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Date: Sat, 01 Nov 2003 06:22:53 -0600
From: "Randall R. Stewart (home)" <randall@stewart.chicago.il.us>
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To: Hanping Long <hanl@ccpu.com>
CC: lksctp-developers@lists.sourceforge.net, tsvwg@ietf.org
Subject: Re: [Tsvwg] one-to-many/one-to-one socket API extension for SCTP
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Hanping Long wrote:

>Hi,
>
>Regarding the one-to-many(UDP style) and one-to-one(tcp style) socket
>interface in socket API, one-to-many is more effiencient in terms of
>exploring the new features of SCTP, can anyone tell me which feature that
>one-to-one style interface CANNOT make use of?
>
>  
>
The big issue (at least I know in KAME-BSD) is that a one-to-one style
socket is not allowed to send data until the connect() or accept() as
completed. This translates into not being able to piggyback data on
the third or fourth leg of the 4-way handshake. The one-to-many uses
implicit connection setup so that when you do a send(data, new-addr); it
will start the connection setup and piggyback the data with the cookie-echo.

At one time there were other features that could not be used by the 
one-to-one style. But
I think for the most mart Kacheong as done a good job of finding ways to 
make them
all work ;->

>Though each association could have its dedicated buffer space for
>one-to-many style API, all the associations will fail if their related
>socket fails. But if one socket fails for one-to-one style API, only one
>association will fail. Is this the benefit of one-to-one style API? or it's
>not possible in reality?
>  
>
What in the world is a "socket fails" .. There is no such thing. You 
either allocate
a socket up front with a
sd = socket(...);
or it just does not work. Once the socket is open the resources are 
dedicated and
I don't see how it could "fail"... if you might give me an example of 
what you
have in mind I might understand what you are driving at.. but I know from
the BSD work in KAME that once I have a socket open.. the socket itself
has nothing in it that can fail.. maybe linux is different.. I don't know...

>But for one-to-one style, the maximum number of associations is limited by
>the number of socket the specific OS allows to open, as well as the select()
>call could handle(the value of FD_SETSIZE).
>
>  
>
Well this is true. In a typical BSD system the max sockets is somewhere 
around
8000. Of course there is almost always a "MAX ASSOC" variable as well.. we
have it in BSD.. (not sure what the default is but I have had it up to 
20,000 before
and brought up 20k assoc)... But back to your point... if  max_socket > 
max_assoc
you have no issue.. however if max_assoc > max_socket (likely) then yep 
..what
you say above is true.

Select would get in the way if you had a lot of assoc in the one-to-one 
catagory.. but
there is something to remember about select and a one-to-many style socket.

1) If you are doing a select for reading when it returns true one of the
    associations is ready to read from.... not a specific one but one of 
them.

2) If you are doing a slect for writing its even more strange :-0 since most
    likely one of the associations under the socket can be written so it 
will
    almost always return true.. however if you want to write a specific
    association it might block (if you don't have non-blocking io set) since
    it might not have been the assoc that the select call said was 
writeable :-0
   Basically select for write just does not work right for the 
one-to-many you
   almost have to do non-blocking io and get the error (EAGAIN) and then
   go forward from there... If you really need to do a select for write 
on a
   specific assoc you might have to peel it off with sctp_peeloff into a
   one-to-one type socket... if theres room :->

R

>Any suggestion or idea is highly appreciated!
>
>hanping
>
>
>
>_______________________________________________
>tsvwg mailing list
>tsvwg@ietf.org
>https://www1.ietf.org/mailman/listinfo/tsvwg
>
>
>  
>


-- 
Randall R. Stewart
randall@stewart.chicago.il.us 815-342-5222 (cell phone)



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Subject: Re: [Tsvwg] one-to-many/one-to-one socket API extension for SCTP
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On Sat, 1 Nov 2003, Randall R. Stewart (home) wrote:

> Hanping Long wrote:
> 
> >Hi,
> >
> >Regarding the one-to-many(UDP style) and one-to-one(tcp style) socket
> >interface in socket API, one-to-many is more effiencient in terms of
> >exploring the new features of SCTP, can anyone tell me which feature that
> >one-to-one style interface CANNOT make use of?
> >
> >  
> >
> The big issue (at least I know in KAME-BSD) is that a one-to-one style
> socket is not allowed to send data until the connect() or accept() as
> completed. This translates into not being able to piggyback data on
> the third or fourth leg of the 4-way handshake. The one-to-many uses
> implicit connection setup so that when you do a send(data, new-addr); it
> will start the connection setup and piggyback the data with the cookie-echo.

In LKSCTP it is possible to do auto connect using sendto()/sendmsg() which
allows bundling of data with cookie-echo. 

> 
> At one time there were other features that could not be used by the 
> one-to-one style. But
> I think for the most mart Kacheong as done a good job of finding ways to 
> make them
> all work ;->

I can think of only one socket option that applies only to one-to-many
style sockets - SCTP_AUTOCLOSE.  

> 
> >Though each association could have its dedicated buffer space for
> >one-to-many style API, all the associations will fail if their related
> >socket fails. But if one socket fails for one-to-one style API, only one
> >association will fail. Is this the benefit of one-to-one style API? or it's
> >not possible in reality?
> >  
> >
> What in the world is a "socket fails" .. There is no such thing. You 
> either allocate
> a socket up front with a
> sd = socket(...);
> or it just does not work. Once the socket is open the resources are 
> dedicated and
> I don't see how it could "fail"... if you might give me an example of 
> what you
> have in mind I might understand what you are driving at.. but I know from
> the BSD work in KAME that once I have a socket open.. the socket itself
> has nothing in it that can fail.. maybe linux is different.. I don't know...

Hanping may have meant the failure of the process/thread owning the socket
as "socket fails".

> 
> >But for one-to-one style, the maximum number of associations is limited by
> >the number of socket the specific OS allows to open, as well as the select()
> >call could handle(the value of FD_SETSIZE).

Yes, there are these limits, but it is possible to increase at least the open
file limit at runtime using 'ulimit -n'.
We don't have a limit on the no. of associations per socket(one-to-many) in
LKSCTP. I guess we should have it as a configurable parameter. 

Also 2.6 Linux is going to support a more scalable event polling mechanism 
than poll or select called epoll.
	http://lse.sourceforge.net/epoll

I haven't tried out this mechanism with SCTP sockets on linux, but hopefully
it should work.

> >
> >  
> >
> Well this is true. In a typical BSD system the max sockets is somewhere 
> around
> 8000. Of course there is almost always a "MAX ASSOC" variable as well.. we
> have it in BSD.. (not sure what the default is but I have had it up to 
> 20,000 before
> and brought up 20k assoc)... But back to your point... if  max_socket > 
> max_assoc
> you have no issue.. however if max_assoc > max_socket (likely) then yep 
> ..what
> you say above is true.
> 
> Select would get in the way if you had a lot of assoc in the one-to-one 
> catagory.. but
> there is something to remember about select and a one-to-many style socket.
> 
> 1) If you are doing a select for reading when it returns true one of the
>     associations is ready to read from.... not a specific one but one of 
> them.
> 
> 2) If you are doing a slect for writing its even more strange :-0 since most
>     likely one of the associations under the socket can be written so it 
> will
>     almost always return true.. however if you want to write a specific
>     association it might block (if you don't have non-blocking io set) since
>     it might not have been the assoc that the select call said was 
> writeable :-0
>    Basically select for write just does not work right for the 
> one-to-many you
>    almost have to do non-blocking io and get the error (EAGAIN) and then
>    go forward from there... If you really need to do a select for write 
> on a
>    specific assoc you might have to peel it off with sctp_peeloff into a
>    one-to-one type socket... if theres room :->
> 
> R
> 
> >Any suggestion or idea is highly appreciated!
> >
> >hanping
> >
> >
> >
> >_______________________________________________
> >tsvwg mailing list
> >tsvwg@ietf.org
> >https://www1.ietf.org/mailman/listinfo/tsvwg
> >
> >
> >  
> >
> 
> 
> 


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From exim@www1.ietf.org  Sun Nov  2 00:10:29 2003
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Subject: [Tsvwg] minimum TCP MSS
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Does any one know the minimum MSS that TCP must support.
I think most TCP implementation, without Path MTU Discovery,
would use the outgoing interface MTU to calculate the MSS
if the destination is local.  If destination is remote, then use the
default MSS.  However, some host may try a smaller MSS even
if locally connected.  So I was wondering what does TCP must
support in order to be compliant with RFC.

It would be great if someone knows this answer.  It would save
me some time from scanning through the RFCs.

Thanks.

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On Thu, 30 Oct 2003, Bill Sommerfeld wrote:
> > Has this been discussed in the transport area?  Is there something to be 
> > done about this, e.g. a BCP document discussing this practice and 
> > suggesting ICMP DU's are OK when establishing connections -- or something 
> > else?
> 
> Maybe the right answer here is a "slewed parallel" connection
> establishment; start with one address, and if you get a soft error or
> a RTO-sized timeout, try a second, third, fourth, etc., connection,
> but leave the prior ones open in the event they complete successfully.
> 
> Soft errors wouldn't kill the first connection -- they'd just be a a
> hint to try another connection in parallel.
> 
> This should probably be buried in a library routine so that
> applications don't have to reinvent this wheel.

This is probably an obvious first thought, as what would be the desirable
behaviour.  But what would that imply in practice?  That at least some
parts of the basic API functions like connect() be augmented by a set of
new functions for managing connectivity?  That could be a forever-project,
while what we'd like is to get a fix that'd be usable in this decade :-).

I'm not saying that's not a probably relevant option to consider, but I 
think folks have been thinking about replacing APIs before and not much 
has come out of it.. so I'm a bit reluctant to start on that particular 
path...

-- 
Pekka Savola                 "You each name yourselves king, yet the
Netcore Oy                    kingdom bleeds."
Systems. Networks. Security. -- George R.R. Martin: A Clash of Kings


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From: "Brian F. G. Bidulock" <bidulock@openss7.org>
To: Van Dang <vdang3@hotmail.com>
Cc: tsvwg@ietf.org
Subject: Re: [Tsvwg] minimum TCP MSS
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Van,

536.

--brian

On Fri, 31 Oct 2003, Van Dang wrote:

> Does any one know the minimum MSS that TCP must support.
> I think most TCP implementation, without Path MTU Discovery,
> would use the outgoing interface MTU to calculate the MSS
> if the destination is local.  If destination is remote, then use the
> default MSS.  However, some host may try a smaller MSS even
> if locally connected.  So I was wondering what does TCP must
> support in order to be compliant with RFC.
>=20
> It would be great if someone knows this answer.  It would save
> me some time from scanning through the RFCs.
>=20
> Thanks.
>=20
> _________________________________________________________________
> Never get a busy signal because you are always connected  with high-spe=
ed=20
> Internet access. Click here to comparison-shop providers. =20
> https://broadband.msn.com
>=20
>=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 exim@www1.ietf.org  Mon Nov  3 16:42:23 2003
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From: The IESG <iesg-secretary@ietf.org>
To: IETF-Announce:;
Cc: Internet Architecture Board <iab@iab.org>,
        RFC Editor <rfc-editor@rfc-editor.org>, <tsvwg@ietf.org>
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Date: Mon, 03 Nov 2003 16:20:04 -0500
Subject: [Tsvwg] Document Action: 'Using TCP DSACKs and SCTP Duplicate TSNs
 to Detect Spurious Retransmissions' to Experimental RFC
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The IESG has approved following document:

- 'Using TCP DSACKs and SCTP Duplicate TSNs to Detect Spurious Retransmissions '
   <draft-ietf-tsvwg-dsack-use-02.txt> as an Experimental RFC

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
 
TCP and SCTP provide notification of duplicate segment receipt through 
DSACK and Duplicate TSN notification, respectively. This document shows
conservative methods of using this information to identify unnecessary 
retransmissions for various applications. This document does not, however
outline what a TCP or SCTP sender should do after a spurious 
retransmission is detected. It is hoped that future work building on the 
detection of spurious transmissions will make TCP and SCTP more robust
to reordererd packets.
 
Working Group Summary
 
The TSVWG supports this protocol going forward, and is investigating 
several proposals that rely on the detection of spurious retransmissions
which might benefit from this work.
 
Protocol Quality
 
This document was reviewed for the IESG by Allison Mankin and Jon Peterson.


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From: Rick Jones <rick.jones2@hp.com>
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Subject: Re: [Tsvwg] minimum TCP MSS
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It is my understanding that there is no minimum MSS specified for TCP (well, one
byte is perhaps implicit in TCP being a byte stream rather than a bit
stream...), although some TCP _implementations_ may have settings to reject
connections advertising an MSS less than some value.

Again, unless I am getting beyond my knowledge, 536 is the _default_ MSS - the
MSS that TCP should assume when it does not send or see an MSS option in the SYN
or SYN|ACK segment.

I _suspect_ but cannot quote chapter and verse to confirm, that 536 stems from
the requirement that a conforming IP stack must be able to reassemble IP
datagram fragments into IP datagrams of at least 576 bytes. That is the
"minimum, maximum IP datagram size" an IP stack must support, not a minimum MTU.
(IIRC, the minimum IPv4 MTU is 68 bytes, and 12XX for v6)

If TCP "knows" that the "minimum, maximum IP datagram size" is 576, then it
"knows" that using an MSS of 536 (576 less 20 bytes of TCP header and 20 bytes
of IP header, options may vary :) means that its traffic has at least a
snowball's chance of getting through.

rick jones
-- 
Wisdom Teeth are impacted, people are affected by the effects of events.
these opinions are mine, all mine; HP might not want them anyway... :)
feel free to post, OR email to raj in cup.hp.com  but NOT BOTH...

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Subject: Re: [Tsvwg] minimum TCP MSS
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Rick Jones wrote:

> It is my understanding that there is no minimum MSS specified for TCP (well, one
> byte is perhaps implicit in TCP being a byte stream rather than a bit
> stream...), although some TCP _implementations_ may have settings to reject
> connections advertising an MSS less than some value.
> 
> Again, unless I am getting beyond my knowledge, 536 is the _default_ MSS - the
> MSS that TCP should assume when it does not send or see an MSS option in the SYN
> or SYN|ACK segment.
> 
> I _suspect_ but cannot quote chapter and verse to confirm, that 536 stems from
> the requirement that a conforming IP stack must be able to reassemble IP
> datagram fragments into IP datagrams of at least 576 bytes. That is the
> "minimum, maximum IP datagram size" an IP stack must support, not a minimum MTU.
> (IIRC, the minimum IPv4 MTU is 68 bytes, and 12XX for v6)
> 
> If TCP "knows" that the "minimum, maximum IP datagram size" is 576, then it
> "knows" that using an MSS of 536 (576 less 20 bytes of TCP header and 20 bytes
> of IP header, options may vary :) means that its traffic has at least a
> snowball's chance of getting through.
> 
> rick jones

MSS = Maximum segment size.

TCP's max-min is based on IP's, as noted before.

There is no min segment size - TCP can, and does, send 0-length segments 
(though it wouldn't be productive if that's all it could send). That 
would mean a min overall packet of 40 bytes, but a 0-length TCP payload.

Joe


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From: David Borman <dab@cray.com>
Subject: Re: [Tsvwg] minimum TCP MSS
Date: Mon, 3 Nov 2003 17:24:07 -0600
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On Nov 3, 2003, at 12:11 PM, Rick Jones wrote:

> It is my understanding that there is no minimum MSS specified for TCP 
> (well, one
> byte is perhaps implicit in TCP being a byte stream rather than a bit
> stream...), although some TCP _implementations_ may have settings to 
> reject
> connections advertising an MSS less than some value.

> Again, unless I am getting beyond my knowledge, 536 is the _default_ 
> MSS - the
> MSS that TCP should assume when it does not send or see an MSS option 
> in the SYN
> or SYN|ACK segment.

Yes, see RFC 1122, section 4.2.2.6

>
> I _suspect_ but cannot quote chapter and verse to confirm, that 536 
> stems from
> the requirement that a conforming IP stack must be able to reassemble 
> IP
> datagram fragments into IP datagrams of at least 576 bytes. That is the
> "minimum, maximum IP datagram size" an IP stack must support, not a 
> minimum MTU.
> (IIRC, the minimum IPv4 MTU is 68 bytes, and 12XX for v6)

For IPv4, RFC 791 defines the minimum IP packet size as 68 bytes.  On 
page 25:

     Every internet module must be able to forward a datagram of 68
     octets without further fragmentation.  This is because an internet
     header may be up to 60 octets, and the minimum fragment is 8 octets.

     Every internet destination must be able to receive a datagram of 576
     octets either in one piece or in fragments to be reassembled.

So, avoiding any IP fragmentation, that puts the minimum TCP MSS at 28 
bytes.  But since every IP must be able to receive a 576 byte packet, 
either as a single piece or in fragments, that's how the TCP default 
MSS of 536 is derived.

> If TCP "knows" that the "minimum, maximum IP datagram size" is 576, 
> then it
> "knows" that using an MSS of 536 (576 less 20 bytes of TCP header and 
> 20 bytes
> of IP header, options may vary :) means that its traffic has at least a
> snowball's chance of getting through.

TCP MSS doesn't include IP or TCP options, it's the underlying MTU 
minus the fixed IP and TCP headers.  Senders need to adjust down the 
amount of TCP data based on any IP or TCP options that it is inserting.

> rick jones

			David Borman, dab@cray.com


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From exim@www1.ietf.org  Tue Nov  4 15:35:26 2003
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From: "Geib, Ruediger" <Ruediger.Geib@t-systems.com>
To: morita.naotaka@lab.ntt.co.jp
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Hello Naotaka,

your revised and new PPS related draft are marked=20
as TSV WG, but haven't been announced yet (at least I=20
couldn't find an announcement in the archive). After=20
reading both, I'm having some remarks, following below.

http://www.ietf.org/internet-drafts/draft-morita-tsvwg-pps-01.txt
http://www.ietf.org/internet-drafts/draft-morita-tsvwg-mfverify-00.txt

Remarks related to morita-tsvwg-pps-01:

While the draft is focused on terminals sending with=20
variable rates, the edge traffic control seems to assume=20
the peak rate as only parameter characterising the=20
controlled traffic. Further, section 8.4 demands=20
continuous sending of packets. Does that mean that=20
terminals sending with variable rates are expected=20
to continuosly create traffic at their peak rate if=20
they desire to use PPS?

Your third draft "Measurable Forwarding: A New per-Hop=20
Behavior (PHB)" wasn't listed in the "individual=20
submissions" section. If IETF didn't accept it this time,=20
a url to it would be useful as I think it helps=20
understanding pps operation.

The framework could put more emphasis on the "stateless=20
core" operation. This feature is a clear and very desirable=20
benefit of PPS against other signaling modes. It is=20
mentioned only once in section 4.2 so far.

I'll stop here and save minor and editorial comments for=20
a time when there's a clearer view on the future of PPS=20
within IETF.

Regards, R=FCdiger


R=FCdiger Geib
T-Systems
Systems Integration
Senior Expert, Technologiezentrum
Am Kavalleriesand 3, 64295 Darmstadt
Germany
Fon: +49 6151 832138
http://www.t-systems.com
=20

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From: Aaron Falk <falk@isi.edu>
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Spencer, Carl-

I just read through draft-dawkins-trigtran-linkup-01.txt and have a
couple of comments.  I'm not sure if this document is on the agenda
for tsvwg this IETF so I'm sending my comments now.

Hope this is helpful,

--aaron

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

In section 2.3, you say:

   Hosts supporting TCP-based applications over subnetwork interfaces
   subject to multi-second outages MAY perform the actions described
   in Section 3. These actions are more attractive for TCP
   implementations used with "human-in-the-loop" applications, but are
   safe for any TCP-based implementation.

Just to be clear: this proposal is only for end-systems, right?  Not
for routers?  The text here is clearly talking about hosts but later
in section 3 you talk about subnetworks.  The proposal seems a little
vague on this.  I feel like you are saying that he retransmitted
packet may come from a router along the path (as suggested in
draft-ietf-pilc-link-design) but the language seems to imply that it
is _hosts_ saving and retransmitting the packets.  A block diagram
would help. :)

Because, if it is the sender saving and retransmitting the packets and
the sender directly connected to the wireless interface, the question
comes up: why not just kick the RTO timer when the interface comes
back up, as Phil Karn suggested?

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

Section 3.1 "Layering  Violation Tradeoffs"

This section is quite confusing.  I don't know what is meant by a
"subnetwork implementation".  At a minimum it's mis-titled.  It
doesn't say why one should be concerned about layering violations, nor
what the tradeoffs are.

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


In section 3. "When a Local Interface Returns to UP", you say:

   When the subnetwork implementation detects that a local interface
   has returned to "UP" status, the subnetwork implementation MAY
   retransmit the last packet stored for each TCP connection.

(Again: the subnetwork vs. end-host issue.)

My concern here is that the proposal in this draft is for links that
are down for long enough that recovering from an RTO is onerous.
Links that are down for only a few RTTs should be allowed to recover
using the existing algorithm.  So, I think you should say something
about how long a link is down before engaging in this behavior.  For
example, you might say 'the local interface has returned to "UP" and
the/one/all connection(s) are in RETRANSMISSION-WAIT.'  Of course,
this means makes implementation a bit harder since you a) need to be
on the end-system and b) need to correlate packets you are saving to
resend with the state of the connection.  A simple approach would be
to have the implementor fix a lower bound in seconds.

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

Finally in Section 3. you need to say something about aging old
buffered packets.  I have this vision of hosts with lots of idle or
even closed connections, a link going down and up triggering a storm
of saved packets from hours/days previous trying to restart
connections that are idle.  The hosts and connections will probably
survive but it seems an un-neighborly thing to do to the network.  So,
I'd propose a setting maximum time that packets can be held, say 1 MSL
(2 minutes, I believe), by any entity doing this algorithm.

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


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> Your third draft "Measurable Forwarding: A New per-Hop 
> Behavior (PHB)" wasn't listed in the "individual 
> submissions" section. If IETF didn't accept it this time, 
> a url to it would be useful as I think it helps 
> understanding pps operation.

Just as an FYI - the IETF secretariat has been using some new tools for
managing Internet-Draft submissions for this IETF, and they're still
hammering out a few bugs. draft-morita-tsvwg-mfbhp-00 was submitted before
the deadline, and we're working to get it into the I-D repository ASAP.

Jon Peterson
NeuStar, Inc. 

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The chairs would like to announce a three-week Working Group Last Call
(allowing an extra week given the intervening IETF meeting) for
draft-ietf-tsvwg-slowstart-00, commencing on November 5th and concluding on
November 26th. Please send any final comments relating to this document to
the TSVWG mailing list.

Jon Peterson
NeuStar, Inc.

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From: "MORITA, Naotaka" <morita.naotaka@lab.ntt.co.jp>
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John,

Thank you for the confirmation.

At 22:03 03/11/04 -0500, Peterson, Jon wrote:
 >
 >> Your third draft "Measurable Forwarding: A New per-Hop
 >> Behavior (PHB)" wasn't listed in the "individual
 >> submissions" section. If IETF didn't accept it this time,
 >> a url to it would be useful as I think it helps
 >> understanding pps operation.
 >
 >Just as an FYI - the IETF secretariat has been using some new tools for
 >managing Internet-Draft submissions for this IETF, and they're still
 >hammering out a few bugs. draft-morita-tsvwg-mfbhp-00 was submitted before
 >the deadline, and we're working to get it into the I-D repository ASAP.
 >
 >Jon Peterson
 >NeuStar, Inc.
 >
 >_______________________________________________
 >tsvwg mailing list
 >tsvwg@ietf.org
 >https://www1.ietf.org/mailman/listinfo/tsvwg

----------------------------------
Naotaka MORITA, Senior Research Engineer, Supervisor
Network Service Systems Laboratories
Information Sharing Laboratory Group
NTT
3-9-11, Midori-Cho, Musashino-Shi, Tokyo 180-8585 JAPAN
Tel. +81-422-59-7464, ISDN-Fax +81-422-60-4012
E-mail: morita.naotaka@lab.ntt.co.jp
MSN messenger: moritanaotaka@hotmail.com
NTT home page: http://www.ntt.co.jp/about/e/index.html 



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Regarding PPS and MF-PHB, could someone give me any hints?

1) Real time video characterisctis
Are there any archive about real time (interactive) video?

2) Constant or Variable
I am personally collecting characteristics of interactive video traffic.
Some of them are quite flat and some of them are quite fluctuating.
Is it possible and reasonable to produce constant rate traffic
by adjusting picuture resolution or quality?

2) In the MF-PHB document, I describe a couple policer behaviors.
I know that there are some variants of the policer description.
Are there any reference in IETF?

3) In the MF-PHB document, I describe two-threshold policer and
two-rate policer.  I guess the two-threshold policer is similar
to single rate Two Rate Markder specified in RFC2697, but
is slightly different.  Are there any discussion about this kind of
new policers?

Naotaka

At 22:03 03/11/04 -0500, Peterson, Jon wrote:
 >
 >> Your third draft "Measurable Forwarding: A New per-Hop
 >> Behavior (PHB)" wasn't listed in the "individual
 >> submissions" section. If IETF didn't accept it this time,
 >> a url to it would be useful as I think it helps
 >> understanding pps operation.
 >
 >Just as an FYI - the IETF secretariat has been using some new tools for
 >managing Internet-Draft submissions for this IETF, and they're still
 >hammering out a few bugs. draft-morita-tsvwg-mfbhp-00 was submitted before
 >the deadline, and we're working to get it into the I-D repository ASAP.
 >
 >Jon Peterson
 >NeuStar, Inc.

----------------------------------
Naotaka MORITA, Senior Research Engineer, Supervisor
Network Service Systems Laboratories
Information Sharing Laboratory Group
NTT
3-9-11, Midori-Cho, Musashino-Shi, Tokyo 180-8585 JAPAN
Tel. +81-422-59-7464, ISDN-Fax +81-422-60-4012
E-mail: morita.naotaka@lab.ntt.co.jp
MSN messenger: moritanaotaka@hotmail.com
NTT home page: http://www.ntt.co.jp/about/e/index.html 



_______________________________________________
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I have question about the RTO.min value.  The draft says that there is one 
RTO.min value per stack.  But, it is possible a host could have one 
association that travels over the public Internet and others that travel 
over a short (a few meters) high speed links.  But, the lower latency local 
links will be forced to take at least the RTO.min value when re-transmitting 
data.

While it may have been shown that a RTO.min value of 1 second is appropriate 
for the public Internet, it seems to unduly slow recovery of low latency 
local links.

Could there not be a RTO.min value per association to accomodate this 
difference in latency that allows each link to operate efficiently?

- Kulwinder.

_________________________________________________________________
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K S A wrote:

>
> I have question about the RTO.min value.  The draft says that there is 
> one RTO.min value per stack.


What draft are you talking about? Socketsapi?

>   But, it is possible a host could have one association that travels 
> over the public Internet and others that travel over a short (a few 
> meters) high speed links.  But, the lower latency local links will be 
> forced to take at least the RTO.min value when re-transmitting data.
>
> While it may have been shown that a RTO.min value of 1 second is 
> appropriate for the public Internet, it seems to unduly slow recovery 
> of low latency local links.
>
> Could there not be a RTO.min value per association to accomodate this 
> difference in latency that allows each link to operate efficiently?
>
I know in KAME we implemented a RTO.min per endpoint.. which is 
inherited from the
RTO.min of the system.. that way you could change things for those 
special cases.

R

> - Kulwinder.
>
> _________________________________________________________________
> Add photos to your e-mail with MSN 8. Get 2 months FREE*.  
> http://join.msn.com/?page=features/photos&pgmarket=en-ca&RU=http%3a%2f%2fjoin.msn.com%2f%3fpage%3dmisc%2fspecialoffers%26pgmarket%3den-ca 
>
>
>
>


-- 
Randall R. Stewart
randall@stewart.chicago.il.us 815-342-5222 (cell phone)



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To: "Randall R. Stewart (home)" <randall@stewart.chicago.il.us>
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On Wed, 5 Nov 2003, Randall R. Stewart (home) wrote:

> K S A wrote:
> 
> >
> > I have question about the RTO.min value.  The draft says that there is 
> > one RTO.min value per stack.
> 
> 
> What draft are you talking about? Socketsapi?
> 
> >   But, it is possible a host could have one association that travels 
> > over the public Internet and others that travel over a short (a few 
> > meters) high speed links.  But, the lower latency local links will be 
> > forced to take at least the RTO.min value when re-transmitting data.
> >
> > While it may have been shown that a RTO.min value of 1 second is 
> > appropriate for the public Internet, it seems to unduly slow recovery 
> > of low latency local links.
> >
> > Could there not be a RTO.min value per association to accomodate this 
> > difference in latency that allows each link to operate efficiently?
> >
> I know in KAME we implemented a RTO.min per endpoint.. which is 
> inherited from the
> RTO.min of the system.. that way you could change things for those 
> special cases.

Even in LK-SCTP, all the RTO parameters(initial, min and max) are tunable
per association or endpoint. This can be done using the new SCTP_RTOINFO 
socket option defined in the sockets api extension draft.

Thanks
Sridhar

> 
> R
> 
> > - Kulwinder.
> >
> > _________________________________________________________________
> > Add photos to your e-mail with MSN 8. Get 2 months FREE*.  
> > http://join.msn.com/?page=features/photos&pgmarket=en-ca&RU=http%3a%2f%2fjoin.msn.com%2f%3fpage%3dmisc%2fspecialoffers%26pgmarket%3den-ca 
> >
> >
> >
> >
> 
> 
> 


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Please send any comments or suggestions on the agenda to myself and Allison.

Jon Peterson
NeuStar, Inc.

----

Transport Area WG (tsvwg)

Monday, November 10 at 0900-1130
=============================

CHAIRS:	Allison Mankin <mankin@psg.com> 
	      Jon Peterson <jon.peterson@neustar.biz>

0900 Charter Discussion (chairs)

0910 Newreno Update (Sally Floyd) (5 mins)
http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-newreno-01.txt

0915 Reorder/Channel Errors (Sally Floyd for Bhandarkar/Reddy) (15 mins)
http://www.ietf.org/internet-drafts/draft-bhandarkar-tcp-dcr-00.txt

0930 FAST TCP (Cheng/Low) (15 mins)
http://www.ietf.org/internet-drafts/draft-jin-wei-low-tcp-fast-01.txt

0945 Early Retransmit (Mark Allman) (15 mins)
http://www.ietf.org/internet-drafts/draft-allman-tcp-early-rexmt-02.txt

1000 IPMP (Matthew Luckie) (10 mins)
http://www.ietf.org/internet-drafts/draft-mcgregor-ipmp-02.txt

1010 RFC1323bis (David Borman) (15 mins)
http://www.ietf.org/internet-drafts/draft-jacobson-tsvwg-1323bis-00.txt

1025 Priority Promotion Scheme (Naotaka MORITA) (15 mins)
http://www.ietf.org/internet-drafts/draft-morita-tsvwg-pps-01.txt

1040 "Packet Drop" chunk type (Randall Stewart) (15 mins)
http://www.ietf.org/internet-drafts/draft-stewart-sctp-pktdrprep-00.txt

1055 SCTP Chunk Authentication (Michael Tuexen) (15 mins)
http://www.ietf.org/internet-drafts/draft-tuexen-sctp-auth-chunk-00.txt

1110 De-correlated Loss Recovery (Yogesh Prem Swami) (10 mins)
http://www.ietf.org/internet-drafts/draft-swami-tsvwg-tcp-dclor-02.txt
http://www.ietf.org/internet-drafts/draft-swami-tcp-lmdr-01.txt

1120 Multi-link Transport (Hosei Matsuoka) (10 mins)
http://www.ietf.org/internet-drafts/draft-matsuoka-multilink-transport-00.tx
t

1130 Close meeting

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On Wednesday, November 5, 2003, at 03:20 PM, Peterson, Jon wrote:
> Please send any comments or suggestions on the agenda to myself and 
> Allison.

At the last meeting it was mentioned that there were plans to create a
new working group to handle the TCP stuff that's been being done in
tsvwg.  I think that's a good idea - what's its status?

Melinda


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Melinda Shore'" <mshore@cisco.com>
Cc: "'tsvwg@ietf.org'" <tsvwg@ietf.org>
Subject: RE: [Tsvwg] proposed agenda for TSVWG Meeting in Minneapolis
Date: Wed, 5 Nov 2003 18:07:10 -0500 
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Yes, this will be discussed during the allotted "charter" time at the
beginning of the session. I'm pleased to say that we do have a proposed
charter, and a plan for migrating existing TSVWG items over to the new WG.

More details will follow before the meeting on Monday.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Melinda Shore [mailto:mshore@cisco.com]
> Sent: Wednesday, November 05, 2003 1:32 PM
> To: Peterson, Jon
> Cc: 'tsvwg@ietf.org'
> Subject: Re: [Tsvwg] proposed agenda for TSVWG Meeting in Minneapolis
> 
> 
> On Wednesday, November 5, 2003, at 03:20 PM, Peterson, Jon wrote:
> > Please send any comments or suggestions on the agenda to myself and 
> > Allison.
> 
> At the last meeting it was mentioned that there were plans to create a
> new working group to handle the TCP stuff that's been being done in
> tsvwg.  I think that's a good idea - what's its status?
> 
> Melinda
> 

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Subject: Re: [Tsvwg] one-to-many/one-to-one socket API extension for SCTP
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>
>On Wed, 5 Nov 2003, Randall R. Stewart (home) wrote:
>
> > K S A wrote:
> >
> > >
> > > I have question about the RTO.min value.  The draft says that there is
> > > one RTO.min value per stack.
> >
> >
> > What draft are you talking about? Socketsapi?

draft-ietf-tsvwg-sctpsocket-07.txt on the WG page.
Where is/are the lastest references? I had read RFC2960 which also has this 
limitation.

- Thanks, Kulwinder.

> >
> > >   But, it is possible a host could have one association that travels
> > > over the public Internet and others that travel over a short (a few
> > > meters) high speed links.  But, the lower latency local links will be
> > > forced to take at least the RTO.min value when re-transmitting data.
> > >
> > > While it may have been shown that a RTO.min value of 1 second is
> > > appropriate for the public Internet, it seems to unduly slow recovery
> > > of low latency local links.
> > >
> > > Could there not be a RTO.min value per association to accomodate this
> > > difference in latency that allows each link to operate efficiently?
> > >
> > I know in KAME we implemented a RTO.min per endpoint.. which is
> > inherited from the
> > RTO.min of the system.. that way you could change things for those
> > special cases.
>
>Even in LK-SCTP, all the RTO parameters(initial, min and max) are tunable

>per association or endpoint. This can be done using the new SCTP_RTOINFO
>socket option defined in the sockets api extension draft.
>
>Thanks
>Sridhar
>
> >
> > R
> >
> > > - Kulwinder.
> > >
> > > _________________________________________________________________
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> > >
> > >
> > >
> >
> >
> >
>
>
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From: "Geib, Ruediger" <Ruediger.Geib@t-systems.com>
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Subject: [Tsvwg] More comments on the pps drafts
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Hello Naotaka,

some more thoughts after having read the MF PHB draft (which I felt to =
be enjoyable reading):

- PPS requires a "customised" output policing=20
  rather than a special queuing discipline. It=20
  may be useful if the framework could=20
  explicitely mention this important fact.=20
- The re-routing and mobility cases will require=20
  more thought. I doubt that a state free attempt=20
  will do. How does a router learn that some MF-H=20
  traffic is exceptional?
  Fast thought: something like a path-token=20
  transmitted within a domain and tracked by=20
  routers?=20
- Also suiting variable rate sources may require=20
  some state. Here, information about the number=20
  of active sources and their peak rate may be=20
  useful. This could help to set the MF-M=20
  threshold in the dynamic case and also be=20
  used for buffer dimensioning.
- Buffer dimensioning needs some more discussion.
- When does an access system decide that a=20
  session is terminated? Without a timer, a=20
  terminal may continue to send MF-H traffic=20
  because it considers the session still to be=20
  up. Again, a limited amount of state would be=20
  required to handle the issue.

The state maintenance requirements I mentioned still aren't as big as =
with RSVP and would refer to aggregates in the core.=20

Regards, R=FCdiger


R=FCdiger Geib
T-Systems
Systems Integration
Senior Expert, Technologiezentrum
Am Kavalleriesand 3, 64295 Darmstadt
Germany
Fon: +49 6151 832138
http://www.t-systems.com
=20

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Subject: Re: [Tsvwg] More comments on the pps drafts
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Ruediger,

At 15:14 03/11/06 +0100, Geib, Ruediger wrote:
 >Hello Naotaka,
 >
 >some more thoughts after having read the MF PHB draft (which I felt to =
be
 >enjoyable reading):

Thank you again.

 >
 >- PPS requires a "customised" output policing
 >  rather than a special queuing discipline. It
 >  may be useful if the framework could
 >  explicitely mention this important fact.

Rate limiting is the required function.
Policer or shaper can be used for the rate limiting.
Due to the short-delay characteristics and
being potential to extend to the elastic mode,
the policer is preferable.
But the shaper, for the quick deployment,
can be used.

 >- The re-routing and mobility cases will require
 >  more thought. I doubt that a state free attempt
 >  will do. How does a router learn that some MF-H
 >  traffic is exceptional?
 >  Fast thought: something like a path-token
 >  transmitted within a domain and tracked by
 >  routers?

Yes, it is a state free attempt.
But the router does not distingush exceptional MF-H traffic from ordinal =
one.
For the mobility case, it will work, because the exceptional traffic
may not be so large.
For the re-routing case, if we limit the topology and anticipate the wors=
t
re-routed traffic, it should work.

 >- Also suiting variable rate sources may require
 >  some state. Here, information about the number
 >  of active sources and their peak rate may be
 >  useful. This could help to set the MF-M
 >  threshold in the dynamic case and also be
 >  used for buffer dimensioning.

I am sorry.  I do not catch your point.
As you have pointed out at the previous mail,
the important feature of PPS is stateless inside the nework,
or inside the core at least.
For the ingress edge, per-flow state may be necessary
to avoid unexpected terminal behavior.

Do you mean the state for varible rate source is at the ingress egde,
or all routers?

 >- Buffer dimensioning needs some more discussion.

I fully agree.  I am now investigating in my labs.

 >- When does an access system decide that a
 >  session is terminated? Without a timer, a
 >  terminal may continue to send MF-H traffic
 >  because it considers the session still to be
 >  up. Again, a limited amount of state would be
 >  required to handle the issue.

Yes, I agree.  For the ingress edge, a limited amout of state
is necessary for unexpected use.
I assume that the terminal send "Bye" message at the SIP level
and the SIP proxy makes the edge router release access control list.

 >
 >The state maintenance requirements I mentioned still aren't as big as w=
ith
 >RSVP and would refer to aggregates in the core.

I agree.

I would like to stress the following points.
1. The state per MF-PHB class is mandatory.  It is the rate limit value a=
t=20
the policer or the shaper.
The rate limit is per-class basis, at the output port at all routers, or=20
even L2 switches.
2. Per-flow states are at terminals.
3. If the operator want to avoid unexpected use such as malicious use or=20
failuer condition,
the edge router should keep per-flow states for per-flow policing at the=20
input port at the ingress router.

Naotaka

 >
 >Regards, R=1B$B=97E=1B(Biger
 >
 >
 >R=1B$B=97E=1B(Biger Geib
 >T-Systems
 >Systems Integration
 >Senior Expert, Technologiezentrum
 >Am Kavalleriesand 3, 64295 Darmstadt
 >Germany
 >Fon: +49 6151 832138
 >http://www.t-systems.com
 >
 >
 >_______________________________________________
 >tsvwg mailing list
 >tsvwg@ietf.org
 >https://www1.ietf.org/mailman/listinfo/tsvwg

----------------------------------
Naotaka MORITA, Senior Research Engineer, Supervisor
Network Service Systems Laboratories
Information Sharing Laboratory Group
NTT
3-9-11, Midori-Cho, Musashino-Shi, Tokyo 180-8585 JAPAN
Tel. +81-422-59-7464, ISDN-Fax +81-422-60-4012
E-mail: morita.naotaka@lab.ntt.co.jp
MSN messenger: moritanaotaka@hotmail.com
NTT home page: http://www.ntt.co.jp/about/e/index.html=20



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You can refer to the following web site for measurement-based admission control.

http://lcn.imit.kth.se/PBAC/


----------------------------------
Naotaka MORITA, Senior Research Engineer, Supervisor
Network Service Systems Laboratories
Information Sharing Laboratory Group
NTT
3-9-11, Midori-Cho, Musashino-Shi, Tokyo 180-8585 JAPAN
Tel. +81-422-59-7464, ISDN-Fax +81-422-60-4012
E-mail: morita.naotaka@lab.ntt.co.jp
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From: "MORITA, Naotaka" <morita.naotaka@lab.ntt.co.jp>
Subject: Re: [Tsvwg] Comments on draft-morita-tsvwg-pps-01
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I thought I had replied to the TSV mailing list, not directly to Ruediger=
,
Naotaka
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Ruediger,

Thank you very much for your comments.

At 16:08 03/11/04 +0100, you wrote:
 >Hello Naotaka,
 >
 >your revised and new PPS related draft are marked
 >as TSV WG, but haven't been announced yet (at least I
 >couldn't find an announcement in the archive). After
 >reading both, I'm having some remarks, following below.
 >
 >http://www.ietf.org/internet-drafts/draft-morita-tsvwg-pps-01.txt
 >http://www.ietf.org/internet-drafts/draft-morita-tsvwg-mfverify-00.txt
 >

Actually I posted three documents.  Those are avilable now.
The most important one is
http://www.ietf.org/internet-drafts/draft-morita-tsvwg-mfphb-00.txt

 >Remarks related to morita-tsvwg-pps-01:
 >
 >While the draft is focused on terminals sending with
 >variable rates, the edge traffic control seems to assume
 >the peak rate as only parameter characterising the
 >controlled traffic. Further, section 8.4 demands
 >continuous sending of packets. Does that mean that
 >terminals sending with variable rates are expected
 >to continuosly create traffic at their peak rate if
 >they desire to use PPS?

I need to check the consistency of the description.
Thanks.
For the edge, I assume peak reate policing per-flow basis.
But the source can be variable bit rate.

 >
 >Your third draft "Measurable Forwarding: A New per-Hop
 >Behavior (PHB)" wasn't listed in the "individual
 >submissions" section. If IETF didn't accept it this time,
 >a url to it would be useful as I think it helps
 >understanding pps operation.

Fortunately, this missing one is now on the web site.

 >
 >The framework could put more emphasis on the "stateless
 >core" operation. This feature is a clear and very desirable
 >benefit of PPS against other signaling modes. It is
 >mentioned only once in section 4.2 so far.

Thank you again.  That is the bottom line of this scheme.

 >
 >I'll stop here and save minor and editorial comments for
 >a time when there's a clearer view on the future of PPS
 >within IETF.

If you make additional comments on MF-PHB according to the
last draft, I am happy.

Regards,
Naotaka

 >
 >Regards, R=1B$B=97E=1B(Biger
 >
 >
 >R=1B$B=97E=1B(Biger Geib
 >T-Systems
 >Systems Integration
 >Senior Expert, Technologiezentrum
 >Am Kavalleriesand 3, 64295 Darmstadt
 >Germany
 >Fon: +49 6151 832138
 >http://www.t-systems.com
 >
 >
 >_______________________________________________
 >tsvwg mailing list
 >tsvwg@ietf.org
 >https://www1.ietf.org/mailman/listinfo/tsvwg

----------------------------------
Naotaka MORITA, Senior Research Engineer, Supervisor
Network Service Systems Laboratories
Information Sharing Laboratory Group
NTT
3-9-11, Midori-Cho, Musashino-Shi, Tokyo 180-8585 JAPAN
Tel. +81-422-59-7464, ISDN-Fax +81-422-60-4012
E-mail: morita.naotaka@lab.ntt.co.jp
MSN messenger: moritanaotaka@hotmail.com
NTT home page: http://www.ntt.co.jp/about/e/index.html=20



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From exim@www1.ietf.org  Fri Nov  7 11:38:23 2003
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I have a comment on draft-jacobson-tsvwg-1323bis-00.txt.

I think the Windows TCP behaviour of sending a zero-valued TSecr on
SYN-ACK rather than using the TSval value sent on the SYN needs to be
addressed in RFC1323bis.  The new APPENDIX E Pseudo-code Summary does
not allow for this behaviour.

It looks to me like at least some implementations do not allow for this
behaviour, probably leading to SRTT and RTTVAR being initialised to very
large values at the SYN-ACK.  Looking at the FreeBSD source and TCP/IP
Illustrated Vol 2 I cannot find any code that disregards zero-valued
TSecr echoes.

It seems to me that the RFC1323 (and RFC1323bis) specification could
be read both to allow and dis-allow this behaviour.  Thus I think this
aspect of the specification should be made clearer.

The pertinent section of RFC1323bis is in 3.2:

    The Timestamp Echo Reply field (TSecr) is only valid if the ACK
    bit is set in the TCP header; if it is valid, it echos a
    timestamp value that was sent by the remote TCP in the TSval
    field of a Timestamps option.  When TSecr is not valid, its
    value must be zero.  The TSecr value will generally be from the
    most recent Timestamp option that was received; however, there
    are exceptions that are explained below.

a) If one takes the sentence "When TSecr is not valid, its value must
be zero." in isolation, it could be taken to mean that whenever an
implementation does not have a valid TSecr to hand (for whatever reason),
it is OK to send a zero TSecr.  This justifies the Windows implementation.

b) If one takes the paragraph as a whole, one can argue that the word
"valid" is specified for this paragraph to have a particular meaning:
"valid" = ACK bit set in header.  Thus all the sentence in a) really
means in this context "The TSecr in a TCP header without the ACK bit set
must have a value of zero".  So "if it is valid [ACK bit set], it echos
a timestamp value that was sent by the remote TCP in the TSval" applies
and Windows TCP is wrong not to echo the TSval it was sent on the SYN.

Interpretation b) can take support from "These TSval values are echoed
in TSecr values in the reverse direction." in 3.3, and the TS.Recent
discussion.

Whatever view is taken on the correct interpretation of RFC1323, at
this stage I do not think a new RFC should disregard at least 3 years
widespread practice by Windows TCP.  Windows probably does this because
it does not set up a full connection control block until the connection
is fully open, and so does not remember the SYN TSval in a way that can
be echoed in the SYN-ACK.  This is not entirely unreasonable behaviour.

If my argument that this area of the specification needs clarifying is
accepted, and the specification will permit the Windows TCP practice,
it has to be decided if it will allow a zero TSecr only on the SYN-ACK,
or on any segment where an implementation wishes to do this.

I do not have a view on this.  However it is worth noting that allowing a
zero TSecr anywhere an implementation chooses is pertinent to the issue
Reiner Ludwig raised ([Tsvwg] on 14 Aug 2003):

    Another open issue with RFC1323, I think, is the question of
    whether *every* segment MUST carry a TS option once that was
    negotiated in the SYN/SYN-ACK handshake.

since it impacts algorithms and testing in a similar way.

	Richard
-- 
Richard Wendland				richard@codeburst.co.uk

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Transport Area WG (tsvwg)

Monday, November 10 at 0900-1130
=============================

CHAIRS:	Allison Mankin <mankin@psg.com> 
	      Jon Peterson <jon.peterson@neustar.biz>

0900 Charter Discussion (chairs)

0910 "Packet Drop" chunk type (Randall Stewart) (15 mins)
http://www.ietf.org/internet-drafts/draft-stewart-sctp-pktdrprep-00.txt

0925 Newreno Update (Sally Floyd) (5 mins)
http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-newreno-01.txt

0930 Reorder/Channel Errors (Sally Floyd for Bhandarkar/Reddy) (15 mins)
http://www.ietf.org/internet-drafts/draft-bhandarkar-tcp-dcr-00.txt

0945 FAST TCP (Cheng/Low) (15 mins)
http://www.ietf.org/internet-drafts/draft-jin-wei-low-tcp-fast-01.txt

1000 LinkUp (Spencer Dawkins) (15 mins)
http://www.ietf.org/internet-drafts/draft-dawkins-trigtran-linkup-01.txt

1015 IPMP (Matthew Luckie) (10 mins)
http://www.ietf.org/internet-drafts/draft-mcgregor-ipmp-02.txt

1025 RFC1323bis (David Borman) (15 mins)
http://www.ietf.org/internet-drafts/draft-jacobson-tsvwg-1323bis-00.txt

1040 Priority Promotion Scheme (Naotaka MORITA) (15 mins)
http://www.ietf.org/internet-drafts/draft-morita-tsvwg-pps-01.txt

1055 SCTP Chunk Authentication (Michael Tuexen) (15 mins)
http://www.ietf.org/internet-drafts/draft-tuexen-sctp-auth-chunk-00.txt

1110 De-correlated Loss Recovery (Yogesh Prem Swami) (10 mins)
http://www.ietf.org/internet-drafts/draft-swami-tsvwg-tcp-dclor-02.txt
http://www.ietf.org/internet-drafts/draft-swami-tcp-lmdr-01.txt

1120 Multi-link Transport (Hosei Matsuoka) (10 mins)
http://www.ietf.org/internet-drafts/draft-matsuoka-multilink-transport-00.tx
t

1130 Close meeting


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Date: Tue, 11 Nov 2003 00:38:55 +0100
From: Roland Bless <bless@tm.uka.de>
To: "MORITA, Naotaka" <morita.naotaka@lab.ntt.co.jp>
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Subject: [Tsvwg] Re: [NSIS] PPS and MF-PHB
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Hi Naotaka,

On Fri, 07 Nov 2003 09:36:44 +0900
"MORITA, Naotaka" <morita.naotaka@lab.ntt.co.jp> wrote:

> I would like to draw QoS experts attention to the following three documents.
> 
> http://www.ietf.org/internet-drafts/draft-morita-tsvwg-pps-01.txt
> http://www.ietf.org/internet-drafts/draft-morita-tsvwg-mfphb-00.txt
> http://www.ietf.org/internet-drafts/draft-morita-tsvwg-mfverify-00.txt
> 
> I am going to make a presentation on Monday morning at TSV WG.

After reading draft-morita-tsvwg-mfphb-00.txt and attending your
presentation, I have some remarks related to this approach:
- I'm not quite sure whether this is really a PHB definition.
  It seems to me, that it is actually a mix of
  * a scheme for measured-based admission control and resource provisioning
  * a Per-Domain Behavior (see RFC 3086)
  * a PHB group description

- The scheme inherently relies on feedback information and it should be made
  clear how the feedback can be provided (you mentioned RTCP as an example,
  what about others?).

- Instead of defining a new PHB, possibly two Class Selector PHBs
  or even an AF class can be used ("green" packets get high priority after
  admission control, "yellow" and "red" packet may be used for probing).
  AF usually gives no guarantees for delay and jitter, however, if AF is
  provisioned properly, "green" packets may get some guarantee.

- For security and robustness, it is essential to police the markings
  for the higher priority. Otherwise the service guarantees for other users
  are violated by a malicious or ill-behaving end-system. Therefore, I think
  first-hop routers should do the policing (based on the feedback
  information?).

- I see some value in the overall approach, but IMHO the proposal may 
  be turned into a description for a measurement-based
  admission control scheme and PDB using Class Selector PHBs.

I copied to the relevant lists (diffserv-interest and tsvwg) as I'm 
interested in other opinions.

Best regards,
 Roland

_______________________________________________
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From: "MORITA, Naotaka" <morita.naotaka@lab.ntt.co.jp>
Subject: Re: [Tsvwg] Re: [NSIS] PPS and MF-PHB
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At 00:38 03/11/11 +0100, Roland Bless wrote:
 >Hi Naotaka,
 >
 >On Fri, 07 Nov 2003 09:36:44 +0900
 >"MORITA, Naotaka" <morita.naotaka@lab.ntt.co.jp> wrote:
 >
 >> I would like to draw QoS experts attention to the following three documents.
 >>
 >> http://www.ietf.org/internet-drafts/draft-morita-tsvwg-pps-01.txt
 >> http://www.ietf.org/internet-drafts/draft-morita-tsvwg-mfphb-00.txt
 >> http://www.ietf.org/internet-drafts/draft-morita-tsvwg-mfverify-00.txt
 >>
 >> I am going to make a presentation on Monday morning at TSV WG.
 >
 >After reading draft-morita-tsvwg-mfphb-00.txt and attending your
 >presentation, I have some remarks related to this approach:
 >- I'm not quite sure whether this is really a PHB definition.
 >  It seems to me, that it is actually a mix of
 >  * a scheme for measured-based admission control and resource provisioning
 >  * a Per-Domain Behavior (see RFC 3086)
 >  * a PHB group description

Your understanding is correct, but I do not think it is a mix.
The key of the PPS is to utilize PHB for admission control.
The proposed new PHB, MF-PHB refers to two-subclasses,
but we do not have to define it as a group description.

 >
 >- The scheme inherently relies on feedback information and it should be made
 >  clear how the feedback can be provided (you mentioned RTCP as an example,
 >  what about others?).

Yes, it is.  As the PPS, I would like to describe the feedback mechanims,
but for a PHB, we do not have to.

 >
 >- Instead of defining a new PHB, possibly two Class Selector PHBs
 >  or even an AF class can be used ("green" packets get high priority after
 >  admission control, "yellow" and "red" packet may be used for probing).
 >  AF usually gives no guarantees for delay and jitter, however, if AF is
 >  provisioned properly, "green" packets may get some guarantee.

This kind of disucssion is what I wanted to do.
Sorry for spending much time on me presentation only.
Are you refering to Three Color Marker in RFC2697 and RFC2698?

 >
 >- For security and robustness, it is essential to police the markings
 >  for the higher priority. Otherwise the service guarantees for other users
 >  are violated by a malicious or ill-behaving end-system. Therefore, I think
 >  first-hop routers should do the policing (based on the feedback
 >  information?).

I agree.  Are there any opinion about the description in pps framework document?
Not sufficient?

 >
 >- I see some value in the overall approach, but IMHO the proposal may
 >  be turned into a description for a measurement-based
 >  admission control scheme and PDB using Class Selector PHBs.
 >
 >I copied to the relevant lists (diffserv-interest and tsvwg) as I'm
 >interested in other opinions.

Me, too.

 >
 >Best regards,
 > Roland
 >
 >_______________________________________________
 >tsvwg mailing list
 >tsvwg@ietf.org
 >https://www1.ietf.org/mailman/listinfo/tsvwg

----------------------------------
Naotaka MORITA, Senior Research Engineer, Supervisor
Network Service Systems Laboratories
Information Sharing Laboratory Group
NTT
3-9-11, Midori-Cho, Musashino-Shi, Tokyo 180-8585 JAPAN
Tel. +81-422-59-7464, ISDN-Fax +81-422-60-4012
E-mail: morita.naotaka@lab.ntt.co.jp
MSN messenger: moritanaotaka@hotmail.com
NTT home page: http://www.ntt.co.jp/about/e/index.html 



_______________________________________________
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Date: Tue, 11 Nov 2003 05:28:27 +0100
From: Roland Bless <bless@tm.uka.de>
To: "MORITA, Naotaka" <morita.naotaka@lab.ntt.co.jp>
Cc: diffserv-interest@ietf.org, "'tsvwg@ietf.org'" <tsvwg@ietf.org>
Subject: Re: [Tsvwg] Re: [NSIS] PPS and MF-PHB
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Hi Naotaka,

>  >After reading draft-morita-tsvwg-mfphb-00.txt and attending your
>  >presentation, I have some remarks related to this approach:
>  >- I'm not quite sure whether this is really a PHB definition.
>  >  It seems to me, that it is actually a mix of
>  >  * a scheme for measured-based admission control and resource provisioning
                     ^^^^^^^^sorry, should have been: measurement-based admission control
>  >  * a Per-Domain Behavior (see RFC 3086)
>  >  * a PHB group description

> Your understanding is correct, but I do not think it is a mix.
> The key of the PPS is to utilize PHB for admission control.

Hmm, I disagree here. A PHB usually describes an
"externally observable forwarding treatment", basically how
a node treats a packet during the forwarding procedure
(typically boils down to queue management and scheduling issues).
In your case, that both PHBs (MF-H,MF-M) get their own assigned 
resources (bw limitation) and that MF-M has a higher drop precedence
than MF-H. On top of that PHB group you define an admission
control procedure with additional traffic conditioning mechanisms
(your token bucket descriptions). This fits better to a PDB 
description in my opinion. And as you state in section 5:

   The Priority Promotion Scheme can be viewed as a kind of admission
   control.

> The proposed new PHB, MF-PHB refers to two-subclasses,
> but we do not have to define it as a group description.

What you call "sub-classes" are effectively two different PHBs: you will
need two distinct DSCPs and each DSCP will be mapped to a distinct PHB
that will cause a specific packet forwarding treatment. RFC 2475 defines
a PHB group as "a set of one or more PHBs that can only be meaningfully
specified and implemented simultaneously,....". Please, see also RFC 3260
for clarification on the AF class as a PHB group. MF-H makes not much
sense without MF-M and vice versa. However, I think your PPS admission control 
scheme can be applied to an AF class as well as to the CSC PHBs, therefore you 
don't need two new PHBs.

>  >- The scheme inherently relies on feedback information and it should be made
>  >  clear how the feedback can be provided (you mentioned RTCP as an example,
>  >  what about others?).
> 
> Yes, it is.  As the PPS, I would like to describe the feedback mechanims,
> but for a PHB, we do not have to.

Yes, for a PHB you don't have to, but that's exactly why admission control 
should be defined not together with the PHB. EF also requires some form of
admission control to achieve the timely forwarding, but admission control 
procedures are not defined within RFC 3246. Your admission control procedure
will not work without feedback, but the PHBs will. Therefore, define a PDB
that defines constraints for corresponding PHBs.

>  >- Instead of defining a new PHB, possibly two Class Selector PHBs
>  >  or even an AF class can be used ("green" packets get high priority after
>  >  admission control, "yellow" and "red" packet may be used for probing).
>  >  AF usually gives no guarantees for delay and jitter, however, if AF is
>  >  provisioned properly, "green" packets may get some guarantee.
> 
> This kind of disucssion is what I wanted to do.
> Sorry for spending much time on me presentation only.
> Are you refering to Three Color Marker in RFC2697 and RFC2698?

Not exactly, because I was not referring to packet marking, 
but as in those RFCs, I used the color code as an equivalent
for the drop precedence (i.e., "green"=AFx1, "yellow"=AFx2, "red"=AFx3).
Note that, the AF class definition allows also to implement only two (instead of 
three) effective different levels of loss probability for an AF class x.

>  >- For security and robustness, it is essential to police the markings
>  >  for the higher priority. Otherwise the service guarantees for other users
>  >  are violated by a malicious or ill-behaving end-system. Therefore, I think
>  >  first-hop routers should do the policing (based on the feedback
>  >  information?).
> 
> I agree.  Are there any opinion about the description in pps framework document?
> Not sufficient?

Yes, I think some issues in 4.4 need more discussion. E.g., how do you ensure
that the feedback information is correct (you only state that "The behavior
in the direction from the destination to the source should also be correct")?
Some aspects should possibly be moved to the security section.

Best regards,
 Roland

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Date: Mon, 10 Nov 2003 23:18:37 -0600
To: Roland Bless <bless@tm.uka.de>
From: "MORITA, Naotaka" <morita.naotaka@lab.ntt.co.jp>
Subject: Re: [Tsvwg] Re: [NSIS] PPS and MF-PHB
Cc: diffserv-interest@ietf.org, "'tsvwg@ietf.org'" <tsvwg@ietf.org>
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Roland,

At 05:28 03/11/11 +0100, Roland Bless wrote:
 >Hi Naotaka,
 >
 >>  >After reading draft-morita-tsvwg-mfphb-00.txt and attending your
 >>  >presentation, I have some remarks related to this approach:
 >>  >- I'm not quite sure whether this is really a PHB definition.
 >>  >  It seems to me, that it is actually a mix of
 >>  >  * a scheme for measured-based admission control and resource provisioning
 >                     ^^^^^^^^sorry, should have been: measurement-based
 >admission control
 >>  >  * a Per-Domain Behavior (see RFC 3086)
 >>  >  * a PHB group description
 >
 >> Your understanding is correct, but I do not think it is a mix.
 >> The key of the PPS is to utilize PHB for admission control.
 >
 >Hmm, I disagree here. A PHB usually describes an
 >"externally observable forwarding treatment", basically how
 >a node treats a packet during the forwarding procedure
 >(typically boils down to queue management and scheduling issues).
 >In your case, that both PHBs (MF-H,MF-M) get their own assigned
 >resources (bw limitation) and that MF-M has a higher drop precedence
 >than MF-H. On top of that PHB group you define an admission
 >control procedure with additional traffic conditioning mechanisms
 >(your token bucket descriptions). This fits better to a PDB
 >description in my opinion. And as you state in section 5:
 >
 >   The Priority Promotion Scheme can be viewed as a kind of admission
 >   control.

That is right.  I will check the meaning of PDP.

 >
 >> The proposed new PHB, MF-PHB refers to two-subclasses,
 >> but we do not have to define it as a group description.
 >
 >What you call "sub-classes" are effectively two different PHBs: you will
 >need two distinct DSCPs and each DSCP will be mapped to a distinct PHB
 >that will cause a specific packet forwarding treatment. RFC 2475 defines
 >a PHB group as "a set of one or more PHBs that can only be meaningfully
 >specified and implemented simultaneously,....". Please, see also RFC 3260
 >for clarification on the AF class as a PHB group. MF-H makes not much
 >sense without MF-M and vice versa. However, I think your PPS admission control
 >scheme can be applied to an AF class as well as to the CSC PHBs, therefore you
 >don't need two new PHBs.

But the important factor is the bandwidth limitation is shared between two 
sub-classes.
Sequence integrity is another one.
How do I describe such things by isolated PHBs?
Anyway, what is CSC?

 >
 >>  >- The scheme inherently relies on feedback information and it should be made
 >>  >  clear how the feedback can be provided (you mentioned RTCP as an example,
 >>  >  what about others?).
 >>
 >> Yes, it is.  As the PPS, I would like to describe the feedback mechanims,
 >> but for a PHB, we do not have to.
 >
 >Yes, for a PHB you don't have to, but that's exactly why admission control
 >should be defined not together with the PHB. EF also requires some form of
 >admission control to achieve the timely forwarding, but admission control
 >procedures are not defined within RFC 3246. Your admission control procedure
 >will not work without feedback, but the PHBs will. Therefore, define a PDB
 >that defines constraints for corresponding PHBs.

I see.  Again, I will check PDB.

 >
 >>  >- Instead of defining a new PHB, possibly two Class Selector PHBs
 >>  >  or even an AF class can be used ("green" packets get high priority after
 >>  >  admission control, "yellow" and "red" packet may be used for probing).
 >>  >  AF usually gives no guarantees for delay and jitter, however, if AF is
 >>  >  provisioned properly, "green" packets may get some guarantee.
 >>
 >> This kind of disucssion is what I wanted to do.
 >> Sorry for spending much time on me presentation only.
 >> Are you refering to Three Color Marker in RFC2697 and RFC2698?
 >
 >Not exactly, because I was not referring to packet marking,
 >but as in those RFCs, I used the color code as an equivalent
 >for the drop precedence (i.e., "green"=AFx1, "yellow"=AFx2, "red"=AFx3).
 >Note that, the AF class definition allows also to implement only two 
(instead of
 >three) effective different levels of loss probability for an AF class x.

I will soon show you the difference between srTCM and my proposed two-threshold.

 >
 >>  >- For security and robustness, it is essential to police the markings
 >>  >  for the higher priority. Otherwise the service guarantees for other users
 >>  >  are violated by a malicious or ill-behaving end-system. Therefore, I think
 >>  >  first-hop routers should do the policing (based on the feedback
 >>  >  information?).
 >>
 >> I agree.  Are there any opinion about the description in pps framework
 >document?
 >> Not sufficient?
 >
 >Yes, I think some issues in 4.4 need more discussion. E.g., how do you ensure
 >that the feedback information is correct (you only state that "The behavior
 >in the direction from the destination to the source should also be correct")?
 >Some aspects should possibly be moved to the security section.

Thank you.  I will do that.
For the correctness of the feedback information is a target of media 
monitor server.

 >
 >Best regards,
 > Roland
 >
 >_______________________________________________
 >tsvwg mailing list
 >tsvwg@ietf.org
 >https://www1.ietf.org/mailman/listinfo/tsvwg 



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Date: Tue, 11 Nov 2003 19:15:12 +0100
From: Roland Bless <bless@tm.uka.de>
To: "MORITA, Naotaka" <morita.naotaka@lab.ntt.co.jp>,
        diffserv-interest@ietf.org, "'tsvwg@ietf.org'" <tsvwg@ietf.org>
Subject: Re: [Diffserv-interest] Re: [Tsvwg] Re: [NSIS] PPS and MF-PHB
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On Mon, 10 Nov 2003 23:18:37 -0600
"MORITA, Naotaka" <morita.naotaka@lab.ntt.co.jp> wrote:
 
> That is right.  I will check the meaning of PDP.

PDB=Per-Domain Behavior, see RFC 3086.

>  >What you call "sub-classes" are effectively two different PHBs: you will
>  >need two distinct DSCPs and each DSCP will be mapped to a distinct PHB
>  >that will cause a specific packet forwarding treatment. RFC 2475 defines
>  >a PHB group as "a set of one or more PHBs that can only be meaningfully
>  >specified and implemented simultaneously,....". Please, see also RFC 3260
>  >for clarification on the AF class as a PHB group. MF-H makes not much
>  >sense without MF-M and vice versa. However, I think your PPS admission control
>  >scheme can be applied to an AF class as well as to the CSC PHBs, therefore you
>  >don't need two new PHBs.
> 
> But the important factor is the bandwidth limitation is shared between two 
> sub-classes.

Sure, no problem. That's why it is a PHB group.

> Sequence integrity is another one.

You mean that micro-flows should not be reordered if their
packets are marked as MF-H and MF-M alternatingly?
This is also a requirement for the PHBs in an AF class.
However, I don't see how you achieve this with the implementation 
described in 4.1 Fig.3: as soon as you have separate queues
for MF-H and MF-M, reordering may occur.

> How do I describe such things by isolated PHBs?

Confer the AF class definition and read RFC3260 in
addition to that.

> Anyway, what is CSC?

Class Selector Compliant PHB, cf. RFC 2474, sec. 4.2.2.2

> I will soon show you the difference between srTCM and my proposed two-threshold.

No, you don't have to, I support that there are differences. I'm not referring to 
the marking scheme, I'm referring to the AF PHB definition.

> For the correctness of the feedback information is a target of media 
> monitor server.

I'm not sure whether it is reasonable that the scheme depends on
the existence of some additional server.

Regards,
 Roland

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	Hi all!

	I have just read the Packet Drop Reporting extension for SCTP =
(http://www.ietf.org/internet-drafts/draft-stewart-sctp-pktdrprep-00.txt)=
 and I have some quite many comments (as usual ;->):

	- Page 3, last paragraph, a typo: "middle box peer can be *can be* =
located..."

	- Page 5, new chunk diagram: This is maybe a matter of style and taste =
but I liked the way flags are described in other SCTP documents (I mean =
in the drawing). What about something like
        0                   1                   2                   3
        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       | Type =3D 0x81   |Reserved |T|B|M|      Chunk Length             =
|
       =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+?

	- Page 5: I think that the field "Truncated Length" is a little bit =
confusing, since it is the length of the datagram before truncating it. =
What about "Original SCTP Datagram Length"?

	- Page 6, beginning, T flag definition: It says that you get the =
original packet length "from the IP header". That's really confusing, as =
later it is clear that only the SCTP datagram is included (without the =
header). I think you should change that and make clear that it is only =
the SCTP datagram what's copied in the PKTDROP chunk. Also, up to this =
point it is not completely clear whether that length also includes the =
SCTP common header...

	- Page 7, Data Field definition: I think you could make clearer here =
that when you speak about the packet you mean just the SCTP packet NOT =
including the header...

	- Section 5.1.1: There are some chunks that a middle box shouldn't send =
back to the sender. These are the INIT ACK, SHUTDOWN COMPLETE and any =
packet containing an ABORT. It shouldn't because that doesn't make =
sense: the sender of such a packet does not have any TCB about that =
association any more. So, the PKTDROP would be an OOTB datagram and that =
would only originate that an ABORT is sent to the other peer.

	In fact, in the case the original packet contained an ABORT, the above =
might make sense.

	If it was a SHUTDOWN COMPLETE, it will basically speed up the fatal =
destiny of the association (the peer would otherwise retransmit the =
SHUTDOWN ACK and get a SHUTDOWN COMPLETE similar to the ABORT it will =
get if the middle box sends the PKTDROP). Receiving an ABORT with the =
*right* VT and still with the T flag set, instead of a SHUTDOWN COMPLETE =
with the T flag set and the reverse VT might be a little bit confusing =
though...

	But if the INIT ACK was lost, that would cause that the establishment =
phase is aborted, and that's not good.

	- Section 5.1.2: The above comment also works for this section. And I =
was also thinking if when receiving a corrupted HEARTBEAT ACK it would =
make much more sense simple sending back a new HEARTBEAT than the =
PKTDROP. Same happens with the SHUTDOWN ACK: better to resend the =
SHUTDOWN instead of the PKTDROP since that would speed up the processing =
of the datagram at the peer side.

	I'm not sure but maybe the same happens with corrupted COOKIE ACK and =
ASCONF ACK chunks... :-/

	- Section 5.2: There are many chunk names written in lower case, such =
as "heartbeat", "shutdown acknowledgement", "cookie-ack", "Packet =
Drop"... and more. I think they should be written with capital letters =
(and the "official" name).

	- Section 5.2: What happens if one receives a PKTDROP containing an =
ASCONF ACK? I think that after carefully checking the serial number, the =
receiver of the PKTDROP could resend the ASCONF ACK (at least in the =
quite probable case that it stores a copy of the last ASCONF ACK sent).

	- Section 5.2, page 9: The calculation of "rtt" is very confusing. What =
are those values there? Besides, there is a bracket missing...

	- Section 5.2., page 10: In the formula for bw_avail, you could mention =
that dividing the result by 1000 is because the rtt is in milliseconds. =
I might be too clumsy but I was wondering whether that value was because =
you wanted to give a share of one thousandth to each peer... :-/

	- Section 6, Security considerations: I suppose that you have already =
though about this, but, as said at the beginning of page 6, middle boxes =
are allowed to send a PKTDROP just to state the bandwidth value. What if =
the peer wants to speed up the transmission and sends a fake PKTDROP as =
being sent by any middle box in between? This can be perfectly done... =
I'm not sure whether you should allow any increase in the cwnd at all... =
:-/



	Ok, that's all (by now! ;->)

	BR Iv=E1n Arias Rodr=EDguez

	PS: I suppose this is the right list to send this mail..

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As mentioned in the meeting on Monday the discussion of IPMP has now
moved to the IMRG list at

https://www1.ietf.org/mail-archive/working-groups/imrg/current/msg00154.html

We invite anyone interested in the topic to participate in the
discussion.

Jon Bennett

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The IESG has received a request from the Transport Area Working Group to 
consider the following document:

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The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by 2003-12-02.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-prsctp-01.txt


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Hi, Tom,

It seems to me that you are basically: 1) questioning the fairness of TCP itself
and, 2) you want to do something in the app level to outmaneuver TCP so as to
achieve your term of fairness.

I am not convinced this is the right thing to do with DCCP.

I remember similar issue was discussed on tsvwg list (I am cc'ing it). Maybe
people in tsvwg can share their thoughts on this as well.

regards,
-Qiaobing

"Phelan, Tom" wrote:

> Hi Phillipe,
>
> The situation I'm worried about is a little different than you describe.
> Say you have a 10M bps link, over which there's a 4M bps video stream and a
> long-lived TCP app.  The TCP app will take the remaining 6M bps, and then
> push for more, which might cause the video stream to lose packets.  The
> video stream then drops to 2M bps, and the TCP app expands to 8M bps.  This
> hardly seems fair.
>
> The video app can protect itself from this by transmitting enough more than
> 4M bps to buffer itself against the TCP probes.  In this case we'd come to a
> relatively steady state where both apps are sending roughly 5M bps.  This
> seems reasonably fair to me, since the TCP app gets what it would get if it
> were competing with other TCP apps, and the video app gets to send at its
> full rate.
>
> Tom P.
>
> > -----Original Message-----
> > From: philippe.gentric@philips.com
> > [mailto:philippe.gentric@philips.com]
> > Sent: Thursday, November 20, 2003 12:23 PM
> > To: Phelan, Tom; 'Rosen, Brian'
> > Cc: 'avt@ietf.org'; 'dccp@ietf.org'; 'mmusic@ietf.org'
> > Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> >
> >
> > Brian, all,
> >
> > It may provide some help in understanding the situation
> > to describe your exemple as follows:
> >
> > * you send video at 4 Mb/s, good.
> >
> > * it happens that you are competing with (a lot of ) TCP
> >
> > AND that you want to be fair to it:
> >
> > * so when that happens you decrease your rate to 2 Mb/s for "a while"
> > because otherwise you are not fair.
> >
> > * but at some moment you will want to come back up to 4 Mb/s right ?
> >
> > => the point is that the **only way**  to know if you "can"
> > do that, is to try it....
> >
> > regards,
> >
> >
> > Philippe Gentric
> > Chief Architect
> > Philips Software
> > philippe.gentric@philips.com
> > http://www.software.philips.com
> >
> >
> >
> >
> >
> >
> >
> >
> >                                                    To:
> > "'Rosen, Brian'" <Brian.Rosen@marconi.com>
> >
> >                                                    cc:
> > "'dccp@ietf.org'" <dccp@ietf.org>
> >
> >
> > "'avt@ietf.org'" <avt@ietf.org>
> >
> >
> > "'mmusic@ietf.org'" <mmusic@ietf.org>
> >
> >                "Phelan, Tom"                        (bcc:
> > Philippe Gentric/MP4-SUR/CE/PHILIPS)
> >
> >                <tphelan@sonusnet.com>              Subject:
> >  RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> >
> >
> >
> >
> >                Sent by:
> > Classification:
> >
> >                mmusic-admin@ietf.org
> >
> >
> >
> >
> >
> >                20/11/2003 17:47
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > Hi Brian,
> >
> > Thanks for the comments.  See inline ...
> >
> > Tom P.
> >
> > > -----Original Message-----
> > > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > > Sent: Thursday, November 20, 2003 11:05 AM
> > > To: 'Phelan, Tom'; 'Jonathan Rosenberg'
> > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> > >
> > >
> > > To me, this is just nonsense.
> > > I can't imagine anyone deploying such a solution.
> > >
> > > This is forcing the problem to fit the solution, and it's not
> > > the right way to go.  If you want it to work, the protocol
> > > has to take the characteristics of the flows as they are, and not
> > > as you would like them to be.  Real time media, streaming
> > or realtime,
> > > is either constant bit rate, or pretty close to it.  That's how it
> > > works.  Our challenge is to make the protocols work with this data,
> > > not change the data, especially silly ideas like doubling the
> > > bandwidth.
> > >
> > > We make video telephones.  They nominally send 4 megabits.  Would
> > > you like us to double that, under any circumstances?
> > >
> > > I thought not.
> >
> > Would you prefer to be forced unnecessarily into packet loss
> > and a lower
> > encoding rate by greedy TCP applications?  Or do you have
> > another solution?
> >
> > I don't think I'm forcing the problem to fit the solution,
> > just perhaps that
> > the problem I'm trying to solve is different than you
> > perceive.  The problem
> > that I'm trying to solve is how can an application that is
> > content with a
> > maximum bandwidth compete with TCP applications that are
> > insatiably greedy,
> > in a way that satisfies some concept of fairness in all directions?
> >
> > The solution that I see is to provide a buffer against TCP
> > probing.  This
> > would involve sending more data than is nominally necessary.
> > If the network
> > has capacity for more than 4M bps, then why shouldn't a media
> > app make use
> > of it?  A TCP app would.
> >
> > I must admit that my initial reactions to suggestions like
> > this were very
> > similar to yours.  I encourage you to look at the draft for
> > more detail on
> > the issues that convinced me otherwise.
> >
> > >
> > > Brian
> > >
> > > > -----Original Message-----
> > > > From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> > > > Sent: Thursday, November 20, 2003 10:36 AM
> > > > To: 'Jonathan Rosenberg'
> > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> > > >
> > > >
> > > > Hi Jonathan,
> > > >
> > > > Thanks for the comments.  The actual proposal is to send up
> > > > to double the
> > > > nominal transmit rate when there's available bandwidth, and
> > > > to reduce the
> > > > rate as necessary if there isn't enough bandwidth.  This
> > > > doubling is to
> > > > allow fair sharing of the chokepoint bandwidth (between DCCP
> > > > media apps and
> > > > TCP apps), and to avoid what appears to me to be the
> > > > potential for bullying
> > > > from TCP applications that want to suck up all the bandwidth
> > > > they can.  It
> > > > doesn't preclude using low-rate codecs, or mean that the
> > > > access link must
> > > > support the double rate for the connection to function well.
> > > >
> > > > This recommendation was meant for best-effort environments,
> > > > where the voice
> > > > traffic receives no special treatment, and must compete with
> > > > TCP apps for
> > > > network resources.  I can see how this would require a
> > > > network that gave
> > > > special treatment to voice packets to provision double the
> > > > capacity, though.
> > > >
> > > > This fairness problem is difficult.  All of the research I'm
> > > > familiar with
> > > > approaches it from the direction of other apps being fair to
> > > > TCP.  I'm not
> > > > aware of any other methods for ensuring that non-TCP apps get
> > > > their fair
> > > > share when competing with TCP.  If you (or anyone else) knows
> > > > of something
> > > > I'd appreciate the pointer.
> > > >
> > > > Tom P.
> > > >
> > > > > -----Original Message-----
> > > > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > > > Sent: Thursday, November 20, 2003 2:32 AM
> > > > > To: Phelan, Tom
> > > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > > Subject: Re: [MMUSIC] DCCP Media Guide
> > > > >
> > > > >
> > > > > Tom,
> > > > >
> > > > > I don't think this solution will be acceptable to many of
> > > > the network
> > > > > providers of interactive voip services. Many of them are
> > > > insistent on
> > > > > using low rate codecs, like g.723, in order to reduce the
> > > costs of
> > > > > capacity in their networks. Doubling the rate would defeat
> > > > > the purpose
> > > > > of using these low rate codecs in the first place.
> > > > >
> > > > > -Jonathan R.
> > > > >
> > > > > Phelan, Tom wrote:
> > > > >
> > > > >
> > > > > > Comments will be appreciated.  In particular, I make the
> > > > > suggestion that
> > > > > > media apps should transmit at up to twice their nominal
> > > > > maximum rate, when
> > > > > > DCCP allows it, in order to provide protection from
> > > competing TCP
> > > > > > applications.  I know that the first time I heard someone
> > > > > make a similar
> > > > > > suggestion I was horrified, but as I developed the issues
> > > > > it seemed to be
> > > > > > necessary.  Reactions to this will be appreciated.
> > > > > >
> > > > > > Tom P.
> > > > > >
> > > > > > _______________________________________________
> > > > > > mmusic mailing list
> > > > > > mmusic@ietf.org
> > > > > > https://www1.ietf.org/mailman/listinfo/mmusic
> > > > > >
> > > > >
> > > > > --
> > > > > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > > > > Chief Technology Officer                    Parsippany, NJ
> > > > 07054-2711
> > > > > dynamicsoft
> > > > > jdrosen@dynamicsoft.com                     FAX:
> > (973) 952-5050
> > > > > http://www.jdrosen.net                      PHONE:
> > (973) 952-5000
> > > > > http://www.dynamicsoft.com
> > > > >
> > > >
> > > > _______________________________________________
> > > > Audio/Video Transport Working Group
> > > > avt@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/avt
> > > >
> > >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> >
> >
> >
> >
> >
>
> _______________________________________________
> dccp IETF mailing list: dccp@ietf.org
> list info:  https://www1.ietf.org/mailman/listinfo/dccp
> wg charter: http://www.ietf.org/html.charters/dccp-charter.html


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        TSV WG list <tsvwg@ietf.org>, "Phelan, Tom" <tphelan@sonusnet.com>,
        avt@ietf.org, dccp@ietf.org, philippe.gentric@philips.com
From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] Re: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Thu, 20 Nov 2003 16:01:47 -0600
To: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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On Nov 20, 2003, at 1:26 PM, Qiaobing Xie wrote:

> Hi, Tom,
>
> It seems to me that you are basically: 1) questioning the fairness of 
> TCP itself
> and, 2) you want to do something in the app level to outmaneuver TCP 
> so as to
> achieve your term of fairness.
>
> I am not convinced this is the right thing to do with DCCP.
>
> I remember similar issue was discussed on tsvwg list (I am cc'ing it). 
> Maybe
> people in tsvwg can share their thoughts on this as well.


The fairness of TCP itself could use some questioning -- but I do not
believe that is a reason to evade TCP friendly rate control.

Eventually, there should be a solution for proper support of predictably
variable rate flows that have fixed delivery requirements. But that 
solution
should be developed considering TCP, SCTP and DCCP as a whole.


_______________________________________________
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https://www1.ietf.org/mailman/listinfo/tsvwg



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Caitlin Bestler'" <cait@asomi.com>,
        Qiaobing Xie
	 <Qiaobing.Xie@motorola.com>
Cc: mmusic@ietf.org, TSV WG list <tsvwg@ietf.org>,
        "Phelan, Tom"
	 <tphelan@sonusnet.com>, avt@ietf.org, dccp@ietf.org,
        philippe.gentric@philips.com
Subject: RE: [Tsvwg] Re: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Thu, 20 Nov 2003 17:14:23 -0500
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The reason to "avoid" TCP friendly rate control is, at least so far,
because there is no way yet proposed to do it without destroying the 
usability of the RTP flow.

Part of the problem is that the only "rate control" that works is to
negotiate an acceptable rate in the signalling before the stream starts.
There really isn't any other way to change the rate, at least for
interactive streams.

For non-interactive streams, you can always buffer more, which allows
you some short term variation.  I don't think that any of the proposed
mechanisms would actually work even in that case, but maybe there is
a way to do something.  Non interactive streams need CBR averaged over
the buffer time.  That can be up to seconds in some cases, hundreds
of milliseconds in other cases, but over that interval, your goodput
must equate to the transmit offer rate times the interval (some small
loss rate acceptable).

It would be nice if there was a way.  I haven't seen one yet.

Brian

> -----Original Message-----
> From: Caitlin Bestler [mailto:cait@asomi.com]
> Sent: Thursday, November 20, 2003 5:02 PM
> To: Qiaobing Xie
> Cc: mmusic@ietf.org; 'Rosen, Brian'; TSV WG list; Phelan, Tom;
> avt@ietf.org; dccp@ietf.org; philippe.gentric@philips.com
> Subject: Re: [Tsvwg] Re: [dccp] RE: [AVT] RE: [MMUSIC] DCCP 
> Media Guide
> 
> 
> 
> On Nov 20, 2003, at 1:26 PM, Qiaobing Xie wrote:
> 
> > Hi, Tom,
> >
> > It seems to me that you are basically: 1) questioning the 
> fairness of 
> > TCP itself
> > and, 2) you want to do something in the app level to 
> outmaneuver TCP 
> > so as to
> > achieve your term of fairness.
> >
> > I am not convinced this is the right thing to do with DCCP.
> >
> > I remember similar issue was discussed on tsvwg list (I am 
> cc'ing it). 
> > Maybe
> > people in tsvwg can share their thoughts on this as well.
> 
> 
> The fairness of TCP itself could use some questioning -- but I do not
> believe that is a reason to evade TCP friendly rate control.
> 
> Eventually, there should be a solution for proper support of 
> predictably
> variable rate flows that have fixed delivery requirements. But that 
> solution
> should be developed considering TCP, SCTP and DCCP as a whole.
> 
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 

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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'Qiaobing Xie'" <Qiaobing.Xie@motorola.com>
Cc: TSV WG list <tsvwg@ietf.org>, "'dccp@ietf.org'" <dccp@ietf.org>
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Subject: [Tsvwg] RE: [dccp] DCCP Media Guide
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Hi Qiaobing [trimmed list a bit],

> It seems to me that you are basically: 1) questioning the 
> fairness of TCP itself
> and, 2) you want to do something in the app level to 
> outmaneuver TCP so as to
> achieve your term of fairness.

The current practice for streaming media is unfair to TCP (well, at least
according to the IAB, an opinion that I buy into).  If we modify the
behavior of streaming media so that it is fair to TCP, we must also be
certain that TCP is fair to this new behavior.  This bi-directional fairness
doesn't seem to have gotten much attention yet.

> I am not convinced this is the right thing to do with DCCP.

I'd love to hear opinions on this.  But if DCCP doesn't give acceptable
service for streaming media, then it isn't going to be used for streaming
media.

Tom Phelan

> 
> I remember similar issue was discussed on tsvwg list (I am 
> cc'ing it). Maybe
> people in tsvwg can share their thoughts on this as well.
> 
> regards,
> -Qiaobing
> 
> "Phelan, Tom" wrote:
> 
> > Hi Phillipe,
> >
> > The situation I'm worried about is a little different than 
> you describe.
> > Say you have a 10M bps link, over which there's a 4M bps 
> video stream and a
> > long-lived TCP app.  The TCP app will take the remaining 6M 
> bps, and then
> > push for more, which might cause the video stream to lose 
> packets.  The
> > video stream then drops to 2M bps, and the TCP app expands 
> to 8M bps.  This
> > hardly seems fair.
> >
> > The video app can protect itself from this by transmitting 
> enough more than
> > 4M bps to buffer itself against the TCP probes.  In this 
> case we'd come to a
> > relatively steady state where both apps are sending roughly 
> 5M bps.  This
> > seems reasonably fair to me, since the TCP app gets what it 
> would get if it
> > were competing with other TCP apps, and the video app gets 
> to send at its
> > full rate.
> >
> > Tom P.
> >
> > > -----Original Message-----
> > > From: philippe.gentric@philips.com
> > > [mailto:philippe.gentric@philips.com]
> > > Sent: Thursday, November 20, 2003 12:23 PM
> > > To: Phelan, Tom; 'Rosen, Brian'
> > > Cc: 'avt@ietf.org'; 'dccp@ietf.org'; 'mmusic@ietf.org'
> > > Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> > >
> > >
> > > Brian, all,
> > >
> > > It may provide some help in understanding the situation
> > > to describe your exemple as follows:
> > >
> > > * you send video at 4 Mb/s, good.
> > >
> > > * it happens that you are competing with (a lot of ) TCP
> > >
> > > AND that you want to be fair to it:
> > >
> > > * so when that happens you decrease your rate to 2 Mb/s 
> for "a while"
> > > because otherwise you are not fair.
> > >
> > > * but at some moment you will want to come back up to 4 
> Mb/s right ?
> > >
> > > => the point is that the **only way**  to know if you "can"
> > > do that, is to try it....
> > >
> > > regards,
> > >
> > >
> > > Philippe Gentric
> > > Chief Architect
> > > Philips Software
> > > philippe.gentric@philips.com
> > > http://www.software.philips.com
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >                                                    To:
> > > "'Rosen, Brian'" <Brian.Rosen@marconi.com>
> > >
> > >                                                    cc:
> > > "'dccp@ietf.org'" <dccp@ietf.org>
> > >
> > >
> > > "'avt@ietf.org'" <avt@ietf.org>
> > >
> > >
> > > "'mmusic@ietf.org'" <mmusic@ietf.org>
> > >
> > >                "Phelan, Tom"                        (bcc:
> > > Philippe Gentric/MP4-SUR/CE/PHILIPS)
> > >
> > >                <tphelan@sonusnet.com>              Subject:
> > >  RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> > >
> > >
> > >
> > >
> > >                Sent by:
> > > Classification:
> > >
> > >                mmusic-admin@ietf.org
> > >
> > >
> > >
> > >
> > >
> > >                20/11/2003 17:47
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > Hi Brian,
> > >
> > > Thanks for the comments.  See inline ...
> > >
> > > Tom P.
> > >
> > > > -----Original Message-----
> > > > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > > > Sent: Thursday, November 20, 2003 11:05 AM
> > > > To: 'Phelan, Tom'; 'Jonathan Rosenberg'
> > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> > > >
> > > >
> > > > To me, this is just nonsense.
> > > > I can't imagine anyone deploying such a solution.
> > > >
> > > > This is forcing the problem to fit the solution, and it's not
> > > > the right way to go.  If you want it to work, the protocol
> > > > has to take the characteristics of the flows as they 
> are, and not
> > > > as you would like them to be.  Real time media, streaming
> > > or realtime,
> > > > is either constant bit rate, or pretty close to it.  
> That's how it
> > > > works.  Our challenge is to make the protocols work 
> with this data,
> > > > not change the data, especially silly ideas like doubling the
> > > > bandwidth.
> > > >
> > > > We make video telephones.  They nominally send 4 
> megabits.  Would
> > > > you like us to double that, under any circumstances?
> > > >
> > > > I thought not.
> > >
> > > Would you prefer to be forced unnecessarily into packet loss
> > > and a lower
> > > encoding rate by greedy TCP applications?  Or do you have
> > > another solution?
> > >
> > > I don't think I'm forcing the problem to fit the solution,
> > > just perhaps that
> > > the problem I'm trying to solve is different than you
> > > perceive.  The problem
> > > that I'm trying to solve is how can an application that is
> > > content with a
> > > maximum bandwidth compete with TCP applications that are
> > > insatiably greedy,
> > > in a way that satisfies some concept of fairness in all 
> directions?
> > >
> > > The solution that I see is to provide a buffer against TCP
> > > probing.  This
> > > would involve sending more data than is nominally necessary.
> > > If the network
> > > has capacity for more than 4M bps, then why shouldn't a media
> > > app make use
> > > of it?  A TCP app would.
> > >
> > > I must admit that my initial reactions to suggestions like
> > > this were very
> > > similar to yours.  I encourage you to look at the draft for
> > > more detail on
> > > the issues that convinced me otherwise.
> > >
> > > >
> > > > Brian
> > > >
> > > > > -----Original Message-----
> > > > > From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> > > > > Sent: Thursday, November 20, 2003 10:36 AM
> > > > > To: 'Jonathan Rosenberg'
> > > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > > Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> > > > >
> > > > >
> > > > > Hi Jonathan,
> > > > >
> > > > > Thanks for the comments.  The actual proposal is to send up
> > > > > to double the
> > > > > nominal transmit rate when there's available bandwidth, and
> > > > > to reduce the
> > > > > rate as necessary if there isn't enough bandwidth.  This
> > > > > doubling is to
> > > > > allow fair sharing of the chokepoint bandwidth (between DCCP
> > > > > media apps and
> > > > > TCP apps), and to avoid what appears to me to be the
> > > > > potential for bullying
> > > > > from TCP applications that want to suck up all the bandwidth
> > > > > they can.  It
> > > > > doesn't preclude using low-rate codecs, or mean that the
> > > > > access link must
> > > > > support the double rate for the connection to function well.
> > > > >
> > > > > This recommendation was meant for best-effort environments,
> > > > > where the voice
> > > > > traffic receives no special treatment, and must compete with
> > > > > TCP apps for
> > > > > network resources.  I can see how this would require a
> > > > > network that gave
> > > > > special treatment to voice packets to provision double the
> > > > > capacity, though.
> > > > >
> > > > > This fairness problem is difficult.  All of the research I'm
> > > > > familiar with
> > > > > approaches it from the direction of other apps being fair to
> > > > > TCP.  I'm not
> > > > > aware of any other methods for ensuring that non-TCP apps get
> > > > > their fair
> > > > > share when competing with TCP.  If you (or anyone else) knows
> > > > > of something
> > > > > I'd appreciate the pointer.
> > > > >
> > > > > Tom P.
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > > > > Sent: Thursday, November 20, 2003 2:32 AM
> > > > > > To: Phelan, Tom
> > > > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > > > Subject: Re: [MMUSIC] DCCP Media Guide
> > > > > >
> > > > > >
> > > > > > Tom,
> > > > > >
> > > > > > I don't think this solution will be acceptable to many of
> > > > > the network
> > > > > > providers of interactive voip services. Many of them are
> > > > > insistent on
> > > > > > using low rate codecs, like g.723, in order to reduce the
> > > > costs of
> > > > > > capacity in their networks. Doubling the rate would defeat
> > > > > > the purpose
> > > > > > of using these low rate codecs in the first place.
> > > > > >
> > > > > > -Jonathan R.
> > > > > >
> > > > > > Phelan, Tom wrote:
> > > > > >
> > > > > >
> > > > > > > Comments will be appreciated.  In particular, I make the
> > > > > > suggestion that
> > > > > > > media apps should transmit at up to twice their nominal
> > > > > > maximum rate, when
> > > > > > > DCCP allows it, in order to provide protection from
> > > > competing TCP
> > > > > > > applications.  I know that the first time I heard someone
> > > > > > make a similar
> > > > > > > suggestion I was horrified, but as I developed the issues
> > > > > > it seemed to be
> > > > > > > necessary.  Reactions to this will be appreciated.
> > > > > > >
> > > > > > > Tom P.
> > > > > > >
> > > > > > > _______________________________________________
> > > > > > > mmusic mailing list
> > > > > > > mmusic@ietf.org
> > > > > > > https://www1.ietf.org/mailman/listinfo/mmusic
> > > > > > >
> > > > > >
> > > > > > --
> > > > > > Jonathan D. Rosenberg, Ph.D.                600 
> Lanidex Plaza
> > > > > > Chief Technology Officer                    Parsippany, NJ
> > > > > 07054-2711
> > > > > > dynamicsoft
> > > > > > jdrosen@dynamicsoft.com                     FAX:
> > > (973) 952-5050
> > > > > > http://www.jdrosen.net                      PHONE:
> > > (973) 952-5000
> > > > > > http://www.dynamicsoft.com
> > > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Audio/Video Transport Working Group
> > > > > avt@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/avt
> > > > >
> > > >
> > >
> > > _______________________________________________
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> > >
> > >
> > >
> > >
> > >
> >
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> > wg charter: http://www.ietf.org/html.charters/dccp-charter.html
> 

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Subject: [Tsvwg] DCCP Media Guide - STOP crossposting
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Hi,

Please don't crosspost to all these lists. It creates problems for the 
list managers and will fragment the discussion.

As long as we are discussing issues related to media and DCCP please 
keep these topics on the AVT mailing list.

Other DCCP related issues goes on the DCCP list.

Best Regards

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


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	Hi all!

	I have read the PR-SCTP extension draft, and I have seen some typos =
(which I hate to comment, but since I saw them anyway...) and also I =
have couple of comments:

	- Section 1.2, page 4, second paragraph, a duplicated word: "... IETF =
so that *that* the designers..."

	- Section 3.1, "Length" definition: "Indicate the size...", shouldn't =
it be "Indicates the size..."?

	- Section 3.2, page 8, "Stream Sequence-N" definition: "... report =
TSN's co*or*esponding to..."

	- Section 3.3.1, title of the section (also in the index): This is kind =
of silly, but, shouldn't be that "param" be "parameter" instead?

	- Section 3.3.2, second paragraph: There is a mistaken reference to =
RFC2960 as [4] instead of [5]...

	- Section 3.3.3, page 9: "... COOKIE or as a** unrecognized...", I =
would say that "a" should be "an"... :-/

	- Section 3.4, page 12, rule C5: Hmm... I think that a very common =
situation will be that we have some TSNs unacknowledged, and then the =
T3-rtx timer expires (the only one that was running), and then you =
decide to abandon all the unacknowledged TSNs and send the FORWARD TSN. =
There is no T3-rtx running, so, C5 says that we "MUST assure that at =
least one T3-rtx timer is running", so, should be restart it? but, which =
one?

	Another thing that is not so clear for me is which address should we =
use for sending the FORWARD TSN... Should we kind of follow the same =
criterion than when retransmitting the TSNs themselves? However, the =
abandoned TSNs might have been sent to different addresses, so, which =
one should be use? Maybe I'm just mistaken but I would say that this is =
not clearly stated in the draft... :-/

	- Section 3.4, page 12, rule A5: As it is written I think it is a =
little bit confusing... I suppose that what you want to mean is that =
after a T3 expiration you check whether you have to abandon or not any =
TSN, and after that you try to advance the Advanced.Peer.Ack.Point, =
right? But that procedure of advancing that variable is stated only in =
C2... The others are mostly related with the arrival of a SACK... :-/


	Ok, that's all this time!

	Have a nice weekend!

	BR Iv=E1n Arias Rodr=EDguez

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From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] RE: [dccp] DCCP Media Guide
Date: Fri, 21 Nov 2003 10:57:51 -0600
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On Nov 21, 2003, at 8:01 AM, Phelan, Tom wrote:

> Hi Qiaobing [trimmed list a bit],
>
>> It seems to me that you are basically: 1) questioning the
>> fairness of TCP itself
>> and, 2) you want to do something in the app level to
>> outmaneuver TCP so as to
>> achieve your term of fairness.
>
> The current practice for streaming media is unfair to TCP (well, at 
> least
> according to the IAB, an opinion that I buy into).  If we modify the
> behavior of streaming media so that it is fair to TCP, we must also be
> certain that TCP is fair to this new behavior.  This bi-directional 
> fairness
> doesn't seem to have gotten much attention yet.
>
>> I am not convinced this is the right thing to do with DCCP.
>
> I'd love to hear opinions on this.  But if DCCP doesn't give acceptable
> service for streaming media, then it isn't going to be used for 
> streaming
> media.
>
> Tom Phelan
>
>

Two very good observations.

But they lead to some very crucial questions:

Is "TCP Friendly Rate Control" unfriendly to an entire class of 
applications?

	I believe that is the case. If so, a solution that is wider that an
	application protocol is called for. If not so, then that application
	should conform to TCP Friendly Rate Control rather than just declaring
	itself to be exempt.

Is it possible to build a two-pronged rate control solution?

	My understanding of the position of those that advocate TCP Friendly
	Rate Control alone is that they do not believe a two-pronged approach
	is plausible.

	I am not certain. What I am in agreement on is that it should
	not be 'n-pronged' where every application that thinks that
	TCP friendly rate control is unfair to it simply exempts
	itself and creates its own *separate* flow control.

	I am a firm believer that *all* applications share the same
	fundamental assumption about network congestion -- namely that
	it is a problem caused by *other* applications.

	If there is to be an alternate to TFRC, it must accommodate
	virtually *all* other connection-oriented traffic. Otherwise
	each little turf will "solve" the network congestion problem
	based upon the above cited universal assumption.


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Hi Caitlin,

See inline ...

Tom Phelan

[snipped]
> Two very good observations.
> 
> But they lead to some very crucial questions:
> 
> Is "TCP Friendly Rate Control" unfriendly to an entire class of 
> applications?

I don't think TFRC in and of itself is unfriendly - it depends on how you
use it.  If you employ strategy 3 from the draft, then you could be subject
to undesirable effects while competing with other TCP (or TFRC) apps.
Strategy 3 bis tries to correct these problems.

> 	I believe that is the case. If so, a solution that is 
> wider that an
> 	application protocol is called for. If not so, then 
> that application
> 	should conform to TCP Friendly Rate Control rather than 
> just declaring
> 	itself to be exempt.

Nothing in the draft declares itself exempt from TFRC - all strategies are
bound by TFRC.  At no time do you transmit faster than TFRC will allow.
Sometimes you might transmit slower.

> Is it possible to build a two-pronged rate control solution?
> 
> 	My understanding of the position of those that advocate 
> TCP Friendly
> 	Rate Control alone is that they do not believe a 
> two-pronged approach
> 	is plausible.
> 
> 	I am not certain. What I am in agreement on is that it should
> 	not be 'n-pronged' where every application that thinks that
> 	TCP friendly rate control is unfair to it simply exempts
> 	itself and creates its own *separate* flow control.

I don't know whether our current two choices for flow control mechanisms
(AIMD, as in TCP, or TFRC) are enough or not.  Maybe they are; maybe we need
one or two more.  Regardless, each application that uses one of these
mechanisms must decide how.  For some apps it will be simple - always
transmit as fast as the flow control allows.  For other apps (streaming
media being the current topic), it's more complicated.

My purpose in the DCCP Media Guide was to ask the question, "Can streaming
media apps use TFRC successfully?"  Personally, I don't think it answers
that question.  I hope it does illuminate some of the issues, though.

> 	I am a firm believer that *all* applications share the same
> 	fundamental assumption about network congestion -- namely that
> 	it is a problem caused by *other* applications.
> 
> 	If there is to be an alternate to TFRC, it must accommodate
> 	virtually *all* other connection-oriented traffic. Otherwise
> 	each little turf will "solve" the network congestion problem
> 	based upon the above cited universal assumption.

I don't see why an alternate to TFRC needs to accommodate all other
connection-oriented traffic.  If we come up with a third congestion control
mechanism that satisfies the needs of some set of applications while
maintaining "fairness" with TCP and TFRC, that seems OK.

That isn't to say that I think we need a third mechanism.

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The chairs would like to announce a two-week Working Group Last Call for
draft-ietf-tsvwg-tcp-eifel-response-04, commencing on November 26th and
concluding on
December 10th. Please send any final comments relating to this document to
the TSVWG mailing list.

http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-tcp-eifel-response-04.t
xt

Jon Peterson
NeuStar, Inc.

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	Hi all!

	Having read the latest version of the extension, I have the following =
comments:

	- Section 3.1.1, page 5, a typo: "in RFC2960 [5], *forall* variable =
parameters."

	- Section 3.2: Here we can read that there are some parameters that can =
be sent in the INIT/INIT ACK and some that should go in the ASCONF =
chunk. But, what it is not stated is what happens if the peer for =
example includes a Delete IP Address parameter. He looks to have a bug, =
so probably the best thing would be sending back an ABORT... :-/

	- Section 3.2: The Error Cause Indication and Success Indication =
parameters cannot be included in the INIT ACK. So, I assume there isn't =
any way the INIT receiver could negatively answer to a Set Primary =
Address... Probably it is not that important, since the receiver of such =
a parameter is not really forced to do anything, but it could be =
interesting to be able to do so...

	- Section 3.3.4, top of page 16: It says "... no more than 2^^31 =
greater than...", I think it should say "... no more than 2^^31 - 1 =
greater than...".

	- Section 4.1, A2): Here we see a SHOULD (about the initialization of =
the serial number) while in section 4.2, C1) we see a MUST (about the =
same value). I imagine that it should state MUST in both places...

	- Section 4.1, A9), a typo: "... it did not recognized the...", if my =
teachers taught me the right English grammar, I suppose it should be =
"... it did not recognize the..." :-)

	- Section 4.1.1, R4): Does it make any sense to be able to send ASCONF =
chunks while in the SHUTDOWN-RECEIVED state (and SHUTDOWN-PENDING state, =
especially if the ASCONF is a retransmission)? And same for ASCONF ACK =
chunks while in the SHUTDOWN-SENT state, and especially, while in the =
SHUTDOWN-PENDING...

	- Section 4.3.1: Maybe, checking whether the ABORT has the T flag set =
or not would help identifying those "false" ABORTs that may be =
received...

	- Section 6, another small typo: "... is a 32 bit *interger* that =
is..."


	Ok, that's all. Thanks!

	BR Iv=E1n Arias Rodr=EDguez

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Ivan.Arias-Rodriguez@nokia.com wrote:

>	Hi all!
>
>	I have read the PR-SCTP extension draft, and I have seen some typos (wh=
ich I hate to comment, but since I saw them anyway...) and also I have co=
uple of comments:
>
>	- Section 1.2, page 4, second paragraph, a duplicated word: "... IETF s=
o that *that* the designers..."
>
>	- Section 3.1, "Length" definition: "Indicate the size...", shouldn't i=
t be "Indicates the size..."?
>
>	- Section 3.2, page 8, "Stream Sequence-N" definition: "... report TSN'=
s co*or*esponding to..."
>
>	- Section 3.3.1, title of the section (also in the index): This is kind=
 of silly, but, shouldn't be that "param" be "parameter" instead?
>
>	- Section 3.3.2, second paragraph: There is a mistaken reference to RFC=
2960 as [4] instead of [5]...
>
>	- Section 3.3.3, page 9: "... COOKIE or as a** unrecognized...", I woul=
d say that "a" should be "an"... :-/
> =20
>

Thanks for the typos.. I found them all ;->=20

Now to your other questions...


>	- Section 3.4, page 12, rule C5: Hmm... I think that a very common situ=
ation will be that we have some TSNs unacknowledged, and then the T3-rtx =
timer expires (the only one that was running), and then you decide to aba=
ndon all the unacknowledged TSNs and send the FORWARD TSN. There is no T3=
-rtx running, so, C5 says that we "MUST assure that at least one T3-rtx t=
imer is running", so, should be restart it? but, which one?
> =20
>

I don't think that it matters. The point of having  the T3 timer running=20
is that if it is NOT running (as you
illustrate above) and the FWD-TSN is lost.. then you have a kind of=20
deadlock scenario.. not really
a dead lock.. but until the sender of the FWD-TSN sends some new data it=20
will never know to resend
the FWD-TSN. Thus the receiver will possibly be stuck with some=20
out-of-order data on queue and
never send it up to the upper layer until new data is sent... not really=20
what you want I think... So, as
long as one tiemr is running the job will get done i.e. it will go off=20
and then cause you to resend
the FWD-TSN.

I suppose we could add a Implementation note like:
"
IMPLEMENTATION NOTE: Any destination's timer may be used for the
purposes of rule C5.
"


>	Another thing that is not so clear for me is which address should we us=
e for sending the FORWARD TSN... Should we kind of follow the same criter=
ion than when retransmitting the TSNs themselves? However, the abandoned =
TSNs might have been sent to different addresses, so, which one should be=
 use? Maybe I'm just mistaken but I would say that this is not clearly st=
ated in the draft... :-/
>
> =20
>

I think again it does not really matter, as long as the address is=20
"confirmed" (see the I-Gv10 which was
sent in during the IETF)...

So maybe a note after C3 stating:
<t>
IMPLEMENTATION NOTE: It is an implemenation decision as to which
destination address is to be sent to, the only restriction
being that the address MUST be one that is CONFIRMED.
</t>


>	- Section 3.4, page 12, rule A5: As it is written I think it is a littl=
e bit confusing... I suppose that what you want to mean is that after a T=
3 expiration you check whether you have to abandon or not any TSN, and af=
ter that you try to advance the Advanced.Peer.Ack.Point, right? But that =
procedure of advancing that variable is stated only in C2... The others a=
re mostly related with the arrival of a SACK... :-/
>
> =20
>
Hmm in looking at the rules I must disagree and agree :->

Rule C1 has to do with a SACK...
Rule C2 has to do with how you figure out if you have
an advance peer ack point.
Rule C3 tells you that you must send a FWD-TSN (possibly)
Rule C4 tells you that you must back it down if you have some data that
is not partially reliable between the last sack point and the advance=20
peer ack point.
Rule C5 tells you that you must maintain your timer..

So really A5 should use the procedures outlined in C2-5 not C1-5..

So I guess we can consider these IETF-Last-Call comments aka..

1) Fix typos.

2) add two implementation notes

3) correct sequecne range in A5 to C2-C5 instead of C1-C5


Does this sound ok to you Ivan?

R

R

>	Ok, that's all this time!
>
>	Have a nice weekend!
>
>	BR Iv=E1n Arias Rodr=EDguez
>
>_______________________________________________
>tsvwg mailing list
>tsvwg@ietf.org
>https://www1.ietf.org/mailman/listinfo/tsvwg
>
>
>
> =20
>


--=20
Randall R. Stewart
randall@stewart.chicago.il.us 815-342-5222 (cell phone)



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Subject: RE: [Tsvwg] Comments about the PR-SCTP extension
Date: Thu, 27 Nov 2003 12:24:21 +0200
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Thread-Topic: [Tsvwg] Comments about the PR-SCTP extension
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To: <randall@stewart.chicago.il.us>
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	Hi all!

	See my comments inline...

> >	- Section 3.4, page 12, rule C5: Hmm... I think that a=20
> very common situation will be that we have some TSNs=20
> unacknowledged, and then the T3-rtx timer expires (the only=20
> one that was running), and then you decide to abandon all the=20
> unacknowledged TSNs and send the FORWARD TSN. There is no=20
> T3-rtx running, so, C5 says that we "MUST assure that at=20
> least one T3-rtx timer is running", so, should be restart it?=20
> but, which one?
> > =20
> >
>=20
> I don't think that it matters. The point of having  the T3=20
> timer running=20
> is that if it is NOT running (as you
> illustrate above) and the FWD-TSN is lost.. then you have a kind of=20
> deadlock scenario.. not really
> a dead lock.. but until the sender of the FWD-TSN sends some=20
> new data it=20
> will never know to resend
> the FWD-TSN. Thus the receiver will possibly be stuck with some=20
> out-of-order data on queue and
> never send it up to the upper layer until new data is sent...=20
> not really=20
> what you want I think... So, as
> long as one tiemr is running the job will get done i.e. it=20
> will go off=20
> and then cause you to resend
> the FWD-TSN.
>=20
> I suppose we could add a Implementation note like:
> "
> IMPLEMENTATION NOTE: Any destination's timer may be used for the
> purposes of rule C5.
> "

	Yeah, you are completely right... In fact, the point of this comment =
(and the next one) was that I was thinking that, since the FWD-TSN is =
acknowledged as the DATA chunks are, you were also performing that =
retransmission counter reset when the SACK arrives, or an increase plus =
cwnd shrink and RTO doubling and so on...

	But since that's not the case (by the way, which is the reason not to =
do so?), then I agree that the destination doesn't really matter =
(however, I think that any implementer will try to send the FWD-TSN to a =
different address than the previously used)...

> >	Another thing that is not so clear for me is which=20
> address should we use for sending the FORWARD TSN... Should=20
> we kind of follow the same criterion than when retransmitting=20
> the TSNs themselves? However, the abandoned TSNs might have=20
> been sent to different addresses, so, which one should be=20
> use? Maybe I'm just mistaken but I would say that this is not=20
> clearly stated in the draft... :-/
> >
> > =20
> >
>=20
> I think again it does not really matter, as long as the address is=20
> "confirmed" (see the I-Gv10 which was
> sent in during the IETF)...
>=20
> So maybe a note after C3 stating:
> <t>
> IMPLEMENTATION NOTE: It is an implemenation decision as to which
> destination address is to be sent to, the only restriction
> being that the address MUST be one that is CONFIRMED.
> </t>

	Same comment also applies here. The NOTE sounds good for me, but mind =
that "implemenation" typo ;-)

> >	- Section 3.4, page 12, rule A5: As it is written I=20
> think it is a little bit confusing... I suppose that what you=20
> want to mean is that after a T3 expiration you check whether=20
> you have to abandon or not any TSN, and after that you try to=20
> advance the Advanced.Peer.Ack.Point, right? But that=20
> procedure of advancing that variable is stated only in C2...=20
> The others are mostly related with the arrival of a SACK... :-/
> >
> > =20
> >
> Hmm in looking at the rules I must disagree and agree :->
>=20
> Rule C1 has to do with a SACK...

	Yes...

> Rule C2 has to do with how you figure out if you have
> an advance peer ack point.

	I agree...

> Rule C3 tells you that you must send a FWD-TSN (possibly)

	Yes but here there is a comment about the received SACK... In fact, =
thinking about it, since you firstly process normally the SACK, I don't =
think you even have to mention the SACK at all, since the value of the =
"Cumulative TSN Ack Point" must have been already updated at this stage, =
being equal to the "Cumulative TSN ACK carried in the received SACK"...

	Hmm... now that I see it, A4) mentions the normal processing procedure =
for SACKs as stated in 6.2.1, but that section basically deals with the =
rwnd only... Weird that 6.2.1 in the RFC doesn't give any pointer to =
sections 7.2.1 and 7.2.2, which depict the "real" steps you take when =
receiving a SACK, and only mentions 7.2.4... :-/ Well, also 8.1 and 8.2 =
are very important sections regarding SACK processing...

	I hadn't taken a look to the section 6.2.1 when I firstly read the =
draft, but now that I have gone through it, it makes me feel kind of =
uncomfortable (not for the draft, but for the RFC itself)... :-/

> Rule C4 tells you that you must back it down if you have some=20
> data that
> is not partially reliable between the last sack point and the advance=20
> peer ack point.

	Yes.

> Rule C5 tells you that you must maintain your timer..

	Exactly.

> So really A5 should use the procedures outlined in C2-5 not C1-5..

	Yes, you are right... Only that the C3 is also SACK-related as it is =
written...

> So I guess we can consider these IETF-Last-Call comments aka..
>=20
> 1) Fix typos.
>=20
> 2) add two implementation notes
>=20
> 3) correct sequecne range in A5 to C2-C5 instead of C1-C5
>=20
>=20
> Does this sound ok to you Ivan?

	It does!

	Thanks!

	BR Iv=E1n Arias Rodr=EDguez

> R

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Subject: [Tsvwg] SCTP and fragmentation at IP level
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Hello all,

In the section 2.16 of draft-ietf-tsvwg-sctpimpguide-09.txt the
fragmentation and reassembly section says (new text):

   ... If an implementation does not support fragmentation of outbound
   user messages, the endpoint must return an error to its upper layer
   and not attempt to send the user message.

   Note: If an implementation that supports fragmentation makes
   available to its upper layer a mechanism to turn off fragmentation
   it may do so. However in so doing, it MUST react just like an
   implementation that does NOT support fragmentation i.e. it MUST
   reject sends that exceed the current P-MTU.

So, if we play the game widely known as "Say it in ALL CAPS" (a
variation of "Simon says", ages 15 to 65) the 2960 could be interpreted
allowing that if the SCTP implementation does not support fragmentation,
it may as well try to send a message exceeding path MTU as a single IP
datagram (that will be fragmented). Now the note in sctpimpguide seems
to contradict the end of first paragraph.

WOuld it be a nice idea not to play the game in first paragraph and do
it in caps, like "the endpoint MUST return an error", "the endpoint
SHOULD return an error" or "the endpoint MAY return an error"?

--Pekka

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