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From: "Jozef Babiarz" <babiarz@nortelnetworks.com>
To: Mpierce1@aol.com, Fil.Dickinson@optus.com.au, tsvwg@ietf.org
Subject: RE: [Tsvwg] draft-baker-diffserv-basic-classes-02
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In a message dated May 27, 2004 Mpierce1@aol.com <mailto:Mpierce1@aol.com>
writes:

[Mike]

I believe this remains as one of the major open questions for the
application of SIP or H.323 in a large, managed network. Some comments: 

I've heard of speech clipping (on answer), but I don't know what
"ring-clipping" is. I presume you are referring to call setup delay (or
delay to ring). The issue is signaling delay in general (call setup and
release), but speech clipping has always been the most serious impact of
delay. 

It seems to me that this issue is really not related to "interfacing to a
TDM telephony switch", but occurs equally within a fully IP environment. 



[Joe]

Ring-clipping can occur if SS7 signaling path from the softswitch to the TDM
CO is much faster than the signaling path from softswitch to IP gateway that
interfaces to the TDM CO were the ringing tone is generated. Speech-clipping
can occur if the signaling path from the call answered phone to the
softswitch and back to the IP gateway that enables the audio path takes
longer then for the person answering the phone to say hello. 

 

[Mike]

No matter what service class is used for signaling (above, below, or same as
speech bearer) there must be guaranteed bandwidth for the expected signaling
traffic to provide sufficient low latency and almost zero discard rate
independent from the amount of speech bearer traffic. In effect, if
signaling is in a separate class and is guaranteed, say 5% of the bandwidth
on an outgoing link and speech bearer (EF) is guaranteeed 50%, then it isn't
really necessary to think of one being above or below or the same as the
other. The forwarding algorithm guarantees a minimum bandwidth to both in
what is effectively "time-division multiplexing", but we don't call it that
since it is not based on strict timing. The result is that the signaling
does not experience "higher latency", but probably has "higher delay
variation", mostly due to the varying packet sizes. 



[Joe]

Agree. Putting signaling traffic into a distinct class and engineering that
class so that queuing dose not occur does work. In next draft (03) I will
total separate telephony signaling from telephony bearer traffic.

 

Regards,

Joe Babiarz 
 


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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial =
PTSIZE=3D10><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>In a message =
dated </span></font><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;
 color:navy'>May 27, 2004</span></font><font size=3D2 color=3Dnavy =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:navy'> <a
href=3D"mailto:Mpierce1@aol.com">Mpierce1@aol.com</a> =
writes:</span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>[Mike]</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I believe this remains as one of =
the major
open questions for the application of SIP or H.323 in a large, managed =
network.
Some comments: <br>
<br>
I've heard of speech clipping (on answer), but I don't know what
&quot;ring-clipping&quot; is. I presume you are referring to call setup =
delay
(or delay to ring). The issue is signaling delay in general (call setup =
and
release), but speech clipping has always been the most serious impact =
of delay.
<br>
<br>
It seems to me that this issue is really not related to =
&quot;interfacing to a
TDM telephony switch&quot;, but occurs equally within a fully IP =
environment. <br>
<br>
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>[Joe]</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Ring-clipping can occur if SS7 =
signaling path
from the softswitch to the TDM CO is much faster than the signaling =
path from softswitch
to IP gateway that interfaces to the TDM CO were the ringing tone is =
generated.
Speech-clipping can occur if the signaling path from the call answered =
phone to
the softswitch and back to the IP gateway that enables the audio path =
takes longer
then for the person answering the phone to say hello. =
</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>[Mike]</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>No matter what service class is =
used for
signaling (above, below, or same as speech bearer) there must be =
guaranteed
bandwidth for the expected signaling traffic to provide sufficient low =
latency
and almost zero discard rate independent from the amount of speech =
bearer
traffic. In effect, if signaling is in a separate class and is =
guaranteed, say
5% of the bandwidth on an outgoing link and speech bearer (EF) is =
guaranteeed
50%, then it isn't really necessary to think of one being above or =
below or the
same as the other. The forwarding algorithm guarantees a minimum =
bandwidth to
both in what is effectively &quot;time-division multiplexing&quot;, but =
we
don't call it that since it is not based on strict timing. The result =
is that
the signaling does not experience &quot;higher latency&quot;, but =
probably has
&quot;higher delay variation&quot;, mostly due to the varying packet =
sizes. </span></font><br>
<br>
</p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>[Joe]</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Agree. Putting signaling traffic =
into a distinct
class and engineering that class so that queuing dose not occur does =
work. In
next draft (03) I will total separate telephony signaling from =
telephony bearer
traffic.</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Regards,</span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
 12.0pt'>Joe Babiarz</span></font><font color=3Dnavy><span =
style=3D'color:navy'> <br>
</span></font>=A0</p>

</div>

</div>

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From: "Jozef Babiarz" <babiarz@nortelnetworks.com>
To: Fil Dickinson <Fil.Dickinson@optus.com.au>, tsvwg@ietf.org
Subject: RE: [Tsvwg] draft-baker-diffserv-basic-classes-02
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In next version 03 of the draft will make the change so that telephony
signaling and telephony bearer traffic are in different classes. Thanks for
your comments.

Regards, Joe.

-----Original Message-----
From: Fil Dickinson [mailto:Fil.Dickinson@optus.com.au] 
Sent: Tuesday, May 25, 2004 12:27 AM
To: tsvwg@ietf.org
Subject: [Tsvwg] draft-baker-diffserv-basic-classes-02

My understanding of this draft is that it proposes the Voice signalling and
Voice Bearer are transported in the same class with the exception being
"signaling flows between high capacity telephony call servers or soft
switches". 
I am concerned with this. During testing of our multi-vendor converged
network I have seen that oversubscribed voice bearer traffic can disrupt the
voice signalling traffic which results in abnormal operation of all voice
calls [re-registration (outage) of IP Phones]. I would like to see this
addressed by a draft that mandates All Voice Signalling and Voice Bearer
traffic operate in distinct classes. Ie, a Telephony Signalling Service
Class and a Telephony Bearer Service Class.

_______________________________________________
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charset=3DISO-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2656.31">
<TITLE>RE: [Tsvwg] draft-baker-diffserv-basic-classes-02</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>In next version 03 of the draft will make the change =
so that telephony signaling and telephony bearer traffic are in =
different classes. Thanks for your comments.</FONT></P>

<P><FONT SIZE=3D2>Regards, Joe.</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Fil Dickinson [<A =
HREF=3D"mailto:Fil.Dickinson@optus.com.au">mailto:Fil.Dickinson@optus.co=
m.au</A>] </FONT>
<BR><FONT SIZE=3D2>Sent: Tuesday, May 25, 2004 12:27 AM</FONT>
<BR><FONT SIZE=3D2>To: tsvwg@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: [Tsvwg] =
draft-baker-diffserv-basic-classes-02</FONT>
</P>

<P><FONT SIZE=3D2>My understanding of this draft is that it proposes =
the Voice signalling and</FONT>
<BR><FONT SIZE=3D2>Voice Bearer are transported in the same class with =
the exception being</FONT>
<BR><FONT SIZE=3D2>&quot;signaling flows between high capacity =
telephony call servers or soft</FONT>
<BR><FONT SIZE=3D2>switches&quot;. </FONT>
<BR><FONT SIZE=3D2>I am concerned with this. During testing of our =
multi-vendor converged</FONT>
<BR><FONT SIZE=3D2>network I have seen that oversubscribed voice bearer =
traffic can disrupt the</FONT>
<BR><FONT SIZE=3D2>voice signalling traffic which results in abnormal =
operation of all voice</FONT>
<BR><FONT SIZE=3D2>calls [re-registration (outage) of IP Phones]. I =
would like to see this</FONT>
<BR><FONT SIZE=3D2>addressed by a draft that mandates All Voice =
Signalling and Voice Bearer</FONT>
<BR><FONT SIZE=3D2>traffic operate in distinct classes. Ie, a Telephony =
Signalling Service</FONT>
<BR><FONT SIZE=3D2>Class and a Telephony Bearer Service Class.</FONT>
</P>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>tsvwg mailing list</FONT>
<BR><FONT SIZE=3D2>tsvwg@ietf.org</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/tsvwg" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/tsvwg</A></FONT=
>
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From: Meyknecht Bernward <bernward.meyknecht@siemens.com>
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Subject: [Tsvwg] what is a  'sufficiently'  long idle period ?
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Hi all,

in SCTP RFC and Implem. Guide there's a sentence telling me how to initialize the 
cwnd after a 'sufficiently' long idle period.

Does anyone have a proposal what could be a good value for 'sufficiently' ?
5, 10, 20 times RTO ? ....

Text from ch. 2.30 / implem. guide:
--------- New text: (Section 7.2.1) --------- o 
The initial cwnd before DATA transmission or after a sufficiently long idle period 
MUST be set to min (4*MTU, max (2*MTU, 4380 bytes)). 

--- --- ---

And how does this match ch. 2.14 from implem guide, where I'm told to set cwnd 
to 2*MTU after one RTO, which could be less than the value from above.
Does the formula from 2.30 replace the one from 2.14?

Text from ch. 2.14 / implem. guide:
--------- New text: (Section 7.2.1) --------- o 
When the association does not transmit data on a given transport address within an RTO, 
the cwnd of the transport address SHOULD be adjusted to 2*MTU. 

Maybe someone has some idea...


Best regards,

Bernward



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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Subject: [Tsvwg] Registration for 7th SCTP Interop is open and ends at June
	18th
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Dear all,

the 7th SCTP interop will be held at University of Applied Sciences,=20
M=FCnster, from
July 25th to July 30th 2004.

Tested specifications include RFC 2960, RFC 3309, RFC 3758, the=20
Implementers Guide,
ADD-IP Extension, PKTDRPREP Extension and the SCTP MIB.

In addition to the 'usual' setup there will be a Gigabit Ethernet for
performance testing.

You can register until June 18th and find more information at
http://sctp.fh-muenster.de

If you have any further questions please do not hesitate to contact me=20=

via
tuexen@fh-muenster.de

Happy SCTPing
Michael


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To: tsvwg@ietf.org
From: Sally Floyd <floyd@icir.org>
Date: Mon, 07 Jun 2004 12:17:32 -0700
Cc: Mark Handley <mjh@icir.org>, Andrei Gurtov <gurtov@cs.helsinki.fi>,
        Jitu Padhye <padhye@microsoft.com>
Subject: [Tsvwg] Errata for RFC 3448, "TCP Friendly Rate Control (TFRC)"
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This is a notice that I am sending in an Errata to the
RFC Editor for RFC 3448.  This is from errata that has been
on the TFRC web page for over a year, but that I have never
managed to send in the RFC Editor for the "RFC Errata" web page.   
FYI.

- Sally
http://www.icir.org/tfrc/

The RFC 3448 Errata:


RFC 3448, TCP Friendly Rate Control (TFRC): Protocol Specification


Reported By: Joerg Widmer
Date: March 4, 2003

* Nofeedback timeout:  Correct an inconsistency in the document.

In Section 4:

"If the sender does not receive a feedback report for two round
trip times, it cuts its sending rate in half."

Should be: 

"If the sender does not receive a feedback report for four round
trip times, it cuts its sending rate in half."


Reported By: Mark Handley, from Wim Heirman
Date: March 13, 2003

In Section 4.4, item (2):

      If the nofeedback timer expires when the sender does not yet have
      an RTT sample, and has not yet received any feedback from the
      receiver,

Should be:

      If the nofeedback timer expires when the sender does not yet have
      an RTT sample, and has not yet received any feedback from the
      receiver, or when p == 0,



Reported By: Sally Floyd, from Michele R.
Date: April 29, 2003

* In Section 5.5, when initializing DF, initialize from 0 to n, not from 
* 1 to n:

In Section 5.5:

      for (i = 1 to n) {
        DF_i = 1;
      }

Should be: 

      for (i = 0 to n) {
        DF_i = 1;
      }



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To: Meyknecht Bernward <bernward.meyknecht@siemens.com>
Subject: Re: [Tsvwg] what is a  'sufficiently'  long idle period ?
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Bernward:

IMO you really do not need to do the cwnd reduction if
you are applying a burst limit... this whole reduction thing (I think)
needs to be revisited with respect to burst limit and of course
how long it takes to degrade the cwnd...

R

Meyknecht Bernward wrote:

>Hi all,
>
>in SCTP RFC and Implem. Guide there's a sentence telling me how to initialize the 
>cwnd after a 'sufficiently' long idle period.
>
>Does anyone have a proposal what could be a good value for 'sufficiently' ?
>5, 10, 20 times RTO ? ....
>
>Text from ch. 2.30 / implem. guide:
>--------- New text: (Section 7.2.1) --------- o 
>The initial cwnd before DATA transmission or after a sufficiently long idle period 
>MUST be set to min (4*MTU, max (2*MTU, 4380 bytes)). 
>
>--- --- ---
>
>And how does this match ch. 2.14 from implem guide, where I'm told to set cwnd 
>to 2*MTU after one RTO, which could be less than the value from above.
>Does the formula from 2.30 replace the one from 2.14?
>
>Text from ch. 2.14 / implem. guide:
>--------- New text: (Section 7.2.1) --------- o 
>When the association does not transmit data on a given transport address within an RTO, 
>the cwnd of the transport address SHOULD be adjusted to 2*MTU. 
>
>Maybe someone has some idea...
>
>
>Best regards,
>
>Bernward
>
>
>
>_______________________________________________
>tsvwg mailing list
>tsvwg@ietf.org
>https://www1.ietf.org/mailman/listinfo/tsvwg
>
>
>  
>


-- 
Randall R. Stewart
815-477-2127 (office)
815-342-5222 (cell phone)



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To: "'Randall R. Stewart (home)'" <randall@stewart.chicago.il.us>,
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Subject: AW: [Tsvwg] what is a  'sufficiently'  long idle period ?
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Hi Randy,

thanks for your reply, yes these cwnd reduction things look strange
for me in some aspects...

As long as my burst limit allows me to send as much data as the actual 
reduced cwnd would allow, in this case, I would be content with 
skipping all these cwnd reduction things, and if you set the 
cwnd to small 2*mtu, this will be the case.

Anyhow I don't like the immediate reduction to 2*mtu after a single 
rto, reduction to min (4*MTU, max (2*MTU, 4380 bytes)) looks 
more logical to me, and... without having a closer look.. I still 
like the idea of setting cwnd to cwnd/2 per rto til 
min (4*MTU, max (2*MTU, 4380 bytes)) is reached. For the case where
a big cwnd/2 still allows to send more than my burst limit 
would allow.

Best regards,
Bernward



-----Ursprungliche Nachricht-----
Von: Randall R. Stewart (home) [mailto:randall@stewart.chicago.il.us]
Gesendet: Dienstag, 8. Juni 2004 14:28
An: Meyknecht Bernward
Cc: tsvwg@ietf.org
Betreff: Re: [Tsvwg] what is a 'sufficiently' long idle period ?


Bernward:

IMO you really do not need to do the cwnd reduction if
you are applying a burst limit... this whole reduction thing (I think)
needs to be revisited with respect to burst limit and of course
how long it takes to degrade the cwnd...

R

Meyknecht Bernward wrote:

>Hi all,
>
>in SCTP RFC and Implem. Guide there's a sentence telling me how to initialize the 
>cwnd after a 'sufficiently' long idle period.
>
>Does anyone have a proposal what could be a good value for 'sufficiently' ?
>5, 10, 20 times RTO ? ....
>
>Text from ch. 2.30 / implem. guide:
>--------- New text: (Section 7.2.1) --------- o 
>The initial cwnd before DATA transmission or after a sufficiently long idle period 
>MUST be set to min (4*MTU, max (2*MTU, 4380 bytes)). 
>
>--- --- ---
>
>And how does this match ch. 2.14 from implem guide, where I'm told to set cwnd 
>to 2*MTU after one RTO, which could be less than the value from above.
>Does the formula from 2.30 replace the one from 2.14?
>
>Text from ch. 2.14 / implem. guide:
>--------- New text: (Section 7.2.1) --------- o 
>When the association does not transmit data on a given transport address within an RTO, 
>the cwnd of the transport address SHOULD be adjusted to 2*MTU. 
>
>Maybe someone has some idea...
>
>
>Best regards,
>
>Bernward
>
>
>
>_______________________________________________
>tsvwg mailing list
>tsvwg@ietf.org
>https://www1.ietf.org/mailman/listinfo/tsvwg
>
>
>  
>


-- 
Randall R. Stewart
815-477-2127 (office)
815-342-5222 (cell phone)


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


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--NextPart

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

	Title		: Stream Control Transmission Protocol (SCTP) Dynamic 
			  Address Reconfiguration
	Author(s)	: R. Stewart, et al.
	Filename	: draft-ietf-tsvwg-addip-sctp-09.txt
	Pages		: 37
	Date		: 2004-6-10
	
This document describes extensions to the Stream Control Transmission
Protocol (SCTP) [RFC2960] that provides a method to reconfigure IP
address information on an existing association.

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

	Title		: Stream Control Transmission Protocol (SCTP) Dynamic 
			  Address Reconfiguration
	Author(s)	: R. Stewart, et al.
	Filename	: draft-ietf-tsvwg-addip-sctp-09.txt
	Pages		: 37
	Date		: 2004-6-10
	
This document describes extensions to the Stream Control Transmission
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Transport Area Working Group Working Group of the IETF.

	Title		: Stream Control Transmission Protocol (SCTP) Dynamic 
			  Address Reconfiguration
	Author(s)	: R. Stewart, et al.
	Filename	: draft-ietf-tsvwg-addip-sctp-09.txt
	Pages		: 37
	Date		: 2004-6-10
	
This document describes extensions to the Stream Control Transmission
Protocol (SCTP) [RFC2960] that provides a method to reconfigure IP
address information on an existing association.

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From: Straub Gilles <gilles.straub@thomson.net>
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Subject: [Tsvwg] Are IP routers transparent to UDP Lite ?
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Hello All,


I am trying to evaluate UDP Lite technique. I had a look to the 2003 draft
on this matter.
I had a question regarding UDB checksum processing in IP routers : although,
in principle a transport layer is processed in end to end (in source and
destination), I wanted to check that no CRC check is performed in IP routers
: is anybody aware of IP routers where UDP or TCP checksum computation
(check & discard) is or can be enabled ? 

If UDP checksum computation is always done in the host only, then UDP lite
applications can be developped on existing internet, otherwise, this would
require router modifications.
Thanks in advance for any feedback I would get on this topic

Best Regards


Gilles STRAUB
gilles.straub@thomson.net
THOMSON


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Straub Gilles wrote:

> Hello All,
> 
> 
> I am trying to evaluate UDP Lite technique. I had a look to the 2003 draft
> on this matter.
> I had a question regarding UDB checksum processing in IP routers : although,
> in principle a transport layer is processed in end to end (in source and
> destination), I wanted to check that no CRC check is performed in IP routers
> : is anybody aware of IP routers where UDP or TCP checksum computation
> (check & discard) is or can be enabled ? 

That would kill IPsec'd packets... a protection that would defeat IPsec 
isn't very protective ;-(

> If UDP checksum computation is always done in the host only, then UDP lite
> applications can be developped on existing internet, otherwise, this would
> require router modifications.

Any router that performed that sort of delayed fill-in or check is 
basically performing as a host anyway. It would need to be updated as 
any host would, if the transport protocols supported changed.

Joe

> Thanks in advance for any feedback I would get on this topic
> 
> Best Regards
> 
> 
> Gilles STRAUB
> gilles.straub@thomson.net
> THOMSON
> 
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg

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To: "'lksctp-developers@lists.sourceforge.net'"
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Subject: [Tsvwg] Registration for 7th SCTP Interop ends at June 18th
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Dear all,

the 7th SCTP interop will be held at University of Applied Sciences,=20
M=FCnster, from
July 25th to July 30th 2004.

Tested specifications include RFC 2960, RFC 3309, RFC 3758, the=20
Implementers Guide,
ADD-IP Extension, PKTDRPREP Extension and the SCTP MIB.

In addition to the 'usual' setup there will be a Gigabit Ethernet for
performance testing.

You can register until June 18th and find more information at
http://sctp.fh-muenster.de

If you have any further questions please do not hesitate to contact me=20=

via
tuexen@fh-muenster.de

Happy SCTPing
Michael


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Two devices that do recompute transport checksums are:

     Network Address Translation (NAT) Midboxes
     and gateways that do some sort of connection splitting

I don't count these as IP routers, although these functions can be 
performed on router hardware platforms... Note that UDP-Lite has a 
different IP protocol type, so these devices will anyway treat the IP 
packets carrying UDP-Lite in a different manner to that of TCP, UDP, etc.

A "normal" router should pass transport checksums as-is.

Gorry Dairhurst


Straub Gilles wrote:

> Hello All,
> 
> 
> I am trying to evaluate UDP Lite technique. I had a look to the 2003 draft
> on this matter.
> I had a question regarding UDB checksum processing in IP routers : although,
> in principle a transport layer is processed in end to end (in source and
> destination), I wanted to check that no CRC check is performed in IP routers
> : is anybody aware of IP routers where UDP or TCP checksum computation
> (check & discard) is or can be enabled ? 
> 
> If UDP checksum computation is always done in the host only, then UDP lite
> applications can be developped on existing internet, otherwise, this would
> require router modifications.
> Thanks in advance for any feedback I would get on this topic
> 
> Best Regards
> 
> 
> Gilles STRAUB
> gilles.straub@thomson.net
> THOMSON
> 
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 
> 


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From: "Lv Qian" <tcp_lq@163.com>
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Subject: [Tsvwg] Inquire problem about TFRC 
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tcpmHi, all,
I met a problem when doing simulation on TFRC these days. I am using ns-2.26, Redhat 9.0, and all NS2 valication tests are passed.
I want to test the friendliness among some TFRC flows in the bottleneck link. The topology is very simple(the parameters such as bandwidth, queue is listed in the script below).
                       s0        k0
                        |        |
                s1 --- r1 ------ r2 --- k1
                        |        |
	                   s2        k2
However, when I use awk to get the throughput data of each flow, I found that when time < 300, the simulation result is satisfactory; however, when time >300, one flow increases abruptly while other flows descend to nearly zero. This is abnormal. I tested other scripts, including the test-suite-friendly.tcl/two-friendly script in ns-2.26/tcl/test. When I changed the relevant bandwidth parameters and simulation time to nearly 1000s, I got the similar result.  

The tfrcsim.tcl script is as follows:
########################################################
set flows 4
set queue  RED
set rate 15
#set ecn 1
set ecn 0
set discount 0

set ns [new Simulator]

proc stop {} {
       global ns s1_tf
       $ns flush-trace
       close $s1_tf
       exit 0
}

set node_(r1) [$ns node]
set node_(r2) [$ns node]
Queue/RED set gentle_ true
if {$ecn==1} {
  Queue/RED set setbit_ true
  Agent/TCP set ecn_ 1
}

$ns duplex-link $node_(r1) $node_(r2) [set rate]Mb 20ms $queue
$ns queue-limit $node_(r1) $node_(r2) 50
$ns queue-limit $node_(r2) $node_(r1) 50

Agent/TFRC set packetSize_ 1000
Agent/TFRCSink set discount_ $discount
Agent/TFRCSink set printLoss_ 1
Agent/TFRCSink set smooth_ 1
Agent/TFRC set printStatus_ 0
Agent/TFRC set df_ 0.95
Agent/TFRC set ca_ 1

for {set i 0} {$i < 3} {incr i} {
    set node_(s$i) [$ns node]
    set node_(k$i) [$ns node]
    $ns duplex-link $node_(s$i) $node_(r1) 100Mb 2ms DropTail
    $ns duplex-link $node_(k$i) $node_(r2) 100Mb 2ms DropTail
    set tf$i [$ns create-connection TFRC $node_(s$i) TFRCSink $node_(k$i) 0]
    set sec [expr $i/10]
    set frac [expr $i%10]
    set starttime $sec.$frac

    $ns at $starttime "[set tf$i] start"
    $ns at 1000 "[set tf$i] stop"
}

set s1_tf [open out.tr w]
$ns trace-queue $node_(r1) $node_(r2) $s1_tf

$ns at 1005 "stop"

$ns run
######################################################

Is there something wrong in my script? Or is there some bugs in the TFRC code in NS2? I am suffered for this problem these days. Can any one help me to solve it? Thank you!



Regards,
Lv Qian (SCE, NTU)
2004-06-17



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To: "Lv Qian" <tcp_lq@163.com>
From: Sally Floyd <floyd@icir.org>
Date: Thu, 17 Jun 2004 13:51:09 -0700
Cc: tcpm <tcpm@ietf.org>, tsvwg <tsvwg@ietf.org>
Subject: [Tsvwg] Re: [tcpm] Inquire problem about TFRC 
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This is a classic case of phase effects.  You could look at
the 1992 paper "On Traffic Phase Effects in Packet-Switched Gateways",
available from "http://www.icir.org/floyd/papers.html".

One solution is to add some randomization in the form of
"overhead_", which dithers TFRC's outgoing packets:
  Agent/TFRC set overhead_ 1

>From tfrc.cc:
                next = size_/xrate;
                //
                // randomize between next*(1 +/- overhead_)
                //
                next = next*(2*overhead_*Random::uniform()-overhead_+1);

Other solutions are to use more realistic scenarios, including
competing TCP traffic, with some web traffic, and reverse-path
traffic (to include small Ack packets in the forward path).

I am enclosing a modified script below.  Instead of using a long
trace file, I modified the script to have NS print out the packet
information that was wanted.

The results can be displayed as follows:

./ns tfrcsim.tcl > data
awk '{if ($4==0) {print $2, $6;}}' data > data.0
awk '{if ($4==1) {print $2, $6;}}' data > data.1
awk '{if ($4==2) {print $2, $6;}}' data > data.2
xgraph -bb -tk -x time -y packets data.0 data.1 data.2 &

There is still a brief period with phase effects, but it is much more 
moderate than before.

You also might be interested in the paper "Internet Research Needs
Better Models", available from
"http://www.icir.org/models/bettermodels.html.

- Sally
http://www.icir.org/floyd/

====================================================================
tfrcsim.tcl (the modified version)
====================================================================
set flows 4
set queue  RED
set rate 15
#set ecn 1
set ecn 0
set discount 0

set ns [new Simulator]

proc stop {} {
       global ns 
       exit 0
}

proc printpkts { time flowid flow } {
       puts "time $time tfrc $flowid data_pkts_sent [$flow set ndatapack_]"
}

set node_(r1) [$ns node]
set node_(r2) [$ns node]
Queue/RED set gentle_ true
if {$ecn==1} {
  Queue/RED set setbit_ true
  Agent/TCP set ecn_ 1
}

$ns duplex-link $node_(r1) $node_(r2) [set rate]Mb 20ms $queue
$ns queue-limit $node_(r1) $node_(r2) 50
$ns queue-limit $node_(r2) $node_(r1) 50

Agent/TFRC set packetSize_ 1000
Agent/TFRCSink set discount_ $discount
#Agent/TFRCSink set printLoss_ 1
Agent/TFRCSink set printLoss_ 0
Agent/TFRCSink set smooth_ 1
Agent/TFRC set printStatus_ 0
Agent/TFRC set df_ 0.95
Agent/TFRC set ca_ 1

#Agent/TFRC set overhead_ 0 
Agent/TFRC set overhead_ 1

for {set i 0} {$i < 3} {incr i} {
    set node_(s$i) [$ns node]
    set node_(k$i) [$ns node]
    $ns duplex-link $node_(s$i) $node_(r1) 100Mb 2ms DropTail
    $ns duplex-link $node_(k$i) $node_(r2) 100Mb 2ms DropTail
    set tf$i [$ns create-connection TFRC $node_(s$i) TFRCSink $node_(k$i) 0]
    set sec [expr $i/10]
    set frac [expr $i%10]
    set starttime $sec.$frac

    $ns at $starttime "[set tf$i] start"
    for {set time 0} {$time <= 1000} {incr time} {
   	 $ns at $time "printpkts $time $i [set tf$i]"
    }
    $ns at 1000 "[set tf$i] stop"
}

$ns at 1005 "stop"

$ns run

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