        =A6 unreasonable man. -- George Bernard Shaw =A6

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From tsvwg-bounces@ietf.org  Fri Oct 22 18:13:54 2004
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Date: Fri, 22 Oct 2004 16:26:43 -0400
From: Randall Stewart <randall@stewart.chicago.il.us>
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Subject: Re: [Tsvwg] I-D ACTION:draft-ietf-tsvwg-sctpimpguide-12.txt
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If no one else objects that seems to me
like a logical way to do it..

SHOULD throw it away

and SHOULD NOT be responding to.....

or something like that :-D

R

Brian F. G. Bidulock wrote:
> Randall,
> 
> Loosen 1) to SHOULD.
> 
> --brian
> 
> On Fri, 22 Oct 2004, Randall Stewart wrote:
> 
> 
>>All:
>>
>>A point and a question..
>>
>>  We probably need to think about last calling this and closing
>>  up the document. I think, based on the last inter-op, we have
>>  things pretty well complete as far as clearifications go..
>>
>>
>>The question..
>>
>>One thing that has off and on nagged at me is the
>>following wording from RFC2960 ...
>>
>> From 8.4:
>>
>>    The receiver of an OOTB packet MUST do the following:
>>
>>    1) If the OOTB packet is to or from a non-unicast address, silently
>>       discard the packet.  Otherwise,
>>*******************************************************************
>>and From 11.2.4.1 Flooding
>>
>>   -  not responding to unexpected packets sent to non-unicast
>>       addresses.
>>
>>*******************************************************************
>>
>>Now I realize that these strictures were put there to keep us
>>from setting up an association with a smurf like attack
>>i.e. src from X (the victim) and send an INIT
>>broadcast  255.255.255.255... or even multi-cast...
>>
>>But I wonder, is the wording too strict here??
>>
>>In the future if you are dealing with "Anycast" .. this is
>>NON-Unicast... and so you have to discard it.. but I could
>>see sometime down the road when we would WANT to respond
>>when sent to an ANYCAST address.... right now V6 limits that to
>>routers... some day anycast may go beyond routers...
>>And SCTP won't be able to use it due to the above
>>wording...
>>
>>
>>I could see...
>>
>>From:IP-A ---INIT(IP-A) -->To:Anycast Address
>>
>><----To:IP-A ---INIT-ACK(IP-Z,AnyCast)from IP-Z
>>--From:IP-A-COOKIE-ECHO------->To IP-Z
>>  From:IP-A<------COOKIE-ACK+ASCONF(Delete Anycast)----From:IP-Z
>>
>>Working just great...and making it so you could
>>establish an SCTP association with an ANYCAST address.... neat
>>idea.. but right now NOT allowed...
>>
>>Soo.. should we just cross that bridge when we come
>>to it... or should we attempt to loosen up the wording
>>for anycast??
>>
>>R
>>
>>
>>Internet-Drafts@ietf.org wrote:
>>
>>>A New Internet-Draft is available from the on-line Internet-Drafts directories.
>>>This draft is a work item of the Transport Area Working Group Working Group of the IETF.
>>>
>>>	Title		: Stream Control Transmission Protocol (SCTP) Implementer's Guide
>>>	Author(s)	: R. Stewart, et al.
>>>	Filename	: draft-ietf-tsvwg-sctpimpguide-12.txt
>>>	Pages		: 96
>>>	Date		: 2004-10-18
>>>	
>>>This document contains a compilation of all defects found up until
>>>   the publishing of this document for the Stream Control Transmission
>>>   Protocol (SCTP) RFC2960 [6].  These defects may be of an editorial or
>>>   technical nature.  This document may be thought of as a companion
>>>   document to be used in the implementation of SCTP to clarify errors
>>>   in the original SCTP document.
>>>
>>>   This document updates RFC2960 [6] and text within this document
>>>   supersedes the text found in RFC2960 [6].
>>>
>>>A URL for this Internet-Draft is:
>>>http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-sctpimpguide-12.txt
>>>
>>>To remove yourself from the I-D Announcement list, send a message to 
>>>i-d-announce-request@ietf.org with the word unsubscribe in the body of the message.  
>>>You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce 
>>>to change your subscription settings.
>>>
>>>
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>>>"anonymous" and a password of your e-mail address. After logging in,
>>>type "cd internet-drafts" and then
>>>	"get draft-ietf-tsvwg-sctpimpguide-12.txt".
>>>
>>>A list of Internet-Drafts directories can be found in
>>>http://www.ietf.org/shadow.html 
>>>or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>>
>>>
>>>Internet-Drafts can also be obtained by e-mail.
>>>
>>>Send a message to:
>>>	mailserv@ietf.org.
>>>In the body type:
>>>	"FILE /internet-drafts/draft-ietf-tsvwg-sctpimpguide-12.txt".
>>>	
>>>NOTE:	The mail server at ietf.org can return the document in
>>>	MIME-encoded form by using the "mpack" utility.  To use this
>>>	feature, insert the command "ENCODING mime" before the "FILE"
>>>	command.  To decode the response(s), you will need "munpack" or
>>>	a MIME-compliant mail reader.  Different MIME-compliant mail readers
>>>	exhibit different behavior, especially when dealing with
>>>	"multipart" MIME messages (i.e. documents which have been split
>>>	up into multiple messages), so check your local documentation on
>>>	how to manipulate these messages.
>>>		
>>>		
>>>Below is the data which will enable a MIME compliant mail reader
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>>>Internet-Draft.
>>>
>>>
>>>------------------------------------------------------------------------
>>>
>>>_______________________________________________
>>>tsvwg mailing list
>>>tsvwg@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/tsvwg
>>
>>
>>-- 
>>Randall Stewart
>>803-345-0369 <or> 815-342-5222(cell)
>>
>>_______________________________________________
>>tsvwg mailing list
>>tsvwg@ietf.org
>>https://www1.ietf.org/mailman/listinfo/tsvwg
> 
> 


-- 
Randall Stewart
803-345-0369 <or> 815-342-5222(cell)

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From tsvwg-bounces@ietf.org  Fri Oct 22 18:19:58 2004
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From: Aaron Falk <falk@isi.edu>
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Hi-

We've published an Internet Draft with a spec of XCP, the eXplicit
Control Protocol, an experimental congestion control protocol.  As the
draft's abstract says:

   XCP is designed to deliver the highest possible end-to-end
   throughput over a broad range of network infrastructure, including
   links with very large bandwidth-delay products, which are not well
   served by the current control algorithms.  XCP is potentially
   applicable to any transport protocol, although initial testing has
   applied it to TCP in particular.  XCP routers are required to
   perform a small calculation on congestion state carried in each
   data packet.  XCP routers also periodically recalculate the local
   parameters required to provide fairness.  On the other hand, there
   is no per-flow congestion state in XCP routers.

Our plan is to publish it as an Experimental RFC.  Allison and Jon
have given us an agenda slot at TSVWG to talk a little about the
protocol, preliminary results, and our plans.

Find the draft at:

  http://www.ietf.org/internet-drafts/draft-falk-xcp-spec-00.txt

Project page (w/code) at:

  http://www.isi.edu/isi-xcp

Regards,

--aaron

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To: Randall Stewart <randall@stewart.chicago.il.us>
Subject: Re: [Tsvwg] I-D ACTION:draft-ietf-tsvwg-sctpimpguide-12.txt
References: <200410191150.HAA14673@ietf.org>	<41795007.1020500@stewart.chicago.il.us>	<20041022133004.A8554@openss7.org>
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Yup, I'd like to see it as SHOULD silently discard the packet in
the first section you excerpted and SHOULD NOT respond to unexpected
packets in the second section.

regards,
--peter

Randall Stewart wrote:
> If no one else objects that seems to me
> like a logical way to do it..
> 
> SHOULD throw it away
> 
> and SHOULD NOT be responding to.....
> 
> or something like that :-D
> 
> R
> 
> Brian F. G. Bidulock wrote:
> 
>> Randall,
>>
>> Loosen 1) to SHOULD.
>>
>> --brian
>>
>> On Fri, 22 Oct 2004, Randall Stewart wrote:
>>
>>
>>> All:
>>>
>>> A point and a question..
>>>
>>>  We probably need to think about last calling this and closing
>>>  up the document. I think, based on the last inter-op, we have
>>>  things pretty well complete as far as clearifications go..
>>>
>>>
>>> The question..
>>>
>>> One thing that has off and on nagged at me is the
>>> following wording from RFC2960 ...
>>>
>>> From 8.4:
>>>
>>>    The receiver of an OOTB packet MUST do the following:
>>>
>>>    1) If the OOTB packet is to or from a non-unicast address, silently
>>>       discard the packet.  Otherwise,
>>> *******************************************************************
>>> and From 11.2.4.1 Flooding
>>>
>>>   -  not responding to unexpected packets sent to non-unicast
>>>       addresses.
>>>
>>> *******************************************************************
>>>
>>> Now I realize that these strictures were put there to keep us
>>> from setting up an association with a smurf like attack
>>> i.e. src from X (the victim) and send an INIT
>>> broadcast  255.255.255.255... or even multi-cast...
>>>
>>> But I wonder, is the wording too strict here??
>>>
>>> In the future if you are dealing with "Anycast" .. this is
>>> NON-Unicast... and so you have to discard it.. but I could
>>> see sometime down the road when we would WANT to respond
>>> when sent to an ANYCAST address.... right now V6 limits that to
>>> routers... some day anycast may go beyond routers...
>>> And SCTP won't be able to use it due to the above
>>> wording...
>>>
>>>
>>> I could see...
>>>
>>> From:IP-A ---INIT(IP-A) -->To:Anycast Address
>>>
>>> <----To:IP-A ---INIT-ACK(IP-Z,AnyCast)from IP-Z
>>> --From:IP-A-COOKIE-ECHO------->To IP-Z
>>>  From:IP-A<------COOKIE-ACK+ASCONF(Delete Anycast)----From:IP-Z
>>>
>>> Working just great...and making it so you could
>>> establish an SCTP association with an ANYCAST address.... neat
>>> idea.. but right now NOT allowed...
>>>
>>> Soo.. should we just cross that bridge when we come
>>> to it... or should we attempt to loosen up the wording
>>> for anycast??
>>>
>>> R
>>>
>>>
>>> Internet-Drafts@ietf.org wrote:
>>>
>>>> A New Internet-Draft is available from the on-line Internet-Drafts 
>>>> directories.
>>>> This draft is a work item of the Transport Area Working Group 
>>>> Working Group of the IETF.
>>>>
>>>>     Title        : Stream Control Transmission Protocol (SCTP) 
>>>> Implementer's Guide
>>>>     Author(s)    : R. Stewart, et al.
>>>>     Filename    : draft-ietf-tsvwg-sctpimpguide-12.txt
>>>>     Pages        : 96
>>>>     Date        : 2004-10-18
>>>>     
>>>> This document contains a compilation of all defects found up until
>>>>   the publishing of this document for the Stream Control Transmission
>>>>   Protocol (SCTP) RFC2960 [6].  These defects may be of an editorial or
>>>>   technical nature.  This document may be thought of as a companion
>>>>   document to be used in the implementation of SCTP to clarify errors
>>>>   in the original SCTP document.
>>>>
>>>>   This document updates RFC2960 [6] and text within this document
>>>>   supersedes the text found in RFC2960 [6].
>>>>
>>>> A URL for this Internet-Draft is:
>>>> http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-sctpimpguide-12.txt 
>>>>
>>>>
>>>> To remove yourself from the I-D Announcement list, send a message to 
>>>> i-d-announce-request@ietf.org with the word unsubscribe in the body 
>>>> of the message.  You can also visit 
>>>> https://www1.ietf.org/mailman/listinfo/I-D-announce to change your 
>>>> subscription settings.
>>>>
>>>>
>>>> Internet-Drafts are also available by anonymous FTP. Login with the 
>>>> username
>>>> "anonymous" and a password of your e-mail address. After logging in,
>>>> type "cd internet-drafts" and then
>>>>     "get draft-ietf-tsvwg-sctpimpguide-12.txt".
>>>>
>>>> A list of Internet-Drafts directories can be found in
>>>> http://www.ietf.org/shadow.html or 
>>>> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>>>
>>>>
>>>> Internet-Drafts can also be obtained by e-mail.
>>>>
>>>> Send a message to:
>>>>     mailserv@ietf.org.
>>>> In the body type:
>>>>     "FILE /internet-drafts/draft-ietf-tsvwg-sctpimpguide-12.txt".
>>>>     
>>>> NOTE:    The mail server at ietf.org can return the document in
>>>>     MIME-encoded form by using the "mpack" utility.  To use this
>>>>     feature, insert the command "ENCODING mime" before the "FILE"
>>>>     command.  To decode the response(s), you will need "munpack" or
>>>>     a MIME-compliant mail reader.  Different MIME-compliant mail 
>>>> readers
>>>>     exhibit different behavior, especially when dealing with
>>>>     "multipart" MIME messages (i.e. documents which have been split
>>>>     up into multiple messages), so check your local documentation on
>>>>     how to manipulate these messages.
>>>>        
>>>>        
>>>> Below is the data which will enable a MIME compliant mail reader
>>>> implementation to automatically retrieve the ASCII version of the
>>>> Internet-Draft.
>>>>
>>>>
>>>> ------------------------------------------------------------------------ 
>>>>
>>>>
>>>> _______________________________________________
>>>> tsvwg mailing list
>>>> tsvwg@ietf.org
>>>> https://www1.ietf.org/mailman/listinfo/tsvwg
>>>
>>>
>>>
>>> -- 
>>> Randall Stewart
>>> 803-345-0369 <or> 815-342-5222(cell)
>>>
>>> _______________________________________________
>>> tsvwg mailing list
>>> tsvwg@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/tsvwg
>>
>>
>>
> 
> 

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To: Brian E Carpenter <brc@zurich.ibm.com>, Fred Baker <fred@cisco.com>,
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Thanks Brian for the review and comments. Your list of changes has been
incorporated into version 04. See below for details.

Fred Baker wrote:
> At 01:41 PM 08/25/04 +0200, Brian E Carpenter wrote:
> 
>> A couple more thoughts, after a side conversation with Kathie.
>>
>> As I said, my comfort level is higher, but this doesn't remove some
>> older basic concerns... maybe I should have mentioned these in the 
>> tsvwg meeting, but I was rushing between sessions.
>>
>> 1. Thirteen classes is about eight too many for real deployments. It
>> will be far too easy for readers to miss the essential comment hidden 
>> in the middle of section 2.1:
>>
>>> It is expected that network administrators will choose the service
>>> classes that they will support based on their need, starting off with 
>>> three or four service classes for user traffic and add others as the 
>>> need arises.
>>
>>
>> This really needs to be mentioned earlier and more prominently, with
>> some emphasis that all classes are OPTIONAL except that the default 
>> class is REQUIRED.
> 
> 
> OK. Specifically where would you like this?

How about the end of section 1.2? I think it needs to be in the reader's
mind before hitting the details.
>
[Joe] Added below paragraph to section 1.2:
The Default Forwarding "Standard" service class is REQUIRED, all other
service classes are OPTIONAL. It is expected that network administrators
will choose the level of service differentiation that they will support
based on their need, starting off with three or four service classes for
user traffic and add others as the need arises.


-----Original Message-----
From: Brian E Carpenter [mailto:brc@zurich.ibm.com] 
Sent: Friday, July 16, 2004 10:45 AM
To: Fred Baker; Babiarz, Jozef [CAR:0S03:EXCH]; Chan, Kwok-Ho
[BL60:470:EXCH]
Subject: Re: I-D ACTION:draft-baker-diffserv-basic-classes-03.txt

Hi,

My comfort level with this is much higher than with the
earlier versions. I still have a couple of substantive
comments, however.

1. I think you need to add a clear statement somewhere in
section 1 like: The facts that four code points are recommended
for AF, and that one code point is recommended for EF, are
arbitrary choices, and the architecture allows any reasonable
number of AF and EF classes simultaneously. The choice of
four AF classes and one EF class in the current document is
also arbitrary, and operators MAY choose to operate more or
fewer of either.

[Joe] Added the above statement to Section 1.4 Key Service Concepts, at the
end of the first paragraph.

2. Going further in the same direction, since all choices
of code points and service classes are arbitrary in the diffserv
architecture, I think that almost all the MUSTs in the document
should be SHOULDs. There are just a few that could reasonably
be left as MUST, namely

> The service MUST be engineered so that EF marked packet flows
>    have sufficient bandwidth in the network to provide guaranteed
>    delivery.  

(which really just repeating part of the definition of EF)

and

>   The probability of loss of AF41 traffic MUST NOT
>    exceed the probability of loss of AF42 traffic, which in turn MUST
>    NOT exceed the probability of loss of AF43 traffic.

(ditto for AF)

and

>    The Standard service class MUST use the Default Forwarding (DF) PHB
>    defined in [RFC2474] 

[Joe] Changed all MUST to SHOULD except as noted above.
(ditto for DF)

You certainly can't have constructs like

>    RECOMMENDED Network Edge Conditioning:
>    o  At peering points (between two DiffServ networks) where SLAs are
>       in place, CS6 marked packets MUST be policed, e.g.  using a single
>       rate with burst size (sr+bs) token bucket policer to keep the CS6
>       marked packet flows to within the traffic rate specified in the
>       SLA.

with a MUST governed by a RECOMMENDED.
[Joe] Changed the MUST to SHOULD throughout the draft.

Regards
     Brian

Regards, Joe.
email: babiarz@nortelnetworks.com


------_=_NextPart_001_01C4BA24.15896781
Content-Type: text/html
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2658.2">
<TITLE>RE: Review comments diffserv-basic-classes-03</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Thanks Brian for the review and comments. Your list =
of changes has been incorporated into version 04. See below for =
details.</FONT></P>

<P><FONT SIZE=3D2>Fred Baker wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; At 01:41 PM 08/25/04 +0200, Brian E Carpenter =
wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&gt; A couple more thoughts, after a side =
conversation with Kathie.</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; As I said, my comfort level is higher, but =
this doesn't remove some</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; older basic concerns... maybe I should have =
mentioned these in the </FONT>
<BR><FONT SIZE=3D2>&gt;&gt; tsvwg meeting, but I was rushing between =
sessions.</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; 1. Thirteen classes is about eight too many =
for real deployments. It</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; will be far too easy for readers to miss =
the essential comment hidden </FONT>
<BR><FONT SIZE=3D2>&gt;&gt; in the middle of section 2.1:</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; It is expected that network =
administrators will choose the service</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; classes that they will support based on =
their need, starting off with </FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; three or four service classes for user =
traffic and add others as the </FONT>
<BR><FONT SIZE=3D2>&gt;&gt;&gt; need arises.</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; This really needs to be mentioned earlier =
and more prominently, with</FONT>
<BR><FONT SIZE=3D2>&gt;&gt; some emphasis that all classes are OPTIONAL =
except that the default </FONT>
<BR><FONT SIZE=3D2>&gt;&gt; class is REQUIRED.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; OK. Specifically where would you like =
this?</FONT>
</P>

<P><FONT SIZE=3D2>How about the end of section 1.2? I think it needs to =
be in the reader's mind before hitting the details.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>[Joe] Added below paragraph to section 1.2:</FONT>
<BR><FONT SIZE=3D2>The Default Forwarding &quot;Standard&quot; service =
class is REQUIRED, all other service classes are OPTIONAL. It is =
expected that network administrators will choose the level of service =
differentiation that they will support based on their need, starting =
off with three or four service classes for user traffic and add others =
as the need arises.</FONT></P>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Brian E Carpenter [<A =
HREF=3D"mailto:brc@zurich.ibm.com">mailto:brc@zurich.ibm.com</A>] =
</FONT>
<BR><FONT SIZE=3D2>Sent: Friday, July 16, 2004 10:45 AM</FONT>
<BR><FONT SIZE=3D2>To: Fred Baker; Babiarz, Jozef [CAR:0S03:EXCH]; =
Chan, Kwok-Ho [BL60:470:EXCH]</FONT>
<BR><FONT SIZE=3D2>Subject: Re: I-D =
ACTION:draft-baker-diffserv-basic-classes-03.txt</FONT>
</P>

<P><FONT SIZE=3D2>Hi,</FONT>
</P>

<P><FONT SIZE=3D2>My comfort level with this is much higher than with =
the</FONT>
<BR><FONT SIZE=3D2>earlier versions. I still have a couple of =
substantive</FONT>
<BR><FONT SIZE=3D2>comments, however.</FONT>
</P>

<P><FONT SIZE=3D2>1. I think you need to add a clear statement =
somewhere in</FONT>
<BR><FONT SIZE=3D2>section 1 like: The facts that four code points are =
recommended</FONT>
<BR><FONT SIZE=3D2>for AF, and that one code point is recommended for =
EF, are</FONT>
<BR><FONT SIZE=3D2>arbitrary choices, and the architecture allows any =
reasonable</FONT>
<BR><FONT SIZE=3D2>number of AF and EF classes simultaneously. The =
choice of</FONT>
<BR><FONT SIZE=3D2>four AF classes and one EF class in the current =
document is</FONT>
<BR><FONT SIZE=3D2>also arbitrary, and operators MAY choose to operate =
more or</FONT>
<BR><FONT SIZE=3D2>fewer of either.</FONT>
</P>

<P><FONT SIZE=3D2>[Joe] Added the above statement to Section 1.4 Key =
Service Concepts, at the end of the first paragraph.</FONT>
</P>

<P><FONT SIZE=3D2>2. Going further in the same direction, since all =
choices</FONT>
<BR><FONT SIZE=3D2>of code points and service classes are arbitrary in =
the diffserv</FONT>
<BR><FONT SIZE=3D2>architecture, I think that almost all the MUSTs in =
the document</FONT>
<BR><FONT SIZE=3D2>should be SHOULDs. There are just a few that could =
reasonably</FONT>
<BR><FONT SIZE=3D2>be left as MUST, namely</FONT>
</P>

<P><FONT SIZE=3D2>&gt; The service MUST be engineered so that EF marked =
packet flows</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; have sufficient bandwidth in =
the network to provide guaranteed</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; delivery.&nbsp; </FONT>
</P>

<P><FONT SIZE=3D2>(which really just repeating part of the definition =
of EF)</FONT>
</P>

<P><FONT SIZE=3D2>and</FONT>
</P>

<P><FONT SIZE=3D2>&gt;&nbsp;&nbsp; The probability of loss of AF41 =
traffic MUST NOT</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; exceed the probability of =
loss of AF42 traffic, which in turn MUST</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; NOT exceed the probability of =
loss of AF43 traffic.</FONT>
</P>

<P><FONT SIZE=3D2>(ditto for AF)</FONT>
</P>

<P><FONT SIZE=3D2>and</FONT>
</P>

<P><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; The Standard service class =
MUST use the Default Forwarding (DF) PHB</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; defined in [RFC2474] </FONT>
</P>

<P><FONT SIZE=3D2>[Joe] Changed all MUST to SHOULD except as noted =
above.</FONT>
<BR><FONT SIZE=3D2>(ditto for DF)</FONT>
</P>

<P><FONT SIZE=3D2>You certainly can't have constructs like</FONT>
</P>

<P><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; RECOMMENDED Network Edge =
Conditioning:</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; o&nbsp; At peering points =
(between two DiffServ networks) where SLAs are</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; in place, =
CS6 marked packets MUST be policed, e.g.&nbsp; using a single</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; rate with =
burst size (sr+bs) token bucket policer to keep the CS6</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; marked =
packet flows to within the traffic rate specified in the</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SLA.</FONT>
</P>

<P><FONT SIZE=3D2>with a MUST governed by a RECOMMENDED.</FONT>
<BR><FONT SIZE=3D2>[Joe] Changed the MUST to SHOULD throughout the =
draft.</FONT>
</P>

<P><FONT SIZE=3D2>Regards</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp; Brian</FONT>
</P>

<P><FONT SIZE=3D2>Regards, Joe.</FONT>
<BR><FONT SIZE=3D2>email: babiarz@nortelnetworks.com</FONT>
</P>

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Hi all,

I'd like to bring up another clarification which didn't make it into the
imp-guide in time (although I understand that the authors in principle
agree to include it). It is a clarification of Karn's algorithm which
sometimes results in wrong RTT samples being taken if not properly
implemented. AFAIK this problem is not clearly spelled out in RFC 2918
or other related IETF docs (please correct me if I'm wrong). I know of
at least one SCTP stack (and a certain simulator :) where this is
currently not correctly implemented. In essence, a cumulative ACK
resulting from a retransmission of an earlier packet can be mistaken for
an ACK for a timed packet. For illustration, consider the following example:

- Host A sends packets 1, 2 and 3 to host B, and choses packet 3 for an
RTT measurement
- Host B receives all packets correctly and sends ACK1, ACK2, and ACK3
- ACK2 and ACK3 are lost on the return path
- Eventually a timeout fires for packet 2, and A retransmits 2
- Upon receipt of 2, B sends a cumulative ACK3 (since it has received 2
and 3 before)
- Since packet 3 has never been retransmitted, even an SCTP
implementation strictly following 2960 could actually accept the ACK for
an RTT measurement, although it was sent in reply to the retransmission
of 2, resulting in a much too high RTT estimate. Since this tends to
happen in case of longer link interruptions, the error is often
amplified by subsequent timer back-offs.

The solution is to either cancel all pending RTT timings upon entering
the recovery phase (which is what e.g. BSD TCP does), or to cancel
timings for packets with TSNs higher than the retransmitted one. (note
that this issue is not a problem for TCP stacks using timestamps)

We propose to include the following clarification in the imp-guide:

Karn's algorithm: RTT measurements MUST only be made using a packet
with TSN r if no packet with TSN less than or equal to r is
retransmitted since r was first sent.

BR,
Stephan.



Randall Stewart wrote:

> All:
> 
> A point and a question..
> 
>   We probably need to think about last calling this and closing
>   up the document. I think, based on the last inter-op, we have
>   things pretty well complete as far as clearifications go..
> 
> 
> The question..
> 
> One thing that has off and on nagged at me is the
> following wording from RFC2960 ...
> 
>  From 8.4:
> 
>     The receiver of an OOTB packet MUST do the following:
> 
>     1) If the OOTB packet is to or from a non-unicast address, silently
>        discard the packet.  Otherwise,
> *******************************************************************
> and From 11.2.4.1 Flooding
> 
>    -  not responding to unexpected packets sent to non-unicast
>        addresses.
> 
> *******************************************************************
> 
> Now I realize that these strictures were put there to keep us
> from setting up an association with a smurf like attack
> i.e. src from X (the victim) and send an INIT
> broadcast  255.255.255.255... or even multi-cast...
> 
> But I wonder, is the wording too strict here??
> 
> In the future if you are dealing with "Anycast" .. this is
> NON-Unicast... and so you have to discard it.. but I could
> see sometime down the road when we would WANT to respond
> when sent to an ANYCAST address.... right now V6 limits that to
> routers... some day anycast may go beyond routers...
> And SCTP won't be able to use it due to the above
> wording...
> 
> 
> I could see...
> 
> From:IP-A ---INIT(IP-A) -->To:Anycast Address
> 
> <----To:IP-A ---INIT-ACK(IP-Z,AnyCast)from IP-Z
> --From:IP-A-COOKIE-ECHO------->To IP-Z
>   From:IP-A<------COOKIE-ACK+ASCONF(Delete Anycast)----From:IP-Z
> 
> Working just great...and making it so you could
> establish an SCTP association with an ANYCAST address.... neat
> idea.. but right now NOT allowed...
> 
> Soo.. should we just cross that bridge when we come
> to it... or should we attempt to loosen up the wording
> for anycast??
> 
> R
> 
> 
> Internet-Drafts@ietf.org wrote:
>> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>> This draft is a work item of the Transport Area Working Group Working Group of the IETF.
>> 
>> 	Title		: Stream Control Transmission Protocol (SCTP) Implementer's Guide
>> 	Author(s)	: R. Stewart, et al.
>> 	Filename	: draft-ietf-tsvwg-sctpimpguide-12.txt
>> 	Pages		: 96
>> 	Date		: 2004-10-18
>> 	
>> This document contains a compilation of all defects found up until
>>    the publishing of this document for the Stream Control Transmission
>>    Protocol (SCTP) RFC2960 [6].  These defects may be of an editorial or
>>    technical nature.  This document may be thought of as a companion
>>    document to be used in the implementation of SCTP to clarify errors
>>    in the original SCTP document.
>> 
>>    This document updates RFC2960 [6] and text within this document
>>    supersedes the text found in RFC2960 [6].
>> 
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-sctpimpguide-12.txt
>> 
>> To remove yourself from the I-D Announcement list, send a message to 
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>> 	"get draft-ietf-tsvwg-sctpimpguide-12.txt".
>> 
>> A list of Internet-Drafts directories can be found in
>> http://www.ietf.org/shadow.html 
>> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>> 
>> 
>> Internet-Drafts can also be obtained by e-mail.
>> 
>> Send a message to:
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From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Stephan Baucke <stephan.baucke@ericsson.com>
Subject: Re: [Tsvwg] I-D ACTION:draft-ietf-tsvwg-sctpimpguide-12.txt
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Stephan,

What damage might occur to the network otherwise that justifies using
MUST here?  What interoperability problems would occur?

Too high an RTT estimate doesn't look like a problem to anybody but
the sender, and appears to be more of a local performance consideration
to me...

It might warrant an implementation note, but a "MUST" level requirement
would need to be justified.

--brian

On Mon, 25 Oct 2004, Stephan Baucke wrote:

> Hi all,
>=20
> I'd like to bring up another clarification which didn't make it into th=
e
> imp-guide in time (although I understand that the authors in principle
> agree to include it). It is a clarification of Karn's algorithm which
> sometimes results in wrong RTT samples being taken if not properly
> implemented. AFAIK this problem is not clearly spelled out in RFC 2918
> or other related IETF docs (please correct me if I'm wrong). I know of
> at least one SCTP stack (and a certain simulator :) where this is
> currently not correctly implemented. In essence, a cumulative ACK
> resulting from a retransmission of an earlier packet can be mistaken fo=
r
> an ACK for a timed packet. For illustration, consider the following exa=
mple:
>=20
> - Host A sends packets 1, 2 and 3 to host B, and choses packet 3 for an
> RTT measurement
> - Host B receives all packets correctly and sends ACK1, ACK2, and ACK3
> - ACK2 and ACK3 are lost on the return path
> - Eventually a timeout fires for packet 2, and A retransmits 2
> - Upon receipt of 2, B sends a cumulative ACK3 (since it has received 2
> and 3 before)
> - Since packet 3 has never been retransmitted, even an SCTP
> implementation strictly following 2960 could actually accept the ACK fo=
r
> an RTT measurement, although it was sent in reply to the retransmission
> of 2, resulting in a much too high RTT estimate. Since this tends to
> happen in case of longer link interruptions, the error is often
> amplified by subsequent timer back-offs.
>=20
> The solution is to either cancel all pending RTT timings upon entering
> the recovery phase (which is what e.g. BSD TCP does), or to cancel
> timings for packets with TSNs higher than the retransmitted one. (note
> that this issue is not a problem for TCP stacks using timestamps)
>=20
> We propose to include the following clarification in the imp-guide:
>=20
> Karn's algorithm: RTT measurements MUST only be made using a packet
> with TSN r if no packet with TSN less than or equal to r is
> retransmitted since r was first sent.
>=20
> BR,
> Stephan.
>=20
>=20

--=20
Brian F. G. Bidulock    =A6 The reasonable man adapts himself to the =A6
bidulock@openss7.org    =A6 world; the unreasonable one persists in  =A6
http://www.openss7.org/ =A6 trying  to adapt the  world  to himself. =A6
                        =A6 Therefore  all  progress  depends on the =A6
                        =A6 unreasonable man. -- George Bernard Shaw =A6

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Brian,

after taking a peek into RFC 2119, I agree that the use of "MUST" seems
inappropriate, since there appears to be no danger of harming the
network or limiting interoperability. The clarification is meant to
point out a potential pitfall to implementers which can lead to
incorrect RTO calculations and bad performance.

BR,
Stephan.


Brian F. G. Bidulock wrote:

> Stephan,
> 
> What damage might occur to the network otherwise that justifies using
> MUST here?  What interoperability problems would occur?
> 
> Too high an RTT estimate doesn't look like a problem to anybody but
> the sender, and appears to be more of a local performance consideration
> to me...
> 
> It might warrant an implementation note, but a "MUST" level requirement
> would need to be justified.
> 
> --brian
> 
> On Mon, 25 Oct 2004, Stephan Baucke wrote:
> 
>> Hi all,
>> 
>> I'd like to bring up another clarification which didn't make it into the
>> imp-guide in time (although I understand that the authors in principle
>> agree to include it). It is a clarification of Karn's algorithm which
>> sometimes results in wrong RTT samples being taken if not properly
>> implemented. AFAIK this problem is not clearly spelled out in RFC 2918
>> or other related IETF docs (please correct me if I'm wrong). I know of
>> at least one SCTP stack (and a certain simulator :) where this is
>> currently not correctly implemented. In essence, a cumulative ACK
>> resulting from a retransmission of an earlier packet can be mistaken for
>> an ACK for a timed packet. For illustration, consider the following example:
>> 
>> - Host A sends packets 1, 2 and 3 to host B, and choses packet 3 for an
>> RTT measurement
>> - Host B receives all packets correctly and sends ACK1, ACK2, and ACK3
>> - ACK2 and ACK3 are lost on the return path
>> - Eventually a timeout fires for packet 2, and A retransmits 2
>> - Upon receipt of 2, B sends a cumulative ACK3 (since it has received 2
>> and 3 before)
>> - Since packet 3 has never been retransmitted, even an SCTP
>> implementation strictly following 2960 could actually accept the ACK for
>> an RTT measurement, although it was sent in reply to the retransmission
>> of 2, resulting in a much too high RTT estimate. Since this tends to
>> happen in case of longer link interruptions, the error is often
>> amplified by subsequent timer back-offs.
>> 
>> The solution is to either cancel all pending RTT timings upon entering
>> the recovery phase (which is what e.g. BSD TCP does), or to cancel
>> timings for packets with TSNs higher than the retransmitted one. (note
>> that this issue is not a problem for TCP stacks using timestamps)
>> 
>> We propose to include the following clarification in the imp-guide:
>> 
>> Karn's algorithm: RTT measurements MUST only be made using a packet
>> with TSN r if no packet with TSN less than or equal to r is
>> retransmitted since r was first sent.
>> 
>> BR,
>> Stephan.
>> 
>> 
> 


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Hi all:
  In the RFC 2960, page 95, section 8.4 Handle "out of the blue" packets, ...

1) If the OOBT packet is to or from a non-unicast address, silently discard 
the packet.

Question:
 Why does the SCTP only support non-unicast address? 

--
Chaoyang University of Technology
WebMail http://webmail.cyut.edu.tw


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On Mon, 25 Oct 2004, Stephan Baucke wrote:

> I'd like to bring up another clarification which didn't make it into the
> imp-guide in time (although I understand that the authors in principle
> agree to include it). It is a clarification of Karn's algorithm which
> sometimes results in wrong RTT samples being taken if not properly
> implemented.

<snip>

> We propose to include the following clarification in the imp-guide:
>
> Karn's algorithm: RTT measurements MUST only be made using a packet
> with TSN r if no packet with TSN less than or equal to r is
> retransmitted since r was first sent.

Yes, that's my fault. I was supposed to put that change in.

Randall: Since I currently can't log into your machine, can you make the
change?

~armando

0--                                              --0
| Armando L. Caro Jr.  |  Protocol Engineering Lab |
| www.armandocaro.net  |    University of Delaware |
0--                                              --0

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On Mon, 25 Oct 2004, Brian F. G. Bidulock wrote:

> What damage might occur to the network otherwise that justifies using
> MUST here?  What interoperability problems would occur?
>
> Too high an RTT estimate doesn't look like a problem to anybody but
> the sender, and appears to be more of a local performance consideration
> to me...
>
> It might warrant an implementation note, but a "MUST" level requirement
> would need to be justified.

Well the wording that Stephan provided is just a modification of the
current wording in RFC2960's Section 6.3.1.C5, which already uses a "MUST"
level requirement.

~armando

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| Armando L. Caro Jr.  |  Protocol Engineering Lab |
| www.armandocaro.net  |    University of Delaware |
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From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: "Armando L. Caro, Jr." <me@armandocaro.net>
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Armando,

On Mon, 25 Oct 2004, Armando L. Caro, Jr. wrote:

> On Mon, 25 Oct 2004, Brian F. G. Bidulock wrote:
>=20
> > What damage might occur to the network otherwise that justifies using
> > MUST here?  What interoperability problems would occur?
> >
> > Too high an RTT estimate doesn't look like a problem to anybody but
> > the sender, and appears to be more of a local performance considerati=
on
> > to me...
> >
> > It might warrant an implementation note, but a "MUST" level requireme=
nt
> > would need to be justified.
>=20
> Well the wording that Stephan provided is just a modification of the
> current wording in RFC2960's Section 6.3.1.C5, which already uses a "MU=
ST"
> level requirement.

Yes, not calculating the RTT correctly can damage the network.  (Just not
in the way Stephan details.)

--brian

>=20
> ~armando
>=20
> 0--                                              --0
> | Armando L. Caro Jr.  |  Protocol Engineering Lab |
> | www.armandocaro.net  |    University of Delaware |
> 0--                                              --0

--=20
Brian F. G. Bidulock    =A6 The reasonable man adapts himself to the =A6
bidulock@openss7.org    =A6 world; the unreasonable one persists in  =A6
http://www.openss7.org/ =A6 trying  to adapt the  world  to himself. =A6
                        =A6 Therefore  all  progress  depends on the =A6
                        =A6 unreasonable man. -- George Bernard Shaw =A6

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To: Armando@mail.eecis.udel.edu
Subject: Re: [Tsvwg] I-D ACTION:draft-ietf-tsvwg-sctpimpguide-12.txt
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Armando@mail.eecis.udel.edu wrote:
> On Mon, 25 Oct 2004, Stephan Baucke wrote:
> 
> 
>>I'd like to bring up another clarification which didn't make it into the
>>imp-guide in time (although I understand that the authors in principle
>>agree to include it). It is a clarification of Karn's algorithm which
>>sometimes results in wrong RTT samples being taken if not properly
>>implemented.
> 
> 
> <snip>
> 
>>We propose to include the following clarification in the imp-guide:
>>
>>Karn's algorithm: RTT measurements MUST only be made using a packet
>>with TSN r if no packet with TSN less than or equal to r is
>>retransmitted since r was first sent.
> 
> 
> Yes, that's my fault. I was supposed to put that change in.
> 
> Randall: Since I currently can't log into your machine, can you make the
> change?
> 

Yep... I will do so.. but I will change the MUST to a SHOULD
per the comments on the list so far..

R

> ~armando
> 
> 0--                                              --0
> | Armando L. Caro Jr.  |  Protocol Engineering Lab |
> | www.armandocaro.net  |    University of Delaware |
> 0--                                              --0
> 


-- 
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From tsvwg-bounces@ietf.org  Mon Oct 25 20:00:24 2004
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To: "Luo(2) Jui(4) Jen(2)" <s9230605@cyut.edu.tw>
Subject: Re: [Tsvwg] why does SCTP only support non-unicast address?
References: <20041025155400.M15181@cyut.edu.tw>
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Because in the normal case
you cannot have an association
with a non-unicast address. If
I send you a broadcast (DATA), how
can you know its me? Also it
presents a DOS attack if you
accept non-unicast addresses. The
exception to this might be
any-cast.. where one and only
one machine is contacted... one
could envision (as my previous post
said) using anycast and SCTP in
combination with the ASCONF() to
delete the anycast address.

R

Luo(2) Jui(4) Jen(2) wrote:
> Hi all:
>   In the RFC 2960, page 95, section 8.4 Handle "out of the blue" packets, ...
> 
> 1) If the OOBT packet is to or from a non-unicast address, silently discard 
> the packet.
> 
> Question:
>  Why does the SCTP only support non-unicast address? 
> 
> --
> Chaoyang University of Technology
> WebMail http://webmail.cyut.edu.tw
> 
> 
> _______________________________________________
> tsvwg mailing list
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> 
> 


-- 
Randall Stewart
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Armando L. Caro, Jr. wrote:

>> What damage might occur to the network otherwise that justifies using
>> MUST here?  What interoperability problems would occur?
>>
>> Too high an RTT estimate doesn't look like a problem to anybody but
>> the sender, and appears to be more of a local performance consideration
>> to me...
>>
>> It might warrant an implementation note, but a "MUST" level requirement
>> would need to be justified.
> 
> Well the wording that Stephan provided is just a modification of the
> current wording in RFC2960's Section 6.3.1.C5, which already uses a "MUST"
> level requirement.

I'd say it depends on whether the new text will be put into the
imp-guide as a _replacement_ for the 2960 text, or as an addendum. In
the first case we will have to keep the MUST, since failure to comply
with the part of the rule already in 2960 may (as Brian correctly
pointed out) harm the network. If it is meant as an addendum containing
just the clarification about retransmitted packets with a smaller TSNs,
I guess a SHOULD would be appropriate.

BR,
Stephan.

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Stephan:

IMO this is something that should be ADDED to the
text as an implementation note:

aka:

IMPLEMENTATION NOTE: RTT measurements should only be made using a packet
with TSN r if no packet with TSN less than or equal to r is
retransmitted since r was first sent.


Does that sound agreeable to everyone... note the original
text in the MUST clause will remain..

R






Stephan Baucke wrote:
> Armando L. Caro, Jr. wrote:
> 
> 
>>>What damage might occur to the network otherwise that justifies using
>>>MUST here?  What interoperability problems would occur?
>>>
>>>Too high an RTT estimate doesn't look like a problem to anybody but
>>>the sender, and appears to be more of a local performance consideration
>>>to me...
>>>
>>>It might warrant an implementation note, but a "MUST" level requirement
>>>would need to be justified.
>>
>>Well the wording that Stephan provided is just a modification of the
>>current wording in RFC2960's Section 6.3.1.C5, which already uses a "MUST"
>>level requirement.
> 
> 
> I'd say it depends on whether the new text will be put into the
> imp-guide as a _replacement_ for the 2960 text, or as an addendum. In
> the first case we will have to keep the MUST, since failure to comply
> with the part of the rule already in 2960 may (as Brian correctly
> pointed out) harm the network. If it is meant as an addendum containing
> just the clarification about retransmitted packets with a smaller TSNs,
> I guess a SHOULD would be appropriate.
> 
> BR,
> Stephan.
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 
> 


-- 
Randall Stewart
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Date: Tue, 26 Oct 2004 05:34:30 -0600
From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Randall Stewart <randall@stewart.chicago.il.us>
Subject: Re: [Tsvwg] I-D ACTION:draft-ietf-tsvwg-sctpimpguide-12.txt
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Randall,

I agree that it is best as a supplemental implementation note
as worded.

--brian

On Tue, 26 Oct 2004, Randall Stewart wrote:

> Stephan:
>=20
> IMO this is something that should be ADDED to the
> text as an implementation note:
>=20
> aka:
>=20
> IMPLEMENTATION NOTE: RTT measurements should only be made using a packe=
t
> with TSN r if no packet with TSN less than or equal to r is
> retransmitted since r was first sent.
>=20
>=20
> Does that sound agreeable to everyone... note the original
> text in the MUST clause will remain..
>=20
> R
>=20

--=20
Brian F. G. Bidulock    =A6 The reasonable man adapts himself to the =A6
bidulock@openss7.org    =A6 world; the unreasonable one persists in  =A6
http://www.openss7.org/ =A6 trying  to adapt the  world  to himself. =A6
                        =A6 Therefore  all  progress  depends on the =A6
                        =A6 unreasonable man. -- George Bernard Shaw =A6

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The 04 version of the draft has all the reviewers comments addressed.

Regards, Joe.
Jozef Babiarz
email: babiarz@nortelnetworks.com
*************************************
A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title		: Configuration Guidelines for DiffServ Service
Classes
	Author(s)	: F. Baker, et al.
	Filename	: draft-baker-diffserv-basic-classes-04.txt
	Pages		: 56
	Date		: 2004-10-25
	
This paper summarizes the recommended correlation between service
   classes and their usage, with references to their corresponding
   recommended Differentiated Service Code Points (DSCP), traffic
   conditioners, Per-Hop Behaviors (PHB) and Active Queue Management
   (AQM) mechanisms. There is no intrinsic requirement that particular
   DSCPs, traffic conditioner PHBs and AQM be used for a certain service
   class, but as a policy it is useful that they be applied consistently
   across the network.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-baker-diffserv-basic-classes-04.tx
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<P><FONT SIZE=3D2>The 04 version of the draft has all the reviewers =
comments addressed.</FONT>
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<P><FONT SIZE=3D2>Regards, Joe.</FONT>
<BR><FONT SIZE=3D2>Jozef Babiarz</FONT>
<BR><FONT SIZE=3D2>email: babiarz@nortelnetworks.com</FONT>
<BR><FONT SIZE=3D2>*************************************</FONT>
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Configuration Guidelines for DiffServ Service Classes</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
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<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
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Randall,

this is fine with me. As the problem description, I propose something
like the following:

The description of Karn's algorithm in RFC 2960 only states that
retransmitted packets must not be used for RTT measurements. But an RTT
measurement should also not be made if other packets with lower TSNs are
retransmitted while a packet is being timed. Otherwise an
acknowledgement resulting from such a retransmission can be mistaken for
the acknowledgement of the packet being timed (if it advances the
cumulative ACK point to the TSN of that packet). This results in an
incorrect RTT measurement.

Stephan.

Randall Stewart wrote:

> Stephan:
> 
> IMO this is something that should be ADDED to the
> text as an implementation note:
> 
> aka:
> 
> IMPLEMENTATION NOTE: RTT measurements should only be made using a packet
> with TSN r if no packet with TSN less than or equal to r is
> retransmitted since r was first sent.
> 
> 
> Does that sound agreeable to everyone... note the original
> text in the MUST clause will remain..
> 
> R
> 
> 
> 
> 
> 
> 
> Stephan Baucke wrote:
>> Armando L. Caro, Jr. wrote:
>> 
>> 
>>>>What damage might occur to the network otherwise that justifies using
>>>>MUST here?  What interoperability problems would occur?
>>>>
>>>>Too high an RTT estimate doesn't look like a problem to anybody but
>>>>the sender, and appears to be more of a local performance consideration
>>>>to me...
>>>>
>>>>It might warrant an implementation note, but a "MUST" level requirement
>>>>would need to be justified.
>>>
>>>Well the wording that Stephan provided is just a modification of the
>>>current wording in RFC2960's Section 6.3.1.C5, which already uses a "MUST"
>>>level requirement.
>> 
>> 
>> I'd say it depends on whether the new text will be put into the
>> imp-guide as a _replacement_ for the 2960 text, or as an addendum. In
>> the first case we will have to keep the MUST, since failure to comply
>> with the part of the rule already in 2960 may (as Brian correctly
>> pointed out) harm the network. If it is meant as an addendum containing
>> just the clarification about retransmitted packets with a smaller TSNs,
>> I guess a SHOULD would be appropriate.
>> 
>> BR,
>> Stephan.
>> 
>> _______________________________________________
>> tsvwg mailing list
>> tsvwg@ietf.org
>> https://www1.ietf.org/mailman/listinfo/tsvwg
>> 
>> 
> 
> 


_______________________________________________
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Yes, it looks good to me.

~armando

0--                                              --0
| Armando L. Caro, Jr. |  Protocol Engineering Lab |
| www.armandocaro.net  |    University of Delaware |
0--                                              --0

On Tue, 26 Oct 2004, Randall Stewart wrote:

> Stephan:
>
> IMO this is something that should be ADDED to the
> text as an implementation note:
>
> aka:
>
> IMPLEMENTATION NOTE: RTT measurements should only be made using a packet
> with TSN r if no packet with TSN less than or equal to r is
> retransmitted since r was first sent.
>
>
> Does that sound agreeable to everyone... note the original
> text in the MUST clause will remain..
>
> R
>
>
>
>
>
>
> Stephan Baucke wrote:
> > Armando L. Caro, Jr. wrote:
> >
> >
> >>>What damage might occur to the network otherwise that justifies using
> >>>MUST here?  What interoperability problems would occur?
> >>>
> >>>Too high an RTT estimate doesn't look like a problem to anybody but
> >>>the sender, and appears to be more of a local performance consideration
> >>>to me...
> >>>
> >>>It might warrant an implementation note, but a "MUST" level requirement
> >>>would need to be justified.
> >>
> >>Well the wording that Stephan provided is just a modification of the
> >>current wording in RFC2960's Section 6.3.1.C5, which already uses a "MUST"
> >>level requirement.
> >
> >
> > I'd say it depends on whether the new text will be put into the
> > imp-guide as a _replacement_ for the 2960 text, or as an addendum. In
> > the first case we will have to keep the MUST, since failure to comply
> > with the part of the rule already in 2960 may (as Brian correctly
> > pointed out) harm the network. If it is meant as an addendum containing
> > just the clarification about retransmitted packets with a smaller TSNs,
> > I guess a SHOULD would be appropriate.
> >
> > BR,
> > Stephan.
> >
> > _______________________________________________
> > tsvwg mailing list
> > tsvwg@ietf.org
> > https://www1.ietf.org/mailman/listinfo/tsvwg
> >
> >
>
>
> --
> Randall Stewart
> 803-345-0369 <or> 815-342-5222(cell)
>
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

_______________________________________________
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I'm resending this email as not everyone has received it when I sent it out
on Oct. 24. Sorry if you get it twice.
Regards, Joe.
************************************************************** 
Please see below answers to David L. Black review comments.
Also, please see Section 7 "Summary of Changes from Previous Drat" for an
outline of all the changes made to
draft-baker-diffserv-basic-classes-04.txt. (04 version should be available
from the on-line Internet-Drafts directory soon).
**********************************************************
Review of draft-baker-diffserv-basic-classes-03.txt
Reviewer: David L. Black, EMC Corporation
For the IETF Transport Area Working Group (tsvwg)
August 5, 2004
-------------------------------------------------

-- Overall comment

I had skimmed an earlier version of this draft and liked it.  Having read
this one in detail, I like it a *lot*.  Diffserv has been described as a
toolkit for constructing QoS services.  This draft is like a set of project
plans (or a do-it-yourself book on plumbing, wiring,
etc.) that explains how to use the tools to build useful things in a fashion
accessible to non-experts.  This draft is a long-overdue addition to the set
of diffserv documents, and does a nice job of weaving together the important
elements of diffserv to show how they can be used together in useful ways.
I'd like to be able say that those of us working on diffserv had such a
coherent vision in mind from the beginning ... ;-).

Most of these comments are details of various sorts.  The important comments
that affect content are:

- Add top down traffic and application classification (Section 2.1).
[Joe]Added Section 2.2 "Categorization of User Service Classes" and reworked
part of section 2.1 to become Section 2.3 "Service Class Characteristics". 

- Examples 2 and 3 need work (Section 2.2).
[Joe]Expanded old Section 2.2, now became Section 2.4. Section 2.4.1 is
Example 1. Section 2.4.2 is detailed Example 2 and Section 2.4.3 is detailed
Example 3.

- AF usage discussion cleanup [(a), (b), and (c) comments on Sections
	4.3, 4.5, 4.7 and 4.8].
[Joe] Implemented changes as discussed. See item 9 in Section 7 "Summary of
Changes from Previous Draft" of version 0.4 of draft for details of changes.

- Remove implied cap on Best Effort traffic (Section 4.9 - probably
	a typo/thinko, but an important one).
[Joe] Modified sentence so that no cap is implied on bandwidth.

- Enrich mapping discussion (Section 5 to the extent not covered
	by top down classification added to Section 2.1).
[Joe]Section 5 was extended and restructured into 4 subsections, 5.1 Mapping
for Signaling, 5.2 Mapping for NTP, 5.3 Explanation of Ring Clipping, and
5.4 VPN Service Mapping.

-- Section 1.4.1

"with no further guarantees" is at odds with the recommendation in Section
4.9 that some minimum amount of bandwidth be guaranteed to best effort.
Some sort of expectation that the network will be usable for best effort
traffic (in some sense) is needed here.
[Joe] Changed wording to be compatible with section 4.9 "Best Effort needs
to provide some minimum bandwidth guarantee ..."

--  Section 2.1

Expand OAM acronym on first use.
[Joe] Done. OAM "Operations, Administration and Management".

Mention of "(traditional telephony)" in description of signaling class is
potentially confusing.  Need to say that this signaling and control traffic
*before* using the T-word.
[Joe] Implemented the change. See section 2.2, etc.

The network control traffic classes look good.  For the user/subscriber
classes, more in the way of top-down rough criteria for sorting traffic into
classes would be a good addition.  Figure 1 is a start, but aspects such as
elasticity, burstiness, and whether the traffic is signaled seem to be
collapsed into the description.  I'm not sure whether another figure with
lots of columns would be better.
[Joe] Added new section and an additional figure.

A couple of things should be added.  First of all, a top-down description of
how the traffic space is divided up that focuses on the traffic
characteristics - in essence more structure in the center column in Figure
1.  What I have in mind is almost there, but breaking out things like but
elasticity, burstiness, and whether the traffic is signaled would probably
help, although a 6+ column table is probably not what's needed.

Second, a top down grouping the user/data classes into a smaller number of
groups by application protocol type would help - to a first approximation,
there are:
- 5 media-oriented classes, split into Interactive (Telephony, Multimedia
	conferencing, RT Interactive) and Streaming (Multimedia, Broadcast
	Video) subgroups.	
- Signaling
- 3 data classes (Low Latency, High Throughput, Low Priority), with
	Low Latency being Interactive, High Throughput being close to
	streaming (but more jitter-tolerant than media, plus TCP-like
	drop response), and Low Priority being Don't Care to some degree.
- Standard (Best Effort)

Signaling needs to be distinguished from Low Latency Data, as Signaling
traffic *is* low latency data.  The essential distinction is something like
"administrative control and management" of the traffic affected as the
protocols in this class tend to be tied to the (usually media)
streams/sessions they signal and control.
[Joe] The above issues/suggestion are addressed in new section 2.2
"Categorization of User Service Classes" and section 2.3 "Service Class
Characteristics".

In Figure 3, add RFC references for the two 3-color markers.
[Joe] Done.

Last bullet in Section 2.1 needs to say "one of" the two services can be
supported.  If Low Priority data is mapped to 000xxx, recommend that xxx not
be 000 to allow this to be distinguished from Standard.
[Joe] Made the above change and recommend that 000xx1.

-- Section 2.2

Editing needed: A network administrator determined that they need
	There are a number of editing problems elsewhere - I didn't
	flag all of them.
[Joe] Reworked section 2.2

"Figure 4, provides a popular industry view" - What's a "popular
	industry view"?  A different phrase should be used.
[Joe] Removed that phrase and replaced it with " Service Provider Network
Configuration Example 1".


Examples 2 & 3 - where did the numbers "five" and "eight" come from?
	Describing these networks in terms of traffic carried/services
	provided and the resulting differentiation goals would be
	better than picking numbers out of thin air.  In general,
	these examples are short on detail/explanation.
[Joe] These two examples have been expanded with service descriptions, etc.
in section 2.4.2 Example 2 and 2.4.3 Example 3.

-- Section 3.2

In the past, I think routing has divided interior vs. exterior routing
protocol traffic into CS7 vs. CS6.  One of the implications of this draft
appears to be that this approach is no longer recommended,
	[ ... snipped due to UPDATE ...]
UPDATE: I'm informed by people who know better than I that routing does not
use CS7, so it suffices to add routing to the list of traffic that is not
appropriate for the Administrative Service class.
[Joe] Left it as was.

Say how to distinguish Network Control signaling flows from user/subscriber
signaling (this is not as easy as it looks, as it may depend on who owns the
softswitch).
[Joe] Addressed in Section 2.2, and 5.1.

-- Section 4.1

IP Telephony "MUST" use Telephony service class is at odds with the "SHOULD"
for the list of protocols - either explain that the listed protocols can be
used for services other than IP Telephony, or downgrade the "MUST" to a
"SHOULD".
[Joe] Change it to SHOULD. 

   o  Packet flows from trusted sources (media gateways inside
      administered network) MAY not require policing.

Can't say "MAY not", and that's also not quite right.  Try "Policing is
OPTIONAL for traffic from trusted sources whose behavior is assured via
other means (e.g., administrative controls on those systems)." The text in
the immediately following bullet has a better structure.
[Joe] Changed it to state that policing is OPTIONAL.

-- Section 4.2

Explain "clipping".  In essence, if the signaling protocol that sets up the
connection fails to complete in the expected time, the result is audio
loss/degradation at the start of the session, which is particularly
noticeable when dealing with unforgiving POTS systems.
[Joe] Added Section 5.3 "Explanation of Ring Clipping".


How is "peer to peer real time control function" distinguished from low
latency data?  Data vs. control is fuzzy at best, so some discussion of what
is controlled and/or the importance of the control function would help.
[Joe] Addressed in several sections 2.2, 2.3 and 5.1.

-- Section 4.3

In first paragraph, expand "Video conferencing" example to "Audio and video
conferencing".
[Joe] Done.

(a) If the end point is not capable of setting
   the DSCP value, then the router topologically closest to the end
   point MUST perform Multifield (MF) Classification as defined in
   [RFC2475] and mark all packets as AF41.

That's peculiar - it looks like it assumes the end-point stays inside the
rate "A" profile/envelope, unless I missed something.  Was AF4x intended
instead of AF41?
[Joe] Changed it to AF4x.

(b) Just to complicate things - running trTCM in Color-Aware mode assumes
that "A" and "B" don't change across domain boundaries.  If they do,
Color-Blind may be necessary if the trTCM rates or burst sizes increase
significantly in the direction of flow (else get too much red & yellow
traffic, by comparison to traffic coming in on interfaces with trTCM running
Color-Blind).
[Joe] Made changes to Conditioning Performed at DiffServ Network Edge
sections to:
o  If packets are marked by a trusted sources or previous trusted DiffServ
domain, and the color marking is to be preserved, then the two rate three
color marker SHOULD be configured to operate in Color-Aware mode.
o  If the packet marking is not trusted or the color marking is not to be
preserved, then the two rate three color marker SHOULD be configured to
operate in Color-Blind mode.

(c) Should say something about use or non-use of marking to distinguish
application-defined content layers in a single [media] stream (e.g.,
complete frames vs. delta frames, additional data for higher resolution).
IMHO, history reveals this example to be an attractive nuisance 
(intellectually appealing, but unworkable in practice due to dynamic
variation in codec output rates) and hence ought to carry a "SHOULD NOT"
recommendation to disfavor it by comparison to the straight
application-level-unaware rate-metering described here.
[Joe] Deleted statement (last sentence in the first paragraph of section
4.3) that: "In the future, it is expected that these services will generate
continuously variable rate traffic with packet marking indicating flow
importance." No discussion was added to draft on "complete frames vs. delta
frames", etc. metering/conditioning scenarios as this draft only addresses
application-level-unaware rate metering.


-- Section 4.5

Section 4.3 comments (a), (b), and (c) apply here (but substitute AF3* for
any AF4* occurrences in those comments).
[Joe] (a) and (b) changed to as above for AF3*. (c) only stated
rate-metering.

-- Section 4.7

Section 4.3 comments (a), (b), and (c) apply here (but substitute AF2* for
any AF4* occurrences in those comments and likewise substitute srTCM for
trTCM).
[Joe] (a) and (b) changed to as above for AF2*. (c) only stated
rate-metering.

-- Section 4.8

Section 4.3 comments (a), (b), and (c) apply here (but substitute AF1* for
any AF4* occurrences in those comments).
[Joe] (a) and (b) changed to as above for AF1*. (c) only stated
rate-metering.

-- Section 4.9

   SHOULD be configured to receive a small
   percentage of forwarding resources (at least 5%).

That can't what was intended, as it places a cap on best effort bandwidth.
Remove the cap by doing something like inserting "at least" before "a small
percentage" and adding "as a guaranteed minimum" at the end.  I suspect this
is a typo/thinko/oversight.
[Joe] Modified sentence so that no cap is implied on bandwidth.

-- Section 5

In general, this section should be enriched/extended, and in particular:
- Please add a VPN service mapping discussion.
[Joe] Added Section 5.4 on support of service classes in VPNs. 

- A discussion of mapping media flows (audio and/or video) should go
    here, unless it's covered by the discussion requested in Section 2.1
[Joe] Addressed in Section 2.2
- Please add a summary of the UMTS traffic class mappings.
[Joe] After a closer analysis, we determined that further work is needed in
this area. We believe that the interworking between DiffServ service classes
and UMTS Traffic Classes should be addressed in a separate document.
Therefore, we removed from this draft any suggested mapping of DSCP values
and DiffServ service classes into UMTS Traffic Classes.


----------------------------------------------------
David L. Black, Senior Technologist
EMC Corporation, 176 South St., Hopkinton, MA  01748
+1 (508) 293-7953             FAX: +1 (508) 293-7786
black_david@emc.com        Mobile: +1 (978) 394-7754
----------------------------------------------------

_______________________________________________
tsvwg mailing list
tsvwg@ietf.org
https://www1.ietf.org/mailman/listinfo/tsvwg


Regards, Joe.
Jozef Babiarz
email: babiarz@nortelnetworks.com

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<P><FONT SIZE=3D2>I'm resending this email as not everyone has received =
it when I sent it out on Oct. 24. Sorry if you get it twice.</FONT>
<BR><FONT SIZE=3D2>Regards, Joe.</FONT>
<BR><FONT =
SIZE=3D2>************************************************************** =
</FONT>
<BR><FONT SIZE=3D2>Please see below answers to David L. Black review =
comments.</FONT>
<BR><FONT SIZE=3D2>Also, please see Section 7 "Summary of Changes from =
Previous Drat" for an outline of all the changes made to =
draft-baker-diffserv-basic-classes-04.txt. (04 version should be =
available from the on-line Internet-Drafts directory soon).</FONT></P>

<P><FONT =
SIZE=3D2>**********************************************************</FON=
T>
<BR><FONT SIZE=3D2>Review of =
draft-baker-diffserv-basic-classes-03.txt</FONT>
<BR><FONT SIZE=3D2>Reviewer: David L. Black, EMC Corporation</FONT>
<BR><FONT SIZE=3D2>For the IETF Transport Area Working Group =
(tsvwg)</FONT>
<BR><FONT SIZE=3D2>August 5, 2004</FONT>
<BR><FONT =
SIZE=3D2>-------------------------------------------------</FONT>
</P>

<P><FONT SIZE=3D2>-- Overall comment</FONT>
</P>

<P><FONT SIZE=3D2>I had skimmed an earlier version of this draft and =
liked it.&nbsp; Having read this one in detail, I like it a =
*lot*.&nbsp; Diffserv has been described as a toolkit for constructing =
QoS services.&nbsp; This draft is like a set of project plans (or a =
do-it-yourself book on plumbing, wiring,</FONT></P>

<P><FONT SIZE=3D2>etc.) that explains how to use the tools to build =
useful things in a fashion accessible to non-experts.&nbsp; This draft =
is a long-overdue addition to the set of diffserv documents, and does a =
nice job of weaving together the important elements of diffserv to show =
how they can be used together in useful ways.&nbsp; I'd like to be able =
say that those of us working on diffserv had such a coherent vision in =
mind from the beginning ... ;-).</FONT></P>

<P><FONT SIZE=3D2>Most of these comments are details of various =
sorts.&nbsp; The important comments that affect content are:</FONT>
</P>

<P><FONT SIZE=3D2>- Add top down traffic and application classification =
(Section 2.1).</FONT>
<BR><FONT SIZE=3D2>[Joe]Added Section 2.2 "Categorization of User =
Service Classes" and reworked part of section 2.1 to become Section 2.3 =
"Service Class Characteristics". </FONT></P>

<P><FONT SIZE=3D2>- Examples 2 and 3 need work (Section 2.2).</FONT>
<BR><FONT SIZE=3D2>[Joe]Expanded old Section 2.2, now became Section =
2.4. Section 2.4.1 is Example 1. Section 2.4.2 is detailed Example 2 =
and Section 2.4.3 is detailed Example 3.</FONT></P>

<P><FONT SIZE=3D2>- AF usage discussion cleanup [(a), (b), and (c) =
comments on Sections</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>4.3, 4.5, =
4.7 and 4.8].</FONT>
<BR><FONT SIZE=3D2>[Joe] Implemented changes as discussed. See item 9 =
in Section 7 "Summary of Changes from Previous Draft" of version 0.4 of =
draft for details of changes.</FONT></P>

<P><FONT SIZE=3D2>- Remove implied cap on Best Effort traffic (Section =
4.9 - probably</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>a =
typo/thinko, but an important one).</FONT>
<BR><FONT SIZE=3D2>[Joe] Modified sentence so that no cap is implied on =
bandwidth.</FONT>
</P>

<P><FONT SIZE=3D2>- Enrich mapping discussion (Section 5 to the extent =
not covered</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>by top =
down classification added to Section 2.1).</FONT>
<BR><FONT SIZE=3D2>[Joe]Section 5 was extended and restructured into 4 =
subsections, 5.1 Mapping for Signaling, 5.2 Mapping for NTP, 5.3 =
Explanation of Ring Clipping, and 5.4 VPN Service Mapping.</FONT></P>

<P><FONT SIZE=3D2>-- Section 1.4.1</FONT>
</P>

<P><FONT SIZE=3D2>&quot;with no further guarantees&quot; is at odds =
with the recommendation in Section 4.9 that some minimum amount of =
bandwidth be guaranteed to best effort.&nbsp; Some sort of expectation =
that the network will be usable for best effort traffic (in some sense) =
is needed here.</FONT></P>

<P><FONT SIZE=3D2>[Joe] Changed wording to be compatible with section =
4.9 &quot;Best Effort needs to provide some minimum bandwidth guarantee =
...&quot;</FONT></P>

<P><FONT SIZE=3D2>--&nbsp; Section 2.1</FONT>
</P>

<P><FONT SIZE=3D2>Expand OAM acronym on first use.</FONT>
<BR><FONT SIZE=3D2>[Joe] Done. OAM "Operations, Administration and =
Management".</FONT>
</P>

<P><FONT SIZE=3D2>Mention of &quot;(traditional telephony)&quot; in =
description of signaling class is potentially confusing.&nbsp; Need to =
say that this signaling and control traffic *before* using the =
T-word.</FONT></P>

<P><FONT SIZE=3D2>[Joe] Implemented the change. See section 2.2, =
etc.</FONT>
</P>

<P><FONT SIZE=3D2>The network control traffic classes look good.&nbsp; =
For the user/subscriber classes, more in the way of top-down rough =
criteria for sorting traffic into classes would be a good =
addition.&nbsp; Figure 1 is a start, but aspects such as elasticity, =
burstiness, and whether the traffic is signaled seem to be collapsed =
into the description.&nbsp; I'm not sure whether another figure with =
lots of columns would be better.</FONT></P>

<P><FONT SIZE=3D2>[Joe] Added new section and an additional =
figure.</FONT>
</P>

<P><FONT SIZE=3D2>A couple of things should be added.&nbsp; First of =
all, a top-down description of how the traffic space is divided up that =
focuses on the traffic characteristics - in essence more structure in =
the center column in Figure 1.&nbsp; What I have in mind is almost =
there, but breaking out things like but elasticity, burstiness, and =
whether the traffic is signaled would probably help, although a 6+ =
column table is probably not what's needed.</FONT></P>

<P><FONT SIZE=3D2>Second, a top down grouping the user/data classes =
into a smaller number of groups by application protocol type would help =
- to a first approximation, there are:</FONT></P>

<P><FONT SIZE=3D2>- 5 media-oriented classes, split into Interactive =
(Telephony, Multimedia</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>conferencing, RT Interactive) and Streaming (Multimedia, =
Broadcast</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Video) =
subgroups.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>- Signaling</FONT>
<BR><FONT SIZE=3D2>- 3 data classes (Low Latency, High Throughput, Low =
Priority), with</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Low =
Latency being Interactive, High Throughput being close to</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>streaming =
(but more jitter-tolerant than media, plus TCP-like</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>drop =
response), and Low Priority being Don't Care to some degree.</FONT>
<BR><FONT SIZE=3D2>- Standard (Best Effort)</FONT>
</P>

<P><FONT SIZE=3D2>Signaling needs to be distinguished from Low Latency =
Data, as Signaling traffic *is* low latency data.&nbsp; The essential =
distinction is something like &quot;administrative control and =
management&quot; of the traffic affected as the protocols in this class =
tend to be tied to the (usually media) streams/sessions they signal and =
control.</FONT></P>

<P><FONT SIZE=3D2>[Joe] The above issues/suggestion are addressed in =
new section 2.2 "Categorization of User Service Classes" and section =
2.3 "Service Class Characteristics".</FONT></P>

<P><FONT SIZE=3D2>In Figure 3, add RFC references for the two 3-color =
markers.</FONT>
<BR><FONT SIZE=3D2>[Joe] Done.</FONT>
</P>

<P><FONT SIZE=3D2>Last bullet in Section 2.1 needs to say &quot;one =
of&quot; the two services can be supported.&nbsp; If Low Priority data =
is mapped to 000xxx, recommend that xxx not be 000 to allow this to be =
distinguished from Standard.</FONT></P>

<P><FONT SIZE=3D2>[Joe] Made the above change and recommend that =
000xx1.</FONT>
</P>

<P><FONT SIZE=3D2>-- Section 2.2</FONT>
</P>

<P><FONT SIZE=3D2>Editing needed: A network administrator determined =
that they need</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>There are =
a number of editing problems elsewhere - I didn't</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>flag all =
of them.</FONT>
<BR><FONT SIZE=3D2>[Joe] Reworked section 2.2</FONT>
</P>

<P><FONT SIZE=3D2>&quot;Figure 4, provides a popular industry =
view&quot; - What's a &quot;popular</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>industry =
view&quot;?&nbsp; A different phrase should be used.</FONT>
<BR><FONT SIZE=3D2>[Joe] Removed that phrase and replaced it with " =
Service Provider Network Configuration Example 1".</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Examples 2 &amp; 3 - where did the numbers =
&quot;five&quot; and &quot;eight&quot; come from?</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Describing these networks in terms of traffic =
carried/services</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>provided =
and the resulting differentiation goals would be</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>better =
than picking numbers out of thin air.&nbsp; In general,</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>these =
examples are short on detail/explanation.</FONT>
<BR><FONT SIZE=3D2>[Joe] These two examples have been expanded with =
service descriptions, etc. in section 2.4.2 Example 2 and 2.4.3 Example =
3.</FONT></P>

<P><FONT SIZE=3D2>-- Section 3.2</FONT>
</P>

<P><FONT SIZE=3D2>In the past, I think routing has divided interior vs. =
exterior routing protocol traffic into CS7 vs. CS6.&nbsp; One of the =
implications of this draft appears to be that this approach is no =
longer recommended,</FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>[ ... =
snipped due to UPDATE ...]</FONT>
<BR><FONT SIZE=3D2>UPDATE: I'm informed by people who know better than =
I that routing does not use CS7, so it suffices to add routing to the =
list of traffic that is not appropriate for the Administrative Service =
class.</FONT></P>

<P><FONT SIZE=3D2>[Joe] Left it as was.</FONT>
</P>

<P><FONT SIZE=3D2>Say how to distinguish Network Control signaling =
flows from user/subscriber signaling (this is not as easy as it looks, =
as it may depend on who owns the softswitch).</FONT></P>

<P><FONT SIZE=3D2>[Joe] Addressed in Section 2.2, and 5.1.</FONT>
</P>

<P><FONT SIZE=3D2>-- Section 4.1</FONT>
</P>

<P><FONT SIZE=3D2>IP Telephony &quot;MUST&quot; use Telephony service =
class is at odds with the &quot;SHOULD&quot; for the list of protocols =
- either explain that the listed protocols can be used for services =
other than IP Telephony, or downgrade the &quot;MUST&quot; to a =
&quot;SHOULD&quot;.</FONT></P>

<P><FONT SIZE=3D2>[Joe] Change it to SHOULD. </FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;&nbsp; o&nbsp; Packet flows from trusted =
sources (media gateways inside</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; administered network) =
MAY not require policing.</FONT>
</P>

<P><FONT SIZE=3D2>Can't say &quot;MAY not&quot;, and that's also not =
quite right.&nbsp; Try &quot;Policing is OPTIONAL for traffic from =
trusted sources whose behavior is assured via other means (e.g., =
administrative controls on those systems).&quot; The text in the =
immediately following bullet has a better structure.</FONT></P>

<P><FONT SIZE=3D2>[Joe] Changed it to state that policing is =
OPTIONAL.</FONT>
</P>

<P><FONT SIZE=3D2>-- Section 4.2</FONT>
</P>

<P><FONT SIZE=3D2>Explain &quot;clipping&quot;.&nbsp; In essence, if =
the signaling protocol that sets up the connection fails to complete in =
the expected time, the result is audio loss/degradation at the start of =
the session, which is particularly noticeable when dealing with =
unforgiving POTS systems.</FONT></P>

<P><FONT SIZE=3D2>[Joe] Added Section 5.3 "Explanation of Ring =
Clipping".</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>How is &quot;peer to peer real time control =
function&quot; distinguished from low latency data?&nbsp; Data vs. =
control is fuzzy at best, so some discussion of what is controlled =
and/or the importance of the control function would help.</FONT></P>

<P><FONT SIZE=3D2>[Joe] Addressed in several sections 2.2, 2.3 and =
5.1.</FONT>
</P>

<P><FONT SIZE=3D2>-- Section 4.3</FONT>
</P>

<P><FONT SIZE=3D2>In first paragraph, expand &quot;Video =
conferencing&quot; example to &quot;Audio and video =
conferencing&quot;.</FONT>
<BR><FONT SIZE=3D2>[Joe] Done.</FONT>
</P>

<P><FONT SIZE=3D2>(a) If the end point is not capable of setting</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; the DSCP value, then the router =
topologically closest to the end</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; point MUST perform Multifield (MF) =
Classification as defined in</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; [RFC2475] and mark all packets as =
AF41.</FONT>
</P>

<P><FONT SIZE=3D2>That's peculiar - it looks like it assumes the =
end-point stays inside the rate &quot;A&quot; profile/envelope, unless =
I missed something.&nbsp; Was AF4x intended instead of AF41?</FONT></P>

<P><FONT SIZE=3D2>[Joe] Changed it to AF4x.</FONT>
</P>

<P><FONT SIZE=3D2>(b) Just to complicate things - running trTCM in =
Color-Aware mode assumes that &quot;A&quot; and &quot;B&quot; don't =
change across domain boundaries.&nbsp; If they do, Color-Blind may be =
necessary if the trTCM rates or burst sizes increase significantly in =
the direction of flow (else get too much red &amp; yellow traffic, by =
comparison to traffic coming in on interfaces with trTCM running =
Color-Blind).</FONT></P>

<P><FONT SIZE=3D2>[Joe] Made changes to Conditioning Performed at =
DiffServ Network Edge sections to:</FONT>
<BR><FONT SIZE=3D2>o&nbsp; If packets are marked by a trusted sources =
or previous trusted DiffServ domain, and the color marking is to be =
preserved, then the two rate three color marker SHOULD be configured to =
operate in Color-Aware mode.</FONT></P>

<P><FONT SIZE=3D2>o&nbsp; If the packet marking is not trusted or the =
color marking is not to be preserved, then the two rate three color =
marker SHOULD be configured to operate in Color-Blind mode.</FONT></P>

<P><FONT SIZE=3D2>(c) Should say something about use or non-use of =
marking to distinguish application-defined content layers in a single =
[media] stream (e.g., complete frames vs. delta frames, additional data =
for higher resolution). IMHO, history reveals this example to be an =
attractive nuisance </FONT></P>

<P><FONT SIZE=3D2>(intellectually appealing, but unworkable in practice =
due to dynamic variation in codec output rates) and hence ought to =
carry a &quot;SHOULD NOT&quot; recommendation to disfavor it by =
comparison to the straight application-level-unaware rate-metering =
described here.</FONT></P>

<P><FONT SIZE=3D2>[Joe] Deleted statement (last sentence in the first =
paragraph of section 4.3) that: &quot;In the future, it is expected =
that these services will generate continuously variable rate traffic =
with packet marking indicating flow importance.&quot; No discussion was =
added to draft on "complete frames vs. delta frames", etc. =
metering/conditioning scenarios as this draft only addresses =
application-level-unaware rate metering.</FONT></P>
<BR>

<P><FONT SIZE=3D2>-- Section 4.5</FONT>
</P>

<P><FONT SIZE=3D2>Section 4.3 comments (a), (b), and (c) apply here =
(but substitute AF3* for any AF4* occurrences in those =
comments).</FONT>
<BR><FONT SIZE=3D2>[Joe] (a) and (b) changed to as above for AF3*. (c) =
only stated rate-metering.</FONT>
</P>

<P><FONT SIZE=3D2>-- Section 4.7</FONT>
</P>

<P><FONT SIZE=3D2>Section 4.3 comments (a), (b), and (c) apply here =
(but substitute AF2* for any AF4* occurrences in those comments and =
likewise substitute srTCM for trTCM).</FONT></P>

<P><FONT SIZE=3D2>[Joe] (a) and (b) changed to as above for AF2*. (c) =
only stated rate-metering.</FONT>
</P>

<P><FONT SIZE=3D2>-- Section 4.8</FONT>
</P>

<P><FONT SIZE=3D2>Section 4.3 comments (a), (b), and (c) apply here =
(but substitute AF1* for any AF4* occurrences in those =
comments).</FONT>
<BR><FONT SIZE=3D2>[Joe] (a) and (b) changed to as above for AF1*. (c) =
only stated rate-metering.</FONT>
</P>

<P><FONT SIZE=3D2>-- Section 4.9</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;&nbsp; SHOULD be configured to receive a =
small</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; percentage of forwarding resources (at =
least 5%).</FONT>
</P>

<P><FONT SIZE=3D2>That can't what was intended, as it places a cap on =
best effort bandwidth.&nbsp; Remove the cap by doing something like =
inserting &quot;at least&quot; before &quot;a small percentage&quot; =
and adding &quot;as a guaranteed minimum&quot; at the end.&nbsp; I =
suspect this is a typo/thinko/oversight.</FONT></P>

<P><FONT SIZE=3D2>[Joe] Modified sentence so that no cap is implied on =
bandwidth.</FONT>
</P>

<P><FONT SIZE=3D2>-- Section 5</FONT>
</P>

<P><FONT SIZE=3D2>In general, this section should be enriched/extended, =
and in particular:</FONT>
<BR><FONT SIZE=3D2>- Please add a VPN service mapping =
discussion.</FONT>
<BR><FONT SIZE=3D2>[Joe] Added Section 5.4 on support of service =
classes in VPNs. </FONT>
</P>

<P><FONT SIZE=3D2>- A discussion of mapping media flows (audio and/or =
video) should go</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; here, unless it's covered by the =
discussion requested in Section 2.1</FONT>
<BR><FONT SIZE=3D2>[Joe] Addressed in Section 2.2</FONT>
<BR><FONT SIZE=3D2>- Please add a summary of the UMTS traffic class =
mappings.</FONT>
<BR><FONT SIZE=3D2>[Joe] After a closer analysis, we determined that =
further work is needed in this area. We believe that the interworking =
between DiffServ service classes and UMTS Traffic Classes should be =
addressed in a separate document. Therefore, we removed from this draft =
any suggested mapping of DSCP values and DiffServ service classes into =
UMTS Traffic Classes.</FONT></P>
<BR>

<P><FONT =
SIZE=3D2>----------------------------------------------------</FONT>
<BR><FONT SIZE=3D2>David L. Black, Senior Technologist</FONT>
<BR><FONT SIZE=3D2>EMC Corporation, 176 South St., Hopkinton, MA&nbsp; =
01748</FONT>
<BR><FONT SIZE=3D2>+1 (508) =
293-7953&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; FAX: +1 (508) 293-7786</FONT>
<BR><FONT =
SIZE=3D2>black_david@emc.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Mobile: +1 (978) 394-7754</FONT>
<BR><FONT =
SIZE=3D2>----------------------------------------------------</FONT>
</P>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>tsvwg mailing list</FONT>
<BR><FONT SIZE=3D2>tsvwg@ietf.org</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/tsvwg" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/tsvwg</A></FONT=
>
</P>
<BR>

<P><FONT SIZE=3D2>Regards, Joe.</FONT>
<BR><FONT SIZE=3D2>Jozef Babiarz</FONT>
<BR><FONT SIZE=3D2>email: babiarz@nortelnetworks.com</FONT>
</P>

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Subject: RE: [Tsvwg] Answers to Review of Diffserv Basic Classes Draft
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		Please see below answers to David L. Black review comments.
		Also, please see Section 7 "Summary of Changes from Previous
Drat" for an outline of all the changes made to
draft-baker-diffserv-basic-classes-04.txt. (04 version should be available
from the on-line Internet-Drafts directory soon).
		**********************************************************
		Review of draft-baker-diffserv-basic-classes-03.txt
		Reviewer: David L. Black, EMC Corporation
		For the IETF Transport Area Working Group (tsvwg)
		August 5, 2004
		-------------------------------------------------

		-- Overall comment

		I had skimmed an earlier version of this draft and liked it.
Having read this one in detail, I like it a *lot*.  Diffserv has been
described as a toolkit for constructing QoS services.  This draft is like a
set of project plans (or a do-it-yourself book on plumbing, wiring,
		etc.) that explains how to use the tools to build useful
things in a fashion accessible to non-experts.  This draft is a long-overdue
addition to the set of diffserv documents, and does a nice job of weaving
together the important elements of diffserv to show how they can be used
together in useful ways.  I'd like to be able say that those of us working
on diffserv had such a coherent vision in mind from the beginning ... ;-).

		Most of these comments are details of various sorts.  The
important comments that affect content are:

		- Add top down traffic and application classification
(Section 2.1).
		[Joe]Added Section 2.2 "Categorization of User Service
Classes" and reworked part of section 2.1 to become Section 2.3 "Service
Class Characteristics". 

		- Examples 2 and 3 need work (Section 2.2).
		[Joe]Expanded old Section 2.2, now became Section 2.4.
Section 2.4.1 is Example 1. Section 2.4.2 is detailed Example 2 and Section
2.4.3 is detailed Example 3.

		- AF usage discussion cleanup [(a), (b), and (c) comments on
Sections
			4.3, 4.5, 4.7 and 4.8].
		[Joe] Implemented changes as discussed. See item 9 in
Section 7 "Summary of Changes from Previous Draft" of version 0.4 of draft
for details of changes.

		- Remove implied cap on Best Effort traffic (Section 4.9 -
probably
			a typo/thinko, but an important one).
		[Joe] Modified sentence so that no cap is implied on
bandwidth.

		- Enrich mapping discussion (Section 5 to the extent not
covered
			by top down classification added to Section 2.1).
		[Joe]Section 5 was extended and restructured into 4
subsections, 5.1 Mapping for Signaling, 5.2 Mapping for NTP, 5.3 Explanation
of Ring Clipping, and 5.4 VPN Service Mapping.

		-- Section 1.4.1

		"with no further guarantees" is at odds with the
recommendation in Section 4.9 that some minimum amount of bandwidth be
guaranteed to best effort.  Some sort of expectation that the network will
be usable for best effort traffic (in some sense) is needed here.
		[Joe] Changed wording to be compatible with section 4.9
"Best Effort needs to provide some minimum bandwidth guarantee ..."

		--  Section 2.1

		Expand OAM acronym on first use.
		[Joe] Done. OAM "Operations, Administration and Management".

		Mention of "(traditional telephony)" in description of
signaling class is potentially confusing.  Need to say that this signaling
and control traffic *before* using the T-word.
		[Joe] Implemented the change. See section 2.2, etc.

		The network control traffic classes look good.  For the
user/subscriber classes, more in the way of top-down rough criteria for
sorting traffic into classes would be a good addition.  Figure 1 is a start,
but aspects such as elasticity, burstiness, and whether the traffic is
signaled seem to be collapsed into the description.  I'm not sure whether
another figure with lots of columns would be better.
		[Joe] Added new section and an additional figure.

		A couple of things should be added.  First of all, a
top-down description of how the traffic space is divided up that focuses on
the traffic characteristics - in essence more structure in the center column
in Figure 1.  What I have in mind is almost there, but breaking out things
like but elasticity, burstiness, and whether the traffic is signaled would
probably help, although a 6+ column table is probably not what's needed.

		Second, a top down grouping the user/data classes into a
smaller number of groups by application protocol type would help - to a
first approximation, there are:
		- 5 media-oriented classes, split into Interactive
(Telephony, Multimedia
			conferencing, RT Interactive) and Streaming
(Multimedia, Broadcast
			Video) subgroups.	
		- Signaling
		- 3 data classes (Low Latency, High Throughput, Low
Priority), with
			Low Latency being Interactive, High Throughput being
close to
			streaming (but more jitter-tolerant than media, plus
TCP-like
			drop response), and Low Priority being Don't Care to
some degree.
		- Standard (Best Effort)

		Signaling needs to be distinguished from Low Latency Data,
as Signaling traffic *is* low latency data.  The essential distinction is
something like "administrative control and management" of the traffic
affected as the protocols in this class tend to be tied to the (usually
media) streams/sessions they signal and control.
		[Joe] The above issues/suggestion are addressed in new
section 2.2 "Categorization of User Service Classes" and section 2.3
"Service Class Characteristics".

		In Figure 3, add RFC references for the two 3-color markers.
		[Joe] Done.

		Last bullet in Section 2.1 needs to say "one of" the two
services can be supported.  If Low Priority data is mapped to 000xxx,
recommend that xxx not be 000 to allow this to be distinguished from
Standard.
		[Joe] Made the above change and recommend that 000xx1.

		-- Section 2.2

		Editing needed: A network administrator determined that they
need
			There are a number of editing problems elsewhere - I
didn't
			flag all of them.
		[Joe] Reworked section 2.2

		"Figure 4, provides a popular industry view" - What's a
"popular
			industry view"?  A different phrase should be used.
		[Joe] Removed that phrase and replaced it with " Service
Provider Network Configuration Example 1".


		Examples 2 & 3 - where did the numbers "five" and "eight"
come from?
			Describing these networks in terms of traffic
carried/services
			provided and the resulting differentiation goals
would be
			better than picking numbers out of thin air.  In
general,
			these examples are short on detail/explanation.
		[Joe] These two examples have been expanded with service
descriptions, etc. in section 2.4.2 Example 2 and 2.4.3 Example 3.

		-- Section 3.2

		In the past, I think routing has divided interior vs.
exterior routing protocol traffic into CS7 vs. CS6.  One of the implications
of this draft appears to be that this approach is no longer recommended,
			[ ... snipped due to UPDATE ...]
		UPDATE: I'm informed by people who know better than I that
routing does not use CS7, so it suffices to add routing to the list of
traffic that is not appropriate for the Administrative Service class.
		[Joe] Left it as was.

		Say how to distinguish Network Control signaling flows from
user/subscriber signaling (this is not as easy as it looks, as it may depend
on who owns the softswitch).
		[Joe] Addressed in Section 2.2, and 5.1.

		-- Section 4.1

		IP Telephony "MUST" use Telephony service class is at odds
with the "SHOULD" for the list of protocols - either explain that the listed
protocols can be used for services other than IP Telephony, or downgrade the
"MUST" to a "SHOULD".
		[Joe] Change it to SHOULD. 

		   o  Packet flows from trusted sources (media gateways
inside
		      administered network) MAY not require policing.

		Can't say "MAY not", and that's also not quite right.  Try
"Policing is OPTIONAL for traffic from trusted sources whose behavior is
assured via other means (e.g., administrative controls on those systems)."
The text in the immediately following bullet has a better structure.
		[Joe] Changed it to state that policing is OPTIONAL.

		-- Section 4.2

		Explain "clipping".  In essence, if the signaling protocol
that sets up the connection fails to complete in the expected time, the
result is audio loss/degradation at the start of the session, which is
particularly noticeable when dealing with unforgiving POTS systems.
		[Joe] Added Section 5.3 "Explanation of Ring Clipping".


		How is "peer to peer real time control function"
distinguished from low latency data?  Data vs. control is fuzzy at best, so
some discussion of what is controlled and/or the importance of the control
function would help.
		[Joe] Addressed in several sections 2.2, 2.3 and 5.1.

		-- Section 4.3

		In first paragraph, expand "Video conferencing" example to
"Audio and video conferencing".
		[Joe] Done.

		(a) If the end point is not capable of setting
		   the DSCP value, then the router topologically closest to
the end
		   point MUST perform Multifield (MF) Classification as
defined in
		   [RFC2475] and mark all packets as AF41.

		That's peculiar - it looks like it assumes the end-point
stays inside the rate "A" profile/envelope, unless I missed something.  Was
AF4x intended instead of AF41?
		[Joe] Changed it to AF4x.

		(b) Just to complicate things - running trTCM in Color-Aware
mode assumes that "A" and "B" don't change across domain boundaries.  If
they do, Color-Blind may be necessary if the trTCM rates or burst sizes
increase significantly in the direction of flow (else get too much red &
yellow traffic, by comparison to traffic coming in on interfaces with trTCM
running Color-Blind).
		[Joe] Made changes to Conditioning Performed at DiffServ
Network Edge sections to:
		o  If packets are marked by a trusted sources or previous
trusted DiffServ domain, and the color marking is to be preserved, then the
two rate three color marker SHOULD be configured to operate in Color-Aware
mode.
		o  If the packet marking is not trusted or the color marking
is not to be preserved, then the two rate three color marker SHOULD be
configured to operate in Color-Blind mode.

		(c) Should say something about use or non-use of marking to
distinguish application-defined content layers in a single [media] stream
(e.g., complete frames vs. delta frames, additional data for higher
resolution). IMHO, history reveals this example to be an attractive nuisance

		(intellectually appealing, but unworkable in practice due to
dynamic variation in codec output rates) and hence ought to carry a "SHOULD
NOT" recommendation to disfavor it by comparison to the straight
application-level-unaware rate-metering described here.
		[Joe] Deleted statement (last sentence in the first
paragraph of section 4.3) that: "In the future, it is expected that these
services will generate continuously variable rate traffic with packet
marking indicating flow importance." No discussion was added to draft on
"complete frames vs. delta frames", etc. metering/conditioning scenarios as
this draft only addresses application-level-unaware rate metering.


		-- Section 4.5

		Section 4.3 comments (a), (b), and (c) apply here (but
substitute AF3* for any AF4* occurrences in those comments).
		[Joe] (a) and (b) changed to as above for AF3*. (c) only
stated rate-metering.

		-- Section 4.7

		Section 4.3 comments (a), (b), and (c) apply here (but
substitute AF2* for any AF4* occurrences in those comments and likewise
substitute srTCM for trTCM).
		[Joe] (a) and (b) changed to as above for AF2*. (c) only
stated rate-metering.

		-- Section 4.8

		Section 4.3 comments (a), (b), and (c) apply here (but
substitute AF1* for any AF4* occurrences in those comments).
		[Joe] (a) and (b) changed to as above for AF1*. (c) only
stated rate-metering.

		-- Section 4.9

		   SHOULD be configured to receive a small
		   percentage of forwarding resources (at least 5%).

		That can't what was intended, as it places a cap on best
effort bandwidth.  Remove the cap by doing something like inserting "at
least" before "a small percentage" and adding "as a guaranteed minimum" at
the end.  I suspect this is a typo/thinko/oversight.
		[Joe] Modified sentence so that no cap is implied on
bandwidth.

		-- Section 5

		In general, this section should be enriched/extended, and in
particular:
		- Please add a VPN service mapping discussion.
		[Joe] Added Section 5.4 on support of service classes in
VPNs. 

		- A discussion of mapping media flows (audio and/or video)
should go
		    here, unless it's covered by the discussion requested in
Section 2.1
		[Joe] Addressed in Section 2.2
-	Please add a summary of the UMTS traffic class mappings.
		[Joe] After a closer analysis, we determined that further
work is needed in this area. We believe that the interworking between
DiffServ service classes and UMTS Traffic Classes should be addressed in a
separate document. Therefore, we removed from this draft any suggested
mapping of DSCP values and DiffServ service classes into UMTS Traffic
Classes.


		----------------------------------------------------
		David L. Black, Senior Technologist
		EMC Corporation, 176 South St., Hopkinton, MA  01748
		+1 (508) 293-7953             FAX: +1 (508) 293-7786
		black_david@emc.com        Mobile: +1 (978) 394-7754
		----------------------------------------------------

		_______________________________________________
		tsvwg mailing list
		tsvwg@ietf.org
		https://www1.ietf.org/mailman/listinfo/tsvwg
<https://www1.ietf.org/mailman/listinfo/tsvwg> 


		Regards, Joe.
		Jozef Babiarz
		email: babiarz@nortelnetworks.com

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<BODY>
<UL><UL>
<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Please see below =
answers to David L. Black review comments.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Also, please see =
Section 7 "Summary of Changes from Previous Drat" for an outline of all =
the changes made to draft-baker-diffserv-basic-classes-04.txt. (04 =
version should be availabl</FONT><FONT SIZE=3D2 FACE=3D"Courier New">e =
from the on-line Internet-Drafts directory soon).</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier =
New">**********************************************************</FONT></=
P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Review of =
draft-baker-diffserv-basic-classes-03.txt</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Reviewer: David L. =
Black, EMC Corporation</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">For the IETF =
Transport Area Working Group (tsvwg)</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">A</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">ugust 5, 2004</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier =
New">-------------------------------------------------</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">-- Overall =
comment</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">I had skimmed an =
earlier version of this draft and liked it.&nbsp; Having read this one =
in detail, I like it a *lot*.&nbsp; Diffserv has been described as a =
toolkit for constructing</FONT><FONT SIZE=3D2 FACE=3D"Courier New"> QoS =
services.&nbsp; This draft is like a set of project plans (or a =
do-it-yourself book on plumbing, wiring,</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">etc.) that explains =
how to use the tools to build useful things in a fashion accessible to =
non-experts.&nbsp; This draft is a long-overdue addition to the =
s</FONT><FONT SIZE=3D2 FACE=3D"Courier New">et of diffserv documents, =
and does a nice job of weaving together the important elements of =
diffserv to show how they can be used together in useful ways.&nbsp; =
I'd like to be able say that those of us working on diffserv had such a =
coherent vision in mind fro</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">m</FONT><FONT SIZE=3D2 FACE=3D"Courier New"> the beginning ... =
;-).</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Most of these =
comments are details of various sorts.&nbsp; The important comments =
that affect content are:</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">- Add top down =
traffic and application classification (Section 2.1).</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe]Added Section =
2.2</FONT> <FONT SIZE=3D2 FACE=3D"Courier New">"Categorization of User =
Service Classes"</FONT><FONT SIZE=3D2 FACE=3D"Courier New"> and =
reworked part of section 2.1 to become Section 2.3 "Service Class =
Characteristics". </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">- Examples 2 and 3 =
need work (Section 2.2).</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe]Expanded old =
Section 2.2, now became Section 2.4. Section 2.4.1 is Example =
1</FONT><FONT SIZE=3D2 FACE=3D"Courier New">. Section 2.4.2 is detailed =
Example 2 and Section 2.4.3 is detailed Example 3.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">- AF usage =
discussion cleanup [(a), (b), and (c) comments on Sections</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">4.3, 4.5, 4.7 and 4.8].</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] Implemented =
changes as discussed. See item 9 in Section 7 "Summary of Chang</FONT><F=
ONT SIZE=3D2 FACE=3D"Courier New">es from Previous Draft" of version =
0.4 of draft for details of changes.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">- Remove implied =
cap on Best Effort traffic (Section 4.9 - probably</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">a typo/thinko, but an important =
one).</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe]</FONT><FONT =
FACE=3D"Times New Roman"></FONT> <FONT SIZE=3D2 FACE=3D"Courier =
New">Modified sentence so that no cap is implied on</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New"></FONT> <FONT SIZE=3D2 FACE=3D"Courier =
New">bandwidth.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">- Enrich =
ma</FONT><FONT SIZE=3D2 FACE=3D"Courier New">pping discussion (Section =
5 to the extent not covered</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">by top down classification added to =
Section 2.1).</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe]Section 5 was =
extended and restructured into 4 subsections, 5.1</FONT> <FONT SIZE=3D2 =
FACE=3D"Courier New">Mapping for Signaling</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New">, 5.2</FONT> <FONT SIZE=3D2 FACE=3D"Courier =
New">Mapping for NTP</FONT><FONT SIZE=3D2 FACE=3D"Courier New">, =
5.3</FONT> <FONT SIZE=3D2 FACE=3D"Courier New">Explanation of Ring =
Clipping</FONT><FONT SIZE=3D2 FACE=3D"Courier New">, and 5.4</FONT> =
<FONT SIZE=3D2 FACE=3D"Courier New">VPN Service Mapping</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">-- Section =
1.4.1</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&quot;with no =
further guarantees&quot; is at odds with the recommendation in Section =
4.9 that some minimum amount of bandwidth be guaranteed to best =
effort.&nbsp; Some sort of expectation that the network will be usable =
f</FONT><FONT SIZE=3D2 FACE=3D"Courier New">or best effort traffic (in =
some sense) is needed here.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe]</FONT> <FONT =
SIZE=3D2 FACE=3D"Courier New">Changed wording to be compatible with =
section 4.9</FONT><FONT SIZE=3D2 FACE=3D"Courier New"></FONT> <FONT =
SIZE=3D2 FACE=3D"Courier New">&quot;Best Effort needs to provide some =
minimum bandwidth guarantee ...&quot;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">--&nbsp; Section =
2.1</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Expand OAM acronym =
on first use.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] Done. OAM =
"Operation</FONT><FONT SIZE=3D2 FACE=3D"Courier New">s, Administration =
and Management".</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Mention of =
&quot;(traditional telephony)&quot; in description of signaling class =
is potentially confusing.&nbsp; Need to say that this signaling and =
control traffic *before* using the T-word.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] Implemented =
the change. See section 2.2, etc.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">The network control =
traffic classes look good.&nbsp; For the user/subscriber classes, more =
in the way of top-down rough criteria for sorting traffic into classes =
would be a good addition.&nbsp; Figure 1 is a start</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">, but aspects such as elasticity, =
burstiness, and whether the traffic is signaled seem to be collapsed =
into the description.&nbsp; I'm not sure whether another figure with =
lots of columns would be better.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] Added new =
section and an additional figure.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">A cou</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">ple of things should be added.&nbsp; =
First of all, a top-down description of how the traffic space is =
divided up that focuses on the traffic characteristics - in essence =
more structure in the center column in Figure 1.&nbsp; What I have in =
mind is almost there, but</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New"></FONT> <FONT SIZE=3D2 FACE=3D"Courier New">breaking out things =
like but elasticity, burstiness, and whether the traffic is signaled =
would probably help, although a 6+ column table is probably not what's =
needed.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Second, a top down =
grouping the user/data classes into a smaller number of groups by =
ap</FONT><FONT SIZE=3D2 FACE=3D"Courier New">plication protocol type =
would help - to a first approximation, there are:</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">- 5 media-oriented =
classes, split into Interactive (Telephony, Multimedia</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">conferencing, RT Interactive) and =
Streaming (Multimedia, Broadcast</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">Video) =
subgroups.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">- =
Signaling</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">- 3 =
data</FONT><FONT SIZE=3D2 FACE=3D"Courier New"> classes (Low Latency, =
High Throughput, Low Priority), with</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">Low Latency being Interactive, High =
Throughput being close to</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">streaming (but more jitter-tolerant than =
media, plus TCP-like</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">drop response), and Low Priority being =
Don't Care to some degree.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">- S</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">tandard (Best Effort)</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Signaling needs to =
be distinguished from Low Latency Data, as Signaling traffic *is* low =
latency data.&nbsp; The essential distinction is something like =
&quot;administrative control and management&quot; of the traffic =
affected as the protocols in t</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">his class tend to be tied to the (usually media) streams/sessions =
they signal and control.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] The above =
issues/suggestion are addressed in new section 2.2 "</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">Categorization of User Service =
Classes</FONT><FONT SIZE=3D2 FACE=3D"Courier New">" and section 2.3 =
"</FONT><FONT SIZE=3D2 FACE=3D"Courier New">Service Class =
Characteristics</FONT><FONT SIZE=3D2 FACE=3D"Courier New">".</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">In Figu</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">re 3, add RFC references for the two =
3-color markers.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] =
Done.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Last bullet in =
Section 2.1 needs to say &quot;one of&quot; the two services can be =
supported.&nbsp; If Low Priority data is mapped to 000xxx, recommend =
that xxx not be 000 to allow this to be distinguishe</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">d from Standard.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] Made the =
above change and recommend that 000xx1.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">-- Section =
2.2</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Editing needed: A =
network administrator determined that they need</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">There are a number of editing problems =
elsewhere - I didn't</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">flag all of them.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] Reworked =
section 2.2</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&quot;Figure 4, =
provides a popular industry view&quot; - What's a =
&quot;popular</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">industry view&quot;?&nbsp; A different =
phrase should be used.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] Removed that =
phrase and replaced it with "</FONT><FONT FACE=3D"Times New =
Roman"></FONT> <FONT SIZE=3D2 FACE=3D"Courier New">Service Provider =
Network Configuration Example 1</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">".</FONT></P>
<BR>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier =
New">Examples</FONT><FONT SIZE=3D2 FACE=3D"Courier New"> 2 &amp; 3 - =
where did the numbers &quot;five&quot; and &quot;eight&quot; come =
from?</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">Describing these networks in terms of =
traffic carried/services</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">provided and the resulting =
differentiation goals would be</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">better than picking numbers out of thin =
air.&nbsp; In general,</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">these exampl</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New">es are short on detail/explanation.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] These two =
examples have been expanded with service descriptions, etc. in section =
2.4.2 Example 2 and 2.4.3 Example 3.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">-- Section =
3.2</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">In the past, I =
think routing has divided interior vs. exterior routing =
protocol</FONT><FONT SIZE=3D2 FACE=3D"Courier New"> traffic into CS7 =
vs. CS6.&nbsp; One of the implications of this draft appears to be that =
this approach is no longer recommended,</FONT></P>

<P ALIGN=3DLEFT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2 FACE=3D"Courier New">[ ... snipped due to UPDATE =
...]</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">UPDATE: I'm =
informed by people who know better than I that routing does not use =
CS7, so it suffic</FONT><FONT SIZE=3D2 FACE=3D"Courier New">es to add =
routing to the list of traffic that is not appropriate for the =
Administrative Service class.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] Left it as =
was.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Say how to =
distinguish Network Control signaling flows from user/subscriber =
signaling (this is not as easy as it looks, as it may</FONT> <FONT =
SIZE=3D2 FACE=3D"Courier New">depend on who owns the =
softswitch).</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] Addressed in =
Section 2.2, and 5.1.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">-- Section =
4.1</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">IP Telephony =
&quot;MUST&quot; use Telephony service class is at odds with the =
&quot;SHOULD&quot; for the list of protocols - either explain that the =
listed protocols can be used for services other than IP Telephony, or =
downgrade the &quot;MUST&quot; to a &quot;SHOULD&quot;.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] Change it to =
SH</FONT><FONT SIZE=3D2 FACE=3D"Courier New">OULD. </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; =
o&nbsp; Packet flows from trusted sources (media gateways =
inside</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; administered network) MAY not =
require policing.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Can't say &quot;MAY =
not&quot;, and that's also not quite right.&nbsp; Try &quot;Policing is =
OPTIONAL for traffic from trusted sources whose behavior i</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">s assured via other means (e.g., =
administrative controls on those systems).&quot; The text in the =
immediately following bullet has a better structure.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] Changed it to =
state that policing is OPTIONAL.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">-- Section =
4.2</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Explain =
&quot;clipping&quot;.&nbsp; In essence, if the</FONT> <FONT SIZE=3D2 =
FACE=3D"Courier New">signaling protocol that sets up the connection =
fails to complete in the expected time, the result is audio =
loss/degradation at the start of the session, which is particularly =
noticeable when dealing with unforgiving POTS systems.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] Added Section =
5.3 "</FONT><FONT SIZE=3D2 FACE=3D"Courier New">E</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New">xplanation of Ring Clipping</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New">".</FONT></P>
<BR>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">How is &quot;peer =
to peer real time control function&quot; distinguished from low latency =
data?&nbsp; Data vs. control is fuzzy at best, so some discussion of =
what is controlled and/or the importance of the control function would =
help.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Jo</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">e] Addressed in several sections 2.2, 2.3 =
and 5.1.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">-- Section =
4.3</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">In first paragraph, =
expand &quot;Video conferencing&quot; example to &quot;Audio and video =
conferencing&quot;.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] =
Done.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">(a) If the end =
point is not capable of setting</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; the =
DSCP value, then the router to</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">pologically closest to the end</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; point =
MUST perform Multifield (MF) Classification as defined in</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; =
[RFC2475] and mark all packets as AF41.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">That's peculiar - =
it looks like it assumes the end-point stays inside the rate =
&quot;A&quot; profile/envelope, unless I missed something.&nbsp; Was =
AF4x intended instead of AF41?</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] Changed it to =
AF4x.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">(b) Just to =
complicate things - running trTCM in Color-Aware mode</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New"> assumes that &quot;A&quot; and =
&quot;B&quot; don't change across domain boundaries.&nbsp; If they do, =
Color-Blind may be necessary if the trTCM rates or burst sizes increase =
significantly in the direction of flow (else get too much red &amp; =
yellow traffic, by comparison to traffic</FONT> <FONT SIZE=3D2 =
FACE=3D"Courier New">c</FONT><FONT SIZE=3D2 FACE=3D"Courier New">oming =
in on interfaces with trTCM running Color-Blind).</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] Made changes =
to Conditioning Performed at DiffServ Network Edge sections =
to:</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">o&nbsp; If packets =
are marked by a trusted sources or previous trusted</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New"></FONT> <FONT SIZE=3D2 FACE=3D"Courier =
New">DiffServ domain, and the color marking is to be pre</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">served, then</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New"></FONT> <FONT SIZE=3D2 FACE=3D"Courier New">the =
two rate three color marker SHOULD be configured to operate =
in</FONT><FONT SIZE=3D2 FACE=3D"Courier New"></FONT> <FONT SIZE=3D2 =
FACE=3D"Courier New">Color-Aware mode.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">o&nbsp; If the =
packet marking is not trusted or the color marking is not</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New"></FONT> <FONT SIZE=3D2 FACE=3D"Courier =
New">to be preserved, then the two rate three color marker SHOULD =
be</FONT><FONT SIZE=3D2 FACE=3D"Courier New"></FONT> <FONT SIZE=3D2 =
FACE=3D"Courier New">configured to operate in C</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New">olor-Blind mode.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">(c) Should say =
something about use or non-use of marking to distinguish =
application-defined content layers in a single [media] stream (e.g., =
complete frames vs. delta frames, additional data for higher =
resolution). IMHO, history reveals t</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New">his example to be an attractive nuisance =
</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">(intellectually =
appealing, but unworkable in practice due to dynamic variation in codec =
output rates) and hence ought to carry a &quot;SHOULD NOT&quot; =
recommendation to disfavor it by comparison to the straight =
application-</FONT><FONT SIZE=3D2 FACE=3D"Courier New">level-unaware =
rate-metering described here.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] D</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">eleted statement (last sentence in =
the</FONT><FONT SIZE=3D2 FACE=3D"Courier New"></FONT> <FONT SIZE=3D2 =
FACE=3D"Courier New">first paragraph of section 4.3) that: &quot;In the =
future, it is</FONT><FONT SIZE=3D2 FACE=3D"Courier New"></FONT> <FONT =
SIZE=3D2 FACE=3D"Courier New">expected that these services will =
generate continuously variable</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New"></FONT> <FONT SIZE=3D2 FACE=3D"Courier New">rate traffic with =
packet marking indicati</FONT><FONT SIZE=3D2 FACE=3D"Courier New">ng =
flow importance.&quot;</FONT><FONT SIZE=3D2 FACE=3D"Courier New"> No =
discussion was added to draft on "complete frames vs. delta frames", =
etc. metering/conditioning scenarios as this draft only addresses =
application-level-unaware rate metering.</FONT></P>
<BR>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">-- Section =
4.5</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Section 4.3 =
comments (a), (b), and (c)</FONT><FONT SIZE=3D2 FACE=3D"Courier New"> =
apply here (but substitute AF3* for any AF4* occurrences in those =
comments).</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] (a) and (b) =
changed to as above for AF3*. (c) only stated rate-metering.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">-- Section =
4.7</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Section 4.3 =
comments (a), (b), and (c) apply here (but substitute AF2* for any =
AF4</FONT><FONT SIZE=3D2 FACE=3D"Courier New">* occurrences in those =
comments and likewise substitute srTCM for trTCM).</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] (a) and (b) =
changed to as above for AF2*. (c) only stated rate-metering.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">-- Section =
4.8</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Section 4.3 =
comments (a), (b), and (c) apply here (but substitu</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">te AF1* for any AF4* occurrences in those =
comments).</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] (a) and (b) =
changed to as above for AF1*. (c) only stated rate-metering.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">-- Section =
4.9</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; SHOULD =
be configured to receive a small</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; =
percentage of forwarding resources (at least 5%).</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">That =
can't</FONT><FONT SIZE=3D2 FACE=3D"Courier New"> what was intended, as =
it places a cap on best effort bandwidth.&nbsp; Remove the cap by doing =
something like inserting &quot;at least&quot; before &quot;a small =
percentage&quot; and adding &quot;as a guaranteed minimum&quot; at the =
end.&nbsp; I suspect this is a typo/thinko/oversight.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe]</FONT> <FONT =
SIZE=3D2 FACE=3D"Courier New">Mod</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">ified sentence so that no cap is implied on bandwidth.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">-- Section =
5</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">In general, this =
section should be enriched/extended, and in particular:</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">- Please add a VPN =
service mapping discussion.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe]</FONT> <FONT =
SIZE=3D2 FACE=3D"Courier New">Added Section 5.4 on support of service =
classes in VPNs</FONT><FONT SIZE=3D2 FACE=3D"Courier New">.</FONT> </P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">-</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New"> A discussion of mapping media flows =
(audio and/or video) should go</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp;&nbsp; =
here, unless it's covered by the discussion requested in Section =
2.1</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] Addressed in =
Section 2.2</FONT></P>
</UL></UL>
<P><FONT SIZE=3D2 FACE=3D"Courier =
New">-&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT> <FONT SIZE=3D2 =
FACE=3D"Courier New">Please add a summary of the UMTS traffic class =
mappings.</FONT>
<UL><UL>
<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">[Joe] After a =
closer anal</FONT><FONT SIZE=3D2 FACE=3D"Courier New">ysis, we =
determined that further work is needed in this area. We believe that =
the interworking between DiffServ service classes and UMTS Traffic =
Classes should be addressed in a separate document. Therefore, we =
removed from this draft any</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New"></FONT> <FONT SIZE=3D2 FACE=3D"Courier New">suggested</FONT> <FONT =
SIZE=3D2 FACE=3D"Courier New">mapping</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New"> of DSCP values and DiffServ service classes into =
UMTS Traffic Classes.</FONT></P>
<BR>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier =
New">----------------------------------------------------</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">David L. Black, =
Senior Technologist</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">EMC Corporation, =
176 South St., Hopkinton, MA&nbsp; 01748</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">+1 (508) =
293-7953&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; FAX</FONT><FONT SIZE=3D2 FACE=3D"Courier New">: +1 (508) =
293-7786</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier =
New">black_david@emc.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Mobile: +1 (978) 394-7754</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier =
New">----------------------------------------------------</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier =
New">_______________________________________________</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">tsvwg mailing =
list</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier =
New">tsvwg@ietf.org</FONT></P>

<P ALIGN=3DLEFT><A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/tsvwg"><U></U><U><FONT =
COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Courier =
New">https://www1.ietf.org/mailman/listinfo/tsvwg</FONT></U></A></P>
<BR>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Arial">Regards, Joe.</FONT></P>

<P ALIGN=3DLEFT><FONT COLOR=3D"#000000" SIZE=3D2 FACE=3D"Arial">Jozef =
Babiarz</FONT></P>

<P ALIGN=3DLEFT><FONT COLOR=3D"#000000" SIZE=3D2 FACE=3D"Arial">email: =
babiarz@nortelnetworks.com</FONT></P>
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Date: Mon, 25 Oct 2004 13:35:57 +0200
From: Brian E Carpenter <brc@zurich.ibm.com>
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Cc: Kwok-Ho Chan <khchan@nortelnetworks.com>, Fred Baker <fred@cisco.com>,
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Subject: [Tsvwg] Re: Review comments diffserv-basic-classes-03
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Thanks. I guess Kathie will still have some objections, but this
seems to take care of my comments.

     Brian

Jozef Babiarz wrote:
> Thanks Brian for the review and comments. Your list of changes has been
> incorporated into version 04. See below for details.
> 
> Fred Baker wrote:
> 
>>At 01:41 PM 08/25/04 +0200, Brian E Carpenter wrote:
>>
>>
>>>A couple more thoughts, after a side conversation with Kathie.
>>>
>>>As I said, my comfort level is higher, but this doesn't remove some
>>>older basic concerns... maybe I should have mentioned these in the 
>>>tsvwg meeting, but I was rushing between sessions.
>>>
>>>1. Thirteen classes is about eight too many for real deployments. It
>>>will be far too easy for readers to miss the essential comment hidden 
>>>in the middle of section 2.1:
>>>
>>>
>>>>It is expected that network administrators will choose the service
>>>>classes that they will support based on their need, starting off with 
>>>>three or four service classes for user traffic and add others as the 
>>>>need arises.
>>>
>>>
>>>This really needs to be mentioned earlier and more prominently, with
>>>some emphasis that all classes are OPTIONAL except that the default 
>>>class is REQUIRED.
>>
>>
>>OK. Specifically where would you like this?
> 
> 
> How about the end of section 1.2? I think it needs to be in the reader's
> mind before hitting the details.
> 
> [Joe] Added below paragraph to section 1.2:
> The Default Forwarding "Standard" service class is REQUIRED, all other
> service classes are OPTIONAL. It is expected that network administrators
> will choose the level of service differentiation that they will support
> based on their need, starting off with three or four service classes for
> user traffic and add others as the need arises.
> 
> 
> -----Original Message-----
> From: Brian E Carpenter [mailto:brc@zurich.ibm.com] 
> Sent: Friday, July 16, 2004 10:45 AM
> To: Fred Baker; Babiarz, Jozef [CAR:0S03:EXCH]; Chan, Kwok-Ho
> [BL60:470:EXCH]
> Subject: Re: I-D ACTION:draft-baker-diffserv-basic-classes-03.txt
> 
> Hi,
> 
> My comfort level with this is much higher than with the
> earlier versions. I still have a couple of substantive
> comments, however.
> 
> 1. I think you need to add a clear statement somewhere in
> section 1 like: The facts that four code points are recommended
> for AF, and that one code point is recommended for EF, are
> arbitrary choices, and the architecture allows any reasonable
> number of AF and EF classes simultaneously. The choice of
> four AF classes and one EF class in the current document is
> also arbitrary, and operators MAY choose to operate more or
> fewer of either.
> 
> [Joe] Added the above statement to Section 1.4 Key Service Concepts, at the
> end of the first paragraph.
> 
> 2. Going further in the same direction, since all choices
> of code points and service classes are arbitrary in the diffserv
> architecture, I think that almost all the MUSTs in the document
> should be SHOULDs. There are just a few that could reasonably
> be left as MUST, namely
> 
> 
>>The service MUST be engineered so that EF marked packet flows
>>   have sufficient bandwidth in the network to provide guaranteed
>>   delivery.  
> 
> 
> (which really just repeating part of the definition of EF)
> 
> and
> 
> 
>>  The probability of loss of AF41 traffic MUST NOT
>>   exceed the probability of loss of AF42 traffic, which in turn MUST
>>   NOT exceed the probability of loss of AF43 traffic.
> 
> 
> (ditto for AF)
> 
> and
> 
> 
>>   The Standard service class MUST use the Default Forwarding (DF) PHB
>>   defined in [RFC2474] 
> 
> 
> [Joe] Changed all MUST to SHOULD except as noted above.
> (ditto for DF)
> 
> You certainly can't have constructs like
> 
> 
>>   RECOMMENDED Network Edge Conditioning:
>>   o  At peering points (between two DiffServ networks) where SLAs are
>>      in place, CS6 marked packets MUST be policed, e.g.  using a single
>>      rate with burst size (sr+bs) token bucket policer to keep the CS6
>>      marked packet flows to within the traffic rate specified in the
>>      SLA.
> 
> 
> with a MUST governed by a RECOMMENDED.
> [Joe] Changed the MUST to SHOULD throughout the draft.
> 
> Regards
>      Brian
> 
> Regards, Joe.
> email: babiarz@nortelnetworks.com
> 
> 

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On Mon, 25 Oct 2004, Stephan Baucke wrote:

> I'd like to bring up another clarification which didn't make it into the
> imp-guide in time (although I understand that the authors in principle
> agree to include it). It is a clarification of Karn's algorithm which
> sometimes results in wrong RTT samples being taken if not properly
> implemented.

<snip>

> We propose to include the following clarification in the imp-guide:
>
> Karn's algorithm: RTT measurements MUST only be made using a packet
> with TSN r if no packet with TSN less than or equal to r is
> retransmitted since r was first sent.

Yes, that's my fault. I was supposed to put that change in.

Randall: Since I currently can't log into your machine, can you make the
change?

~armando

0--                                              --0
| Armando L. Caro Jr.  |  Protocol Engineering Lab |
| www.armandocaro.net  |    University of Delaware |
0--                                              --0

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Hi,

I'd like to request a short time slot to present our "SCTP NAT Transverse Considerations" 
draft at tsvwg in DC.

The draft is at <http://www.ietf.org/internet-drafts/draft-xie-tsvwg-sctp-nat-00.txt> by
Q. Xie, R. Stewart, and M. Holdrege.

We'd also like to ask if it can be made a tsvwg work item for an informational RFC.

regards,
-Qiaobing





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