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Hello everyone,

We are a group of 3 students, Anup Deshpande, Prasad
Edlabadkar and Amit Somanai, from Pune Institute of
Computer Technology ,Pune , India. We are studying in
the 4th year(i.e. final year) of Electronics &
Telecommunication Engineering. We have a thrust for
cutting edge technology and love working on
challenging projects. We also like working on
innovative ideas. Right now we are working on our
final year project. The basic Project idea was to
transport SS7 signals using TCP. But then we came
across the disadvantages of TCP.
So we have three options with us :
1. Using SCTP instead of TCP at the transport level
2. Using Sigtran stack
3. Using UDP and develop some application level on it 

The problem is we don`t know which one is actually
feasible. We are implementing the project in Linux. Is
there any Linux Kernel that is supporting SCTP. Can
you suggest any small module that is still to
implemented?

Plz do reply 


________________________________________________________________________
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Amit:

A couple of suggestions for you:

1) Yes you can get a SCTP stack for linux (sourceforge I think) lk-sctp
    is the project name I think.

2) I do think this question is best addressed to the sigtran mailing
    list... (I think sigtran@ietf.org .. but please check on this). They
    specialize in SS7/IP ... tsvwg usually does not discuss SS7 but end
    to end transport issues..

R

amit somani wrote:
> Hello everyone,
> 
> We are a group of 3 students, Anup Deshpande, Prasad
> Edlabadkar and Amit Somanai, from Pune Institute of
> Computer Technology ,Pune , India. We are studying in
> the 4th year(i.e. final year) of Electronics &
> Telecommunication Engineering. We have a thrust for
> cutting edge technology and love working on
> challenging projects. We also like working on
> innovative ideas. Right now we are working on our
> final year project. The basic Project idea was to
> transport SS7 signals using TCP. But then we came
> across the disadvantages of TCP.
> So we have three options with us :
> 1. Using SCTP instead of TCP at the transport level
> 2. Using Sigtran stack
> 3. Using UDP and develop some application level on it 
> 
> The problem is we don`t know which one is actually
> feasible. We are implementing the project in Linux. Is
> there any Linux Kernel that is supporting SCTP. Can
> you suggest any small module that is still to
> implemented?
> 
> Plz do reply 
> 
> 
> ________________________________________________________________________
> Missed your favourite TV serial last night? Try the new, Yahoo! TV.
>        visit http://in.tv.yahoo.com
> _______________________________________________
> 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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On Thu, 2 Jan 2003, amit somani wrote:

> Hello everyone,
> 
> We are a group of 3 students, Anup Deshpande, Prasad
> Edlabadkar and Amit Somanai, from Pune Institute of
> Computer Technology ,Pune , India. We are studying in
> the 4th year(i.e. final year) of Electronics &
> Telecommunication Engineering. We have a thrust for
> cutting edge technology and love working on
> challenging projects. We also like working on
> innovative ideas. Right now we are working on our
> final year project. The basic Project idea was to
> transport SS7 signals using TCP. But then we came
> across the disadvantages of TCP.
> So we have three options with us :
> 1. Using SCTP instead of TCP at the transport level
> 2. Using Sigtran stack
> 3. Using UDP and develop some application level on it 
> 
> The problem is we don`t know which one is actually
> feasible. We are implementing the project in Linux. Is
> there any Linux Kernel that is supporting SCTP. Can
> you suggest any small module that is still to
> implemented?

Yes. The 2.5 beta version of the linux kernel includes SCTP support.
You can download it from www.kernel.org. The latest release is 2.5.54. 
The SCTP implementation is currently under development but is in a usable
state. It supports the UDP-style SCTP sockets interface as defined
in the the following internet draft
   http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-sctpsocket-05.txt
and will be compliant to RFC2960 when complete. 
You can find more information about the linux kernel SCTP project at
   http://sourceforge.net/projects/lksctp
 
Thanks
Sridhar


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From mailnull@www1.ietf.org  Mon Jan  6 10:23:04 2003
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In a article I read :
"...many existing TCP implementations reset snd_max (the highest byte sent) to snd_una (the highest byte acknowledged) when they exit persist state........."

I recall that TCP enters persist state whenever a zero advertised window has been received and exit persist state when a successive non-zero advertised window has been received.

Please, could someone tell me which existing TCP implementations follow the behaviour described upon? (reset snd_max to snd_una )

Thanks in advance,
Pierluigi


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Subject: RE: [Tsvwg] Re: Last Call: Robust ECN Signaling with Nonces to Experimental 
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Hello,

[Sorry about the delay in responding]

> The security concerns you mention are cetainly valid.  But, one does
> not need the nonce to exploit things if some man-in-the-middle is
> willing to simply generate ACKs indicating "congestion experienced"
> for the end receiver.  

Yes, I agree there are other ways to do DoS attacks (none of
the attacks in my previous e-mail was Man in the Middle attack
though; they are active/passive eavesdroppers). When people
are trying to orchestrate a DoS attack, they need as many DoS
vulnerabilities as possible so that they can generate an innocent
looking DoS attack-traffic that will be harder even for a human
observer to distinguish from a non-attack traffic. For example, if
a web site sees 1000 SYN packets per seconds, it can easily figure
out that the server is under attack (unless it's CNN or New York
Times's web site). On the other hand, however, if the same web 
server sees 100 false nounce, 100 ECN-experienced, 100 run-away 
SYN messages, etcetera, it cannot immediately conclude if it 
is under attack or if the receivers just happen to be on a bad 
network path...

(ECN nounce + ECN) is therefore worse than ECN alone. That's
the point I was trying to make. [I guess I was not very clear]

BTW, IPSec is not an option here. The present key exchange
protocol (IKE) takes 3-5 round-trips to just establish a shared
secret (newer protocols like JFK are better--2 round
trips--but they are still heavyweight). On top of that, the users
will have to wait for at least another 3/2 round trip before they
can see the first byte of "data." This is just not a viable option
for many servers (it would require 3 IPSec enabled servers where
one tradition TCP server would have done--by Little's  theorem
assuming short lived connections). In addition, IPSec does a whole
lot of "other things" (identity protection, non-repudiation ...)
which are not needed in most of the cases.

If we are serious about security, we should stop solving the problem
in an ad hoc manner and make TCP self-secure once and for all. Or we 
can wait until there is a spectacular attack (after which 
every web server will have a hack to prevent such attacks. Then 
we can design protocols to make these hacks work together...).  
 
> 
> And, if the penalty imposed by the sender is worse for blowing the
> nonce than for receiving "congestion experienced" then I might agree
> that the nonce allows for a bigger and better attack than without


As I said in the last paragraph, the penalty is higher because 
any non-trivial policy will open *one more* door for DOS attack.


<...>

> The nonce prevents receivers from lying and protects the network

<...>

I don't buy this:

If users want to lie, they don't need to first send a ECN-setup
SYN and then later start cheating. They have a simpler option: Not
to send ECN-setup at all. Every operating system provides a way to
do this. Every OS *has to* provide this because there are some ultra 
smart firewalls which drop all packets with CWR bit set and never 
bother to even information the receiver with a reset. Besides, to 
cheat this way, one will need the source code of the operating system.
(If operating system vendors want to cheat, they will just not 
implement ECN, and probably even save a few pennies too) 

I am of course assuming here that packets with ECT code points
share the same network queues as traditional TCP packets 
(if they don't, then there are other related problems which 
need to be solved by DiffServ framework. I don't know all the 
details :-( ).

So, I don't see how ECN nounces are helping. The only place where 
it might help is with crazy middle boxes. But that problem is again 
a generic problem that DiffServ should solve (at IP layer)--not TCP.

> there are bugs in boxen in the network.  The nonce does not protect
> against malicious third-parties, I agree -- it was not intended to.
> But, I don't think it hurts that case either.
> 

A cheating TCP receiver is a malicious user (if not a third
party), and 1 bit nounce can only protect in 50% of the cases.
That's why I said the security is weak--it's weak even for the
problem it's trying to solve--not because it cannot protect TCP
from all possible attacks. I know people have difficulty getting
good security with even 2048 bits or more...


If people are happy with the nounce placebo, that's fine with me.
My only suggestion is that we should not waste (I'm sorry, I could
not come with a better word) the bit in the TCP header.
Tomorrow if we need those bits for some functionality (instead of for
protection), the only option would be to have a new transport 
layer protocol. In addition, putting the nounce in a TCP option 
is better since one can use more bits and detect cheating receivers
with a higher confidence level than 50%. I bit entropy is too little.


--yogesh


BTW, just curious to know if someone's designing a protocol police 
to ensure that TCP senders halve their data rate after packet
loss. To the best of my knowledge, TCP receivers are not the only
ones who are capable of cheating; even TCP senders can cheat. Just
curious know.
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Subject: Re: [Tsvwg] Re: Last Call: Robust ECN Signaling with Nonces to Experimental 
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Yogesh-

> > The nonce prevents receivers from lying and protects the network
> 
> I don't buy this:
> 
> If users want to lie, they don't need to first send a ECN-setup
> SYN and then later start cheating. They have a simpler option: Not
> to send ECN-setup at all.

This case is not "cheating".  Everything is still congestion
responsive in this case.  The cheating that the nonce prevents is
when the receiver can conceal a congestion indication from the
network.

If you don't setup ECN then your packets will not be marked, but
rather dropped -- which means either you have to tell the truth
about congestion or you don't get all your data (if you ACK
something that has not arrived).  I don't think the incentive for
covering up congestion is there in the non-ECN case.

> A cheating TCP receiver is a malicious user (if not a third
> party), and 1 bit nounce can only protect in 50% of the cases.
> That's why I said the security is weak--it's weak even for the
> problem it's trying to solve--not because it cannot protect TCP
> from all possible attacks. I know people have difficulty getting
> good security with even 2048 bits or more...

This is FUD.  The nonce is *much* better than 50%.  You have a 50%
shot at guessing the nonce right on every ACK.  But, if you guess
wrong once then the sender is on to you -- so, the probability of
guessing right is 0.5^N (where N is the number of ACKs
transmitted).  So, clearly as the connection progresses and N grows
large the probability of keeping your game going gets pretty small
pretty quickly.

> Tomorrow if we need those bits for some functionality (instead of
> for protection), the only option would be to have a new
> transport layer protocol.

(Actually, when we run out I am sure that the first thing that will
happen will be someone generating an RFC defining a TCP "more bits"
option.) 

> BTW, just curious to know if someone's designing a protocol
> police=20 to ensure that TCP senders halve their data rate after
> packet loss. To the best of my knowledge, TCP receivers are not
> the only ones who are capable of cheating; even TCP senders can
> cheat. Just curious know.

TCP senders could be an even bigger problem -- because they Just
Control the sending rate (i.e., they don't need manipulative tricks
like the reciever does).  See some of the ICIR/AT&T work on
pushback.  Or, Sally Floyd and Kevin Fall's notion of a penalty
box.

allman


--
Mark Allman -- BBN/NASA GRC -- http://roland.grc.nasa.gov/~mallman/
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Subject: Re: [Tsvwg] Re: Last Call: Robust ECN Signaling with Nonces to Experimental 
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Hi Mark, all,

> This case is not "cheating".  Everything is still congestion
> responsive in this case.  The cheating that the nonce prevents is
> when the receiver can conceal a congestion indication from the
> network.

You are right. I don't know what I was thinking while making that
argument. Apologies.

> This is FUD.  The nonce is *much* better than 50%.  You have a 50%
> shot at guessing the nonce right on every ACK.  But, if you guess
> wrong once then the sender is on to you -- so, the probability of
> guessing right is 0.5^N (where N is the number of ACKs
> transmitted).  So, clearly as the connection progresses and N
> grows large the probability of keeping your game going gets pretty
> small pretty quickly.

This is just half of the story. The other half is that the
probability of an attacker being successful in perpetrating 
a DoS attack is (1-0.5^n). But an attacker does not have 
to send N ACKs, total of four ACKs messages for 2 different TCP 
segments will suffice (the sequence of messages
 
(time=0, ACK-1, Nounce=0)
(time=0, ACK-1, Nounce=1)
(time=1, ACK-2, Nounce=1)
(time=1, ACK-2, Nounce=0), 

is all that an attacker has to send to trigger the sender's 
policy--assuming FIFO queues.)

> (Actually, when we run out I am sure that the first thing that
> will happen will be someone generating an RFC defining a TCP "more
> bits" option.)

Sure. The other option is to let the 4 bits in the TCP be the MSbs
of DOFF, and let the Nounce be an option.

The reason we differ on nounces is probably because I am too
pessimistic about DoS attacks, and the draft is too pessimistic
about cheating receivers and middle boxes.

To cheat an attacker needs the source code of the OS, technical 
expertise of the internal working of the OS etc. On the other hand, 
to do a DoS attack, all one needs is a script that can snoop and 
construct TCP ACKs. I don't know which of the two is more dangerous 
in a realistic scenario. 

--yogesh
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From: Mark Allman <mallman@grc.nasa.gov>
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Subject: Re: [Tsvwg] Re: Last Call: Robust ECN Signaling with Nonces to Experimental 
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> > > A cheating TCP receiver is a malicious user (if not a third
> > > party), and 1 bit nounce can only protect in 50% of the cases.
> > > That's why I said the security is weak--it's weak even for the
> > > problem it's trying to solve--not because it cannot protect
> > > TCP from all possible attacks. I know people have difficulty
> > > getting good security with even 2048 bits or more...
> > 
> > This is FUD.  The nonce is *much* better than 50%.  You have a 50%
> > shot at guessing the nonce right on every ACK.  But, if you guess
> > wrong once then the sender is on to you -- so, the probability of
> > guessing right is 0.5^N (where N is the number of ACKs
> > transmitted).  So, clearly as the connection progresses and N
> > grows large the probability of keeping your game going gets pretty
> > small pretty quickly.
> 
> This is just half of the story. 

I have requoted the statement you made (at the top) that I responded
to.  I was considering only the case when we were considering
*receivers* (not third parties) lying.

I stick by the above argument.  The case of someone snooping traffic
and hosing the data rate by lying is as you outlined (not quoted).
But, that was not the comment I responded to.  We agree that
eavesdroppers can exploit ECN and the ECN nonce as a mechanism for
DoS.  I simply wanted to apply pushback to your statement that the
nonce does not offer good protection for the problem it purports to
solve.  In my opinion, it offers good protection for its intent.

allman


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Subject: Re: [Tsvwg] Re: Last Call: Robust ECN Signaling with Nonces to Experimental
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On Wed, Jan 08, 2003 at 12:53:29PM -0500, Mark Allman wrote:
> We agree that eavesdroppers can exploit ECN and the ECN
> nonce as a mechanism for DoS.

Can either of you explain this statement a little bit more?

If an evesdropper is able to spoof an acknowledgement with a
faulty nonce, as described in Yogesh's prior mail, it would
seem just as capable of spoofing a RST to tear down the
connection in one packet (not probabilistically four).
While the nonce doesn't solve the problem of spoofing
evesdroppers, I don't yet see how the nonce increases the
potential for misbehavior in the network.

-neil

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Yeah, what Neil said. 

In addition, I Would Have Thought that an attacker who couldn't spoof an ACK (or RST) would just be SYN-flooding anyway. I'm not sure why we're not holding ECN Nonces to this standard. 

Do we think third-party attackers would rather twiddle with TCP sending rates than (1) overwhelm a network segment entirely via flooding, or (2) RST TCP connections?

Sounds like an awfully polite attacker to me - or is that the answer (twiddling with TCP sending rates would be less likely to set off intrusion detection systems)?

Spencer

> -----Original Message-----
> From: Neil Spring [mailto:nspring@cs.washington.edu]
> Sent: Wednesday, January 08, 2003 1:19 PM
> To: tsvwg@ietf.org
> Subject: Re: [Tsvwg] Re: Last Call: Robust ECN Signaling with 
> Nonces to
> Experimental
> 
> 
> On Wed, Jan 08, 2003 at 12:53:29PM -0500, Mark Allman wrote:
> > We agree that eavesdroppers can exploit ECN and the ECN
> > nonce as a mechanism for DoS.
> 
> Can either of you explain this statement a little bit more?
> 
> If an evesdropper is able to spoof an acknowledgement with a
> faulty nonce, as described in Yogesh's prior mail, it would
> seem just as capable of spoofing a RST to tear down the
> connection in one packet (not probabilistically four).
> While the nonce doesn't solve the problem of spoofing
> evesdroppers, I don't yet see how the nonce increases the
> potential for misbehavior in the network.
> 
> -neil> 
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Subject: RE: [Tsvwg] Concerns about SCTP
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> -----Original Message-----
> From: Randall Stewart [mailto:randall@stewart.chicago.il.us]
> 
> Some comments for you to think upon... :>
> 
> Williams, Jim wrote:
> > [...]
> > 
> > I regard the following as two basic
> > rules of protocol design.
> > 
> > Rule 1:  Two implementations that are fully
> >          compliant with the protocol definition
> >          must interoperate.  If they don't,
> >          then that is a defect in the protocol
> >          definition.
> > 
> > Rule 2:  One end of an association should not
> >          be able to consume an unbounded amount
> >          of resources on the other end without
> >          negotiation and agreement from the other
> >          end.
> > 
> > I believe in this respect, the multi-homing feature
> > of SCTP comes up short.  Each remote address that
> > is supported consumes resources on the local end.
> > SCTP places no limit and requires no negotiation
> > as to the number addresses that can be included
> > in an INIT.  The receiver must either allocate
> > the resources needed to support this number of
> > addresses (violating rule 2), or must abort
> > the association if too many are requested
> > (violating rule 1).
> > 
> 
> I don't see how this violates rule 1. SCTP provides a
> basic "I can't do that .. not enough resources" in response
> to something that can't be done. Yes the association will
> not come up until the peer reduces the resources request...

If one views it from a system perspective, a user may
be running a file system over RDDP over SCTP.  If the
file system fails to mount, the user would regard this
a failure to interoperate.  However this failure can
be prevented in multiple ways.

1.  At the RDDP over SCTP adaption layer, specify that
    if an endpoint specifies multiple IP addresses and
    the association setup is aborted with a resource error,
    the endpoint MUST retry the association with fewer
    IP addresses. (The draft could also remain silent
    on this and trust the good sense of the upper layers
    to do this anyway, but that seems like asking
    for interoperability woes unnecessarily.)

    Although this works, I regard it as more of a 
    work around than a well designed negotiation.
    An example of a well designed negotiation is
    the mechanism use by SCTP to determine the number
    of streams per association.

2.  Treat this as a flaw in SCTP and fix it.  Set
    aside practical and political considerations for
    the moment.  I regard this a flaw (perhaps minor
    but real) that affects not only RDDP, but all
    low end, high speed, embedded implementations of
    SCTP.  If the goal of SCTP is to become a
    universal transport, not just a high end
    transport, then this may make sense.

    I acknowledge that responders to my original
    post do not agree that there is any flaw.  But
    my own opinion is not yet changed by reading
    the responses.

3.  Acknowledge that SCTP is not an optimum transport
    for RDDP, and use a different transport.  If one
    were to design a transport optimized for high
    speed hardware accelerated applications, it would
    look quite different from SCTP.

    The counter arguments here, though, are that there
    is advantage to standardizing on a transport across
    many applications even if not optimum for each
    application.  And also that transports should
    be designed to optimize data flow through the
    network, not to optimize end point implementation,
    (at least that seems to be the IETF philosophy).


Anyway, this is the direction I was hoping to steer
the discussion.  For now, #1 seems like the preferred
direction.  But I though #2 and #3 were worth at least
brief discussion.

> I know, for me, I truly have
> a hard time understanding how ECN would work better for
> flow control than the rwnd. aka
> 
> ------DATA-----> (rcver realizes its is running low
>                    on resources (chunks/structures/?)
>   <------SACK(rwnd=0)

I agree with this.  But I was unaware that this is allowed.
TCP specifically forbids retracting a previously advertised
window.

I am not clear on where SCTP stands on this.  It appears
that in RFC2960 there is no prohibition on this, but
it does seem contrary to a few SHOULDs regarding rwnd
calculation.  Anyway, this may be the right
answer unless the SHOULDs evolve into de facto MUSTs.

Interestingly, though, RDDP has the property that all
received data can be immediately processed.  There is
never any need to stop the remote sender.  The only
problem occurs when data comes in faster than it
can be processed.  In this case the desired behavior
is to slow down the sender a bit, not stop it.
Therefore ECN might be a better functional fit
for this requirement than rwnd.  

If the receiver knew how much
data the sender currently had outstanding (usually
equal to its cwnd) then it could set rwnd to
a half (or some other fraction) of that and 
accomplish the same thing, but the receiver
does not have this knowledge.  This creates
a problem using rwnd to throttle the sender's
rate.  But this problem doesn't appear
insurmountable.

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Date: Thu,  9 Jan 2003 12:13:47 -0600
From: Caitlin Bestler <cait@asomi.com>
Subject: RE: [Tsvwg] Concerns about SCTP
To: "'tsvwg@ietf.org'" <tsvwg@ietf.org>
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        "Williams, Jim" <Jim.Williams@emulex.com>, rddp@ietf.org
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On 1/9/03, Williams, Jim wrote:


>
>Interestingly, though, RDDP has the property that all
>received data can be immediately processed.  There is
>never any need to stop the remote sender.  The only
>problem occurs when data comes in faster than it
>can be processed.  In this case the desired behavior
>is to slow down the sender a bit, not stop it.
>Therefore ECN might be a better functional fit
>for this requirement than rwnd.  
>
That property is intrinsic to RDDP no matter what transport
it runs over. The ULP is responsible for pacing untagged
messages, but only the LLP paces tagged messages.

RDDP releases all LLP buffers "immediately", meaning as
quickly as the placement of the tagged/untagged data can be
completed. Given that characteristic it is highly unlikely
that there would be buffering resources dedicated to
specific assocation or connection.

Therefore the challenge, whether over MPA/TCP and/or SCTP,
is to ensure that the internal bandwidth required to support
DDP placement will not be swamped by all incoming data.

Specifically the windows SHOULD be set so that the risk of
exhausting the placement bandwith is as low as possible.

However those windoes CANNOT beset so that said risk is
non-existent. LLP Credits are delivered asynchronously,
uncertainty on placemnt speed because of potential host bus
contention or retries, and more importantly in integral
amounts. Those integral amounts, spread over hundreds of
thousands of connections, either force a severe
under-allocation or an over-allocation.

And certainly the available to-host bandwidth over a
conventional host bus is statistically reliable but
defintely not guaranteed. Limiting your RNIC to the host bus
*guaranteed* rates would be a severe limitation.

I believe most implementers will properly choose to take a
slight risk of over-allocation, knowing that in the worst
case the result is that a packet is dropped. A properly
credited LLP packet arrives and there is no available buffer
for it. No LLP ack is generated, and it is eventually
retransmitted. The same thing can happen at any number of
routers and switches along the route for the very same
reason.

Also consider that in some cases the host memory placement
capacity may actually exceed the capacity of the external
link layer. In that case LLP flow control deals only with
network congestion, RNIC and host buffering is irrelevant.
  

Caitlin Bestler
http://asomi.com/CaitlinBestler/
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From: "Andrei Gurtov" <andrei.gurtov@sonera.com>
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Subject: [Tsvwg] Is a TFRC sender allowed to increase the send rate more than twice?
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Hi,

In draft-ietf-tsvwg-tfrc-05.txt on p. 11 a TFRC sender computes its send
rate as

If (p > 0)
                   Calculate X_calc using the TCP throughput equation.
                   X = max(min(X_calc, 2*X_recv), s/t_mbi);

when there are some losses detected. A question is, when X_recv and X_calc
suddenly increase, is the sender allowed to increase its transmission rate X
more than twice? Such situation could appear when a number of delayed or
reordered packets show up at the receiver causing the loss rate become low
and the receive rate higher than the current transmit rate. Alternatively,
the receiver or PEP having good knowledge of the bottleneck link could
adjust the loss rate and X_recv in feedback packets to make the sender adapt
faster to significant changes in available bandwidth. Yet another
possibility is a spoofing attack doing the same.

According to the equation above more than twofold increase of the send rate
is ok, but the NS implementation clamps the send rate increase at twice the
current in all cases.

Off topic:

Given the recent discussion on TFRC response to application-limited
connections, I would be cautious to say that TFRC is suitable for telephony
applications.

A typo in p. 21 pervious -> previous?

Andrei




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Comments in-line


Williams, Jim wrote:
> 
>>-----Original Message-----
>>From: Randall Stewart [mailto:randall@stewart.chicago.il.us]
>>
>>Some comments for you to think upon... :>
>>
>>Williams, Jim wrote:
>>
>>>[...]
>>>
>>>I regard the following as two basic
>>>rules of protocol design.
>>>
>>>Rule 1:  Two implementations that are fully
>>>         compliant with the protocol definition
>>>         must interoperate.  If they don't,
>>>         then that is a defect in the protocol
>>>         definition.
>>>
>>>Rule 2:  One end of an association should not
>>>         be able to consume an unbounded amount
>>>         of resources on the other end without
>>>         negotiation and agreement from the other
>>>         end.
>>>
>>>I believe in this respect, the multi-homing feature
>>>of SCTP comes up short.  Each remote address that
>>>is supported consumes resources on the local end.
>>>SCTP places no limit and requires no negotiation
>>>as to the number addresses that can be included
>>>in an INIT.  The receiver must either allocate
>>>the resources needed to support this number of
>>>addresses (violating rule 2), or must abort
>>>the association if too many are requested
>>>(violating rule 1).
>>>
>>
>>I don't see how this violates rule 1. SCTP provides a
>>basic "I can't do that .. not enough resources" in response
>>to something that can't be done. Yes the association will
>>not come up until the peer reduces the resources request...
> 
> 
> If one views it from a system perspective, a user may
> be running a file system over RDDP over SCTP.  If the
> file system fails to mount, the user would regard this
> a failure to interoperate.  However this failure can
> be prevented in multiple ways.
> 
> 1.  At the RDDP over SCTP adaption layer, specify that
>     if an endpoint specifies multiple IP addresses and
>     the association setup is aborted with a resource error,
>     the endpoint MUST retry the association with fewer
>     IP addresses. (The draft could also remain silent
>     on this and trust the good sense of the upper layers
>     to do this anyway, but that seems like asking
>     for interoperability woes unnecessarily.)
> 
>     Although this works, I regard it as more of a 
>     work around than a well designed negotiation.
>     An example of a well designed negotiation is
>     the mechanism use by SCTP to determine the number
>     of streams per association.
> 
> 2.  Treat this as a flaw in SCTP and fix it.  Set
>     aside practical and political considerations for
>     the moment.  I regard this a flaw (perhaps minor
>     but real) that affects not only RDDP, but all
>     low end, high speed, embedded implementations of
>     SCTP.  If the goal of SCTP is to become a
>     universal transport, not just a high end
>     transport, then this may make sense.
> 
>     I acknowledge that responders to my original
>     post do not agree that there is any flaw.  But
>     my own opinion is not yet changed by reading
>     the responses.
> 
> 3.  Acknowledge that SCTP is not an optimum transport
>     for RDDP, and use a different transport.  If one
>     were to design a transport optimized for high
>     speed hardware accelerated applications, it would
>     look quite different from SCTP.
> 
>     The counter arguments here, though, are that there
>     is advantage to standardizing on a transport across
>     many applications even if not optimum for each
>     application.  And also that transports should
>     be designed to optimize data flow through the
>     network, not to optimize end point implementation,
>     (at least that seems to be the IETF philosophy).
> 
> 
> Anyway, this is the direction I was hoping to steer
> the discussion.  For now, #1 seems like the preferred
> direction.  But I though #2 and #3 were worth at least
> brief discussion.
> 

As you may have guessed I to am in camp 1 :> I don't beleive
it is a flaw... and even if we did (which I know I don't) the
solution would require multiple message round trips to setup
the addresses... this is quite a bit unlike the streams issue.

I just don't see this as a problem.. the DDP adaption is very
unlikely to allow multiple IP addresses... if so the NIC's
will need to all be related and cross-configured together...

DDP clients will just have to be aware of this issue...

On top of that I also think multi-homing is not as numerous
as you think for clients... two maybe three addresses is what
most clients will have.. not the 100's you seem to invision.. in
fact we could even add to the SCTP-DDP adaption something like
"A user of SCTP MUST NOT bind more than N addresses"... and then
put in verbage to say how the receiver an association with more
than N addresses aborts it... very simple.. and I believe there
is a user defined reason for abort that I know the socket API
supports... so we define a SCTP-DDP adatption ABORT reason that
comes in with the abort.. and your problem is solved...maybe not
what you want but I don't see anything wrong with this..



> 
>>I know, for me, I truly have
>>a hard time understanding how ECN would work better for
>>flow control than the rwnd. aka
>>
>>------DATA-----> (rcver realizes its is running low
>>                   on resources (chunks/structures/?)
>>  <------SACK(rwnd=0)
> 
> 
> I agree with this.  But I was unaware that this is allowed.
> TCP specifically forbids retracting a previously advertised
> window.
> 

I suggest you take a close look at the code snippet below:
"
#define	sbspace(sb) \
     ((long) imin((int)((sb)->sb_hiwat - (sb)->sb_cc), \
	 (int)((sb)->sb_mbmax - (sb)->sb_mbcnt)))
"


This is one of the key macro's used out of sys/socketvar.h which
is used to caclulate space and the windows.. notice the use
of sb_mbmax and sb_mbcnt... these track the amount of mbuf
useage...

The reason I know this is I have been bitten by the above when we were
doing our SCTP implementation... Since the mbcnt also comes in to play I
was seeing the rwnd drop sharply at the end unexpectedly.. this is
because the mbuf limits were being hit before the sb_cc limits...

I won't go in to the details of how I worked around this ... my point
is that TCP has factored into its rwnd (at least in bsd) the amount
of mbuf's (structures to track things) used for years...

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

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Hi

Are there any studies that have measured or attempted to measure the
percentage of out-of-order TCP traffic  on the Internet?  Similar to the
studies that folks have done on fragmented IP traffic in the Internet,
admittedly  it is much  more difficult to measure the former.

Thanks,
Murali

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Subject: RE: [Tsvwg] Out of order  TCP traffic on the Internet
Date: Fri, 10 Jan 2003 13:40:25 -0600
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Thread-Topic: [Tsvwg] Out of order  TCP traffic on the Internet
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From: "Teisberg, Robert" <robert.teisberg@hp.com>
To: "Murali Bashyam" <mbashyam@cisco.com>, <tsvwg@ietf.org>
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Here's one (from the abstract: "We have found, however, that
parallelism in Internet components and links is causing packet
reordering under normal operation and that the incidence of packet
reordering appears to be substantially higher than previously
reported"):

IEEE/ACM Transactions on Networking (TON) Volume 7, Issue 6 (December 1999)

Packet reordering is not pathological network behavior 

Authors 
  Jon C. R. Bennett  Harvard Univ., Cambridge, MA 
  Craig Partridge  BBN Technologies, Cambridge, MA 
  Nicholas Shectman  BBN Technologies, Cambridge, MA 

Publisher
 ACM Press   New York, NY, USA 

  Pages: 789 - 798  Periodical-Issue-Article 
  Year of Publication: 1999 
  ISSN:1063-6692  

> -----Original Message-----
> From: Murali Bashyam [mailto:mbashyam@cisco.com]
> Sent: Friday, January 10, 2003 12:41 PM
> To: tsvwg@ietf.org
> Subject: [Tsvwg] Out of order TCP traffic on the Internet
> 
> 
> Hi
> 
> Are there any studies that have measured or attempted to measure the
> percentage of out-of-order TCP traffic  on the Internet?  
> Similar to the
> studies that folks have done on fragmented IP traffic in the Internet,
> admittedly  it is much  more difficult to measure the former.
> 
> Thanks,
> Murali
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 
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cc: tsvwg@ietf.org, "Sally Floyd" <floyd@icir.org>
Subject: Re: [Tsvwg] Is a TFRC sender allowed to increase the send rate more than twice? 
In-reply-to: Your message of "Fri, 10 Jan 2003 11:47:46 +0200."
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>Hi,
>
>In draft-ietf-tsvwg-tfrc-05.txt on p. 11 a TFRC sender computes its send
>rate as
>
>If (p > 0)
>                   Calculate X_calc using the TCP throughput equation.
>                   X = max(min(X_calc, 2*X_recv), s/t_mbi);
>
>when there are some losses detected. A question is, when X_recv and X_calc
>suddenly increase, is the sender allowed to increase its transmission rate X
>more than twice? Such situation could appear when a number of delayed or
>reordered packets show up at the receiver causing the loss rate become low
>and the receive rate higher than the current transmit rate. Alternatively,
>the receiver or PEP having good knowledge of the bottleneck link could
>adjust the loss rate and X_recv in feedback packets to make the sender adapt
>faster to significant changes in available bandwidth. Yet another
>possibility is a spoofing attack doing the same.
>
>According to the equation above more than twofold increase of the send rate
>is ok, but the NS implementation clamps the send rate increase at twice the
>current in all cases.

In practice, I believe the difference has no effect on behaviour.

 - During slowstart, the send rate increase is capped to twice the
   send rate explicitly.

 - During normal operation, the normal increase rate due to the expiry
   of old higher-loss history is (from memory) about 0.12 packets per
   RTT.  Thus to double the sending rate would take about 6 RTTs.  It
   seems rate unlikely that any bunching would cause more than 6
   RTTs of packets to arrive at once without having previously causing
   the receiver to signal the sender to reduce rate because the
   receive rate was so low while the packets were being delayed.  

 - If history discounting has cut in, the 6 RTTs above would be 3
   RTTs, but delaying of 3 RTTs of traffic would still likely trigger
   the receiver to clamp the senders rate to twice the (very low) receive
   rate before the delayed traffic arrives.  

Limiting the send rate to twice the receive rate is a safety measure
that prevents serious overshoot on links with low levels of
statistical multiplexing.  In the case where you can delay several
RTTs worth of packets without losing any, and then deliver them all in
a high-rate burst, it seems rather likely that the level of
statistical multiplexing isn't all that low, so this additional safety
mechanism isn't required.  In the worst and very unlikely case, you
take some losses in the next RTT and slow down again pretty quickly.

So, to summarise, restricting the increase to twice the send rate (as
ns does) is a reasonable thing to do, but not doing it is very
unlikely to have any significant effect, and even less likely to have
any significant adverse effect.

Does this answer your question?

Cheers,
	Mark
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From: Sourabh Ladha <ladha@mail.eecis.udel.edu>
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Subject: [Tsvwg] TCP-sack implementations
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Hi,
  I have been searching for freebsd- tcp-sack implementations. Could
anyone please direct me to some of the implementations other than Luigi Rizzo's .


Thanks
Sourabh

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From: Marco Mellia <mellia@prezzemolo.polito.it>
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Organization: Politecnico di Torino
To: Murali Bashyam <mbashyam@cisco.com>, tsvwg@ietf.org
Subject: Re: [Tsvwg] Out of order  TCP traffic on the Internet
Date: Mon, 13 Jan 2003 13:38:55 +0100
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We have done some measurement, and about 3/4% of TCP connections seems to 
suffer from packet reordering.

For more details, 

M. Mellia, A. Carpani, R. Lo Cigno
Measuring IP and TCP behavior on Edge Nodes
IEEE Globecom 2002
Taipei, TW, November 17-21

> Are there any studies that have measured or attempted to measure the
> percentage of out-of-order TCP traffic  on the Internet?  Similar to the
> studies that folks have done on fragmented IP traffic in the Internet,
> admittedly  it is much  more difficult to measure the former.

-- 
Ciao,                    /\/\/\rco

+-----------------------------------+  
| Marco Mellia - Assistant Professor|
| Tel: 39-011-5644173               |
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+-----------------------------------+
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Subject: [Tsvwg] Last Call: A Conservative SACK-based Loss Recovery Algorithm
 for TCP to Proposed Standard
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The IESG has received a request from the Transport Area Working Group 
to consider 'A Conservative SACK-based Loss Recovery Algorithm for TCP' 
<draft-allman-tcp-sack-13.txt> as a Proposed Standard.  

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 2003-2-13.

Files can be obtained via http://www.ietf.org/internet-drafts/draft-allman-tcp-sack-13.txt



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From: "Andrei Gurtov" <andrei.gurtov@sonera.com>
To: "Mark Handley" <mjh@icir.org>
Cc: <tsvwg@ietf.org>, "Sally Floyd" <floyd@icir.org>
References: <65115.1042318072@vulture.icir.org>
Subject: Re: [Tsvwg] Is a TFRC sender allowed to increase the send rate more than twice? 
Date:   Mon, 13 Jan 2003 10:54:42 +0200
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> >Hi,
> >
> >In draft-ietf-tsvwg-tfrc-05.txt on p. 11 a TFRC sender computes its send
> >rate as
> >
> >If (p > 0)
> >                   Calculate X_calc using the TCP throughput equation.
> >                   X = max(min(X_calc, 2*X_recv), s/t_mbi);
> >
> >when there are some losses detected. A question is, when X_recv and
X_calc
> >suddenly increase, is the sender allowed to increase its transmission
rate X
> >more than twice? Such situation could appear when a number of delayed or
> >reordered packets show up at the receiver causing the loss rate become
low
> >and the receive rate higher than the current transmit rate.
Alternatively,
> >the receiver or PEP having good knowledge of the bottleneck link could
> >adjust the loss rate and X_recv in feedback packets to make the sender
adapt
> >faster to significant changes in available bandwidth. Yet another
> >possibility is a spoofing attack doing the same.
> >
> >According to the equation above more than twofold increase of the send
rate
> >is ok, but the NS implementation clamps the send rate increase at twice
the
> >current in all cases.
>
> In practice, I believe the difference has no effect on behaviour.
>
>  - During slowstart, the send rate increase is capped to twice the
>    send rate explicitly.
>
>  - During normal operation, the normal increase rate due to the expiry
>    of old higher-loss history is (from memory) about 0.12 packets per
>    RTT.  Thus to double the sending rate would take about 6 RTTs.  It
>    seems rate unlikely that any bunching would cause more than 6
>    RTTs of packets to arrive at once without having previously causing
>    the receiver to signal the sender to reduce rate because the
>    receive rate was so low while the packets were being delayed.
>
>  - If history discounting has cut in, the 6 RTTs above would be 3
>    RTTs, but delaying of 3 RTTs of traffic would still likely trigger
>    the receiver to clamp the senders rate to twice the (very low) receive
>    rate before the delayed traffic arrives.
>
> Limiting the send rate to twice the receive rate is a safety measure
> that prevents serious overshoot on links with low levels of
> statistical multiplexing.  In the case where you can delay several
> RTTs worth of packets without losing any, and then deliver them all in
> a high-rate burst, it seems rather likely that the level of
> statistical multiplexing isn't all that low, so this additional safety
> mechanism isn't required.  In the worst and very unlikely case, you
> take some losses in the next RTT and slow down again pretty quickly.
>
> So, to summarise, restricting the increase to twice the send rate (as
> ns does) is a reasonable thing to do, but not doing it is very
> unlikely to have any significant effect, and even less likely to have
> any significant adverse effect.
>
> Does this answer your question?

Thank you, this answers the first part of the question.

I do think that links with delays and with low level of statistical
multiplexing exist. For example GPRS has a dedicated buffer per user and has
delay spikes of x10 RTT due to handovers.

My own interest in asking this question is a possibility to explicitly
control the rate of the TFRC sender from the receiver or from the network.
Currently it takes more than 1 min for TFRC to reach the full rate after a
handover from GPRS to UMTS. It seems that some push to the TFRC sender is
needed to make it transmit at the full rate faster. My guess is that one
leap change in transmission rate is preferrable by applications than many
small incremental changes.

After a handover the receiver can set X_rcv to the new link's rate and p to
some small none-zero value which will cause the TFRC sender to send at the
higher rate immediately.
Of couse this shouldn't be used to the sender over the shared Internet, but
for servers connected by LAN it seems quite appropriate.

Now that I'm certain that this is supported by specs, I can go on with
experiments.

Andrei

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From mailnull@www1.ietf.org  Wed Jan 15 13:25:23 2003
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hello everyone,
we are a group of 3 students from pune institute of
computer technology,pune ,india. we are doing a
project on SS7. for that we want to know how the SS7
signals look? Though we`ve read many tutorials from
net and also read a book on SS7 by travis russel but
still could not get the exact details of the SS7
signals ( especially Message Signal Units). My exact
doubt is how does SS7 signals look exactly on a SS7
link?
Can someone send us the message which are practically
extracted from SS7 link as we do not have access to
SS7 link and give us the details of the message? or
else can someone give the signals which might be used
for demo purpose like some signal xxxxxxxxxxx for call
establishment , xxxxxx..xx for canceling a call and so
on.

plz do reply.

thank you.
amit. 

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From mailnull@www1.ietf.org  Wed Jan 15 14:04:48 2003
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From: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>
To: "'rohc@ietf.org'" <rohc@ietf.org>
Cc: "'sigtran@ietf.org'" <sigtran@ietf.org>,
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Subject: [Tsvwg] Removing SCTP header compression from the ROHC charter
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ROHCers (cc sigtran & tsvwg),

The current ROHC WG charter has SCTP header compression listed as
one of our work items.  When this item was added, activity within
the ROHC WG was high, and the addition was made as a prediction of
future expectations rather than an outcome of well justified
immediate requirements.

Trying to fulfill this work item during the last year, we have
repeatedly asked for contributions to SCTP header compression, but
we have not received a single response.

Given that this work item was originally added without immediate
market pull, and considering the lack of current interest in doing
the work (or even having it done at all), we intend to remove SCTP
header compression from the ROHC WG charter.  To keep it in the
charter, justifications for developing SCTP header compression now
would have to be presented, as well as industry support and
commitments for doing the actual work.

Barring further input, we will send a request to the ADs by Jan 24,
so please reply before this date if you believe SCTP header
compression should be on the agenda now.

/Lars-Erik & Carsten
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From: Anil Kumar <anil.vydyam@wipro.com>
To: amit somani <amitksomani@yahoo.co.in>,
        lksctp developers
	 <sctp-developers-list@cig.mot.com>,
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I hope this will help, if not, please let me know ...

- Anil.V
 <<ISUP_MSGS.doc>> 

> -----Original Message-----
> From:	amit somani [SMTP:amitksomani@yahoo.co.in]
> Sent:	Wednesday, January 15, 2003 11:55 PM
> To:	lksctp developers; Sigtran; tsvwg@ietf.org
> Subject:	[Sigtran] SS7 signal
> 
> hello everyone,
> we are a group of 3 students from pune institute of
> computer technology,pune ,india. we are doing a
> project on SS7. for that we want to know how the SS7
> signals look? Though we`ve read many tutorials from
> net and also read a book on SS7 by travis russel but
> still could not get the exact details of the SS7
> signals ( especially Message Signal Units). My exact
> doubt is how does SS7 signals look exactly on a SS7
> link?
> Can someone send us the message which are practically
> extracted from SS7 link as we do not have access to
> SS7 link and give us the details of the message? or
> else can someone give the signals which might be used
> for demo purpose like some signal xxxxxxxxxxx for call
> establishment , xxxxxx..xx for canceling a call and so
> on.
> 
> plz do reply.
> 
> thank you.
> amit. 
> 
> ________________________________________________________________________
> Missed your favourite TV serial last night? Try the new, Yahoo! TV.
>        visit http://in.tv.yahoo.com
> _______________________________________________
> Sigtran mailing list
> Sigtran@ietf.org
> https://www1.ietf.org/mailman/listinfo/sigtran

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0M8R4KGxGuEAAAAAAAAAAAAAAAAAAAAAPgADAP7/CQAGAAAAAAAAAAAAAAABAAAAJgAAAAAAAAAA
EAAAKAAAAAEAAAD+////AAAAACUAAAD/////////////////////////////////////////////
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------_=_NextPart_000_01C2BD13.B9B30260--
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From: Pasi Sarolahti <pasi.sarolahti@nokia.com>
Organization: Nokia Research Center
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Date: Thu, 16 Jan 2003 17:46:59 +0200
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Subject: [Tsvwg] Update on F-RTO draft
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Hi,

We submitted an update to F-RTO draft for dealing with spurious TCP 
retransmission timeouts. The draft can be found at
http://www.ietf.org/internet-drafts/draft-sarolahti-tsvwg-tcp-frto-03.txt

Based on the feedback in Atlanta, the focus is now on detecting 
spurious RTO, and the response related text is taken out of the 
algorithm description, although possible response alternatives are 
discussed in the end of the draft.

I'd like to have advise on how to proceed with the draft. Personally, I 
would like to see it become a tsvwg work item, which hopefully would 
attract a thorough review on it, and make its path towards experimental 
RFC easier. So I'm asking for input on would you approve it to become a 
WG item? If not, why not? Based on the earlier discussion on this list 
a few months ago, I got the feeling that generally folks were taking 
the idea of F-RTO quite positively, right?

Btw, F-RTO is included as an option in Linux 2.5 TCP implementation, so 
the Linux users can try it out by installing some of the recent Linux 
kernel versions. I can also give a pointer to separate patch for anyone 
interested.

- Pasi
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hi all!
  we r a group of student from pune institute of
computer technology,pune,india. we r working on ss7 as
our final year project.
 i've some basic questions
1. is it possible to have ss7 signal on a EPABX? does
telephone exchange provides it?
2. if possible then will it be possible that we can
get those ss7 signals on computer by inserting some
kind of card in ISA slot.
3. are such kind of cards available ?
4. how much such card will cost?
5. what kind of hardware is used at exchange to deal
with ss7 signals?
6. is there any kind of software tool available which
will simulate/
    generate ss7 signals as actual.
  if u've any material available will u please mail
it.
  thank u.
         anup   
 

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From mailnull@www1.ietf.org  Fri Jan 17 10:09:31 2003
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hi all!
  we r a group of student from pune institute of
computer technology,pune,india. we r working on ss7 as
our final year project.
 i've some basic questions
1. is it possible to have ss7 signal on a EPABX? does
telephone exchange provides it?
2. if possible then will it be possible that we can
get those ss7 signals on computer by inserting some
kind of card in ISA slot.
3. are such kind of cards available ?
4. how much such card will cost?
5. what kind of hardware is used at exchange to deal
with ss7 signals?
6. is there any kind of software tool available which
will simulate/
    generate ss7 signals as actual.
  if u've any material available will u please mail
it.
  thank u.
         amit
 

________________________________________________________________________
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       visit http://in.tv.yahoo.com
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From mailnull@www1.ietf.org  Fri Jan 17 10:09:55 2003
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hi all!
  we r a group of student from pune institute of
computer technology,pune,india. we r working on ss7 as
our final year project.
 i've some basic questions
1. is it possible to have ss7 signal on a EPABX? does
telephone exchange provides it?
2. if possible then will it be possible that we can
get those ss7 signals on computer by inserting some
kind of card in ISA slot.
3. are such kind of cards available ?
4. how much such card will cost?
5. what kind of hardware is used at exchange to deal
with ss7 signals?
6. is there any kind of software tool available which
will simulate/
    generate ss7 signals as actual.
  if u've any material available will u please mail
it.
  thank u.
         amit
 

________________________________________________________________________
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       visit http://in.tv.yahoo.com
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From mailnull@www1.ietf.org  Fri Jan 17 10:32:26 2003
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hi all!
  we r a group of student from pune institute of
computer technology,pune,india. we r working on ss7 as
our final year project.
 i've some basic questions
1. is it possible to have ss7 signal on a EPABX? does
telephone exchange provides it?
2. if possible then will it be possible that we can
get those ss7 signals on computer by inserting some
kind of card in ISA slot.
3. are such kind of cards available ?
4. how much such card will cost?
5. what kind of hardware is used at exchange to deal
with ss7 signals?
6. is there any kind of software tool available which
will simulate/
    generate ss7 signals as actual.
  if u've any material available will u please mail
it.
  thank u.
         amit


________________________________________________________________________
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       visit http://in.tv.yahoo.com
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From mailnull@www1.ietf.org  Tue Jan 21 10:53:27 2003
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From: Matt Mathis <mathis@psc.edu>
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        IPv6 MIB team <ipv6mib@ibr.cs.tu-bs.de>
cc: "Peter O'Neil" <poneil@ucar.edu>
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Subject: [Tsvwg] TCP ESTATS design team
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We are in the process of assembling a design team to update the TCP extended 
statistics MIB, draft-ietf-tsvwg-tcp-mib-extension-01.txt.    If you would like 
to help with this effort please send a note to Peter O'Neil or me.

The extended statistics MIB will endow TCP with substantial diagnostic 
capabilities.

We will have a new draft for review by the TSVWG in time for the ID cutoff.
This draft will incorporate the detailed per connection statistics from the 
web100 project and the good transport work that was done by the IPv6 team and 
later determined to be out of scope.

The IPv6 team is finishing up update to rfc2012.  Our document will be an
extension to it.

Thanks,
--MM--
-------------------------------------------
Matt Mathis      http://www.psc.edu/~mathis
Work:412.268.3319    Home/Cell:412.654.7529
-------------------------------------------


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From mailnull@www1.ietf.org  Tue Jan 21 16:27:04 2003
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To: "Andrei Gurtov" <andrei.gurtov@sonera.com>
cc: tsvwg@ietf.org, "Sally Floyd" <floyd@icir.org>
From: Sally Floyd <floyd@icir.org>
Subject: Re: [Tsvwg] Is a TFRC sender allowed to increase the send rate more than twice? 
Date: Tue, 21 Jan 2003 13:26:14 -0800
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Andrei -

>After a handover the receiver can set X_rcv to the new link's rate and p to
>some small none-zero value which will cause the TFRC sender to send at the
>higher rate immediately.
>Of couse this shouldn't be used to the sender over the shared Internet, but
>for servers connected by LAN it seems quite appropriate.

It would be good to have additional mechanisms in place to ensure
that something like this would *only* be used for servers connected
by a LAN, with no competing traffic on the LAN, and would not be
used over the shared Internet...

(Apologies for the late reply.)

- Sally
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Date: Tue, 21 Jan 2003 16:44:52 -0600
From: Qiaobing Xie <Qiaobing.Xie@Motorola.com>
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CC: "'Randall Stewart'" <randall@stewart.chicago.il.us>, tsvwg@ietf.org
Subject: Re: [Tsvwg] Concerns about SCTP
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...
> If one views it from a system perspective, a user may
> be running a file system over RDDP over SCTP.  If the
> file system fails to mount, the user would regard this
> a failure to interoperate.  However this failure can
> be prevented in multiple ways.
> 
> 1.  At the RDDP over SCTP adaption layer, specify that
>     if an endpoint specifies multiple IP addresses and
>     the association setup is aborted with a resource error,
>     the endpoint MUST retry the association with fewer
>     IP addresses. (The draft could also remain silent
>     on this and trust the good sense of the upper layers
>     to do this anyway, but that seems like asking
>     for interoperability woes unnecessarily.)

As you indicated above that you are essentially talking about system level
inter-operability (ie. what a user may see), "the good sense of the upper layers"
is not only desirable, but is fundamental. One can not expect system level
inter-operability without "the good sense" in system engineering at all the
layers.

> 
>     Although this works, I regard it as more of a
>     work around than a well designed negotiation.
>     An example of a well designed negotiation is
>     the mechanism use by SCTP to determine the number
>     of streams per association.

As a matter of fact, the SCTP design team had discussion on this and deliberately
choose to use *minimal negotiation* at the association setup. The reason is that,
nothing comes free and lower setup latency was considered more important to most
applications. Moreover, the stream negotiation mechanism (that you seem to agree
with) also need "the good sense of the upper layers" to work well - e.g., the
initiator side's upper layer will need to allow its SCTP stack to re-try with a
lowered number of outbound streams.

-Qiaobing
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Subject: Re: [Tsvwg] queries on draft-ietf-tsvwg-sctpimpguide-05.txt
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Hi, Armando,

"Armando L. Caro Jr." wrote:
....
> >       Originally, Max.Burst was applied only in the case of a SACK
> > acknowledging fast retransmitted DATA Chunks. However, Max.Burst is
> > now used always. Note that this is a modification to the general
> > congestion control algorithm.
> 
> Yes, this change was made to avoid bursts at any time. Max.Burst was left
> as a parameter, because we don't really know what an acceptable burst
> is... 4 was a reasonable number chosen earlier, so we stuck with it.

Do you have a reference to any analysis/discussion on the impact of this change to
the data throughput over a long, fat SCTP connection? (or similar analysis for
long fat TCP connections).

regards,
-Qiaobing
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> Trying to fulfill this work item during the last year, we have
> repeatedly asked for contributions to SCTP header compression, but
> we have not received a single response.

We published a first version of an SCTP compression profile in
draft-west-sctp-epic-00.txt. This draft is based on the formal header
compression notation (see draft-ietf-rohc-formal-notation-00.txt),
so we intend to update the SCTP draft once the formal notation draft
is finalised.

> Given that this work item was originally added without immediate
> market pull, and considering the lack of current interest in doing
> the work (or even having it done at all), we intend to remove SCTP
> header compression from the ROHC WG charter.  To keep it in the
> charter, justifications for developing SCTP header compression now
> would have to be presented, as well as industry support and
> commitments for doing the actual work.

This work item is important because, unlike for TCP, there is currently
no published scheme for SCTP compression. This will hinder the
deployment of SCTP, particularly in the mobile environment.

Regarding the industry support, we're currently working on SCTP
compression at the moment in the form of the notation draft. Once this
work is done we'll publish an updated version of our SCTP profile.

> Barring further input, we will send a request to the ADs by Jan 24,
> so please reply before this date if you believe SCTP header
> compression should be on the agenda now.

I believe that SCTP compression should remain on the ROHC charter.

Regards,

Richard
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From: Janardhan Iyengar <iyengar@mail.eecis.udel.edu>
To: tsvwg@ietf.org
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Subject: [Tsvwg] TCP SACK/FACK implementations on BSD
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Hi all,

We are doing some experiments using the OpenBSD implementation of TCP SACK
and FACK. Could anyone validate for us whether these are the docs that
these implementations conform to (or are closest to)?

OpenBSD TCP-SACK: A Conservative SACK-based Loss Recovery Algorithm for
TCP (draft-allman-tcp-sack-13.txt)

OpenBSD TCP-FACK: SIGCOMM'96 Paper "Forward Acknowledgment: Refining TCP
Congestion Control"

If not, then which are the implementations available right now that do?

Further, do the ns-2 implementations of TCP-SACK/FACK conform to the docs
above?

thanks and regards,
jana/armando

-----------------------------------------------------------------------
Janardhan R. Iyengar				   iyengar@cis.udel.edu
University of Delaware		       http://www.cis.udel.edu/~iyengar
-----------------------------------------------------------------------

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From: Ethan Blanton <eblanton@cs.ohiou.edu>
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Janardhan Iyengar spake unto us the following wisdom:
> We are doing some experiments using the OpenBSD implementation of TCP SACK
> and FACK. Could anyone validate for us whether these are the docs that
> these implementations conform to (or are closest to)?
>=20
> OpenBSD TCP-SACK: A Conservative SACK-based Loss Recovery Algorithm for
> TCP (draft-allman-tcp-sack-13.txt)

We have not heard anything about the OpenBSD people trying to
implement our algorithm.  That doesn't preclude that they might have
done so, but I suspect that their SACK support predates
draft-allman-tcp-sack-13.

> Further, do the ns-2 implementations of TCP-SACK/FACK conform to the docs
> above?

The ns-2 SACK algorithm is similar to draft-allman-tcp-sack-13 (and is
in fact the basis we worked from), but it is not the same.

Ethan

--=20
And if I claim to be a wise man / it surely means that I don't know.
                -- Kansas, "Carry on Wayward Son"

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Subject: [Tsvwg] RFC 3448 on TFRC: Protocol Specification
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3448

        Title:      TCP Friendly Rate Control (TFRC):
                    Protocol Specification
        Author(s):  M. Handley, S. Floyd, J. Padhye, J. Widmer
        Status:     Standards Track
        Date:       January 2003
        Mailbox:    mjh@icir.org, floyd@icir.org,
                    padhye@microsoft.com,
                    widmer@informatik.uni-mannheim.de
        Pages:      24
        Characters: 52657
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-tsvwg-tfrc-05.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3448.txt


This document specifies TCP-Friendly Rate Control (TFRC).  TFRC is a
congestion control mechanism for unicast flows
operating in a best-effort Internet environment.  It is
reasonably fair when competing for bandwidth with TCP flows,
but has a much lower variation of throughput over time
compared with TCP, making it more suitable for applications
such as telephony or streaming media where a relatively smooth
sending rate is of importance.

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

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
standardization state and status of this protocol.  Distribution
of this memo is unlimited.

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Submissions for Requests for Comments should be sent to
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Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

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Subject: [Tsvwg] Re: [dccp] DCCP and UDP Lite
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See in-line comments.

Michael Welzl wrote:
> 
> Hi everybody,
> 
> I just noticed the following sentences in the DCCP specification
> (page 11, section 5.1, "checksum length"):
> 
> "The checksum length field specifies what parts of the packet
> are covered by the checksum field. The checksum always covers
> at least the DCCP header, DCCP options, and a pseudoheader taken
> from the network-layer header (see below). If the checksum
> length field is zero, that is all the checksum covers."
> 
> The functionality of setting the checksum length field
> to zero closely resembles the capability of UDP Lite to
> receive erroneous payload packets and deliver them to the
> application.

The current rev of the ID (with IESG) for UDP-Lite is:
http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-udp-lite-01.txt

This contains advice on this topic, that had not been previously
been included in draft-00. There is nothing dangerous with the
transport layer employing partial checksums providing:

   a) The checksum always protect the appropriate headers.

   b) It is the application that requests partial protection,
      it is not something that the transport layer decides to 
      use/not use.

> 
> I would say that:
> 
> EITHER:
> 
> - a comment should be included that states that erroneous
>   payload MUST NOT be delivered to the applications and may
>   only be used for congestion control purposes (or, possibly
>   better, the sentence about values other than 0 or 15 being
>   experimental should be changed to "values other than 15").
> 
> OR:
> 
> - UDP Lite should be accepted upon acceptance of DCCP.
> 

Option (1) declares this experimental.

Option (2)
Since UDP-Lite -01 is now with the IESG, awaiting AD action,
UDP-Lite should be published soon. Perhaps then, we should
look again at whether/how DCCP should follow this?
 
> This has nothing to do with personal preference - it's just
> similar functionality that caused a lot of discussions in
> the case of UDP Lite ...
> 
> Cheers,
> Michael
> 
> _______________________________________________
> 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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Hi everybody,

I just noticed the following sentences in the DCCP specification
(page 11, section 5.1, "checksum length"):

"The checksum length field specifies what parts of the packet
are covered by the checksum field. The checksum always covers
at least the DCCP header, DCCP options, and a pseudoheader taken
from the network-layer header (see below). If the checksum
length field is zero, that is all the checksum covers."


The functionality of setting the checksum length field
to zero closely resembles the capability of UDP Lite to
receive erroneous payload packets and deliver them to the
application.


I would say that:

EITHER:

- a comment should be included that states that erroneous
  payload MUST NOT be delivered to the applications and may
  only be used for congestion control purposes (or, possibly
  better, the sentence about values other than 0 or 15 being
  experimental should be changed to "values other than 15").

OR:

- UDP Lite should be accepted upon acceptance of DCCP.


This has nothing to do with personal preference - it's just
similar functionality that caused a lot of discussions in
the case of UDP Lite ...

Cheers,
Michael

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From mailnull@www1.ietf.org  Fri Jan 31 12:14:57 2003
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Date: Fri, 31 Jan 2003 09:13:20 -0800 (PST)
From: Michael Kirkham <mikek@muonics.com>
To: Matt Mathis <mathis@psc.edu>
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Subject: [Tsvwg] Re: TCP ESTATS design team
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I thought I'd take a quick look through the MIB and make a few comments.
I assume this is a new MIB and not one that's currently deployed (given
the 'experimental' branch placement and the MODULE-IDENTITY's only
revision DESCRIPTION).  If I'm wrong, feel free to ignore these comments
as they aren't things that can be changed due to revision policy.

1. It is recommended (although not required) that objects with 'Counter'
types (Counter32, Counter64) denote plurality by ending in 's'.  Of the
the objects in this MIB with 'Counter' types, several do denote plurality,
although by ending in (for example) 'AcksOut' versus 'OutAcks', and many
do not.  You might consider reversing the suffix where plurality is
denoted, and modifying the descriptors of those that don't denote
plurality to do.  There are quite a few, so for the sake of brevety I
haven't listed them.

2. The indexing scheme of the various tables seems a little confusing to
me.  Most of the tables are indexed solely on tcpEStatsConnectIndex except,
oddly enough, the table that actually defines tcpEStatsConnectIndex:

   tcpEStatsConnectIdEntry  OBJECT-TYPE
       SYNTAX       TcpEStatsConnectIdEntry
       MAX-ACCESS   not-accessible
       STATUS       current
       DESCRIPTION
           "Connection ID table."
       INDEX   { tcpEStatsConnectLocalAddressType,
                 tcpEStatsConnectLocalAddress,
                 tcpEStatsConnectLocalPort,
                 tcpEStatsConnectRemAddressType,
                 tcpEStatsConnectRemAddress,
                 tcpEStatsConnectRemPort }
       ::= { tcpEStatsConnectIdTable 1 }

   TcpEStatsConnectIdEntry ::= SEQUENCE {
                 tcpEStatsConnectLocalAddressType  InetAddressType,
                 tcpEStatsConnectLocalAddress      InetAddress,
                 tcpEStatsConnectLocalPort         InetPortNumber,
                 tcpEStatsConnectRemAddressType    InetAddressType,
                 tcpEStatsConnectRemAddress        InetAddress,
                 tcpEStatsConnectRemPort           InetPortNumber,
                 tcpEStatsConnectIndex                Integer32
                 }

I'm guessing that the intent here was to make it easy to find the
tcpEStatsConnectIndex values for a particular 'local' address by
e.g. doing a get-next walk of tcpEStatsConnectIndex.<AddressType>.<Address>
and because all but tcpEStatsConnectIndex are not-accessible (and at
least one column of a table must be accessible).  But I don't see that
this provides much benefit to the other tables.

Why not have the other tables indexed by the same objects if they
are going to be indexed externally anyway?  Or, perhaps better,
have the tcpEStatsConnectIdEntry table map only a single type/address/port
triplet per row, and use a pair of tcpEStatsConnectIndex values
(one for each side of the connection) to index the other tables?
That way you locate statistics from either vantage point, that way
you could quickly locate statistics from either direction (what
defines 'local' versus 'remote' in the context of this MIB?)

Disadvantage: using this method to allow for statistics to be located from
either direction means doubling the data.  Though maybe the data is
already doubled, if the intent is for tcpEStatsConnectIdTable to contain
two rows for every connection, with different values for
tcpEStatsConnectIndex.  Is the intended meaning behind the DESCRIPTION for
tcpEStatsConnectIndex that there be two rows in TcpEStatsConnectIdTable
for each connection, and for each of those two rows to have the same
tcpEStatsConnectIndex value?

I believe this could use some clarification.

That's all for now.  I appologize if I've overlooked something in my
cursory viewing or if these issues have already been hashed out (I've only
just subscribed to this list, in response to your mibs list posting).

On Tue, 21 Jan 2003, Matt Mathis wrote:

> Date: Tue, 21 Jan 2003 15:15:21 -0500 (EST)
> From: Matt Mathis <mathis@psc.edu>
> To: mibs@ops.ietf.org
> Subject: TCP ESTATS design team
>
> We are in the process of assembling a design team to update the TCP extended
> statistics MIB, draft-ietf-tsvwg-tcp-mib-extension-02.txt.    If you would like
> to help with this effort please send a note to Peter O'Neil or me.
>
> The extended statistics MIB will endow TCP with substantial diagnostic
> capabilities.
>
> We will have a new draft for review by the TSVWG in time for the ID cutoff.
> This draft will incorporate the detailed per connection statistics from the
> web100 project and the good transport work that was done by the IPv6 team and
> later determined to be out of scope.
>
> The IPv6 team is finishing up update to rfc2012.  Our document will be an
> extension to it.
>
> Thanks,
> --MM--
> -------------------------------------------
> Matt Mathis      http://www.psc.edu/~mathis
> Work:412.268.3319    Home/Cell:412.654.7529
> -------------------------------------------
>
>
>
>
>

--
Michael Kirkham
www.muonics.com

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In-reply-to: Your message of "31 Jan 2003 09:34:08 +0100."
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Subject: [Tsvwg] Re: [dccp] DCCP and UDP Lite
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>Hi everybody,
>
>I just noticed the following sentences in the DCCP specification
>(page 11, section 5.1, "checksum length"):
>
>"The checksum length field specifies what parts of the packet
>are covered by the checksum field. The checksum always covers
>at least the DCCP header, DCCP options, and a pseudoheader taken
>from the network-layer header (see below). If the checksum
>length field is zero, that is all the checksum covers."
>
>
>The functionality of setting the checksum length field
>to zero closely resembles the capability of UDP Lite to
>receive erroneous payload packets and deliver them to the
>application.
>
>
>I would say that:
>
>EITHER:
>
>- a comment should be included that states that erroneous
>  payload MUST NOT be delivered to the applications and may
>  only be used for congestion control purposes (or, possibly
>  better, the sentence about values other than 0 or 15 being
>  experimental should be changed to "values other than 15").
>
>OR:
>
>- UDP Lite should be accepted upon acceptance of DCCP.
>
>
>This has nothing to do with personal preference - it's just
>similar functionality that caused a lot of discussions in
>the case of UDP Lite ...

It's not just similar - it was entirely inspired by UDP Lite.  We're
still rather undecided as to whether this is a useful function.  It
can easily be removed if there's pushback, but so far the comments
we've received have mostly been in favour.

It's clearly not a dangerous thing for a transport to allow this.
It's just rather pointless unless there are link-layers that might
take advantage of it, and applications that can make use of it.  

In our case it comes down to whether we have a better use for those
four bits.  Currently we think it's worth spending those 4 bits in
this way, given that the unreliable applications DCCP is primarily
intended to support are the sort of applications that might be able to
use this functionality.

 - Mark
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From: Matt Mathis <mathis@psc.edu>
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Thanks for the comments.

About "denote plurality", yes, we got it wrong.   The current document follows 
a prototype which doesn't use SNMP at all.  We were unaware of the naming 
convention and I am sure we missed others too.  This will get fixed.

About indexing - we know we have a problem here, but I don't know enough about 
SNMP (I am a TCP guy) to grasp the subtleties.  There are two main issues:

1) We want to support a fast way to poll on a known connection. The "generic"
approach (again, used by the non-SNMP prototype) is to have a general (but
slow) way to search a table to find a specific 4-tuple.  This yields 
a "Connection ID" which is an opaque int.  The Connection ID can then be used 
for fast polling on any variables in the logical row.

2) We want to be able to support sparse tables to reduce the memory load on a 
busy server.   The current draft has controls per group of columns 
(which are actually separate smaller tables.)   We are considering putting in 
per-connection controls as well.

I will be defering to people with more MIB expertics on the details.

BTW there is a document status page at http://www.web100.org/mib

Thanks again for taking the time to look at the document,
--MM--
-------------------------------------------
Matt Mathis      http://www.psc.edu/~mathis
Work:412.268.3319    Home/Cell:412.654.7529
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