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From: "vishal verma" <vishalve@rediffmail.com>
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Subject: [rohc] SigComp related query
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Hi ,
   I have following questions regarding SigComp

1. How is SigComp message differs from normal SIP text based 
message?
2. When using static dictionary based compression ,how will text
content and a keyword offset of dictionary be differentiated ?


regards, vishal




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ROHCers,

Last year in Minneapolis, an individual draft for a UDP-Lite profile
was discussed, and we also agreed to make it a WG draft. However, the
work got delayed for various reasons, and the author also wanted to
make significant changes to the document. Therefore, we decided the
updated draft should stay as an individual draft in this version,
although the intention is to soon make it a WG document, and the work
should hereafter continue on the list.

Please express your opinions about this draft, especially opinions 
about using this document as the starting point for the ROHC UDP-Lite
profile development.

Thanks!
/Lars-Erik



-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: den 3 februari 2003 12:44
Subject: I-D ACTION:draft-pelletier-rohc-udplite-01.txt


A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: RObust Header Compression (ROHC):Profiles for UDP Lite
	Author(s)	: G. Pelletier
	Filename	: draft-pelletier-rohc-udplite-01.txt
	Pages		: 19
	Date		: 2003-1-31
	
This document defines ROHC (Robust Header Compression) profiles for
compression of RTP/UDP-Lite/IP packets (Real-Time Transport Protocol,
User Datagram Protocol Lite, Internet Protocol) and UDP-Lite/IP.
These profiles are defined based on their differences with the
profiles specified in [RFC-3095] for UDP [RFC-768].
Although both transport protocols are very similar, ROHC profiles
must be defined separately for robust compression of UDP-Lite headers
because it does not share protocol identifier with UDP. Also, the
UDP-Lite Checksum Coverage field does not share the semantics of the
corresponding UDP Length field and as a consequence cannot always be
inferred anymore.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-pelletier-rohc-udplite-01.txt

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	 <Ghyslain.Pelletier@epl.ericsson.se>
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Subject: RE: [rohc] FW: I-D ACTION:draft-pelletier-rohc-udplite-01.txt
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Hi,

The updated draft is quite a bit longer than the old one - a list of
what's changed since the previous version would be much appreciated!

One point I noticed is that the new draft supports the compression
of multiple IP headers. Is this something that's particularly useful
for UDP-Lite (as compared with vanilla UDP), or just a feature that
would be good to have in general?

Thanks,

Richard

> -----Original Message-----
> From: Lars-Erik Jonsson (EAB) 
> [mailto:Lars-Erik.Jonsson@epl.ericsson.se]
> Sent: Monday, February 03, 2003 1:30 PM
> To: Rohc (E-mail)
> Subject: [rohc] FW: I-D ACTION:draft-pelletier-rohc-udplite-01.txt
> 
> 
> ROHCers,
> 
> Last year in Minneapolis, an individual draft for a UDP-Lite profile
> was discussed, and we also agreed to make it a WG draft. However, the
> work got delayed for various reasons, and the author also wanted to
> make significant changes to the document. Therefore, we decided the
> updated draft should stay as an individual draft in this version,
> although the intention is to soon make it a WG document, and the work
> should hereafter continue on the list.
> 
> Please express your opinions about this draft, especially opinions 
> about using this document as the starting point for the ROHC UDP-Lite
> profile development.
> 
> Thanks!
> /Lars-Erik
> 
> 
> 
> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Sent: den 3 februari 2003 12:44
> Subject: I-D ACTION:draft-pelletier-rohc-udplite-01.txt
> 
> 
> A New Internet-Draft is available from the on-line 
> Internet-Drafts directories.
> 
> 
> 	Title		: RObust Header Compression 
> (ROHC):Profiles for UDP Lite
> 	Author(s)	: G. Pelletier
> 	Filename	: draft-pelletier-rohc-udplite-01.txt
> 	Pages		: 19
> 	Date		: 2003-1-31
> 	
> This document defines ROHC (Robust Header Compression) profiles for
> compression of RTP/UDP-Lite/IP packets (Real-Time Transport Protocol,
> User Datagram Protocol Lite, Internet Protocol) and UDP-Lite/IP.
> These profiles are defined based on their differences with the
> profiles specified in [RFC-3095] for UDP [RFC-768].
> Although both transport protocols are very similar, ROHC profiles
> must be defined separately for robust compression of UDP-Lite headers
> because it does not share protocol identifier with UDP. Also, the
> UDP-Lite Checksum Coverage field does not share the semantics of the
> corresponding UDP Length field and as a consequence cannot always be
> inferred anymore.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-pelletier-rohc-udplite-01.txt

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type "cd internet-drafts" and then
	"get draft-pelletier-rohc-udplite-01.txt".

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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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From: Ghyslain Pelletier <Ghyslain.Pelletier@epl.ericsson.se>
Organization: Ericsson Erisoft AB, Ericsson Research Corporate Unit
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Subject: Re: [rohc] FW: I-D ACTION:draft-pelletier-rohc-udplite-01.txt
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Hi Richard,

> The updated draft is quite a bit longer than the old one - a list of
> what's changed since the previous version would be much appreciated!

Most of the content of the draft has changed as the assumptions made
regarding the behavior of UDP-Lite flows have evolved, leading to a
totally different approach taken for compressing the headers. 

Also, the previous version was suggesting an approach and was written
with the intent to "probe" the group's opinion regarding directions for
the work, while this draft is written with the intent to be as much as
possible a complete specification of a ROHC profile for UDP-Lite.

With this in mind, I suggest that the draft be read as a totally new
document, as not much from version 00 is left in version 01.

> One point I noticed is that the new draft supports the compression
> of multiple IP headers. Is this something that's particularly useful
> for UDP-Lite (as compared with vanilla UDP), or just a feature that
> would be good to have in general?

Something good to have in general.

Regards,

/Ghyslain

Ghyslain Pelletier
Wireless IP Optimizations
AWARE - Advanced Wireless Algorithm Research
Ericsson Research, Corporate Unit
Ericsson AB
Luleå, Sweden
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From: "Price, Richard" <richard.price@roke.co.uk>
To: "'Ghyslain Pelletier'" <Ghyslain.Pelletier@epl.ericsson.se>
Cc: "'Lars-Erik Jonsson (EAB)'" <Lars-Erik.Jonsson@epl.ericsson.se>,
        "Rohc (E-mail)" <rohc@ietf.org>
Subject: RE: [rohc] FW: I-D ACTION:draft-pelletier-rohc-udplite-01.txt
Date: Mon, 3 Feb 2003 17:19:36 -0000 
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> With this in mind, I suggest that the draft be read as a totally new
> document, as not much from version 00 is left in version 01.

Ok, I guess I'll have to read the whole draft ;-)

> > One point I noticed is that the new draft supports the compression
> > of multiple IP headers. Is this something that's particularly useful
> > for UDP-Lite (as compared with vanilla UDP), or just a feature that
> > would be good to have in general?
> 
> Something good to have in general.

In this case, would it be worth thinking about some other features that
are good to have in general? For the ROHC-TCP work we've developed a
number of enhancements (e.g. context replication) that are equally
applicable to other protocols such as UDP-Lite. If we're taking the
opportunity to improve the profile relative to UDP then these techniques
might be worth considering.

Regards,

Richard
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Subject: RE: [rohc] RE: DAP issues
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[Abbie wrote:]

> I think Zhigang clarified it in his mail on Tues, but we 
> could add the following
> line to section 2, paragraph 2.
> 
> "The entire DAP message (message-header + CRLF + message-body)
> is returned. This allows the sender to compare what it sent 
> with what the
> receiver decompressed."

[Zacco wrote:]
 
> >> This is the message header I would not like to have in the 
> returned message. The uncompressed message is just enough. Do 
> we need the header information for some special purpose? If 
> yes, I promise I will not resist.

Zacco,

Sorry, I couldn't follow you. Do you mean the returned uncompressed message should not include the message-header? But then how can the sender be sure that the *entire* DAP message it sent has been correctly decompressed by the remote endpoint? Note, what SigComp processes is the DAP message, not just the DAP message-body. 

I thought this was obvious, but I will add the clarification as Abbie suggested above.

Zhigang
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> Either (as it states in DAP 1b) the receiver can set the 
> need-response field to TRUE and  both sender and receiver control the flow, allowing the 
> possibility of infinite sending (fixed by one end setting the field to FALSE)
> or
> (as I suggested in my last mail) the sender is in control of 
> the message flow, the receiver must NOT set the flag to TRUE. 
> 
> One thing to take into account is whether either approach 
> will enable/prevent specific test patterns.  The latter seems simpler to me, and I don't 
> think it prevents anything.  What does anyone else think?

Abbie,

I used the former in the document for two reasons:

1) It allows arbitrary message exchange pattern, e.g., 3-way message exchange: m1 from A to B, m2 from B to A, then m3 from A to B. B can control in this case.

2) Again, I don't think the dead loop is an issue. I mentioned it in the document just as a reminder. The bottom line is the common sense in implementation. If that is not there, no protocol is bullet proof. (No protocol can prevent a bad or incorrect implementation anyway.) In this case, the DAP clearly provides ways to avoid dead loop (I'll make it more redundantly clear). One should not blame DAP for a bad/incorrect implementation.

Zhigang
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This is a multi-part message in MIME format.

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Hi,

Here is version 1c. Based on comments from Zacco and Abbie, I added=20
clarifications on returned uncompressed DAP messages and "need-feedback"
header field.=20

As planned, I would like to freeze this as the DAP version 1 for now.
At least it's good enough for the first SigComp interop.=20

(Of course, any questions related to the understanding of the document
are still welcome. I've been trying to be concise, but may not have
covered every fine details ...)

Thanks for Abbie, Adam and Zacco for their valuable comments.

BR,
Zhigang

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Hi all,

Having been asked to have a read the MIB for ROHC, I finally got around to it.  Here are my comments.  Mostly these are just a few typos.  Anyway...

4.1.2. 'notIn&Use' -> 'notInUse'?

4.1.4. the state of the CID seems to be missing an entry... 
	'conrtext' -> 'context'
	'mean packets size' -> 'mean packet size'
Since a row in the rohcContextTable is created when a context is initialised, I was wondering whether it makes sense to create a contxt in the 'unused' state?  I suppose if you view creating a pool of unassigned contexts to use, they would be 'unused'.  I don't think anything needs to change -- I'm just thinking out loud!

4.3. This is just a random thought, but... I noticed that the LLA profiles are bundled in with the ROHC-RTP profiles.  On one level this clearly makes sense, since they compressing the same thing.  On the other hand, there a quite a lot of things that are LLA specific and have no relevance to the 'regular' RTP profiles.  If it's not possible/too much of a change, I wouldn't bother, but... is it possible to have the LLA bits as an extension of the original RTP profiles, rather than included.

5.

Definition of rohcChannelEntry seems suspicious -- referring to 'scripts in non-volatile memory'...

rohcChannelType: 'ight' -> 'might'

rohcInstanceVersion: I assume that this a 'vendor specific' version?  (i.e. I'm not aware of any 'version' of rohc, per se, but only of the profiles...

rohcInstanceLargeCIDs: 'retireved' -> 'retrieved'

rohcInstanceMRRU: 'according to .' According to what?!

rohcInstanceContextStorageTime: 'rohcInstanceContextExpireTime' doesn't appear anywhere in the MIB..!

rohcInstancePackets: 'Counter of passing' -> 'Counter of packets passing'

rohcInstanceCompressionRatio: would it be useful to have the compression ratio for just the headers as well as the overall compression ratio?

rohcProfileTable: I can understand this storing a list of the profiles supported by the decompressor (and, I assume, this would cover the compressor for that instance?) but I'm not sure about the injunction that the compressor MUST NOT use a profile not in the list.  A compressor can only use a profile that has been negotiated (in some way) with the peer decompressor.  I guess you could use the MIB to do that, but...

rohcProfileVersion: I assume that this some 'vendor specific' version, since rohc profiles have a version encoded into the profile identifier...

rohcContextCIDState: can a CID change state from 'active' to 'active'..?

rohcContextStorageTime: 'rochContextCIDState' -> 'rohcContextCIDState'

rohcContextFeedbacks: 'feddbacks' -> 'feedbacks'

rohcContextAllPacjetsMeanSize: 'in byte rounded' -> 'in bytes rounded'

rohcContextAllHeadersMeanSize: 'in byte rounded' -> 'in bytes rounded'

rohcContextLastHeadersMeanSize: 'in byte rounded' -> 'in bytes rounded'

rohcRtpContextState 'noContex' -> 'noContext'

6. Well, ok, not directly related to the text in the security considerations but...  Is there no way of using the MIB to control the mode in which a decompressor operates?  There aren't many changes that you might like to be able to make to a rohc instance, but that struck me as a possibility.

Cheers,

Mark.

-- 
Mark A. West, Consultant Engineer
Roke Manor Research Ltd., Romsey, Hants.  SO51 0ZN
Phone +44 (0)1794 833311   Fax  +44 (0)1794 833433


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From: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>
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Subject: [rohc] WGLC comments addressed: Terminology and Examples
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Hi all,

The last-call comments on the Terminology and Examples document
have been addressed and a diff can be found at:
http://standards.ericsson.net/larsman/draft-ietf-rohc-terminology-and-examples-02diff.pdf

The title has been slightly changed based on Juergen's comment, but
alternative suggestions are still welcome. The comment about numbering
and adding captions to figures has not been addressed, since there is
exactly one figure per section, each mapping directly to the section
title. Therefore, additional numbers and/or captions would just be
redundant.

The updated draft will be posted when also the MIB has been updated
to address its last-call comments.

Cheers,
/L-E


-----------------------------------
Lars-Erik Jonsson, M.Sc
Wireless IP Optimizations
AWARE - Advanced Wireless Algorithm Research
Ericsson Research, Corporate Unit
Ericsson AB
Box 920, S-971 28 Luleå, Sweden
E-mail: lars-erik.jonsson@ericsson.com
Phone: +46 920 20 21 07
Fax: +46 920 20 20 99
Home: +46 920 999 57

My opinions are my personal opinions and should not be considered
as the opinions of my employer, if not explicitly stated. 
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From: "Valerie Kenneally" <vkenneally@cit.ie>
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Subject: [rohc] Sigcomp and Feedback
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Hi All,

	I am trying to understand the feedback mechanism provided by SIGCOMP.  My
question is based on what happens if a remote endpoint receives a message
which it cannot decompress for whatever reason so we have a DECOMPRESSION
FAILURE.

Can the remote end-point notify the sending end-point that decompression has
failed.  The reason I ask this is that the RFC states that feedback is
piggybacked onto a Sigcomp message.

I don't know if I am completely misunderstanding the format of a Sigcomp
message but should it not always contain a compressed message.  If a
DECOMPRESSION FAILURE occurs a follow-up message cannot be sent.  So can a
Sigcomp message be sent for the sole purpose of DECOMPRESSION FAILURE
notification without being accompanied by a compressed message?


Any help at all would be greatly appreciated
Regards,
Val

------------------------
Val Kenneally,
Post-Graduate,
Electronic Eng. Dept.,
CIT,
Rossa Ave.,
Bishopstown,
Cork,
Ireland.
------------------------
Tel: +353 87 2273668
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From mailnull@www1.ietf.org  Tue Feb  4 10:38:51 2003
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To: "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>, rohc@ietf.org
Subject: RE: [rohc] a dummy application protocol (DAP) for SigComp interop
Date: Tue, 4 Feb 2003 15:35:52 -0000 
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Hi Zhigang, all,

Thanks for your hard work on DAP.  I think we have a good interoperable protocol.  I have a couple of questions about its behaviour though.

If the sender sets the need-response header field to TRUE but doesn't receive anything back within a certain time, what should it do?  Should it retransmit the message it just sent or carry on and transmit the next one it was planning to send anyway?

I think to answer this we have to look at the test scenarios.  I think the main one where the possibility of retransmission may be interesting is the case of dropped packets.

As this is not a huge issue I think that it would be good if we could configurably either retransmit or continue but that continuing with the next message is simplest, and sufficient if configurable isn't possible (dropped packet retransmission can be dealt with by duplicate messages).

A similar question is about compartments and repeatability of the test.  There are several possible behaviours for an endpoint sending a return message:
- send back what it received,
- send an arbitrary message,
- send a specific message

If I am a receiver and I have specific messages to send to separate compartments at the same remote endpoint, does it matter which I send to which compartment?  If it does (possibly for repeatability of the test?), then can this be agreed out of band e.g. configured compartment ids or must it be worked out (if so, how)?
Again I think we need to look at the scenarios we want to test, to see how much of an issue this is.
I think that automatically working this out, or sending the information to the receiver would require changes/enhancements to DAP so I suggest that we are able to configure compartment ids and exchange the information of what they are verbally.

Any thoughts/comments welcome.

Best regards,

Abbie


>Hi,
>
>Here is version 1c. Based on comments from Zacco and Abbie, I added 
>clarifications on returned uncompressed DAP messages and 
>"need-feedback"
>header field. 
>
>As planned, I would like to freeze this as the DAP version 1 for now.
>At least it's good enough for the first SigComp interop. 
>
>(Of course, any questions related to the understanding of the document
>are still welcome. I've been trying to be concise, but may not have
>covered every fine details ...)
>
>Thanks for Abbie, Adam and Zacco for their valuable comments.
>
>BR,
>Zhigang
>
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From mailnull@www1.ietf.org  Tue Feb  4 11:52:59 2003
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Hi,

Some comments on draft-pelletier-rohc-udplite-01.txt:

General:

UDP-Lite is still given as an Internet Draft - any ideas
on whether it is stable or likely to change in the future?

Page 4: "For UDP-Lite, the checksum has also completely
random values"

The checksum doesn't take random values (in fact its value
could be inferred from the uncompressed packet). We do
however need to transmit it "as-is" to avoid breaking the
end-to-end error protection.

Page 5: Inter-packet behavior

Can we say more about the behaviour of the Checksum
Coverage field than the three cases given? For example,
even in behaviour pattern 3 (changing, but does not follow
any specific pattern) the checksum will almost certainly
cover far fewer than the maximum of 65535 bytes. So a
version of LSB encoding will get a higher compression
ratio than just sending the field in full.

Page 6: "the field is CHANGING and the value must be sent
along with every packet."

CHANGING doesn't necessarily imply that the field value
must be sent in full (e.g. LSB encoding will often be
sufficient to reconstruct the field at the decompressor).

Page 6: "Simplicity is a strong motivation for the design
of the UDP-Lite header compression profiles."

I'd like to clarify what is meant by the above sentence.
Are we looking for a profile that's simple in general, or
one that's simple given a full implementation of RFC 3095?

Page 7: "These profiles build on the specifications found
in [RFC-3095] with as little modifications as possible."

I think that the proposed profile contains at least one
enhancement that isn't strictly related to UDP-Lite,
namely the support for multiple IP headers. Whilst I
think this is a good idea, we should probably state
explicitly when enhancements of this nature are made.

Page 9: Packet formats

I'd recommend nailing down the behaviour of UDP-Lite more
precisely before defining the packet formats. This will
save time in the long run, because a small change to our
understanding of UDP-Lite may require the packet formats
to be completely rewritten. In particular I'd like to
investigate whether we can get more aggressive compression
of the Checksum Coverage field in behaviour pattern 3.

Regards,

Richard

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Hi Richard,

Thanks for your review and comments, that's certainly appreciated. Some
thoughts inline.

/Ghyslain

"Price, Richard" wrote:

> UDP-Lite is still given as an Internet Draft - any ideas
> on whether it is stable or likely to change in the future?

My understanding is that the UDP-Lite draft was recently edited with
minor modifications, and is now very stable. I expect it to be published
without any substancial modifications.
 
> Page 4: "For UDP-Lite, the checksum has also completely
> random values"
> 
> The checksum doesn't take random values (in fact its value
> could be inferred from the uncompressed packet). We do
> however need to transmit it "as-is" to avoid breaking the
> end-to-end error protection.

I used the term "random" from a header compression perspective, in the
sense that it cannot be compressed once one does not find acceptable to
break its end-to-end nature using a recomputation of its value at the
decompressor. The end-to-end nature of the checksum is mentionned two
paragraphs above the quoted text, thus leading to this classification.

Do you not agree that "random" is the correct classification for the
UDP-Lite checksum field?

> Page 5: Inter-packet behavior
> 
> Can we say more about the behaviour of the Checksum
> Coverage field than the three cases given? For example,
> even in behaviour pattern 3 (changing, but does not follow
> any specific pattern) the checksum will almost certainly
> cover far fewer than the maximum of 65535 bytes. So a
> version of LSB encoding will get a higher compression
> ratio than just sending the field in full.

I did consider this before, but my understanding is that this would
require more changes to packets formats as defined for UDP than might be
desirable, due to the more extreme cases. One objective with the
definition of profiles for UDP-Lite as stated in the draft is to deviate
as little as possible from what's already specified for UDP in RFC-3095.
However I am certainly open into exploring this possibility once more. I
would however appreciate if you could elaborate on this (i.e.
motivations, expected behavior of the CC field, what is needed/tradeoffs
of including LSB encoding for this field, mainly with respect to the
resulting packet format).

> Page 6: "the field is CHANGING and the value must be sent
> along with every packet."
> 
> CHANGING doesn't necessarily imply that the field value
> must be sent in full (e.g. LSB encoding will often be
> sufficient to reconstruct the field at the decompressor).

Agreed. But see above.
 
> Page 6: "Simplicity is a strong motivation for the design
> of the UDP-Lite header compression profiles."
> 
> I'd like to clarify what is meant by the above sentence.
> Are we looking for a profile that's simple in general, or
> one that's simple given a full implementation of RFC 3095?

Simple, given an existing implementation of profiles 1 and 2.

If I reword section 4.1 as follow:

"A strong motivation for the design of a header compression
scheme for UDP-Lite based on ROHC is to define profiles that
entail only a few simple modifications to the corresponding
profiles defined for UDP in [RFC-3095]."

> Page 7: "These profiles build on the specifications found
> in [RFC-3095] with as little modifications as possible."
> 
> I think that the proposed profile contains at least one
> enhancement that isn't strictly related to UDP-Lite,
> namely the support for multiple IP headers. Whilst I
> think this is a good idea, we should probably state
> explicitly when enhancements of this nature are made.

The next sentence after the text quoted above begins with "unless stated
otherwise". The addition of support for multiple IP headers is stated in
the introduction (informative) and in section 5.3.1 (normative). It is
possible that the way the text is written it might still be useful to
bring the specific mention of that addition in Section 5.

> Page 9: Packet formats
> 
> I'd recommend nailing down the behaviour of UDP-Lite more
> precisely before defining the packet formats. This will
> save time in the long run, because a small change to our
> understanding of UDP-Lite may require the packet formats
> to be completely rewritten. In particular I'd like to
> investigate whether we can get more aggressive compression
> of the Checksum Coverage field in behaviour pattern 3.

As said earlier, I will appreciate any input you can provide me on the
expected behaviour of UDP-Lite from your perspective. I think that
draft-pelletier-rohc-udplite-01.txt already provides a rather detailed
discussion on this which should hopefully be at least a good basis.
Also, I would definitely prefer that the profiles for UDP-Lite not
require the packet formats to be completely rewritten.
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Subject: RE: [rohc] a dummy application protocol (DAP) for SigComp interop
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Thread-Topic: [rohc] a dummy application protocol (DAP) for SigComp interop
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Hi Abbie,

Please see my answers inline. 

Just one comment first. DAP is a simple tool to facilitate 
the tests of SigComp. Except some required behavior on the receiving
side, the sender's behavior is pretty much open. I.e. how to *use* 
DAP is the sender's choice. This flexibility allows us to construct 
various test cases to fully test the interoperability of SigComp.

Zhigang

> If the sender sets the need-response header field to TRUE but 
> doesn't receive anything back within a certain time, what 
> should it do?  Should it retransmit the message it just sent 
> or carry on and transmit the next one it was planning to send anyway?
> 
> I think to answer this we have to look at the test scenarios. 
>  I think the main one where the possibility of retransmission 
> may be interesting is the case of dropped packets.

I guess you are talking about SigComp over UDP. Also, you must
mean the sender does not even receive the uncompressed message. 
(Otherwise, the sender will know the message has been received). 
There are 2 possibilities: a) the message sent by the sender 
is lost; b) both the uncompressed and compressed message returned by the 
receiver are lost.

It's up to the sender to decide which case it is dealing with. If it
is "conservative", a retransmission may be a good idea especially
when the possibly lost message carries state creation request that
may be used by the next message. Or it can continue to the next one. 
It should do whatever it needs to do.

> As this is not a huge issue I think that it would be good if 
> we could configurably either retransmit or continue but that 
> continuing with the next message is simplest, and sufficient 
> if configurable isn't possible (dropped packet retransmission 
> can be dealt with by duplicate messages).

This is a local decision by the sender. Sorry, I  what you mean
by "configurable".

> A similar question is about compartments and repeatability of 
> the test.  There are several possible behaviours for an 
> endpoint sending a return message:
> - send back what it received,
> - send an arbitrary message,
> - send a specific message

The main reason to have a response message - at least for
now - is to allow piggybacked sigcomp feedback. The message-body
and some (but not all) message-header fields are not specified. However,
note the "peer compartment rule" specified in DAP. 

> If I am a receiver and I have specific messages to send to 
> separate compartments at the same remote endpoint, does it 
> matter which I send to which compartment? If it does 
> (possibly for repeatability of the test?), then can this be 
> agreed out of band e.g. configured compartment ids or must it 
> be worked out (if so, how)?

See above. If it is a response requested by the sender, you
must send the message from a local compartment that is
peer the remote compartment. Also see the DAP document.

If the receiver wants to send a message to a different 
compartment, it must send another message on its own.

> Again I think we need to look at the scenarios we want to 
> test, to see how much of an issue this is.
> I think that automatically working this out, or sending the 
> information to the receiver would require 
> changes/enhancements to DAP so I suggest that we are able to 
> configure compartment ids and exchange the information of 
> what they are verbally.

I'm not sure I understand the issue. As described in current DAP, 
the sender decides the compressor compartment ID of a message it sends.
Then the receiver will use the (endpoint-ID, compressor compartment ID)
pair to derive the corresponding decompressor compartment. Do you
mean you also want to control the local representation of the
decompressor compartment ID? Probably description of some particular
scenarios would help. I certainly agree that we can/should add whatever 
enhancements to DAP when needed.
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From: "Price, Richard" <richard.price@roke.co.uk>
To: "'Ghyslain Pelletier'" <Ghyslain.Pelletier@epl.ericsson.se>
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Subject: [rohc] RE: Comments on draft-pelletier-rohc-udplite-01.txt
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Hi Ghyslain,

Many thanks for your reply. I've added some further thoughts inline:

> > Page 4: "For UDP-Lite, the checksum has also completely
> > random values"
> > 
> > The checksum doesn't take random values (in fact its value
> > could be inferred from the uncompressed packet). We do
> > however need to transmit it "as-is" to avoid breaking the
> > end-to-end error protection.
> 
> I used the term "random" from a header compression perspective, in the
> sense that it cannot be compressed once one does not find acceptable to
> break its end-to-end nature using a recomputation of its value at the
> decompressor. The end-to-end nature of the checksum is mentionned two
> paragraphs above the quoted text, thus leading to this classification.
> 
> Do you not agree that "random" is the correct classification for the
> UDP-Lite checksum field?

I'd like to avoid the term "random", which generally refers to some
data that has no discernable pattern whatsoever (thus the term is
inapplicable to the UDP checksum).

RFC 3095 coins the more general term "irregular" to describe fields
"for which no useful change pattern can be identified". As well as
truly random fields, this definition also applies to fields that do
have a discernable pattern, but which isn't useful for compression
(e.g. because it would violate end-to-end integrity). How about
using this classification instead?

> > Page 5: Inter-packet behavior
> > 
> > Can we say more about the behaviour of the Checksum
> > Coverage field than the three cases given? For example,
> > even in behaviour pattern 3 (changing, but does not follow
> > any specific pattern) the checksum will almost certainly
> > cover far fewer than the maximum of 65535 bytes. So a
> > version of LSB encoding will get a higher compression
> > ratio than just sending the field in full.
> 
> I did consider this before, but my understanding is that this would
> require more changes to packets formats as defined for UDP than might be
> desirable, due to the more extreme cases. One objective with the
> definition of profiles for UDP-Lite as stated in the draft is to deviate
> as little as possible from what's already specified for UDP in RFC-3095.

I think it would be helpful to cleanly separate the chapter
on UDP-Lite behaviour from the proposed compression profile.
The behaviour chapter should just describe how we expect UDP-Lite
to behave, without taking into account how we intend to compress
it (much like the behaviour draft we did for ROHC-TCP). If we
expect the CC field to exhibit non-random behaviour then we
should say so, even if we then ignore this when actually
compressing the field.

> However I am certainly open into exploring this possibility once more. I
> would however appreciate if you could elaborate on this (i.e.
> motivations, expected behavior of the CC field, what is needed/tradeoffs
> of including LSB encoding for this field, mainly with respect to the
> resulting packet format).

Ok. I think the first step is to have a look at the three
behaviour patterns for UDP-Lite in more detail:

Behaviour pattern 1 occurs when UDP-Lite is used in the same
way as vanilla UDP. As stated in the draft, this behaviour
allows the CC field to be inferred from the overall packet
length.

Behaviour pattern 2 occurs when the UDP-Lite checksum covers
a simple fixed-length header (e.g. RTP/UDP-Lite only). In this
case the CC field is static relative to the context, just as
is stated in the draft.

Behaviour pattern 3 occurs when the UDP-Lite checksum covers
a more complex variable-length portion of the packet (e.g.
the RTP/UDP-Lite header and also the error-sensitive part
of the payload). The draft states that the CC field has
completely random behaviour in this case, but I disagree
with this - there's a lot we can say about the CC field even
if we can't predict its behaviour exactly. For example, it
will typically be much smaller than the maximum value of
65535 bytes, and only the last few LSBs are likely to change
relative to the previous packet.

So a more complex encoding for the CC field will save 1-2
bytes per header whenever behaviour pattern 3 occurs. I
think that this saving is well worth making: UDP-Lite is
a new protocol and we can't predict exactly how it will be
used, so we need to make sure we efficiently support all
three behaviour patterns.

The tradeoff is that the packet formats will need to be
different from those in RFC 3095. However I believe that
the increased efficiency and future-proofing outweighs
this drawback (see below).

> > Page 6: "Simplicity is a strong motivation for the design
> > of the UDP-Lite header compression profiles."
> > 
> > I'd like to clarify what is meant by the above sentence.
> > Are we looking for a profile that's simple in general, or
> > one that's simple given a full implementation of RFC 3095?
> 
> Simple, given an existing implementation of profiles 1 and 2.
> 
> If I reword section 4.1 as follow:
> 
> "A strong motivation for the design of a header compression
> scheme for UDP-Lite based on ROHC is to define profiles that
> entail only a few simple modifications to the corresponding
> profiles defined for UDP in [RFC-3095]."

Ok, I understand the design goal but I'm not sure I agree
with it ;-)

For the original RTP/UDP/IP profile, compression efficiency
was a much more important goal than simplicity (as anyone
who's read RFC 3095 will no doubt be aware!). For UDP-Lite
the efficiency of the solution should be equally important,
because the lossy, bandwidth-poor links where efficiency is
paramount are exactly the kind of links where UDP-Lite is
expected to be used in preference to regular UDP.

> > Page 7: "These profiles build on the specifications found
> > in [RFC-3095] with as little modifications as possible."
> > 
> > I think that the proposed profile contains at least one
> > enhancement that isn't strictly related to UDP-Lite,
> > namely the support for multiple IP headers. Whilst I
> > think this is a good idea, we should probably state
> > explicitly when enhancements of this nature are made.
> 
> The next sentence after the text quoted above begins with "unless stated
> otherwise". The addition of support for multiple IP headers is stated in
> the introduction (informative) and in section 5.3.1 (normative). It is
> possible that the way the text is written it might still be useful to
> bring the specific mention of that addition in Section 5.

The current text makes it clear that support for multiple
IP headers has been added, but doesn't explain the motivation
behind this addition. It seems to contradict the goal of
making as little modifications to RFC 3095 as possible.

From my perspective I'd like the final solution to be as
good as possible, so I strongly support making improvements
of this nature. However we might need to rewrite the
original sentence to clarify that RFC 3095 can be modified
if we believe the performance gain to be worthwhile.

> > Page 9: Packet formats
> > 
> > I'd recommend nailing down the behaviour of UDP-Lite more
> > precisely before defining the packet formats. This will
> > save time in the long run, because a small change to our
> > understanding of UDP-Lite may require the packet formats
> > to be completely rewritten. In particular I'd like to
> > investigate whether we can get more aggressive compression
> > of the Checksum Coverage field in behaviour pattern 3.
> 
> As said earlier, I will appreciate any input you can provide me on the
> expected behaviour of UDP-Lite from your perspective. I think that
> draft-pelletier-rohc-udplite-01.txt already provides a rather detailed
> discussion on this which should hopefully be at least a good basis.
> Also, I would definitely prefer that the profiles for UDP-Lite not
> require the packet formats to be completely rewritten.

In general I'm a big fan of reusing our existing work as far as
possible (in fact the whole purpose of the Formal Notation draft
is to provide a library of encoding methods that can be reused
between profiles). However I'd never do this at the expense of
a key goal like compression efficiency or future-proofing.

Also, I suspect that rewriting the packet formats wouldn't be
too difficult in practice. The hard part is defining the
encoding methods that compress each field, and we can reuse
what's been done in RFC 3095 for this. Once we know how to
compress each field the packet formats are basically trivial.

Regards,

Richard
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From: "Surtees, Abigail" <abigail.surtees@roke.co.uk>
To: "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>, rohc@ietf.org
Subject: RE: [rohc] a dummy application protocol (DAP) for SigComp interop
Date: Wed, 5 Feb 2003 14:54:56 -0000 
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Hi Zhigang

>
>Just one comment first. DAP is a simple tool to facilitate 
>the tests of SigComp. Except some required behavior on the receiving
>side, the sender's behavior is pretty much open. I.e. how to *use* 
>DAP is the sender's choice. This flexibility allows us to construct 
>various test cases to fully test the interoperability of SigComp.

I think then that we need to look at some test cases in more detail to see how we would need to use DAP.

The test document says:
"   This test should check that Endpoint 1 can request a feedback item to be returned by Endpoint 2, and that it can make use of the returned feedback to
   improve the overall compression ratio (e.g. by accessing states acknowledged by the returned feedback).

   Support required at Endpoint 1: Uploading bytecode that requests a feedback item to be returned, and making use of the resulting feedback.
   Support required at Endpoint 2: Basic SigComp only.
"
How exactly should we test this?

Obviously the most basic test is a sequence of messages where A requests a feedback in Sigcomp and a response in DAP.  If we do this with multiple messages, should B know to request a response too or will A automatically send another message on receipt of a message from B?

There are several other possible tests:

That if a message doesn't authenticate, then any feedback received with it is ignored.  Requires the receiver to set the authenticate field to reject.

Which feedback item is sent if multiple messages request feedback but there is no return message, then eventually there is?
Sender must set response to false for several messages even though sending with compressor that expects feedback.

The message arrives but the feedback is lost.  If the sender retransmits then identical state will be created - a feature which should be tested.  If the sender continues and sends its next message new state will be created (the compressor should not reference the lost message).
Which are we trying to test?

The message doesn't arrive so the sender never receives any feedback.  (This could be simulated by the sender instructing the receiver to reject the message.)  Again the choice between retransmission or continuation could have an effect on the compressor.

I think that in order to be clear how DAP needs to behave, we have to be clear on what exactly we are trying to test and how - maybe to the level of specifying, for example,
"A and B exchange 6 messages with feedback" or
"A sends 4 messages to B, requesting a response, the 3rd being a retransmission of the second because that response is not received"

My second point was as a result of thinking about tests at this level:

If A and B are exchanging messages between 2 compartments (and A sends the first message for both) does it matter, in any test case, what is in the body of the messages returned.  If it doesn't then that's fine.  If it does then B needs some way of knowing which of the messages it has corresponds to which compartment.

e.g.

A has msg_2a and msg_3a to send from compartments 2 and 3, both going to B and asking for a response.  B has ready prepared messages msg_2b and msg_3b to go to A.  Does it matter whether msg_2b is the response to msg_2a or msg_3a?  If it does, then B must have some way of distinguishing/knowing that.

The only case I can think of where it might matter is shared mode?  Is this the case and are there any others, or can all the responses be arbitrary? 

I hope this clarifies the points I was trying to raise.

Best regards,

Abbie

>> If the sender sets the need-response header field to TRUE but 
>> doesn't receive anything back within a certain time, what 
>> should it do?  Should it retransmit the message it just sent 
>> or carry on and transmit the next one it was planning to send anyway?
>> 
>> I think to answer this we have to look at the test scenarios. 
>>  I think the main one where the possibility of retransmission 
>> may be interesting is the case of dropped packets.
>
>I guess you are talking about SigComp over UDP. Also, you must
>mean the sender does not even receive the uncompressed message. 
>(Otherwise, the sender will know the message has been received). 
>There are 2 possibilities: a) the message sent by the sender 
>is lost; b) both the uncompressed and compressed message 
>returned by the 
>receiver are lost.
>
>It's up to the sender to decide which case it is dealing with. If it
>is "conservative", a retransmission may be a good idea especially
>when the possibly lost message carries state creation request that
>may be used by the next message. Or it can continue to the next one. 
>It should do whatever it needs to do.
>
>> As this is not a huge issue I think that it would be good if 
>> we could configurably either retransmit or continue but that 
>> continuing with the next message is simplest, and sufficient 
>> if configurable isn't possible (dropped packet retransmission 
>> can be dealt with by duplicate messages).
>
>This is a local decision by the sender. Sorry, I  what you mean
>by "configurable".
>
>> A similar question is about compartments and repeatability of 
>> the test.  There are several possible behaviours for an 
>> endpoint sending a return message:
>> - send back what it received,
>> - send an arbitrary message,
>> - send a specific message
>
>The main reason to have a response message - at least for
>now - is to allow piggybacked sigcomp feedback. The message-body
>and some (but not all) message-header fields are not 
>specified. However,
>note the "peer compartment rule" specified in DAP. 
>
>> If I am a receiver and I have specific messages to send to 
>> separate compartments at the same remote endpoint, does it 
>> matter which I send to which compartment? If it does 
>> (possibly for repeatability of the test?), then can this be 
>> agreed out of band e.g. configured compartment ids or must it 
>> be worked out (if so, how)?
>
>See above. If it is a response requested by the sender, you
>must send the message from a local compartment that is
>peer the remote compartment. Also see the DAP document.
>
>If the receiver wants to send a message to a different 
>compartment, it must send another message on its own.
>
>> Again I think we need to look at the scenarios we want to 
>> test, to see how much of an issue this is.
>> I think that automatically working this out, or sending the 
>> information to the receiver would require 
>> changes/enhancements to DAP so I suggest that we are able to 
>> configure compartment ids and exchange the information of 
>> what they are verbally.
>
>I'm not sure I understand the issue. As described in current DAP, 
>the sender decides the compressor compartment ID of a message it sends.
>Then the receiver will use the (endpoint-ID, compressor compartment ID)
>pair to derive the corresponding decompressor compartment. Do you
>mean you also want to control the local representation of the
>decompressor compartment ID? Probably description of some particular
>scenarios would help. I certainly agree that we can/should add 
>whatever 
>enhancements to DAP when needed.
>
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Hi Abbie,

You've raised many valid questions any person going to the interop
will have to answer. But it seems to me they are beyond the scope
of DAP. It's just a dummy protocol. :-) 

The unspoken philosophy is: write your test code using DAP in the 
best way you see, following the test cases described in the document
Richard sent. DAP guarantees the receiver will understand the message 
and perform operations as the sender instructed. But it is the 
sender that leads.

Detailed answers inline.

Zhigang

> Obviously the most basic test is a sequence of messages where 
> A requests a feedback in Sigcomp and a response in DAP.  

Yes.

> If we do this with multiple messages, should B know to request a 
> response too or will A automatically send another message on 
> receipt of a message from B?

If B wants to do its own test and need a DAP message coming back,
it should explicitly request so by setting need-response to TRUE.
Basically, each sender takes care of its own business and asks 
what it needs.
 
> That if a message doesn't authenticate, then any feedback 
> received with it is ignored.  Requires the receiver to set 
> the authenticate field to reject.

Not exactly, it is the sender that tells the receiver whether 
a message should be accepted or rejected.

> Which feedback item is sent if multiple messages request 
> feedback but there is no return message, then eventually there is?
> Sender must set response to false for several messages even 
> though sending with compressor that expects feedback.

For each message received with need-response set to TRUE, the receive
must send a DAP message back. It doesn't matter what it carries
(the message-body can even be empty). Its purpose is only to allow
the piggyback of sigcomp feedback.
 
> The message arrives but the feedback is lost.  If the sender 
> retransmits then identical state will be created - a feature 
> which should be tested.  If the sender continues and sends 
> its next message new state will be created (the compressor 
> should not reference the lost message).
> Which are we trying to test?

Both are interesting cases.

> The message doesn't arrive so the sender never receives any 
> feedback.  (This could be simulated by the sender instructing 
> the receiver to reject the message.)  Again the choice 
> between retransmission or continuation could have an effect 
> on the compressor.

True, just as any sigcomp implementation has to make its own choice.

> I think that in order to be clear how DAP needs to behave, we 
> have to be clear on what exactly we are trying to test and 
> how - maybe to the level of specifying, for example,
> "A and B exchange 6 messages with feedback" or
> "A sends 4 messages to B, requesting a response, the 3rd 
> being a retransmission of the second because that response is 
> not received"

What you describe here is the sender's behaviours, not the DAP
itself. Probably something can be added to the test case document
sent by Richard? 
 
> A has msg_2a and msg_3a to send from compartments 2 and 3, 
> both going to B and asking for a response.  B has ready 
> prepared messages msg_2b and msg_3b to go to A.  Does it 
> matter whether msg_2b is the response to msg_2a or msg_3a?  
> If it does, then B must have some way of distinguishing/knowing that.

For piggybacking sigcomp feedback, the message-body of a response
doesn't matter. It's just a DAP message. Note, B must generate
a response message if there is none "ready". That is why I put the 
wording about encapsulating the original message in the message-body
of the response.
 
> The only case I can think of where it might matter is shared 
> mode?  

It matters only in terms of compression ratio. (Obviously if
the response message-body contains the original message, the ratio
will be higher.) But in terms of testing the interoperability, no.

> Is this the case and are there any others, or can all 
> the responses be arbitrary? 

At this moment, I don't see a need to specify the message-body of
the response. But there may be. We can always update the DAP when
we have a clearer view.
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Hi Zhigang, all,

Sorry for jumping in, but Abbie's finished for today, and I think it would be helpful to try and resolve this.  So, here goes...

I think that the issue is largely one of what we are describing.  DAP, as you point out, is a dummy application protocol (the name says it all!) and I believe that we have agreed on the structure of the DAP header.  So far, so good...

However, we believe that we also need to define at least some classes of application behaviour (i.e. how to use DAP) in order to perform useful, repeatable tests.  From one of Zhigang's comments, you seem to be suggesting documenting the application behaviour as part of the test description, and that makes sense to me.  I think that there are only a few sensible behaviours that need to be considered.

Anyway, comments inline.  Hopefully the nesting isn't getting *too* confusing!

Cheers,

Mark.


> 
> The unspoken philosophy is: write your test code using DAP in the 
> best way you see, following the test cases described in the document
> Richard sent. DAP guarantees the receiver will understand the message 
> and perform operations as the sender instructed. But it is the 
> sender that leads.

That sounds like a perfectly reasonable philosophy.  And we should all agree on it, otherwise I think we'll have problems.  And I really hope that we don't end up testing our applications, instead of sigcomp..! ;-)

> 
> > If we do this with multiple messages, should B know to request a 
> > response too or will A automatically send another message on 
> > receipt of a message from B?
> 
> If B wants to do its own test and need a DAP message coming back,
> it should explicitly request so by setting need-response to TRUE.
> Basically, each sender takes care of its own business and asks 
> what it needs.

Ok, so that's applying the philosophy above.

When you talk about B setting need-response to TRUE to 'do its own test', it sounds like we have 2 tests running in parallel here, and I'm not sure that's a good idea...

>  
> > That if a message doesn't authenticate, then any feedback 
> > received with it is ignored.  Requires the receiver to set 
> > the authenticate field to reject.
> 
> Not exactly, it is the sender that tells the receiver whether 
> a message should be accepted or rejected.

Apologies -- the point is slightly unclear.  The scenario is, in more detail:
	A sends message to B, requesting response
	B responds to A with a message that fails authentication

A is 'in charge', but B needs to know that it should respond (in this case) with a message that will fail authentication.  Nothing in A's message will tell it this.

> 
> > Which feedback item is sent if multiple messages request 
> > feedback but there is no return message, then eventually there is?
> > Sender must set response to false for several messages even 
> > though sending with compressor that expects feedback.
> 
> For each message received with need-response set to TRUE, the receive
> must send a DAP message back. It doesn't matter what it carries
> (the message-body can even be empty). Its purpose is only to allow
> the piggyback of sigcomp feedback.

I think another slight misunderstanding here...
The simple answer is to take your 'sender-centric' view, in which the sender controls what happens.
This test requires A to send several messages to B without requesting a response, and then sends a message that does request a response.  This clearly requires the sender to be in control.

>  
> > The message arrives but the feedback is lost.  If the sender 
> > retransmits then identical state will be created - a feature 
> > which should be tested.  If the sender continues and sends 
> > its next message new state will be created (the compressor 
> > should not reference the lost message).
> > Which are we trying to test?
> 
> Both are interesting cases.

Indeed, and so our view is that both should be tested.  However, this suggests that a DAP endpoint should be configurable to retransmit or continue after a timeout.  (You can 'fake' the retransmission case).

> 
> > The message doesn't arrive so the sender never receives any 
> > feedback.  (This could be simulated by the sender instructing 
> > the receiver to reject the message.)  Again the choice 
> > between retransmission or continuation could have an effect 
> > on the compressor.
> 
> True, just as any sigcomp implementation has to make its own choice.
> 

I think that this raises an important distinction.  We are *not* defining sigcomp behaviour, we *are* defining the behaviour of the application that uses sigcomp.  And I think that, as I said earlier, we need to define the range of behaviours that are useful in order to carry out a reasonable set of tests.

> > I think that in order to be clear how DAP needs to behave, we 
> > have to be clear on what exactly we are trying to test and 
> > how - maybe to the level of specifying, for example,
> > "A and B exchange 6 messages with feedback" or
> > "A sends 4 messages to B, requesting a response, the 3rd 
> > being a retransmission of the second because that response is 
> > not received"
> 
> What you describe here is the sender's behaviours, not the DAP
> itself. Probably something can be added to the test case document
> sent by Richard? 

Well, it's the behaviour of the application using DAP, yes.  And that's what we are trying to describe.  Apologies if this 'layer' issue has confused things..!

If we agree that the tests should describe the application behaviour then I'm all in favour of that.

>  
> > A has msg_2a and msg_3a to send from compartments 2 and 3, 
> > both going to B and asking for a response.  B has ready 
> > prepared messages msg_2b and msg_3b to go to A.  Does it 
> > matter whether msg_2b is the response to msg_2a or msg_3a?  
> > If it does, then B must have some way of 
> distinguishing/knowing that.
> 
> For piggybacking sigcomp feedback, the message-body of a response
> doesn't matter. It's just a DAP message. Note, B must generate
> a response message if there is none "ready". That is why I put the 
> wording about encapsulating the original message in the message-body
> of the response.
>  
> > The only case I can think of where it might matter is shared 
> > mode?  
> 
> It matters only in terms of compression ratio. (Obviously if
> the response message-body contains the original message, the ratio
> will be higher.) But in terms of testing the interoperability, no.
> 

I'm not convinced that this is necessarily true.  The contents of a message may affect not only the compression ratio but also have an effect on the state management function at both ends.  Now, it may be that this isn't an issue, but I'm not absolutely convinced that the nature of the returned message is completely irrelevant.  I probably need to think of a concrete example, though!

> > Is this the case and are there any others, or can all 
> > the responses be arbitrary? 
> 
> At this moment, I don't see a need to specify the message-body of
> the response. But there may be. We can always update the DAP when
> we have a clearer view.

It's not really a DAP issue, but it is an issue for the application using DAP.  See previous comment for more rambling..!

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Hi Mark,

> However, we believe that we also need to define at least some 
> classes of application behaviour (i.e. how to use DAP) in 
> order to perform useful, repeatable tests.  

I'm not sure it is necessary, but certainly helpful for people who
cannot make up their own mind on how to use DAP.

> From one of 
> Zhigang's comments, you seem to be suggesting documenting the 
> application behaviour as part of the test description, and 
> that makes sense to me.  

Great, at least you agree with me. It's like the distinction
between a document defining a script language and another document
defines a sets of "meaningful" scripts written in the language. 
Another practical reason is that I don't have much bandwith for this.
 
> When you talk about B setting need-response to TRUE to 'do 
> its own test', it sounds like we have 2 tests running in 
> parallel here, and I'm not sure that's a good idea...

It is more challenging than the scenario where A is active
but B only responds passively. On the other hand, it is part 
of what should be tested, since an sigcomp implementation 
should be able to handle both incoming and outgoing messages anyway.

In practice, one can deal with the response request by returning
a dummy DAP message with empty message-body, or put the received
message in the message-body. It's easy and should not interrupt the 
"normal" tests it want to conduct. But I may miss some particular 
scenarios.

> Apologies -- the point is slightly unclear.  The scenario is, 
> in more detail:
> 	A sends message to B, requesting response
> 	B responds to A with a message that fails authentication
> 
> A is 'in charge', but B needs to know that it should respond 
> (in this case) with a message that will fail authentication.  
> Nothing in A's message will tell it this.

You're right, this particular case is not supported by current DAP. 
We can add the support (e.g. sender has more control over the 
response) in the next version. For now, A can "locally"
reject the message even though B asked A to accept.

> This test requires A to send several messages to B without 
> requesting a response, and then sends a message that does 
> request a response.  This clearly requires the sender to be 
> in control.

I think this is supported by current DAP, i.e, the sender
has full control over the need-response field of every message
it sends.

> I think that this raises an important distinction.  We are 
> *not* defining sigcomp behaviour, we *are* defining the 
> behaviour of the application that uses sigcomp.  And I think 
> that, as I said earlier, we need to define the range of 
> behaviours that are useful in order to carry out a reasonable 
> set of tests.

I'm still confused about the "range" and "reasonable set" of DAP
application behaviors. We have a set of sigcomp operations/features
we want to test. So, everyone just bring their DAP application
that targets at testing those sigcomp features (hopefully complete).
The application behaviors may be different. But that is OK. In
fact, it is even better for the interop test since that means 
a sigcomp implementation is subject to more diversified conditions.

That being said, I do agree that documentation of the some *example*
behaviors of DAP application is useful. That can serve as a starting 
point, but not as a boundary. We can populate the document as we go.

> I'm not convinced that this is necessarily true.  The 
> contents of a message may affect not only the compression 
> ratio but also have an effect on the state management 
> function at both ends. (snip)  I probably 
> need to think of a concrete example, though!

Yes, an example would help. I could miss things, given such a short time.

Zhigang
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From: "West, Mark (ITN)" <mark.a.west@roke.co.uk>
To: "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>,
        "Surtees, Abigail" <abigail.surtees@roke.co.uk>, rohc@ietf.org
Subject: RE: [rohc] a dummy application protocol (DAP) for SigComp interop
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Hi Zhigang, all,

I've commented inline, but the key point that you are raising seems to be that the sender should be in complete control of the test that it is running.  This appears to be manageable, but we should obviously agree that this is the case.

Up to now we have been assuming that there is some 'co-operative' element between the 2 enpoints in running a test.  Clearly we will have some difficulties in running tests if this is not a valid assumption!

Cheers,

Mark.


> 
> Hi Mark,
> 
> > However, we believe that we also need to define at least some 
> > classes of application behaviour (i.e. how to use DAP) in 
> > order to perform useful, repeatable tests.  
> 
> I'm not sure it is necessary, but certainly helpful for people who
> cannot make up their own mind on how to use DAP.
> 
> > From one of 
> > Zhigang's comments, you seem to be suggesting documenting the 
> > application behaviour as part of the test description, and 
> > that makes sense to me.  
> 
> Great, at least you agree with me. It's like the distinction
> between a document defining a script language and another document
> defines a sets of "meaningful" scripts written in the language. 
> Another practical reason is that I don't have much bandwith for this.
> 

Of course I agree ;-)
And your analogy is helpful.  I don't think that you need commit bandwidth to this: DAP is agreed.  I'm only thinking about how we use it...
 
> > When you talk about B setting need-response to TRUE to 'do 
> > its own test', it sounds like we have 2 tests running in 
> > parallel here, and I'm not sure that's a good idea...
> 
> It is more challenging than the scenario where A is active
> but B only responds passively. On the other hand, it is part 
> of what should be tested, since an sigcomp implementation 
> should be able to handle both incoming and outgoing messages anyway.
> 

That's a good point.

> In practice, one can deal with the response request by returning
> a dummy DAP message with empty message-body, or put the received
> message in the message-body. It's easy and should not interrupt the 
> "normal" tests it want to conduct. But I may miss some particular 
> scenarios.
> 

True -- ok, we should cover this in at least some of the test cases.

> > Apologies -- the point is slightly unclear.  The scenario is, 
> > in more detail:
> > 	A sends message to B, requesting response
> > 	B responds to A with a message that fails authentication
> > 
> > A is 'in charge', but B needs to know that it should respond 
> > (in this case) with a message that will fail authentication.  
> > Nothing in A's message will tell it this.
> 
> You're right, this particular case is not supported by current DAP. 
> We can add the support (e.g. sender has more control over the 
> response) in the next version. For now, A can "locally"
> reject the message even though B asked A to accept.
> 

Hmmm -- this is certainly an example of where it's necessary to describe application layer behaviour!  It's not necessarily something that has to be supported by DAP.  However, you raise an interesting point.

If we assume that the sender is completely in control of the test, then I'm not sure that 'response' messages need to contain the authentication details, for example, since the 'controlling' endpoint (A, in this example) knows what it should do for each message that it receives in response to a message that it has sent.

> > This test requires A to send several messages to B without 
> > requesting a response, and then sends a message that does 
> > request a response.  This clearly requires the sender to be 
> > in control.
> 
> I think this is supported by current DAP, i.e, the sender
> has full control over the need-response field of every message
> it sends.
> 

Agreed.

> > I think that this raises an important distinction.  We are 
> > *not* defining sigcomp behaviour, we *are* defining the 
> > behaviour of the application that uses sigcomp.  And I think 
> > that, as I said earlier, we need to define the range of 
> > behaviours that are useful in order to carry out a reasonable 
> > set of tests.
> 
> I'm still confused about the "range" and "reasonable set" of DAP
> application behaviors. We have a set of sigcomp operations/features
> we want to test. So, everyone just bring their DAP application
> that targets at testing those sigcomp features (hopefully complete).
> The application behaviors may be different. But that is OK. In
> fact, it is even better for the interop test since that means 
> a sigcomp implementation is subject to more diversified conditions.
> 
> That being said, I do agree that documentation of the some *example*
> behaviors of DAP application is useful. That can serve as a starting 
> point, but not as a boundary. We can populate the document as we go.
> 

I think I basically agree that arbitrary application behaviour is worthwhile testing.  However, the only reason I suggest documenting something (as you say -- 'example' behaviours) is to give us a baseline set of repeatable tests that we all agree on.  That doesn't mean that we cannot/should not test other things.  I agree!

> > I'm not convinced that this is necessarily true.  The 
> > contents of a message may affect not only the compression 
> > ratio but also have an effect on the state management 
> > function at both ends. (snip)  I probably 
> > need to think of a concrete example, though!
> 
> Yes, an example would help. I could miss things, given such a 
> short time.
> 

Still haven't thought of one, but I'll see if I can..!

> Zhigang
> 

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From mailnull@www1.ietf.org  Thu Feb  6 06:25:09 2003
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From: "Lajos Zaccomer (ETH)" <Lajos.Zaccomer@eth.ericsson.se>
To: "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>,
        mark.a.west@roke.co.uk, abigail.surtees@roke.co.uk, rohc@ietf.org
Subject: RE: [rohc] a dummy application protocol (DAP) for SigComp interop
Date: Thu, 6 Feb 2003 12:25:36 +0100 
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Hi,

I have two questions:

1. How can endpoint A now if decompression failure occured on endpoint B? Empty message is not good for this purpose, as it may be a valid output. For negative tests, it is required to know somehow.

2. When the uncompressed message is sent back with the same header, what length should I use, the original (compressed) or the actual (uncompressed) message length?

BR/Zacco

-----Original Message-----
From: zhigang.c.liu@nokia.com [mailto:zhigang.c.liu@nokia.com]
Sent: Thursday, February 06, 2003 2:14 AM
To: mark.a.west@roke.co.uk; abigail.surtees@roke.co.uk; rohc@ietf.org
Subject: RE: [rohc] a dummy application protocol (DAP) for SigComp interop


Hi Mark,

> However, we believe that we also need to define at least some 
> classes of application behaviour (i.e. how to use DAP) in 
> order to perform useful, repeatable tests.  

I'm not sure it is necessary, but certainly helpful for people who
cannot make up their own mind on how to use DAP.

> From one of 
> Zhigang's comments, you seem to be suggesting documenting the 
> application behaviour as part of the test description, and 
> that makes sense to me.  

Great, at least you agree with me. It's like the distinction
between a document defining a script language and another document
defines a sets of "meaningful" scripts written in the language. 
Another practical reason is that I don't have much bandwith for this.
 
> When you talk about B setting need-response to TRUE to 'do 
> its own test', it sounds like we have 2 tests running in 
> parallel here, and I'm not sure that's a good idea...

It is more challenging than the scenario where A is active
but B only responds passively. On the other hand, it is part 
of what should be tested, since an sigcomp implementation 
should be able to handle both incoming and outgoing messages anyway.

In practice, one can deal with the response request by returning
a dummy DAP message with empty message-body, or put the received
message in the message-body. It's easy and should not interrupt the 
"normal" tests it want to conduct. But I may miss some particular 
scenarios.

> Apologies -- the point is slightly unclear.  The scenario is, 
> in more detail:
> 	A sends message to B, requesting response
> 	B responds to A with a message that fails authentication
> 
> A is 'in charge', but B needs to know that it should respond 
> (in this case) with a message that will fail authentication.  
> Nothing in A's message will tell it this.

You're right, this particular case is not supported by current DAP. 
We can add the support (e.g. sender has more control over the 
response) in the next version. For now, A can "locally"
reject the message even though B asked A to accept.

> This test requires A to send several messages to B without 
> requesting a response, and then sends a message that does 
> request a response.  This clearly requires the sender to be 
> in control.

I think this is supported by current DAP, i.e, the sender
has full control over the need-response field of every message
it sends.

> I think that this raises an important distinction.  We are 
> *not* defining sigcomp behaviour, we *are* defining the 
> behaviour of the application that uses sigcomp.  And I think 
> that, as I said earlier, we need to define the range of 
> behaviours that are useful in order to carry out a reasonable 
> set of tests.

I'm still confused about the "range" and "reasonable set" of DAP
application behaviors. We have a set of sigcomp operations/features
we want to test. So, everyone just bring their DAP application
that targets at testing those sigcomp features (hopefully complete).
The application behaviors may be different. But that is OK. In
fact, it is even better for the interop test since that means 
a sigcomp implementation is subject to more diversified conditions.

That being said, I do agree that documentation of the some *example*
behaviors of DAP application is useful. That can serve as a starting 
point, but not as a boundary. We can populate the document as we go.

> I'm not convinced that this is necessarily true.  The 
> contents of a message may affect not only the compression 
> ratio but also have an effect on the state management 
> function at both ends. (snip)  I probably 
> need to think of a concrete example, though!

Yes, an example would help. I could miss things, given such a short time.

Zhigang
_______________________________________________
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From mailnull@www1.ietf.org  Thu Feb  6 07:05:12 2003
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From: "Price, Richard" <richard.price@roke.co.uk>
To: "'rohc@ietf.org'" <rohc@ietf.org>
Date: Thu, 6 Feb 2003 12:06:48 -0000 
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Subject: [rohc] Question on RFC 3095 terminology
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Hi,

I've a quick question on the terminology used in RFC 3095:
what exactly is meant by the term "encoding method"?

Section 4.5 of the RFC includes a library of encoding
methods such as LSB encoding and timer-based compression.
Do any of the other compression techniques used by the
RFC (e.g. inference rules for reconstructing the fields
that don't need to be sent) also count as encoding
methods?

Also, do encoding methods only apply to a single field in
the header or can they include more comprehensive
techniques such as list compression?

Thanks,

Richard
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From: "Surtees, Abigail" <abigail.surtees@roke.co.uk>
To: "'Lajos Zaccomer (ETH)'" <Lajos.Zaccomer@eth.ericsson.se>,
        "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>,
        "West, Mark (ITN)"
	 <mark.a.west@roke.co.uk>, rohc@ietf.org
Subject: RE: [rohc] a dummy application protocol (DAP) for SigComp interop
Date: Thu, 6 Feb 2003 14:51:53 -0000 
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Hi Zacco,

>I have two questions:
>
>1. How can endpoint A now if decompression failure occured on 
>endpoint B? Empty message is not good for this purpose, as it 
>may be a valid output. For negative tests, it is required to 
>know somehow.
>
DAP states that endpoint B should return its decompressed message to A whatever the value of the authenticate field.
If there is decompression failure then there may or may not be some output but whatever it is, it isn't going to be of any use to endpoint A, so there is no point sending it.
In terms of A actually knowing with regard to the next message to compress that there has been decompression failure, it doesn't know (nor can it because SigComp doesn't have a NACK mechanism (at the moment, at any rate)).
If endpoint B has unexpected decompression failure then the results of the test won't be as expected and it is likely that there is a bug somewhere, in which case the implementers of endpoint A and B need to work out where it is.

>2. When the uncompressed message is sent back with the same 
>header, what length should I use, the original (compressed) or 
>the actual (uncompressed) message length?

As I understand it, a DAP message (that is header + CRLF + body) is compressed and sent.
The decompressor receives, decompresses and passes it to the application, which checks it and returns it "as is" to the endpoint from which it came (without putting a dap header on it).  
The original endpoint can distinguish it from a new DAP message because it will have the endpoint's own IP address.  
As there is no new DAP header being added there is no need to worry about what length to give it.

I hope that clarifies things.

Best regards,

Abbie
>
>BR/Zacco
>
>-----Original Message-----
>From: zhigang.c.liu@nokia.com [mailto:zhigang.c.liu@nokia.com]
>Sent: Thursday, February 06, 2003 2:14 AM
>To: mark.a.west@roke.co.uk; abigail.surtees@roke.co.uk; rohc@ietf.org
>Subject: RE: [rohc] a dummy application protocol (DAP) for 
>SigComp interop
>
>
>Hi Mark,
>
>> However, we believe that we also need to define at least some 
>> classes of application behaviour (i.e. how to use DAP) in 
>> order to perform useful, repeatable tests.  
>
>I'm not sure it is necessary, but certainly helpful for people who
>cannot make up their own mind on how to use DAP.
>
>> From one of 
>> Zhigang's comments, you seem to be suggesting documenting the 
>> application behaviour as part of the test description, and 
>> that makes sense to me.  
>
>Great, at least you agree with me. It's like the distinction
>between a document defining a script language and another document
>defines a sets of "meaningful" scripts written in the language. 
>Another practical reason is that I don't have much bandwith for this.
> 
>> When you talk about B setting need-response to TRUE to 'do 
>> its own test', it sounds like we have 2 tests running in 
>> parallel here, and I'm not sure that's a good idea...
>
>It is more challenging than the scenario where A is active
>but B only responds passively. On the other hand, it is part 
>of what should be tested, since an sigcomp implementation 
>should be able to handle both incoming and outgoing messages anyway.
>
>In practice, one can deal with the response request by returning
>a dummy DAP message with empty message-body, or put the received
>message in the message-body. It's easy and should not interrupt the 
>"normal" tests it want to conduct. But I may miss some particular 
>scenarios.
>
>> Apologies -- the point is slightly unclear.  The scenario is, 
>> in more detail:
>> 	A sends message to B, requesting response
>> 	B responds to A with a message that fails authentication
>> 
>> A is 'in charge', but B needs to know that it should respond 
>> (in this case) with a message that will fail authentication.  
>> Nothing in A's message will tell it this.
>
>You're right, this particular case is not supported by current DAP. 
>We can add the support (e.g. sender has more control over the 
>response) in the next version. For now, A can "locally"
>reject the message even though B asked A to accept.
>
>> This test requires A to send several messages to B without 
>> requesting a response, and then sends a message that does 
>> request a response.  This clearly requires the sender to be 
>> in control.
>
>I think this is supported by current DAP, i.e, the sender
>has full control over the need-response field of every message
>it sends.
>
>> I think that this raises an important distinction.  We are 
>> *not* defining sigcomp behaviour, we *are* defining the 
>> behaviour of the application that uses sigcomp.  And I think 
>> that, as I said earlier, we need to define the range of 
>> behaviours that are useful in order to carry out a reasonable 
>> set of tests.
>
>I'm still confused about the "range" and "reasonable set" of DAP
>application behaviors. We have a set of sigcomp operations/features
>we want to test. So, everyone just bring their DAP application
>that targets at testing those sigcomp features (hopefully complete).
>The application behaviors may be different. But that is OK. In
>fact, it is even better for the interop test since that means 
>a sigcomp implementation is subject to more diversified conditions.
>
>That being said, I do agree that documentation of the some *example*
>behaviors of DAP application is useful. That can serve as a starting 
>point, but not as a boundary. We can populate the document as we go.
>
>> I'm not convinced that this is necessarily true.  The 
>> contents of a message may affect not only the compression 
>> ratio but also have an effect on the state management 
>> function at both ends. (snip)  I probably 
>> need to think of a concrete example, though!
>
>Yes, an example would help. I could miss things, given such a 
>short time.
>
>Zhigang
>_______________________________________________
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>
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Hi all,

I thought that it might be worth (at the risk of running this into the ground) listing a couple of example message flows so that we can hopefully agree on how things might work at the application layer...

I'll base this on the sender-centric view, and to start with I'll assume an 'active' sender (controlling the test) and a 'passive' receiver (doing what the sender tells it to).

(note:
  NR = need-response
  AU = message-auth
  MS = message-seq)

Thus, we have the trivial case of A as the active sender and B as the passive receiver doing a simple exchange with a message and a response:

A                                    B
|---- Comp=A:3, NR=T, AU=A, MS=1 --->|
|                                    |
|<--- Comp=B:3, NR=F, AU=A, MS=1 ----|
|                                    |

I assume, although I don't think that it is explicit in DAP, that the message sequence is echoed in the returned header?  This allows scenarios (still sender-driven) like the following to work properly:

A                                    B
|                                    |
|---- Comp=A:3, NR=T, AU=A, MS=1 --->|
|                                    |
|---- Comp=A:3, NR=T, AU=A, MS=2 --->|
|                                    |
| X<- Comp=B:3, NR=F, AU=A, MS=1 ----|
|                                    |
|<--- Comp=B:3, NR=F, AU=A, MS=2 ----|
|                                    |
|---- Comp=A:3, NR=T, AU=A, MS=1 --->|
|                                    |

and the application at endpoint A knows which message to retransmit (if that's what it wants to do...)

It's clear that this can cover a wide variety of test cases.  There's one issue that I'll return to at the end of the mail, though.

Now consider the case where we have two active senders - both A and B are controlling things.  This can actually work fine, but we need to be careful about compartment ID spaces, otherwise we get a situation such as:

A                                    B
|                                    |
|---- Comp=A:1, NR=T, AU=A, MS=1 --->| } simultaneous
|<--- Comp=B:1, NR=T, AU=A, MS=1 ----| } (i.e. messages cross)
|                                    |

Now, A will think B's message is a response to its message (although it isn't) and vice versa!  However, if the compartment ID spaces are disjoint, both sides can control their own flow(s) and it all works just fine:

A                                    B
|                                    |
|---- Comp=A:1, NR=T, AU=A, MS=1 --->| } simultaneous
|<--- Comp=B:7, NR=T, AU=A, MS=1 ----| } (i.e. messages cross)
|                                    |
|<--- Comp=B:1, NR=F, AU=A, MS=1 ----|
|---- Comp=A:7, NR=F, AU=A, MS=1 --->|
|                                    |

How about the case with 2 active senders that share a compartment?  Is this desirable?  Well, I think it raises some issues...

A                                    B
|                                    |
|---- Comp=A:3, NR=T, AU=A, MS=1 --->|
|                                    |
|<--- Comp=B:3, NR=?, AU=?, MS=? ----|
|                                    |
|---- Comp=A:3, NR=?, AU=?, MS=? --->|
|                                    |

A sends a message to B.  Fine!  B responds to A, but is also acting as an active sender.  
So B can ask for a response (but A might be planning to send one anyway!).  B can also indicate whether the message should be accepted or not, but A has its own plan for whether to accept the message.  Should B echo A's message sequence (so A knows what it's a response to), or should it put its own number in?  The same applies to A's next message.  

I think that the responses have to be kept distinct, so we get:

A                                    B
|                                    |
|---- Comp=A:3, NR=T, AU=A, MS=1 --->|
|                                    |
|<--- Comp=B:3, NR=F, AU=A, MS=1 ----| [response to A]
|<--- Comp=B:3, NR=T, AU=A, MS=7 ----| [B initiated message]
|                                    |
|---- Comp=A:3, NR=T, AU=A, MS=2 --->| [A initiated message]
|---- Comp=A:3, NR=F, AU=A, MS=7 --->| [response to B]
|                                    |

And finally (phew!), I'd just like to consider if this case is interesting and, if so, how can it be tested:

A                                    B
|                                    |
|---- Comp=A:3, NR=T, AU=A, MS=1 --->|
|                                    |
|<--- Comp=B:3, NR=?, AU=A, MS=1 ----|
|                                    |
|---- Comp=A:3, NR=F, AU=A, MS=2 ->X |
|                                    |
|<--- Comp=B:3, NR=?, AU=A, MS=1 ----|
|                                    |

So, the test is A sends a message to B and gets a response.  A responds, but this message is lost and B should retransmit its message.  It sounds like B should be an active sender, but this has problems, as outlined above.

I don't know if this is a distinct test case - since if you cut out the first message (from A to B), you can write this as a test with B as the single active sender.  But, to do it as written requires B to be at least 'semi-active'!

(There are other examples where it would be nice for B to be able to send multiple messages in response to A, for example, but I'll save describing these for the sake of brevity!)

I hope that this helps more than it hinders, but..!

;-)

Mark.

-- 
Mark A. West, Consultant Engineer
Roke Manor Research Ltd., Romsey, Hants.  SO51 0ZN
Phone +44 (0)1794 833311   Fax  +44 (0)1794 833433
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Subject: RE: [rohc] a dummy application protocol (DAP) for SigComp interop
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Hi Mark and all,

It looks we are pretty much in agreement. Due to my
bandwidth constraints, I suggest we divide the work as follows:

- You guys will go ahead with the example test cases using DAP.
- During that process and interop tests, we probably will 
discover cases where DAP needs to modified/enhanced (e.g. more
instructions from sender to receiver). I can then take care of 
updating DAP with new versions.

BR,
Zhigang
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Hi Zacco,

It seems you may be confused between a DAP message and its
message-body. Just in case, it is the entire DAP message 
(header + CRLF + [body]) that goes through SigComp layer 
(both the sending side and the receving side).

Detailed answer below.

Zhigang

> 1. How can endpoint A now if decompression failure occured on 
> endpoint B? 

It knows if it doesn't receive the uncompressed message retured
from B. Of course, this assumes we don't drop the returned message.

> Empty message is not good for this purpose, as it 
> may be a valid output. For negative tests, it is required to 
> know somehow.

There is no empty DAP message, only DAP message with empty 
message-body. The message-header is alwasy carried in each DAP message.
See above.

> 2. When the uncompressed message is sent back with the same 
> header, what length should I use, the original (compressed) 
> or the actual (uncompressed) message length?

The DAP receiver simply takes whatever is delivered by the local
SigComp decompressor, check its format and then return it back 
without any modifications and without going through the local 
sigcomp compressor.
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Subject: RE: [rohc] a dummy application protocol (DAP) for SigComp interop
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Zhigang,

This seems like an effective use of resources!

Glad to see we're converging on an answer... ;-)
(And that we definitely have enough to do for the interop)

Cheers,

Mark.

> -----Original Message-----
> From: zhigang.c.liu@nokia.com [mailto:zhigang.c.liu@nokia.com]
> Sent: 06 February 2003 19:45
> To: West, Mark (ITN); Surtees, Abigail; rohc@ietf.org
> Subject: RE: [rohc] a dummy application protocol (DAP) for SigComp
> interop
> 
> 
> Hi Mark and all,
> 
> It looks we are pretty much in agreement. Due to my
> bandwidth constraints, I suggest we divide the work as follows:
> 
> - You guys will go ahead with the example test cases using DAP.
> - During that process and interop tests, we probably will 
> discover cases where DAP needs to modified/enhanced (e.g. more
> instructions from sender to receiver). I can then take care of 
> updating DAP with new versions.
> 
> BR,
> Zhigang
> 
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Hi Richard,

A quick question. Does the following mean a sender can
assume DMS on the receiving side = 131072?

-----------------------------
iii) Long messages (up to 64K over UDP)
------------------------------

Zhigang
 
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Hi,

(This is not specified on RFC 3320, so I want to double check.)

If a message carries state creation request but the receiving 
endpoint has SMS = 0, I understand that the message can still be
accepted but the state creation request will be rejected.

Now, what about the requested feedback carried in the message?
Will it be returned? If yes, that means the sender must not 
rely on feedback to determine if the state has been successfully
created. Then, is the answer no?

Zhigang

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From mailnull@www1.ietf.org  Fri Feb  7 02:27:43 2003
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hi,

According to SigComp user guide, my interpretation is that the receiving 
endpoint returns the acked_state_id
in the "returned feedback item" field of SigComp message. However, when 
referring to RFC 3321, section
6.1, Figure 5 shows that the acked_state_id is in the "Remaining SigComp 
message" field.

Does it mean that acked_state_id can exist in either location in a SigComp 
message ?

regards,
Kwang Mien




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From: "Lajos Zaccomer (ETH)" <Lajos.Zaccomer@eth.ericsson.se>
To: "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>,
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Hi,

No, I'm not confused about what to send. The fact is that some parts are simply not clear enough. See my inline comments below!
 
BR/Zacco

-----Original Message-----
From: zhigang.c.liu@nokia.com [mailto:zhigang.c.liu@nokia.com]
Sent: Thursday, February 06, 2003 9:02 PM
To: Lajos.Zaccomer@eth.ericsson.se; mark.a.west@roke.co.uk; abigail.surtees@roke.co.uk; rohc@ietf.org
Subject: RE: [rohc] a dummy application protocol (DAP) for SigComp interop


Hi Zacco,

It seems you may be confused between a DAP message and its
message-body. Just in case, it is the entire DAP message 
(header + CRLF + [body]) that goes through SigComp layer 
(both the sending side and the receving side).

Detailed answer below.

Zhigang

> 1. How can endpoint A now if decompression failure occured on 
> endpoint B? 

It knows if it doesn't receive the uncompressed message retured
from B. Of course, this assumes we don't drop the returned message.

[L.Z.] Explicit notification would be better for testing.We could do it very simply in the verification message that is always sent back.

> Empty message is not good for this purpose, as it 
> may be a valid output. For negative tests, it is required to 
> know somehow.

There is no empty DAP message, only DAP message with empty 
message-body. The message-header is alwasy carried in each DAP message.
See above.

[L.Z.] You misunderstood me. I wanted to say: an empty string output of the UDVM did't mean decompression failure. Of course, it didn't. But it's useless to build a timeout mechanism to check decompression failure, when we have the possibility of an explicit notification. Abbie, this explicit notificatrion is nothing to do with NACK, as it is not part of SigComp but DAP.

> 2. When the uncompressed message is sent back with the same 
> header, what length should I use, the original (compressed) 
> or the actual (uncompressed) message length?

The DAP receiver simply takes whatever is delivered by the local
SigComp decompressor, check its format and then return it back 
without any modifications and without going through the local 
sigcomp compressor.

[L.Z.] I asked something else. Abbie had an answer that I already knew: it didn't matter. I expected DAP a simple protocol without having optional things in it. For testing, it would have been favourable. I fear we have to spend long hours with adjusting our DAP versions because of the freedom DAP allows.
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From mailnull@www1.ietf.org  Fri Feb  7 04:39:33 2003
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From: "Price, Richard" <richard.price@roke.co.uk>
To: "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>, rohc@ietf.org
Subject: RE: [rohc] SigComp Interop
Date: Fri, 7 Feb 2003 09:41:59 -0000 
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Hi Zhigang,

In general, the sender should assume that DMS = 2048 unless
the receiver specifically agrees to offer more (either by
prior agreement, or inband using the SigComp feedback mechanism).

For this particular case, it would be useful to test the
sender behaviour for both extremes (DMS = 2048 and DMS = 131072).
If the receiver offers insufficient memory then the sender
should just send the long message uncompressed.

Regards,

Richard

> -----Original Message-----
> From: zhigang.c.liu@nokia.com [mailto:zhigang.c.liu@nokia.com]
> Sent: Thursday, February 06, 2003 11:38 PM
> To: Price, Richard; rohc@ietf.org
> Subject: RE: [rohc] SigComp Interop
> 
> 
> Hi Richard,
> 
> A quick question. Does the following mean a sender can
> assume DMS on the receiving side = 131072?
> 
> -----------------------------
> iii) Long messages (up to 64K over UDP)
> ------------------------------
> 
> Zhigang
>  
> 
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From mailnull@www1.ietf.org  Fri Feb  7 05:11:04 2003
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From: "Price, Richard" <richard.price@roke.co.uk>
To: "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>, rohc@ietf.org
Subject: RE: [rohc] SigComp state creation request, requested  feedback, a
	nd SMS = 0
Date: Fri, 7 Feb 2003 09:58:55 -0000 
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> Hi,
> 
> (This is not specified on RFC 3320, so I want to double check.)
> 
> If a message carries state creation request but the receiving 
> endpoint has SMS = 0, I understand that the message can still be
> accepted but the state creation request will be rejected.

Hi Zhigang,

That's right - state creation requests can be rejected due to lack
of memory, even if the decompressed message is accepted.

> Now, what about the requested feedback carried in the message?
> Will it be returned?

The requested feedback can still be returned even if SMS = 0.

> If yes, that means the sender must not rely on feedback to determine
> if the state has been successfully created. Then, is the answer no?

The sender can still rely on the feedback to determine if the state
has been successfully created, but only if it takes into account the
returned SigComp parameters as well. In particular it needs to check
that the receiver has advertised SMS > 0.

SigComp-Extended has the following to say on the subject:

   Note: There is a possibility that state(S) is discarded due to lack
   of state memory even though the announcement information is
   successfully forwarded.  This possibility must be taken into account
   (otherwise a decompression failure may occur); this can be done by
   using the SigComp parameter state_memory_size which is announced by
   the SigComp feedback mechanism.  The endpoint can use this parameter
   to infer if a state creation request has failed due to lack of
   memory.

Hope this helps!

Regards,

Richard
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From: "Price, Richard" <richard.price@roke.co.uk>
To: "'Chan Kwang Mien'" <chankm@icr.a-star.edu.sg>, rohc@ietf.org
Subject: RE: [rohc] Query about Returned Feedback
Date: Fri, 7 Feb 2003 10:12:53 -0000 
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> hi,
> 
> According to SigComp user guide, my interpretation is that 
> the receiving endpoint returns the acked_state_id
> in the "returned feedback item" field of SigComp message. 
> However, when referring to RFC 3321, section
> 6.1, Figure 5 shows that the acked_state_id is in the 
> "Remaining SigComp message" field.
> 
> Does it mean that acked_state_id can exist in either location 
> in a SigComp message ?

Hi Kwang Mien,

That's exactly right - the acked_state_id can be carried either
in the "returned feedback item", or in the main body of the
SigComp message. Where the acked_state_id goes is chosen by the
bytecode that is uploaded to the UDVM.

The SigComp User Guide contains bytecode to implement both
cases (see Section 6.1 and Section 6.5 for more details).

Regards,

Richard
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From mailnull@www1.ietf.org  Fri Feb  7 05:17:23 2003
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From: "Lajos Zaccomer (ETH)" <Lajos.Zaccomer@eth.ericsson.se>
To: "'Price, Richard'" <richard.price@roke.co.uk>,
        "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>, rohc@ietf.org
Subject: RE: [rohc] SigComp state creation request, requested  feedback, a
	 nd SMS = 0
Date: Fri, 7 Feb 2003 11:17:30 +0100 
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Hi,

This may help, but sending the returned parameters is not mandatory, moreover not recommended!

BR/Zacco

-----Original Message-----
From: Price, Richard [mailto:richard.price@roke.co.uk]
Sent: Friday, February 07, 2003 10:59 AM
To: 'zhigang.c.liu@nokia.com'; rohc@ietf.org
Subject: RE: [rohc] SigComp state creation request, requested feedback, a nd SMS = 0


> Hi,
> 
> (This is not specified on RFC 3320, so I want to double check.)
> 
> If a message carries state creation request but the receiving 
> endpoint has SMS = 0, I understand that the message can still be
> accepted but the state creation request will be rejected.

Hi Zhigang,

That's right - state creation requests can be rejected due to lack
of memory, even if the decompressed message is accepted.

> Now, what about the requested feedback carried in the message?
> Will it be returned?

The requested feedback can still be returned even if SMS = 0.

> If yes, that means the sender must not rely on feedback to determine
> if the state has been successfully created. Then, is the answer no?

The sender can still rely on the feedback to determine if the state
has been successfully created, but only if it takes into account the
returned SigComp parameters as well. In particular it needs to check
that the receiver has advertised SMS > 0.

SigComp-Extended has the following to say on the subject:

   Note: There is a possibility that state(S) is discarded due to lack
   of state memory even though the announcement information is
   successfully forwarded.  This possibility must be taken into account
   (otherwise a decompression failure may occur); this can be done by
   using the SigComp parameter state_memory_size which is announced by
   the SigComp feedback mechanism.  The endpoint can use this parameter
   to infer if a state creation request has failed due to lack of
   memory.

Hope this helps!

Regards,

Richard
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From mailnull@www1.ietf.org  Fri Feb  7 05:49:28 2003
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From: "Price, Richard" <richard.price@roke.co.uk>
To: "'Lajos Zaccomer (ETH)'" <Lajos.Zaccomer@eth.ericsson.se>,
        "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>, rohc@ietf.org
Subject: RE: [rohc] SigComp state creation request, requested  feedback, a
	 nd SMS = 0
Date: Fri, 7 Feb 2003 10:51:26 -0000 
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Hi Zacco,

It's an implementation decision at the receiver whether to
announce that SMS > 0 or not. Note however that if the
receiver chooses not to do this, then the sender MUST
assume that SMS = 0 (in other words it has to assume that
the receiver can't save state).

Regards,

Richard

> -----Original Message-----
> From: Lajos Zaccomer (ETH) [mailto:Lajos.Zaccomer@eth.ericsson.se]
> Sent: Friday, February 07, 2003 10:18 AM
> To: Price, Richard; 'zhigang.c.liu@nokia.com'; rohc@ietf.org
> Subject: RE: [rohc] SigComp state creation request, requested 
> feedback,
> a nd SMS = 0
> 
> 
> Hi,
> 
> This may help, but sending the returned parameters is not 
> mandatory, moreover not recommended!
> 
> BR/Zacco
> 
> -----Original Message-----
> From: Price, Richard [mailto:richard.price@roke.co.uk]
> Sent: Friday, February 07, 2003 10:59 AM
> To: 'zhigang.c.liu@nokia.com'; rohc@ietf.org
> Subject: RE: [rohc] SigComp state creation request, requested 
> feedback, a nd SMS = 0
> 
> 
> > Hi,
> > 
> > (This is not specified on RFC 3320, so I want to double check.)
> > 
> > If a message carries state creation request but the receiving 
> > endpoint has SMS = 0, I understand that the message can still be
> > accepted but the state creation request will be rejected.
> 
> Hi Zhigang,
> 
> That's right - state creation requests can be rejected due to lack
> of memory, even if the decompressed message is accepted.
> 
> > Now, what about the requested feedback carried in the message?
> > Will it be returned?
> 
> The requested feedback can still be returned even if SMS = 0.
> 
> > If yes, that means the sender must not rely on feedback to determine
> > if the state has been successfully created. Then, is the answer no?
> 
> The sender can still rely on the feedback to determine if the state
> has been successfully created, but only if it takes into account the
> returned SigComp parameters as well. In particular it needs to check
> that the receiver has advertised SMS > 0.
> 
> SigComp-Extended has the following to say on the subject:
> 
>    Note: There is a possibility that state(S) is discarded due to lack
>    of state memory even though the announcement information is
>    successfully forwarded.  This possibility must be taken into account
>    (otherwise a decompression failure may occur); this can be done by
>    using the SigComp parameter state_memory_size which is announced by
>    the SigComp feedback mechanism.  The endpoint can use this parameter
>    to infer if a state creation request has failed due to lack of
>    memory.
> 
> Hope this helps!
> 
> Regards,
> 
> Richard
> _______________________________________________
> Rohc mailing list
> Rohc@ietf.org
> https://www1.ietf.org/mailman/listinfo/rohc
> 
_______________________________________________
Rohc mailing list
Rohc@ietf.org
https://www1.ietf.org/mailman/listinfo/rohc



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Message-ID: <F005CD411D18D3119C8F00508B0874800A1446EC@ehubunt100.eth.ericsson.se>
From: "Lajos Zaccomer (ETH)" <Lajos.Zaccomer@eth.ericsson.se>
To: "'Price, Richard'" <richard.price@roke.co.uk>,
        "Lajos Zaccomer (ETH)"
	 <Lajos.Zaccomer@eth.ericsson.se>,
        "'zhigang.c.liu@nokia.com'"
	 <zhigang.c.liu@nokia.com>, rohc@ietf.org
Subject: RE: [rohc] SigComp state creation request, requested  feedback, a
	 nd SMS = 0
Date: Fri, 7 Feb 2003 12:07:44 +0100 
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Hi,

I am aware of this, but I think it is worth to indicate there may be potential problems or low compression efficiency issues because of this. Think of what happens if B does not send the returned parameters: A has to send the bytecode in every message. In such a situation B proves to perform better than than A, unless A is brave enough not to send the bytecode again.
Anyway, the RFC does not mandate that "it has to assume that the receiver can't save state." (As far as I can remember, it is not even recommended.) It may be logical, but in situations when every byte spared is a success, the temptation is great to make "reckless" implementations that do not care of such things. Assumptions amongst "friend" softwares work well, which may pave the way before these, handicapping others.

BR/Zacco

-----Original Message-----
From: Price, Richard [mailto:richard.price@roke.co.uk]
Sent: Friday, February 07, 2003 11:51 AM
To: 'Lajos Zaccomer (ETH)'; 'zhigang.c.liu@nokia.com'; rohc@ietf.org
Subject: RE: [rohc] SigComp state creation request, requested feedback, a nd SMS = 0


Hi Zacco,

It's an implementation decision at the receiver whether to
announce that SMS > 0 or not. Note however that if the
receiver chooses not to do this, then the sender MUST
assume that SMS = 0 (in other words it has to assume that
the receiver can't save state).

Regards,

Richard

> -----Original Message-----
> From: Lajos Zaccomer (ETH) [mailto:Lajos.Zaccomer@eth.ericsson.se]
> Sent: Friday, February 07, 2003 10:18 AM
> To: Price, Richard; 'zhigang.c.liu@nokia.com'; rohc@ietf.org
> Subject: RE: [rohc] SigComp state creation request, requested 
> feedback,
> a nd SMS = 0
> 
> 
> Hi,
> 
> This may help, but sending the returned parameters is not 
> mandatory, moreover not recommended!
> 
> BR/Zacco
> 
> -----Original Message-----
> From: Price, Richard [mailto:richard.price@roke.co.uk]
> Sent: Friday, February 07, 2003 10:59 AM
> To: 'zhigang.c.liu@nokia.com'; rohc@ietf.org
> Subject: RE: [rohc] SigComp state creation request, requested 
> feedback, a nd SMS = 0
> 
> 
> > Hi,
> > 
> > (This is not specified on RFC 3320, so I want to double check.)
> > 
> > If a message carries state creation request but the receiving 
> > endpoint has SMS = 0, I understand that the message can still be
> > accepted but the state creation request will be rejected.
> 
> Hi Zhigang,
> 
> That's right - state creation requests can be rejected due to lack
> of memory, even if the decompressed message is accepted.
> 
> > Now, what about the requested feedback carried in the message?
> > Will it be returned?
> 
> The requested feedback can still be returned even if SMS = 0.
> 
> > If yes, that means the sender must not rely on feedback to determine
> > if the state has been successfully created. Then, is the answer no?
> 
> The sender can still rely on the feedback to determine if the state
> has been successfully created, but only if it takes into account the
> returned SigComp parameters as well. In particular it needs to check
> that the receiver has advertised SMS > 0.
> 
> SigComp-Extended has the following to say on the subject:
> 
>    Note: There is a possibility that state(S) is discarded due to lack
>    of state memory even though the announcement information is
>    successfully forwarded.  This possibility must be taken into account
>    (otherwise a decompression failure may occur); this can be done by
>    using the SigComp parameter state_memory_size which is announced by
>    the SigComp feedback mechanism.  The endpoint can use this parameter
>    to infer if a state creation request has failed due to lack of
>    memory.
> 
> Hope this helps!
> 
> Regards,
> 
> Richard
> _______________________________________________
> Rohc mailing list
> Rohc@ietf.org
> https://www1.ietf.org/mailman/listinfo/rohc
> 
_______________________________________________
Rohc mailing list
Rohc@ietf.org
https://www1.ietf.org/mailman/listinfo/rohc



From mailnull@www1.ietf.org  Fri Feb  7 06:26:30 2003
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Reply-To: <vkenneally@cit.ie>
From: "Valerie Kenneally" <vkenneally@cit.ie>
To: <rohc@ietf.org>
Subject: RE: [rohc] SigComp state creation request, requested  feedback, a nd SMS = 0
Date: Fri, 7 Feb 2003 11:28:58 -0000
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Hi all,

	I am trying to understand the feedback mechanism provided by SIGCOMP.  My
question is based on what happens if a remote endpoint receives a message
which it cannot decompress for whatever reason so we have a DECOMPRESSION
FAILURE.

Can the remote end-point notify the sending end-point that decompression has
failed.  The reason I ask this is that the RFC states that feedback is
piggybacked onto a Sigcomp message.

I don't know if I am completely misunderstanding the format of a Sigcomp
message but should it not always contain a compressed message.  If a
DECOMPRESSION FAILURE occurs a follow-up message/response cannot be sent.
So can a Sigcomp message be sent for the sole purpose of DECOMPRESSION
FAILURE
notification without being accompanied by a compressed message?


Any help at all would be greatly appreciated
Regards,
Val

-----Original Message-----
From: rohc-admin@ietf.org [mailto:rohc-admin@ietf.org]On Behalf Of Lajos
Zaccomer (ETH)
Sent: 07 February 2003 11:08
To: 'Price, Richard'; Lajos Zaccomer (ETH); 'zhigang.c.liu@nokia.com';
rohc@ietf.org
Subject: RE: [rohc] SigComp state creation request, requested feedback,
a nd SMS = 0


Hi,

I am aware of this, but I think it is worth to indicate there may be
potential problems or low compression efficiency issues because of this.
Think of what happens if B does not send the returned parameters: A has to
send the bytecode in every message. In such a situation B proves to perform
better than than A, unless A is brave enough not to send the bytecode again.
Anyway, the RFC does not mandate that "it has to assume that the receiver
can't save state." (As far as I can remember, it is not even recommended.)
It may be logical, but in situations when every byte spared is a success,
the temptation is great to make "reckless" implementations that do not care
of such things. Assumptions amongst "friend" softwares work well, which may
pave the way before these, handicapping others.

BR/Zacco

-----Original Message-----
From: Price, Richard [mailto:richard.price@roke.co.uk]
Sent: Friday, February 07, 2003 11:51 AM
To: 'Lajos Zaccomer (ETH)'; 'zhigang.c.liu@nokia.com'; rohc@ietf.org
Subject: RE: [rohc] SigComp state creation request, requested feedback, a nd
SMS = 0


Hi Zacco,

It's an implementation decision at the receiver whether to
announce that SMS > 0 or not. Note however that if the
receiver chooses not to do this, then the sender MUST
assume that SMS = 0 (in other words it has to assume that
the receiver can't save state).

Regards,

Richard

> -----Original Message-----
> From: Lajos Zaccomer (ETH) [mailto:Lajos.Zaccomer@eth.ericsson.se]
> Sent: Friday, February 07, 2003 10:18 AM
> To: Price, Richard; 'zhigang.c.liu@nokia.com'; rohc@ietf.org
> Subject: RE: [rohc] SigComp state creation request, requested
> feedback,
> a nd SMS = 0
>
>
> Hi,
>
> This may help, but sending the returned parameters is not
> mandatory, moreover not recommended!
>
> BR/Zacco
>
> -----Original Message-----
> From: Price, Richard [mailto:richard.price@roke.co.uk]
> Sent: Friday, February 07, 2003 10:59 AM
> To: 'zhigang.c.liu@nokia.com'; rohc@ietf.org
> Subject: RE: [rohc] SigComp state creation request, requested
> feedback, a nd SMS = 0
>
>
> > Hi,
> >
> > (This is not specified on RFC 3320, so I want to double check.)
> >
> > If a message carries state creation request but the receiving
> > endpoint has SMS = 0, I understand that the message can still be
> > accepted but the state creation request will be rejected.
>
> Hi Zhigang,
>
> That's right - state creation requests can be rejected due to lack
> of memory, even if the decompressed message is accepted.
>
> > Now, what about the requested feedback carried in the message?
> > Will it be returned?
>
> The requested feedback can still be returned even if SMS = 0.
>
> > If yes, that means the sender must not rely on feedback to determine
> > if the state has been successfully created. Then, is the answer no?
>
> The sender can still rely on the feedback to determine if the state
> has been successfully created, but only if it takes into account the
> returned SigComp parameters as well. In particular it needs to check
> that the receiver has advertised SMS > 0.
>
> SigComp-Extended has the following to say on the subject:
>
>    Note: There is a possibility that state(S) is discarded due to lack
>    of state memory even though the announcement information is
>    successfully forwarded.  This possibility must be taken into account
>    (otherwise a decompression failure may occur); this can be done by
>    using the SigComp parameter state_memory_size which is announced by
>    the SigComp feedback mechanism.  The endpoint can use this parameter
>    to infer if a state creation request has failed due to lack of
>    memory.
>
> Hope this helps!
>
> Regards,
>
> Richard
> _______________________________________________
> Rohc mailing list
> Rohc@ietf.org
> https://www1.ietf.org/mailman/listinfo/rohc
>
_______________________________________________
Rohc mailing list
Rohc@ietf.org
https://www1.ietf.org/mailman/listinfo/rohc

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From mailnull@www1.ietf.org  Fri Feb  7 07:30:36 2003
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From: "Price, Richard" <richard.price@roke.co.uk>
To: "'Lajos Zaccomer (ETH)'" <Lajos.Zaccomer@eth.ericsson.se>,
        "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>, rohc@ietf.org
Subject: RE: [rohc] SigComp state creation request, requested  feedback, a
	 nd SMS = 0
Date: Fri, 7 Feb 2003 12:32:54 -0000 
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> Hi,
> 
> I am aware of this, but I think it is worth to indicate there 
> may be potential problems or low compression efficiency 
> issues because of this. Think of what happens if B does not 
> send the returned parameters: A has to send the bytecode in 
> every message. In such a situation B proves to perform better 
> than than A, unless A is brave enough not to send the bytecode again.

Hi Zacco,

It's worth pointing out that getting a good compression ratio
is the responsibility of both the sender and the receiver. Thus,
if the compression ratio is worse in the direction A -> B than in
the direction B -> A, it's wrong to claim that B is performing
better than A. It might be B's fault that the compression ratio
is low.

> Anyway, the RFC does not mandate that "it has to assume that 
> the receiver can't save state." (As far as I can remember, it 
> is not even recommended.)

In fact, from Section 5 of RFC 3320:

   "The compressor MUST ensure that the message can be decompressed
   using the resources available at the remote endpoint."

If the compressor assumes that the receiver can save state without
first checking that SMS > 0, then it's broken!

> It may be logical, but in situations when every byte spared
> is a success, the temptation is great to make "reckless"
> implementations that do not care of such things. Assumptions
> amongst "friend" softwares work well, which may pave the way
> before these, handicapping others.

I agree completely with the above - the temptation is great to
cut corners just to save a few bytes. The result is an
implementation that only interoperates with "friendly" software,
which is a Bad Thing!

We took care to specifically ban this kind of behaviour in
RFC 3320.

Regards,

Richard
_______________________________________________
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From: "Surtees, Abigail" <abigail.surtees@roke.co.uk>
To: "'Lajos Zaccomer (ETH)'" <Lajos.Zaccomer@eth.ericsson.se>,
        "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>,
        "West, Mark (ITN)"
	 <mark.a.west@roke.co.uk>, rohc@ietf.org
Subject: RE: [rohc] a dummy application protocol (DAP) for SigComp interop
Date: Fri, 7 Feb 2003 12:46:08 -0000 
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Hi Zacco,

Comments in-line (hopefully we're not getting too indented).

Best regards,

Abbie
>> 1. How can endpoint A now if decompression failure occured on 
>> endpoint B? 
>
>It knows if it doesn't receive the uncompressed message retured
>from B. Of course, this assumes we don't drop the returned message.
>
>[L.Z.] Explicit notification would be better for testing.We 
>could do it very simply in the verification message that is 
>always sent back.
>
>> Empty message is not good for this purpose, as it 
>> may be a valid output. For negative tests, it is required to 
>> know somehow.
>
>There is no empty DAP message, only DAP message with empty 
>message-body. The message-header is alwasy carried in each DAP message.
>See above.
>
>[L.Z.] You misunderstood me. I wanted to say: an empty string 
>output of the UDVM did't mean decompression failure. Of 
>course, it didn't. But it's useless to build a timeout 
>mechanism to check decompression failure, when we have the 
>possibility of an explicit notification. Abbie, this explicit 
>notificatrion is nothing to do with NACK, as it is not part of 
>SigComp but DAP.
>
[A.S.] I think we are getting confused between empty messages and DECOMPRESSION_FAILURES occurring in the torture tests and those occurring during interoperation between 2 endpoints.
In the first case we are not using DAP and these occurrences are either expected (in which case it's fine) or they're not (in which case the implementer has something to debug).  In the second case (which is where we will be at SIPit) there are, as Zhigang said, no empty messages because there will always be a DAP header and if there is DECOMPRESSION_FAILURE then (assuming no dropped packets) there is a bug at one endpoint or the other, at which point both implementers need to look at their log files and work out what has gone wrong.

From my experience at 3095 interop tests I would say that endpoint A doesn't need to be told there has been a DECOMPRESSION_FAILURE.  I would expect that the decompressing endpoint would flag up a DECOMPRESSION_FAILURE to the implementer either in an on-screen message or in a log file.  If it's the former then the implementer will spot it and the test can be stopped and debugged or if it's the latter then test may go slightly awry but this will be noted by one end or the other and it can be debugged.  As both implementers will be present there can be discussions rather than requiring one endpoint to indicate everything to the other.
>> 2. When the uncompressed message is sent back with the same 
>> header, what length should I use, the original (compressed) 
>> or the actual (uncompressed) message length?
>
>The DAP receiver simply takes whatever is delivered by the local
>SigComp decompressor, check its format and then return it back 
>without any modifications and without going through the local 
>sigcomp compressor.
>
>[L.Z.] I asked something else. Abbie had an answer that I 
>already knew: it didn't matter. I expected DAP a simple 
>protocol without having optional things in it. For testing, it 
>would have been favourable. I fear we have to spend long hours 
>with adjusting our DAP versions because of the freedom DAP allows.
>
[A.S.]  I'm not sure I understand the question.  There aren't any options in DAP.  The length is always the uncompressed length because the DAP header is part of what gets compressed (in the sending direction) and the message is just returned "as is" in the uncompressed direction.
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From: "Lajos Zaccomer (ETH)" <Lajos.Zaccomer@eth.ericsson.se>
To: "'Surtees, Abigail'" <abigail.surtees@roke.co.uk>,
        "Lajos Zaccomer (ETH)" <Lajos.Zaccomer@eth.ericsson.se>,
        "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>,
        "West, Mark (ITN)"
	 <mark.a.west@roke.co.uk>, rohc@ietf.org
Subject: RE: [rohc] a dummy application protocol (DAP) for SigComp interop
Date: Fri, 7 Feb 2003 14:05:44 +0100 
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Hi,

Regarding decomp failure:

In negative tests I may send such messages _on purpose_. This is the case in several of the torture tests. For automatic test suits that can be run without any user intervention, it is essential to have all the information to set  verdict without having a look at the other implementation's log file or consulting its owner.

Regarding length I quote what you wrote:

"As there is no new DAP header being added there is no need to worry about what length to give it."
I don't think that "The length is always the uncompressed length" especially when sending the compressed message. I guess you did not mean this.

BR/Zacco



-----Original Message-----
From: Surtees, Abigail [mailto:abigail.surtees@roke.co.uk]
Sent: Friday, February 07, 2003 1:46 PM
To: 'Lajos Zaccomer (ETH)'; 'zhigang.c.liu@nokia.com'; West, Mark (ITN); rohc@ietf.org
Subject: RE: [rohc] a dummy application protocol (DAP) for SigComp interop


Hi Zacco,

Comments in-line (hopefully we're not getting too indented).

Best regards,

Abbie
>> 1. How can endpoint A now if decompression failure occured on 
>> endpoint B? 
>
>It knows if it doesn't receive the uncompressed message retured
>from B. Of course, this assumes we don't drop the returned message.
>
>[L.Z.] Explicit notification would be better for testing.We 
>could do it very simply in the verification message that is 
>always sent back.
>
>> Empty message is not good for this purpose, as it 
>> may be a valid output. For negative tests, it is required to 
>> know somehow.
>
>There is no empty DAP message, only DAP message with empty 
>message-body. The message-header is alwasy carried in each DAP message.
>See above.
>
>[L.Z.] You misunderstood me. I wanted to say: an empty string 
>output of the UDVM did't mean decompression failure. Of 
>course, it didn't. But it's useless to build a timeout 
>mechanism to check decompression failure, when we have the 
>possibility of an explicit notification. Abbie, this explicit 
>notificatrion is nothing to do with NACK, as it is not part of 
>SigComp but DAP.
>
[A.S.] I think we are getting confused between empty messages and DECOMPRESSION_FAILURES occurring in the torture tests and those occurring during interoperation between 2 endpoints.
In the first case we are not using DAP and these occurrences are either expected (in which case it's fine) or they're not (in which case the implementer has something to debug).  In the second case (which is where we will be at SIPit) there are, as Zhigang said, no empty messages because there will always be a DAP header and if there is DECOMPRESSION_FAILURE then (assuming no dropped packets) there is a bug at one endpoint or the other, at which point both implementers need to look at their log files and work out what has gone wrong.

From my experience at 3095 interop tests I would say that endpoint A doesn't need to be told there has been a DECOMPRESSION_FAILURE.  I would expect that the decompressing endpoint would flag up a DECOMPRESSION_FAILURE to the implementer either in an on-screen message or in a log file.  If it's the former then the implementer will spot it and the test can be stopped and debugged or if it's the latter then test may go slightly awry but this will be noted by one end or the other and it can be debugged.  As both implementers will be present there can be discussions rather than requiring one endpoint to indicate everything to the other.
>> 2. When the uncompressed message is sent back with the same 
>> header, what length should I use, the original (compressed) 
>> or the actual (uncompressed) message length?
>
>The DAP receiver simply takes whatever is delivered by the local
>SigComp decompressor, check its format and then return it back 
>without any modifications and without going through the local 
>sigcomp compressor.
>
>[L.Z.] I asked something else. Abbie had an answer that I 
>already knew: it didn't matter. I expected DAP a simple 
>protocol without having optional things in it. For testing, it 
>would have been favourable. I fear we have to spend long hours 
>with adjusting our DAP versions because of the freedom DAP allows.
>
[A.S.]  I'm not sure I understand the question.  There aren't any options in DAP.  The length is always the uncompressed length because the DAP header is part of what gets compressed (in the sending direction) and the message is just returned "as is" in the uncompressed direction.
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From mailnull@www1.ietf.org  Fri Feb  7 08:22:41 2003
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From: "Price, Richard" <richard.price@roke.co.uk>
To: "'vkenneally@cit.ie'" <vkenneally@cit.ie>, rohc@ietf.org
Subject: RE: [rohc] SigComp state creation request, requested  feedback, a
	 nd SMS = 0
Date: Fri, 7 Feb 2003 13:24:57 -0000 
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Hi Val,

Basic SigComp (RFC 3320) doesn't have the ability to notify the remote
endpoint when decompression failure has occurred. The reason for this is
simply that if a SigComp message fails to decompress, how can we work
out which remote endpoint sent it? We can't successfully decompress the
"From:" header, so we don't know where to send the feedback.

In some cases it might be possible to work around this problem by
using the IP address and port - a proposal to do this can be found in:

http://www.ietf.org/internet-drafts/draft-roach-sigcomp-nack-00.txt

However, we haven't got around to standardising the above mechanism
just yet...

Regards,

Richard

> -----Original Message-----
> From: Valerie Kenneally [mailto:vkenneally@cit.ie]
> Sent: Friday, February 07, 2003 11:29 AM
> To: rohc@ietf.org
> Subject: RE: [rohc] SigComp state creation request, requested 
> feedback,
> a nd SMS = 0
> 
> 
> Hi all,
> 
> 	I am trying to understand the feedback mechanism provided by SIGCOMP.  My
> question is based on what happens if a remote endpoint receives a message
> which it cannot decompress for whatever reason so we have a DECOMPRESSION
> FAILURE.
> 
> Can the remote end-point notify the sending end-point that decompression has
> failed.  The reason I ask this is that the RFC states that feedback is
> piggybacked onto a Sigcomp message.
> 
> I don't know if I am completely misunderstanding the format of a Sigcomp
> message but should it not always contain a compressed message.  If a
> DECOMPRESSION FAILURE occurs a follow-up message/response cannot be sent.
> So can a Sigcomp message be sent for the sole purpose of DECOMPRESSION
> FAILURE notification without being accompanied by a compressed message?
> 
> 
> Any help at all would be greatly appreciated
> Regards,
> Val
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From: "Surtees, Abigail" <abigail.surtees@roke.co.uk>
To: "'Lajos Zaccomer (ETH)'" <Lajos.Zaccomer@eth.ericsson.se>,
        "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>,
        "West, Mark (ITN)"
	 <mark.a.west@roke.co.uk>, rohc@ietf.org
Subject: RE: [rohc] a dummy application protocol (DAP) for SigComp interop
Date: Fri, 7 Feb 2003 13:40:42 -0000 
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Hi Zacco,

>Regarding decomp failure:
>
>In negative tests I may send such messages _on purpose_. This 
>is the case in several of the torture tests. For automatic 
>test suits that can be run without any user intervention, it 
>is essential to have all the information to set  verdict 
>without having a look at the other implementation's log file 
>or consulting its owner.

I agree that the torture tests can quite legitimately produce decompression failure.  However, these don't run with DAP, they are stand alone messages.  

When we are running with DAP (assuming no corruption or packet loss) there shouldn't be any decompression failure.  Why do we need to run without any intervention?  The implementers are all going to be at SIPit.
As we said before, we can extend DAP in the future if we want to, but we have enough for the interop tests which are fast approaching, so we should freeze DAP in order to implement it.

>
>Regarding length I quote what you wrote:
>
>"As there is no new DAP header being added there is no need to 
>worry about what length to give it."
>I don't think that "The length is always the uncompressed 
>length" especially when sending the compressed message. I 
>guess you did not mean this.
>
I'm afraid I still don't understand the question you are asking. 

The diagram below shows how I see things working (correct me if I'm wrong) and there is know need to know about the length to give the returned uncompressed header.

(I am assuming that endpoint A has a body_msg of length x already known from somewhere (e.g. file))

Endpoint A

 				overall msg			overall length

                        <----x-------> 
				|body_msg    |            	x


DAP layer: adds a DAP header (length y) with line 8: "body-length: x"

			<--y--><-------x--->
			|DAP	|body_msg    |           	x + y

Sigcomp layer: compresses (x+y) bytes to z bytes

			        <------z---->
			        |Sigcomp msg| 			z


z bytes are sent to endpoint B.

endpoint B:
	 		        <------z---->
			        |Sigcomp msg| 			z

Sigcomp layer: decompresses z bytes to (x + y) bytes

			<--y--><-------x--->
			|DAP	|body_msg    |           	x + y

DAP layer: reads and verifies DAP header (length y) with line 8: "body-length: x"
		returns the (x + y) bytes it received uncompressed to endpoint A.

			<--y--><-------x--->
			|DAP	|body_msg    |           	x + y

Endpoint A receives the original message it compressed.

Nowhere has the receiving endpoint had to fill in a length value.  Hence my not understanding the question.  There is nothing optional in DAP.

Best regards,

Abbie

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Subject: RE: [rohc] SigComp Interop
Date: Fri, 7 Feb 2003 11:24:53 -0600
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So, the answer is that just for this test case, the sender
can assume the receiver support DMS = 131072?

Zhigang

> -----Original Message-----
> From: ext Price, Richard [mailto:richard.price@roke.co.uk]
> Sent: February 07, 2003 3:42 AM
> To: Liu Zhigang.C (NRC/Dallas); rohc@ietf.org
> Subject: RE: [rohc] SigComp Interop
> 
> 
> Hi Zhigang,
> 
> In general, the sender should assume that DMS = 2048 unless
> the receiver specifically agrees to offer more (either by
> prior agreement, or inband using the SigComp feedback mechanism).
> 
> For this particular case, it would be useful to test the
> sender behaviour for both extremes (DMS = 2048 and DMS = 131072).
> If the receiver offers insufficient memory then the sender
> should just send the long message uncompressed.
> 
> Regards,
> 
> Richard
> 
> > -----Original Message-----
> > From: zhigang.c.liu@nokia.com [mailto:zhigang.c.liu@nokia.com]
> > Sent: Thursday, February 06, 2003 11:38 PM
> > To: Price, Richard; rohc@ietf.org
> > Subject: RE: [rohc] SigComp Interop
> > 
> > 
> > Hi Richard,
> > 
> > A quick question. Does the following mean a sender can
> > assume DMS on the receiving side = 131072?
> > 
> > -----------------------------
> > iii) Long messages (up to 64K over UDP)
> > ------------------------------
> > 
> > Zhigang
> >  
> > 
> 
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From: Chan Kwang Mien <chankm@icr.a-star.edu.sg>
Subject: RE: [rohc] Query about Returned Feedback
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hi Richard,

Can the shared_state_id also be included in the "returned feedback item" 
field ?
for e.g  A sends a compressed message to B. B decompresses and saves the state.
In the next SigComp message to B, A includes the shared_state_id of the 
previous message
in the "returned feedback item" field.

i have the impression that the "returned feedback item" field should only 
contain items requested by the
other remote host. is that true ?

regards,
Kwang Mien


> > hi,
> >
> > According to SigComp user guide, my interpretation is that
> > the receiving endpoint returns the acked_state_id
> > in the "returned feedback item" field of SigComp message.
> > However, when referring to RFC 3321, section
> > 6.1, Figure 5 shows that the acked_state_id is in the
> > "Remaining SigComp message" field.
> >
> > Does it mean that acked_state_id can exist in either location
> > in a SigComp message ?
>
>Hi Kwang Mien,
>
>That's exactly right - the acked_state_id can be carried either
>in the "returned feedback item", or in the main body of the
>SigComp message. Where the acked_state_id goes is chosen by the
>bytecode that is uploaded to the UDVM.
>
>The SigComp User Guide contains bytecode to implement both
>cases (see Section 6.1 and Section 6.5 for more details).
>
>Regards,
>
>Richard

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From mailnull@www1.ietf.org  Mon Feb 10 12:19:03 2003
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From: "Price, Richard" <richard.price@roke.co.uk>
To: "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>, rohc@ietf.org
Subject: RE: [rohc] SigComp Interop
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Hi Zhigang,

If possible I'd like to test both cases. Firstly, the receiver
should offer DMS = 131072 and the sender should compress the
long message using as much memory as it likes.

Secondly, the receiver should offer DMS = 2048 to see what
happens. If the sender is implemented correctly then
decompression failure won't occur (if the sender can't compress
the long message with DMS = 2048 then it should just send it
uncompressed).

Regards,

Richard

> -----Original Message-----
> From: zhigang.c.liu@nokia.com [mailto:zhigang.c.liu@nokia.com]
> Sent: Friday, February 07, 2003 5:25 PM
> To: Price, Richard; rohc@ietf.org
> Subject: RE: [rohc] SigComp Interop
> 
> 
> So, the answer is that just for this test case, the sender
> can assume the receiver support DMS = 131072?
> 
> Zhigang
> 
> > -----Original Message-----
> > From: ext Price, Richard [mailto:richard.price@roke.co.uk]
> > Sent: February 07, 2003 3:42 AM
> > To: Liu Zhigang.C (NRC/Dallas); rohc@ietf.org
> > Subject: RE: [rohc] SigComp Interop
> > 
> > 
> > Hi Zhigang,
> > 
> > In general, the sender should assume that DMS = 2048 unless
> > the receiver specifically agrees to offer more (either by
> > prior agreement, or inband using the SigComp feedback mechanism).
> > 
> > For this particular case, it would be useful to test the
> > sender behaviour for both extremes (DMS = 2048 and DMS = 131072).
> > If the receiver offers insufficient memory then the sender
> > should just send the long message uncompressed.
> > 
> > Regards,
> > 
> > Richard
> > 
> > > -----Original Message-----
> > > From: zhigang.c.liu@nokia.com [mailto:zhigang.c.liu@nokia.com]
> > > Sent: Thursday, February 06, 2003 11:38 PM
> > > To: Price, Richard; rohc@ietf.org
> > > Subject: RE: [rohc] SigComp Interop
> > > 
> > > 
> > > Hi Richard,
> > > 
> > > A quick question. Does the following mean a sender can
> > > assume DMS on the receiving side = 131072?
> > > 
> > > -----------------------------
> > > iii) Long messages (up to 64K over UDP)
> > > ------------------------------
> > > 
> > > Zhigang
> > >  
> > > 
> > 
> 
_______________________________________________
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From: "Price, Richard" <richard.price@roke.co.uk>
To: "'Chan Kwang Mien'" <chankm@icr.a-star.edu.sg>, rohc@ietf.org
Subject: RE: [rohc] Query about Returned Feedback
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Hi Kwang Mien,

The "returned feedback item" field sent in the direction A -> B is
controlled by the endpoint B. So it will contain the shared_state_id
if and only if endpoint B requests that this should be the case
(by sending the appropriate bytecode to endpoint A).

Endpoint A must do exactly what B tells it, so it MUST NOT include
the shared_state_id unless B asks it to.

Regards,

Richard

> -----Original Message-----
> From: Chan Kwang Mien [mailto:chankm@icr.a-star.edu.sg]
> Sent: Monday, February 10, 2003 3:28 AM
> To: Price, Richard; rohc@ietf.org
> Subject: RE: [rohc] Query about Returned Feedback
> 
> 
> 
> hi Richard,
> 
> Can the shared_state_id also be included in the "returned feedback item" 
> field ? for e.g  A sends a compressed message to B. B decompresses 
> and saves the state. In the next SigComp message to B, A includes the 
> shared_state_id of the previous message in the "returned feedback item"
> field.
> 
> i have the impression that the "returned feedback item" field should only 
> contain items requested by the other remote host. is that true ?
> 
> regards,
> Kwang Mien
_______________________________________________
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https://www1.ietf.org/mailman/listinfo/rohc



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Subject: RE: [rohc] SigComp Interop
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Thread-Topic: [rohc] SigComp Interop
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> Secondly, the receiver should offer DMS = 2048 to see what
> happens. If the sender is implemented correctly then
> decompression failure won't occur (if the sender can't compress
> the long message with DMS = 2048 then it should just send it
> uncompressed).

I don't think sender can ever compress a 64KB message, unless
it can compress it down to something like 1KB. By definition,
part of DMS is used to buffer the compressed message.

So, basically if the receiver does not advertise any DMS,
this test is to check if the sender is "smart" enough to 
send the message uncompressed.

Zhigang
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Hi
Can some one tell me how the packet is updated on the decompressor side
when there is packet loss in the transit. I would really appreciate some
pointers in this regard.

thanks
alphonse






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Subject: RE: [rohc] context update in case of packet loss
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Thread-Topic: [rohc] context update in case of packet loss
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From: "Ashok Kumar" <ashok.kum@wipro.com>
To: <alreddy@hss.hns.com>, <rohc@ietf.org>
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Hi Alphonse,

When a packet is lost in transit, decompressor will not receive the
packet. So nothing needs to be done at decompressor. The subsequent
packets received by decompressor will indicate how the context should be
updated at decompressor.

-Ashok

-----Original Message-----
From: alreddy@hss.hns.com [mailto:alreddy@hss.hns.com] 
Sent: Tuesday, February 11, 2003 11:44 AM
To: rohc@ietf.org
Subject: [rohc] context update in case of packet loss




Hi
Can some one tell me how the packet is updated on the decompressor side
when there is packet loss in the transit. I would really appreciate some
pointers in this regard.

thanks
alphonse






This message is proprietary to Hughes Software Systems Limited (HSS) and
is
intended solely for the use of the individual to whom it is addressed.
It
may contain privileged or confidential information and should not be
circulated or used for any purpose other than for what it is intended.
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you are strictly prohibited from using, copying, altering, or disclosing
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damage arising from the use of the information transmitted by this email
including damage from virus.


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From: "Lajos Zaccomer (ETH)" <Lajos.Zaccomer@eth.ericsson.se>
To: "'zhigang.c.liu@nokia.com'" <zhigang.c.liu@nokia.com>,
        richard.price@roke.co.uk, rohc@ietf.org
Subject: RE: [rohc] SigComp Interop
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Hi,

I don't think this is an interoperability issue. It can be checked fine alone.
It is true that decomp failure does not happen, but compression failure does. That is why it is an internal affair.

BR/Zacco

-----Original Message-----
From: zhigang.c.liu@nokia.com [mailto:zhigang.c.liu@nokia.com]
Sent: Monday, February 10, 2003 6:42 PM
To: richard.price@roke.co.uk; rohc@ietf.org
Subject: RE: [rohc] SigComp Interop


> Secondly, the receiver should offer DMS = 2048 to see what
> happens. If the sender is implemented correctly then
> decompression failure won't occur (if the sender can't compress
> the long message with DMS = 2048 then it should just send it
> uncompressed).

I don't think sender can ever compress a 64KB message, unless
it can compress it down to something like 1KB. By definition,
part of DMS is used to buffer the compressed message.

So, basically if the receiver does not advertise any DMS,
this test is to check if the sender is "smart" enough to 
send the message uncompressed.

Zhigang
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Hi Ashok
Please can you tell me where exactly does the compressor and decompressor
fit in a traditional layer 7 model. My understanding is that it fits
between layer 2 (data link layer) and layer 3 (network layer or IP). If
that is the case, if a solution has to be implemented then dont we have to
tweak into layer2, layer3 implementation of the stacks on respective
platforms??
This doubt is specific to wireline implementatio of ROHC.
since IP packet is also compressed. how is the data transmission done
incase there is a router or some entity in between which doesnt support
ROHC. Basically the question is should all the entities in the route
support ROHC or is it suffcient if ROHC is implemented just on the
endpoints??

i would be really glad to hear back from you

thanks
alphonse




"Ashok Kumar" <ashok.kum@wipro.com> on 02/11/2003 11:56:35 AM

To:   alreddy, rohc@ietf.org
cc:

Subject:  RE: [rohc] context update in case of packet loss





Hi Alphonse,

When a packet is lost in transit, decompressor will not receive the
packet. So nothing needs to be done at decompressor. The subsequent
packets received by decompressor will indicate how the context should be
updated at decompressor.

-Ashok

-----Original Message-----
From: alreddy@hss.hns.com [mailto:alreddy@hss.hns.com]
Sent: Tuesday, February 11, 2003 11:44 AM
To: rohc@ietf.org
Subject: [rohc] context update in case of packet loss




Hi
Can some one tell me how the packet is updated on the decompressor side
when there is packet loss in the transit. I would really appreciate some
pointers in this regard.

thanks
alphonse






This message is proprietary to Hughes Software Systems Limited (HSS) and
is
intended solely for the use of the individual to whom it is addressed.
It
may contain privileged or confidential information and should not be
circulated or used for any purpose other than for what it is intended.
If
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Subject: RE: [rohc] context update in case of packet loss
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From: "kamaldeep dham" <kamaldeep.dham@wipro.com>
To: <alreddy@hss.hns.com>, "Ashok Kumar" <ashok.kum@wipro.com>,
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I guess, RFC 3095 does not specify, whether ROHC is a layer concept
between L2 and L3 or 
It's a sub layer in side L2 (data link). With respect to PDCP, ROHC
should be a part
of PDCP implementation. Basically your ROHC implementation will provide
rcv and snd
iinterfaces to the PDCP where it receives & sends IP packets. 

In my understanding ROHC is designed for point to point links only.
That's why you can have redundant information in the IP header.

-Kamal

-----Original Message-----
From: alreddy@hss.hns.com [mailto:alreddy@hss.hns.com] 
Sent: Tuesday, February 11, 2003 4:12 PM
To: Ashok Kumar; rohc@ietf.org
Subject: RE: [rohc] context update in case of packet loss




Hi Ashok
Please can you tell me where exactly does the compressor and
decompressor fit in a traditional layer 7 model. My understanding is
that it fits between layer 2 (data link layer) and layer 3 (network
layer or IP). If that is the case, if a solution has to be implemented
then dont we have to tweak into layer2, layer3 implementation of the
stacks on respective platforms?? This doubt is specific to wireline
implementatio of ROHC. since IP packet is also compressed. how is the
data transmission done incase there is a router or some entity in
between which doesnt support ROHC. Basically the question is should all
the entities in the route support ROHC or is it suffcient if ROHC is
implemented just on the endpoints??

i would be really glad to hear back from you

thanks
alphonse




"Ashok Kumar" <ashok.kum@wipro.com> on 02/11/2003 11:56:35 AM

To:   alreddy, rohc@ietf.org
cc:

Subject:  RE: [rohc] context update in case of packet loss





Hi Alphonse,

When a packet is lost in transit, decompressor will not receive the
packet. So nothing needs to be done at decompressor. The subsequent
packets received by decompressor will indicate how the context should be
updated at decompressor.

-Ashok

-----Original Message-----
From: alreddy@hss.hns.com [mailto:alreddy@hss.hns.com]
Sent: Tuesday, February 11, 2003 11:44 AM
To: rohc@ietf.org
Subject: [rohc] context update in case of packet loss




Hi
Can some one tell me how the packet is updated on the decompressor side
when there is packet loss in the transit. I would really appreciate some
pointers in this regard.

thanks
alphonse






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Hi Alphanso,

Please see the comments inline.

Regards,
Ashok


Hi Ashok
Please can you tell me where exactly does the compressor and
decompressor
fit in a traditional layer 7 model. My understanding is that it fits
between layer 2 (data link layer) and layer 3 (network layer or IP). If
that is the case, if a solution has to be implemented then dont we have
to
tweak into layer2, layer3 implementation of the stacks on respective
platforms??

Ashok >>  Since ROHC involves compression of Network Level PDU, so
certainly the compression can only be done once you have got network
level PDU from the upper layer. ROHC can be implemented as part of Data
Link Layer. How you implement ROHC to fit in a particular stack is
responsibility of the specific implementaion. >>

This doubt is specific to wireline implementatio of ROHC.
since IP packet is also compressed. how is the data transmission done
incase there is a router or some entity in between which doesnt support
ROHC. Basically the question is should all the entities in the route
support ROHC or is it suffcient if ROHC is implemented just on the
endpoints??

Ashok >> ROHC is primarily aimed for Point to Point Links. Both ends of
link should support ROHC. >>

i would be really glad to hear back from you

thanks
alphonse




"Ashok Kumar" <ashok.kum@wipro.com> on 02/11/2003 11:56:35 AM

To:   alreddy, rohc@ietf.org
cc:

Subject:  RE: [rohc] context update in case of packet loss





Hi Alphonse,

When a packet is lost in transit, decompressor will not receive the
packet. So nothing needs to be done at decompressor. The subsequent
packets received by decompressor will indicate how the context should be
updated at decompressor.

-Ashok

-----Original Message-----
From: alreddy@hss.hns.com [mailto:alreddy@hss.hns.com]
Sent: Tuesday, February 11, 2003 11:44 AM
To: rohc@ietf.org
Subject: [rohc] context update in case of packet loss




Hi
Can some one tell me how the packet is updated on the decompressor side
when there is packet loss in the transit. I would really appreciate some
pointers in this regard.

thanks
alphonse






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To: "'alreddy@hss.hns.com'" <alreddy@hss.hns.com>, rohc@ietf.org
Subject: RE: [rohc] context update in case of packet loss
Date: Tue, 11 Feb 2003 12:38:39 +0100
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> Please can you tell me where exactly does the compressor and 
> decompressor fit in a traditional layer 7 model. My understanding
> is that it fits between layer 2 (data link layer) and layer 3
> (network layer or IP). If that is the case, if a solution has to
> be implemented then dont we have to tweak into layer2, layer3
> implementation of the stacks on respective platforms??

I would say that ROHC is "on top of" layer 2, i.e. the layer 2
technology must provide means to support ROHC. In the ROHC WG, we
have defined this for PPP as L2, see RFC 3241, which is the only
L2 protocol we have considered us (the IETF) responsible for. Take
a look at RFC 3241, it is a good example of L2 support for ROHC.

BR
/L-E

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Hi all,

For those interested in the (fast approaching) SigComp test at SIPit, we said that we would put together some example DAP message flows.  So, hopefully (!), the following link will give you some indications of the sort of things that we've been thinking of:

    http://sigcomp.srmr.co.uk/cgi-bin/extract_flow.cgi

Obviously, comments are welcome!

Cheers,

Mark.

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Richard, Ghyslain, and others,

> > Do you not agree that "random" is the correct classification for the
> > UDP-Lite checksum field?
> 
> I'd like to avoid the term "random", which generally refers to some
> data that has no discernable pattern whatsoever (thus the term is
> inapplicable to the UDP checksum). RFC 3095 coins the more general
> term "irregular" to describe fields "for which no useful change
> pattern can be identified". As well as truly random fields, this
> definition also applies to fields that do have a discernable pattern,
> but which isn't useful for compression (e.g. because it would violate
> end-to-end integrity). How about using this classification instead?

I agree that IRREGULAR might be the right word to use when we discuss
the classification, as that is the classification we used in 3095, and
when we now address the differences in UDP-Lite we should be consistent.

 
> I think it would be helpful to cleanly separate the chapter
> on UDP-Lite behavior from the proposed compression profile.
> The behavior chapter should just describe how we expect UDP-Lite
> to behave, without taking into account how we intend to compress
> it (much like the behavior draft we did for ROHC-TCP). If we
> expect the CC field to exhibit non-random behavior then we
> should say so, even if we then ignore this when actually
> compressing the field.

I agree, and that is what the current draft seems to do by providing
an analysis of the behavior in chapter 3, listing the design rationales
for the profile in chapter 4, and defining the actual profile in
chapter 5.


> The current text makes it clear that support for multiple
> IP headers has been added, but doesn't explain the motivation
> behind this addition. It seems to contradict the goal of
> making as little modifications to RFC 3095 as possible.

Not necessarily. One could actually say that this is a very simple
fix to allow compression of more than 2 IP headers. The profiles in
3095 would be able to handle that in IR/IR-DYN headers, but it has
not been defined how to carry additional information in compressed
headers. The solution (as defined by the IP profile) imposes really
no change to the profiles in 3095, it just requires a minor addition.
The header field structure for the added field is the same as the one
used in IR/IR-DYN headers. The actual compressed header formats are
thus not changed.


In general, it should be noted that UDP-Lite is very similar to UDP, 
and the difference is only about the checksum coverage (the fact that
the checksum can not be disabled does not affect the compression, it 
is already supported by the RTP/UDP and UPD profiles). One of the
reasons UDP-Lite has been defined as it is, is to simplify deployment
by making it implementable through a rather small modification to an
existing UDP implementation, and the same argument apply as well to
the UDP-Lite compression profile. We should make sure to minimize the
effort needed to modify existing ROHC profile 1,2 implementations to
get UDP-Lite compression. This will make life easier for implementers,
as well as it will simplify deployment of UDP-Lite compression. This
way, we should also be able to get interoperability verified easily,
as we already have had several interoperability tests of the current
profiles, and almost all parts have now been tested.

The WG has already completed its UDP and UDP/RTP tasks, and what is
left here is only to address the differences in UDP-Lite. Of course,
we should try to do our best to anticipate how UDP-Lite usage could
have implications for the coverage field behavior, and use this when
we incorporate coverage field handling in the profile(s). The current
draft has the means to exclude the coverage field or include it as a
two-octet field. However, I agree with Richard that it might make 
sense to use some variable length approach so that (when sufficient)
only one byte can be sent as W-LSB coded bits, and we should
investigate how to achieve this. The current draft shows that we can
do this without having to require significant changes to the UDP
profiles, we just have to focus on getting the last bit of the puzzle
right without breaking the rest of it.

Cheers,
/L-E


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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 Robust Header Compression Working Group of the IETF.

	Title		: RObust Header Compression (ROHC):Terminology and 
                          Channel Mapping Examples
	Author(s)	: L. Jonsson
	Filename	: draft-ietf-rohc-terminology-and-examples-02.txt
	Pages		: 20
	Date		: 2003-2-13
	
RFC 3095 defines a Proposed Standard framework with profiles for
RObust Header Compression (ROHC). The standard introduces various
concepts which might be difficult to understand and especially to
relate correctly to the surrounding environments where header
compression may be used. This document aims at clarifying these
aspects of ROHC, discussing terms such as ROHC instances, ROHC
channels, ROHC feedback, and ROHC contexts, and how these terms
relate to other terms like network elements and IP interfaces,
commonly used for example when addressing MIB issues.

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To: rohc@ietf.org
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Subject: [rohc] Use of User-specific Dictionary
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hi,

i would like to refer to the test on Use of User-Specific Dictionary.

i) Should the User-Specific Dictionary be uploaded as plain text or included
as part of a SigComp message which means it has to be compressed ?

ii) How does endpoint 1 inform endpoint 2 that it is uploading the 
Dictionary to it ?
is there a special flag to indicate this ?

Thanks !!

regards,
Kwang Mien

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To: "'Chan Kwang Mien'" <chankm@icr.a-star.edu.sg>, rohc@ietf.org
Subject: RE: [rohc] Use of User-specific Dictionary
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> i would like to refer to the test on Use of User-Specific Dictionary.
> 
> i) Should the User-Specific Dictionary be uploaded as plain text or included
> as part of a SigComp message which means it has to be compressed ?

Hi Kwang Mien,

The choice of how to upload the User-Specific Dictionary is made by the
compressor. The dictionary will always be part of a SigComp message,
but it's up to the compressor to decide whether to send the dictionary
compressed or as plain text.

> ii) How does endpoint 1 inform endpoint 2 that it is uploading the
> Dictionary to it ?
> is there a special flag to indicate this ?

Endpoint 1 informs endpoint 2 that it is uploading the dictionary by
sending a piece of UDVM bytecode. This code should tell endpoint 2
to decompress the dictionary (if necessary) and then to save it as
state, so that it can be retrieved when later messages need to be
decompressed.

Regards,

Richard

> Thanks !!
> 
> regards,
> Kwang Mien
> 
> _______________________________________________
> Rohc mailing list
> Rohc@ietf.org
> https://www1.ietf.org/mailman/listinfo/rohc
> 
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FYI.

==========================================================================
INTERNET DRAFT                                              V. Madisetti
Date: Jan 30, 2003                                                S. Rao  
Expires: June 30, 2003                                         N. Suresh 
                                          Georgia Institute of Technology


           
    Enhancements to ECRTP with Applications to Robust Header 
                   Compression for Wireless 
            <draft-madisetti-rao-suresh-rohc-00.txt>






Abstract

Enhanced Compressed RTP (ECRTP) and Robust Header Compression
(ROHC) are IP header compression schemes that have been 
proposed for use in wireless applications, primarily for
voice and audio applications.  While ROHC offers a high
degree of compression and is robust, it is also computationally
expensive. On the other hand, ECRTP does not achieve as high
a compression, while being less expensive computationally. 
Our proposal, E2CRTP, is an enhanced version of ECRTP that
offers a middleground between the ROHC and ECRTP, providing
robustness, compression efficiency, and moderate computational
complexity. 


TABLE OF CONTENTS

   1. Introduction ..................................... 2 

   2. Proposed enhancements to ECRTP - E2CRTP .......... 2 
       2.1 W-LSB encoding of fields
           2.1.1 Packet Structure         .............. 2
       2.2 Reduced frequency of Checksum transmissions
           2.2.1 Packet Structure
   3. Assumptions. ..................................... 6 
   4. Simulation Results ............................... 7
   5. Observations and Conclusions ..................... 8 
   6. Summary & Future Work ............................ 9 
   7. Security Considerations .......................... 9 
   8. References ....................................... 10 
   9. Authors' Addresses ............................... 10 



http://www.ietf.org/internet-drafts/draft-madisetti-rao-suresh-rohc-00.txt

===================


-- 
Vijay K. Madisetti
Professor, ECE
Georgia Institute of Technology
http://www.ece.gatech.edu/~vkm/

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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 Robust Header Compression Working Group of the IETF.

	Title		: Requirements for RoHC IP/SCTP Robust Header 
                          Compression
	Author(s)	: C. Schmidt, M. Tuexen
	Filename	: draft-ietf-rohc-sctp-requirements-02.txt
	Pages		: 13
	Date		: 2003-2-14
	
This document contains requirements for the IP/SCTP header
compression scheme (profile) to be developed by the ROHC WG.  The
structure of this document is inherited from the document defining
IP/TCP requirements for ROHC.

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This draft is a work item of the Robust Header Compression Working Group of the IETF.

	Title		: Interoperability of RFC 3095
	Author(s)	: L. Jonsson
	Filename	: draft-ietf-rohc-rtp-rfc3095-interoperability-01.txt
	Pages		: 30
	Date		: 2003-2-18
	
RFC 3095 defines a Proposed Standard protocol for RObust Header
Compression (ROHC). In order to move the standard further to Draft
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From mailnull@www1.ietf.org  Mon Feb 24 03:51:24 2003
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Hi all,

Whilst there are a few nits, random references and other odd bits to finalise on the behaviour draft, the main outstanding item appears to be the discussion of 'shareable' TCP fields.  (Although, at the risk of making words up, I've changed the name to 'replicable', since I don't think that 'sharing' [old argument!] is an accurate description of how they are used).

Having finally got around to updating this, I thought I'd post some updated text for comments, prior to submitting the revised draft.  So, here it is...


4. Classification of replicable header fields

   Where multiple flows either overlap in time or occur sequentially
   within a short space of time there can be a great deal of similarity
   in header field values.  Such commonality of field values is
   reflected in the compression context.  Thus, it should be possible to
   utilise links between fields across different flows to improve the
   compression ratio.  In order to do this, it is important to
   understand the 'replicable' characteristics of the various header
   fields.

   The key concept is that of 'replication', where an existing context
   is used as a baseline and replicated to initialise a new context.
   Those fields that are the same are then automatically initialised in
   the new context.  Those that have changed will be updated or
   overwritten with values from the initialisation packet that triggered
   the replication.  This section considers the commonality between
   fields in different flows.

   It should be noted, however, that replication is based on contexts
   (rather than just field values) and so 'meta-fields' that are part of
   the context should also be included.  These, of course, are dependent
   upon the nature of the compression algorithm being applied.

   A brief analysis of the relationship of TCP/IP fields among
   'replicable' packet streams follows.

         'N/A' -- The field need not be considered in the replication
            process as it is inferred or known 'a priori' (and,
            therefore, does not appear in the context).

         'No' -- The field cannot be replicated since its change pattern
            between two packet flows is uncorrelated.

         'Yes' -- The field may be replicated.  This does not guarantee
            that the field value will be the same across two candidate
            streams, only that it might be possible to exploit
            replication to increase the compression ratio.  Specific
            encoding methods can be used to improve the compression
            efficiency.


4.1 IPv4 Header (inner and/or outer)

          +-----------------------+---------------+------------+
          | Field                 | Class         | Replicable |
          +-----------------------+---------------+------------+
          | Version               | STATIC        | N/A        |
          | Header Length         | STATIC-KNOWN  | N/A        |
          | DSCP                  | CHANGING      | No  (1)    |
          | ECT flag              | CHANGING      | No  (2)    |
          | CE flag               | CHANGING      | No  (2)    |
          | Packet Length         | INFERRED      | N/A        |
          | Identification        | CHANGING      | Yes (3)    |
          | Reserved flag         | CHANGING      | No  (4)    |
          | Don't Fragment flag   | CHANGING      | No         |
          | More Fragments flag   | STATIC-KNOWN  | N/A        |
          | Fragment Offset       | STATIC-KNOWN  | N/A        |
          | Time To Live          | CHANGING      | Yes        |
          | Protocol              | STATIC        | N/A        |
          | Header Checksum       | INFERRED      | N/A        |
          | Source Address        | STATIC-DEF    | Yes        |
          | Destination Address   | STATIC-DEF    | Yes        |
          +-----------------------+---------------+------------+

   (1) The DSCP is marked based on the application's requirements.  If
      it can be assumed that replicable connections often carry the same
      type of traffic, the DSCP may be regarded as replicable.  However,
      issues such as re-marking will need to be taken into account.

   (2) It is not possible for the ECN bits to be replicated (note that
      use of the ECN nonce scheme is anticipated).  However, it seems
      likely that all TCP flows between ECN-capable hosts will use ECN,
      the use (or not) of ECN for flows between the same end-points
      might be considered replicable.  See also note (4).

   (3) The replicable context for this field includes the IP-ID, NBO,
      and RND flags (as described in ROHC RTP).  This highlights that
      the replication is of the context, rather than just the header
      field values and, as such, needs to be considered based on the
      exact nature of compression applied to each field.

   (4) Since the possible future behavior of the 'Reserved Flag' cannot
      be predicted, it is not considered as replicable.  However, it
      might be expected that the behaviour of the reserved flag between
      the same end-points will be similar.  In this case, any selection
      of packet formats (for example) based on this behaviour might
      carry across to the new flow.  In the case of packet formats, this
      can probably be considered as a compressor-local decision.


4.2 IPv6 Header (inner and/or outer)

          +-----------------------+---------------+------------+
          | Field                 | Class         | Replicable |
          +-----------------------+---------------+------------+
          | Version               | STATIC        | N/A        |
          | Traffic Class         | CHANGING      | Yes (1)    |
          | ECT flag              | CHANGING      | No  (2)    |
          | CE flag               | CHANGING      | No  (2)    |
          | Flow Label            | STATIC-DEF    | N/A        |
          | Payload Length        | INFERRED      | N/A        |
          | Next Header           | STATIC        | N/A        |
          | Hop Limit             | CHANGING      | Yes        |
          | Source Address        | STATIC-DEF    | Yes        |
          | Destination Address   | STATIC-DEF    | Yes        |
          +-----------------------+---------------+------------+

   (1) See comment for IPv4, above.

   (2) See comment for IPv4, above.


4.3 TCP Header

          +-----------------------+---------------+------------+
          | Field                 | Class         | Replicable |
          +-----------------------+---------------+------------+
          | Source Port           | STATIC-DEF    |  Yes (1)   |
          | Destination Port      | STATIC-DEF    |  Yes (1)   |
          | Sequence Number       | CHANGING      |  No  (2)   |
          | Acknowledgement Number| CHANGING      |  No        |
          | Data Offset           | INFERRED      |  N/A       |
          | Reserved Bits         | CHANGING      |  No  (3)   |
          | Flags                 |               |            |
          |         CWR           | CHANGING      |  No  (4)   |
          |         ECE           | CHANGING      |  No  (4)   |
          |         URG           | CHANGING      |  No        |
          |         ACK           | CHANGING      |  No        |
          |         PSH           | CHANGING      |  No        |
          |         RST           | CHANGING      |  No        |
          |         SYN           | CHANGING      |  No        |
          |         FIN           | CHANGING      |  No        |
          | Window                | CHANGING      |  Yes       |
          | Checksum              | CHANGING      |  No        |
          | Urgent Pointer        | CHANGING      |  Yes (5)   |
          +-----------------------+---------------+------------+

   (1) On the server side, the port number is likely to be a well-known
      value.  On the client side, the port number is generally selected
      by the stack automatically.  Whether the port number is replicable
      depends upon how the stack chooses the port number.  However, most
      implementations use a simple scheme which sequentially picks the
      next available port number.  This is clearly exploitable in a
      compression scheme.

   (2) With the recommendation (and expected deployment) of TCP Initial
      Sequence Number randomization, defined in RFC 1948 [13], it will
      be impossible to share the sequence number.  Thus, this field will
      not be regarded as replicable.

   (3) See comment (4) for the IPv4 header, above.

   (4) See comment (2) on ECN flags for the IPv4 header, above.

   (5) The urgent pointer is very rarely used.  This means that, in
      practice, the field may be considered replicable.


4.4 TCP Options

          +---------------------------+--------------+------------+
          | Option                    | SYN-only (1) | Replicable |
          +---------------------------+--------------+------------+
          | End of Option List        | No           | No   (2)   |
          | No-Operation              | No           | No   (2)   |
          | Maximum Segment Size      | Yes          | Yes        |
          | Window Scale              | Yes          | Yes        |
          | SACK-Permitted            | Yes          | Yes        |
          | SACK                      | No           | No         |
          | Timestamp                 | No           | No         |
          +---------------------------+--------------+------------+

   (1) This indicates whether the option only appears in SYN packet or
      not.  Options that are not 'SYN-only' may appear in any packet.

      Many TCP options are used only in SYN packets.  Some options, such
      as MSS, Window Scale, SACK-Permitted etc., will tend to have the
      same value among replicable packet streams.
      Thus, to support context sharing, the compressor should maintain
      such TCP options in the context (even though they only appear in
      the SYN segment).

   (2) Since these options have fixed values, they could be regarded as
      replicable.  However, the only interesting thing to convey about
      these options is their presence: if it is known that such an
      option exists, its value is defined.


4.5 Summary of replication

   From the above analysis, it can be seen that there are reasonable
   grounds for exploiting redundancy between flows, as well as between
   packets within a flow.  Simply consider the advantage of being able
   to elide the source and destination addresses for a repeated
   connection between two IPv6 endpoints.

   The final point to note for the use of replication is that it
   requires the compressor to have a suitable degree of confidence that
   the source data is present and correct at the decompressor.  This may
   place some restrictions on which of the 'changing' fields, in
   particular, can be utilised during replication.


... comments welcome!

Cheers,

Mark.

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From: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>
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Subject: [rohc] Updating the ROHC impl. guide
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Hi all,

The Berlin interop generated a couple of questions on 3095, 
and some new issues have also been raised on the mailing list
since October when we made the last update of the impl. guide.

I've tried to collect these, address the questions and find
answers. My proposals for how to capture these issues in the
implementer's guide can be found below. Please let me know if
something is not clear or if there are ideas for improving the
clarifying texts.

Thanx!
/L-E


NOTE: All section references in this document refer to 3095,
if not explicitly stated otherwise.


1) Section 4.9 should clarify that a UO-1-ID packet with an
   Extension 3 must never be sent with RND flags that changes
   the packet interpretation, i.e. that would violate the base
   condition for UO-1-ID (at least one RND value must be 0).
   The last line of the first paragraph is also redundant and
   unclear.

 * Proposed modified text for the first paragraph of section 4.9:

  "Extension-3 is applied to give values and indicate changes to
   fields other than SN, TS and IP-ID in IP header(s) and RTP
   header.  In case of UO-1-ID packets, it should be noted that
   values provided in extensions do not update the context, with
   an exception for SN, TS and IP-ID fields, which always update
   the context (Section 5.7.3.). It should also be noted that a
   UO-1-ID packet with Extension 3 must never be sent with RND
   flags that changes the packet interpretation, i.e. that would
   violate the base condition for UO-1-ID (at least one RND value
   must be 0)."


2) It should be clarified that W-LSB encoding is used for TS, both
   if it is sent unscaled or scaled. Further, it should be noted
   that TS_OFFSET must be recalculated when an unscaled TS is
   received.

 * Proposed modification of section 4.2:
   - Change 4.2 to become "TS encoding considerations"
   - Renumber current 4.2 to 4.2.2.
   - Add 4.2.1., "Encoding of TS values"
  "RTP Timestamp values (TS) are always encoded using W-LSB
   encoding, both when sent scaled and when sent unscaled. For
   TS values sent in Extension 3, W-LSB encoded values are sent
   using the self-describing variable-length format (section
   4.5.6), and this applies to both scaled and unscaled values."

   - Add 4.2.4, "Recalculating TS_OFFSET"
  "TS can be sent unscaled if the TS value change does not match
   the established TS_STRIDE, but the TS_STRIDE might still stay
   unchanged. To ensure correct decompression of subsequent packets,
   the decompressor should therefore always recalculate the RTP TS
   modulo, TS_OFFSET, when a packet with an unscaled TS value is
   received."
   

3) It has been noted that there is an ambiguity about the
   "CC" and "Generic CSRC list" in 5.7.7.6. The question is
   whether CC=0 actually means that there are no CSRC items
   present at all (as the CC field means in RTP), or if an
   "empty" generic compressed CSRC list of at least 1 octet
   must still be present. The former should be the obvious
   interpretation as the CC field would otherwise have no
   meaning at all, and as the latter interpretation would
   just waste useless octets. However, this is not totally
   clear from the text, and it is currently not discussed 
   in the implementer's guide. Another list compression question
   is whether a 7-bit bit mask can be used also when there are
   more than 7 items in the list. 

 * Proposed addition to the impl. guide:
   - Change 4.3 to become "List compression issues"
   - Rename current section 4.5 to "CSRC list items in RTP
     dynamic chain"
   - Renumber current sections 4.3-4.5 to 4.3.1-4.3.3
   - Renumber current section 4.15 to 4.3.4
   - Renumber subsequent sections 4.6-4.14 to 4.4-4.12
   - Rewrite the beginning of current section 4.5. to say:
  "Section 5.7.7.6 defines the static and dynamic parts of
   the RTP header.  In the dynamic part, a 'CC' field indicates
   the number of CSRC items present in the 'Generic CSRC list'.
   It should be noted that when the value of this CC field is
   equal to zero, there is no Generic CSRC list present in the
   dynamic chain, i.e. the field comment should have said
   "variable length, present if CC > 0". A 'CC' field can also
   be found within the 'Generic CSRC list' (when present), and
   these fields would then have the same meaning. In order..."
   - Add subsection 4.3.5, "Bit masks in list compression"
  "A 7-bit or 15-bit mask may be used in the insertion and/or
   removal schemes for compressed lists. It should be noted that
   even if a list has more than 7 items, a 7-bit mask can be
   used as long as changes are only applied in the first part of
   the reference list, and all items with an index not covered
   by the 7-bit mask are then kept unchanged."


4) Section 4.11 seems to be totally incorrect, as RTP SN is only
   16 bits long, and feedback packets can carry 16 bits. I do 
   not know where this section comes from, but it should be removed.


5) For the UDP profile, all formats are the same as for profile 1,
   if not explicitly stated otherwise. However, it is rather unclear
   from section 5.11.3 whether "New format:" for UOR-2 means the
   format is changed for UOR-2-only, or if this new format replaces
   UOR-2, UOR-2-ID, and UOR-2-TS. In fact, it does replace all of
   them.

 * Proposed addition to the impl. guide:
   - Add new section 4.xx, "UOR-2 in profile 2 (UDP)":
  "One single new format is defined for UOR-2 in profile 2, which
   replaces all three (UOR-2, UOR-2-ID, UOR-2-TS) formats from
   profile 1. The same UOR-2 format is thus used independent of if
   there are IP headers with a corresponding RND=1 or not."


6) As suggested by Zhigang and agreed on the list, we should give
   an additional clarification in section 4.7.

 * Proposed additional paragraph in 4.7:
  "An updating packet thus updates the reference values of all
   header fields, either explicitly or implicitly. In UO-mode,
   all packets are updating packets (except for UO-1-ID, which
   only updates TS, SN and IP-ID, see 5.7.3), while in R-mode
   all packets with a CRC are updating packets."


7) As Carsten noted on the list, the logical union of suboptions
   for IPCP and IPV6CP negotiations, as specified by RFC 3241,
   can not be used for the PROFILES suboption, as the whole 
   union must then be considered within each of the two IPCP
   negotiations to avoid getting an ambiguous profile set.

 * Proposed new section 6, "PROFILES suboption in ROHC-over-PPP"
  "The logical union of suboptions for IPCP and IPV6CP
   negotiations, as specified by RFC 3241, can not be used for
   the PROFILES suboption, as the whole union would then have
   to be considered within each of the two IPCP negotiations, to
   avoid getting an ambiguous profile set. An implementation of
   RFC 3241 must therefore make sure the same profile set is
   negotiated for both IPv4 and IPv6 (IPCP and IPV6CP)."
   


-----------------------------------
Lars-Erik Jonsson, M.Sc
Wireless IP Optimizations
AWARE - Advanced Wireless Algorithm Research
Ericsson Research, Corporate Unit
Ericsson AB
Box 920, S-971 28 Luleå, Sweden
E-mail: lars-erik.jonsson@ericsson.com
Phone: +46 920 20 21 07
Fax: +46 920 20 20 99
Home: +46 920 999 57

My opinions are my personal opinions and should not be considered
as the opinions of my employer, if not explicitly stated. 
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From: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>
To: "'West, Mark'" <mark.a.west@roke.co.uk>,
        "'rohc@ietf.org'"
	 <rohc@ietf.org>
Subject: RE: [rohc] TCP 'shareable' fields
Date: Tue, 25 Feb 2003 14:13:32 +0100
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Mark,

This is a nice update and I do not have many comments. I think
the classification looks good, and overall we now seem to be
consistent with what we have agreed to during the last 9 months.

A few comments below!

Rgds,
/L-E



> Whilst there are a few nits, random references and other odd 
> bits to finalise on the behaviour draft, the main outstanding 
> item appears to be the discussion of 'shareable' TCP fields.  
> (Although, at the risk of making words up, I've changed the 
> name to 'replicable', since I don't think that 'sharing' [old 
> argument!] is an accurate description of how they are used).

Yes, this seems to be the common understanding since May -02.
Let's stick to that term, hopefully we will now at least be 
consistent at some places...=)


>    It should be noted, however, that replication is based on
>    contexts (rather than just field values) and so 'meta-fields'
>    that > are part of the context should also be included.
>    These, of course, are dependent upon the nature of the
>    compression algorithm being applied.

The last line would become clearer in our context if we modify and
extend it to:
"...the compression protocol (ROHC profile) being applied."

By the way, is it clear what is meant with "meta-fields"? It is
not to me. I guess you mean "data not part of the original 
headers", but does it make sense to replicate that? Do we need
to analyze what information might be considerable for that
purpose?


> 4.2 IPv6 Header (inner and/or outer)
> 
>           +-----------------------+---------------+------------+
>           | Field                 | Class         | Replicable |
>           +-----------------------+---------------+------------+
>           | Version               | STATIC        | N/A        |
>           | Traffic Class         | CHANGING      | Yes (1)    |
>           | ECT flag              | CHANGING      | No  (2)    |
>           | CE flag               | CHANGING      | No  (2)    |
>           | Flow Label            | STATIC-DEF    | N/A        |
>           | Payload Length        | INFERRED      | N/A        |
>           | Next Header           | STATIC        | N/A        |
>           | Hop Limit             | CHANGING      | Yes        |
>           | Source Address        | STATIC-DEF    | Yes        |
>           | Destination Address   | STATIC-DEF    | Yes        |
>           +-----------------------+---------------+------------+
> 
>    (1) See comment for IPv4, above.

Which comment for IPv4? We should probably clarify that the field
has a different name, i.e. refer to the actual v4 field.


> 4.5 Summary of replication
> 
>    From the above analysis, it can be seen that there are reasonable
>    grounds for exploiting redundancy between flows, as well as between
>    packets within a flow.  Simply consider the advantage of being able
>    to elide the source and destination addresses for a repeated
>    connection between two IPv6 endpoints.

This summary is fine, but maybe we should also note already in this
summary that it is important to find a suitable trade-off, providing
the means of replicating a field will always imply a cost, and must
thus be justified by a reasonably gain.
_______________________________________________
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Subject: RE: [rohc] TCP 'shareable' fields
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Hi from SIPit...

I agree with and will incorporate your comments -- thanks!

BTW - yes 'meta-fields' means what you think it means; I will try to clarify.  My understanding is that you would actually replicate *context*, and the context is more than just the fields (e.g. the IP-ID Network Byte Order).  However, the behaviour of these additional things in the context is slightly outside of just the protocol behaviour.  I think...  Although, having said that, we do discuss the way the IP-ID behaves.  So perhaps I can bound the description a bit better.

Cheers,

Mark.


-----Original Message-----
From: Lars-Erik Jonsson (EAB)
To: West, Mark; 'rohc@ietf.org'
Sent: 25/02/03 13:13
Subject: RE: [rohc] TCP 'shareable' fields

Mark,

This is a nice update and I do not have many comments. I think
the classification looks good, and overall we now seem to be
consistent with what we have agreed to during the last 9 months.

A few comments below!

Rgds,
/L-E



> Whilst there are a few nits, random references and other odd 
> bits to finalise on the behaviour draft, the main outstanding 
> item appears to be the discussion of 'shareable' TCP fields.  
> (Although, at the risk of making words up, I've changed the 
> name to 'replicable', since I don't think that 'sharing' [old 
> argument!] is an accurate description of how they are used).

Yes, this seems to be the common understanding since May -02.
Let's stick to that term, hopefully we will now at least be 
consistent at some places...=)


>    It should be noted, however, that replication is based on
>    contexts (rather than just field values) and so 'meta-fields'
>    that > are part of the context should also be included.
>    These, of course, are dependent upon the nature of the
>    compression algorithm being applied.

The last line would become clearer in our context if we modify and
extend it to:
"...the compression protocol (ROHC profile) being applied."

By the way, is it clear what is meant with "meta-fields"? It is
not to me. I guess you mean "data not part of the original 
headers", but does it make sense to replicate that? Do we need
to analyze what information might be considerable for that
purpose?


> 4.2 IPv6 Header (inner and/or outer)
> 
>           +-----------------------+---------------+------------+
>           | Field                 | Class         | Replicable |
>           +-----------------------+---------------+------------+
>           | Version               | STATIC        | N/A        |
>           | Traffic Class         | CHANGING      | Yes (1)    |
>           | ECT flag              | CHANGING      | No  (2)    |
>           | CE flag               | CHANGING      | No  (2)    |
>           | Flow Label            | STATIC-DEF    | N/A        |
>           | Payload Length        | INFERRED      | N/A        |
>           | Next Header           | STATIC        | N/A        |
>           | Hop Limit             | CHANGING      | Yes        |
>           | Source Address        | STATIC-DEF    | Yes        |
>           | Destination Address   | STATIC-DEF    | Yes        |
>           +-----------------------+---------------+------------+
> 
>    (1) See comment for IPv4, above.

Which comment for IPv4? We should probably clarify that the field
has a different name, i.e. refer to the actual v4 field.


> 4.5 Summary of replication
> 
>    From the above analysis, it can be seen that there are reasonable
>    grounds for exploiting redundancy between flows, as well as between
>    packets within a flow.  Simply consider the advantage of being able
>    to elide the source and destination addresses for a repeated
>    connection between two IPv6 endpoints.

This summary is fine, but maybe we should also note already in this
summary that it is important to find a suitable trade-off, providing
the means of replicating a field will always imply a cost, and must
thus be justified by a reasonably gain.
_______________________________________________
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https://www1.ietf.org/mailman/listinfo/rohc



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From: "Price, Richard" <richard.price@roke.co.uk>
To: "West, Mark" <mark.a.west@roke.co.uk>,
        "'Lars-Erik Jonsson (EAB) '"
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Subject: RE: [rohc] TCP 'shareable' fields
Date: Tue, 25 Feb 2003 14:31:40 -0000
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Hi,

The so-called 'meta-fields' are mentioned in ROHC-TCP and also
the Formal Notation, but in these drafts they're known as
'control fields'.  It would be nice to use a consistent term
between all of the drafts.

I'm not sure that either of the current terms is particularly
descriptive - how about adopting a more explicit term such as
'compressor-created fields'?

Regards,

Richard

> -----Original Message-----
> From: West, Mark 
> Sent: Tuesday, February 25, 2003 1:37 PM
> To: 'Lars-Erik Jonsson (EAB) '; ''rohc@ietf.org' '
> Subject: RE: [rohc] TCP 'shareable' fields
> 
> 
> 
> Hi from SIPit...
> 
> I agree with and will incorporate your comments -- thanks!
> 
> BTW - yes 'meta-fields' means what you think it means; I will 
> try to clarify.  My understanding is that you would actually 
> replicate *context*, and the context is more than just the 
> fields (e.g. the IP-ID Network Byte Order).  However, the 
> behaviour of these additional things in the context is 
> slightly outside of just the protocol behaviour.  I think...  
> Although, having said that, we do discuss the way the IP-ID 
> behaves.  So perhaps I can bound the description a bit better.
> 
> Cheers,
> 
> Mark.
> 
> 
> -----Original Message-----
> From: Lars-Erik Jonsson (EAB)
> To: West, Mark; 'rohc@ietf.org'
> Sent: 25/02/03 13:13
> Subject: RE: [rohc] TCP 'shareable' fields
> 
> Mark,
> 
> This is a nice update and I do not have many comments. I think
> the classification looks good, and overall we now seem to be
> consistent with what we have agreed to during the last 9 months.
> 
> A few comments below!
> 
> Rgds,
> /L-E
> 
> 
> 
> > Whilst there are a few nits, random references and other odd 
> > bits to finalise on the behaviour draft, the main outstanding 
> > item appears to be the discussion of 'shareable' TCP fields.  
> > (Although, at the risk of making words up, I've changed the 
> > name to 'replicable', since I don't think that 'sharing' [old 
> > argument!] is an accurate description of how they are used).
> 
> Yes, this seems to be the common understanding since May -02.
> Let's stick to that term, hopefully we will now at least be 
> consistent at some places...=)
> 
> 
> >    It should be noted, however, that replication is based on
> >    contexts (rather than just field values) and so 'meta-fields'
> >    that > are part of the context should also be included.
> >    These, of course, are dependent upon the nature of the
> >    compression algorithm being applied.
> 
> The last line would become clearer in our context if we modify and
> extend it to:
> "...the compression protocol (ROHC profile) being applied."
> 
> By the way, is it clear what is meant with "meta-fields"? It is
> not to me. I guess you mean "data not part of the original 
> headers", but does it make sense to replicate that? Do we need
> to analyze what information might be considerable for that
> purpose?
> 
> 
> > 4.2 IPv6 Header (inner and/or outer)
> > 
> >           +-----------------------+---------------+------------+
> >           | Field                 | Class         | Replicable |
> >           +-----------------------+---------------+------------+
> >           | Version               | STATIC        | N/A        |
> >           | Traffic Class         | CHANGING      | Yes (1)    |
> >           | ECT flag              | CHANGING      | No  (2)    |
> >           | CE flag               | CHANGING      | No  (2)    |
> >           | Flow Label            | STATIC-DEF    | N/A        |
> >           | Payload Length        | INFERRED      | N/A        |
> >           | Next Header           | STATIC        | N/A        |
> >           | Hop Limit             | CHANGING      | Yes        |
> >           | Source Address        | STATIC-DEF    | Yes        |
> >           | Destination Address   | STATIC-DEF    | Yes        |
> >           +-----------------------+---------------+------------+
> > 
> >    (1) See comment for IPv4, above.
> 
> Which comment for IPv4? We should probably clarify that the field
> has a different name, i.e. refer to the actual v4 field.
> 
> 
> > 4.5 Summary of replication
> > 
> >    From the above analysis, it can be seen that there are reasonable
> >    grounds for exploiting redundancy between flows, as well as between
> >    packets within a flow.  Simply consider the advantage of being able
> >    to elide the source and destination addresses for a repeated
> >    connection between two IPv6 endpoints.
> 
> This summary is fine, but maybe we should also note already in this
> summary that it is important to find a suitable trade-off, providing
> the means of replicating a field will always imply a cost, and must
> thus be justified by a reasonably gain.
> _______________________________________________
> Rohc mailing list
> Rohc@ietf.org
> https://www1.ietf.org/mailman/listinfo/rohc
> 
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Subject: Re: [rohc] Updating the ROHC impl. guide
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Couple of comments below.

----- Original Message ----- 
From: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>
To: "Rohc (E-mail)" <rohc@ietf.org>
Sent: Tuesday, February 25, 2003 5:30 AM
Subject: [rohc] Updating the ROHC impl. guide


<snip>

> 4) Section 4.11 seems to be totally incorrect, as RTP SN is only
>    16 bits long, and feedback packets can carry 16 bits. I do 
>    not know where this section comes from, but it should be removed.

I don't have the document in front of me, but the ESP SN is 32 bits, 
so that may explain it. 
 

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Micke,

> -----Original Message-----
> From: Mikael Degermark [mailto:micke@cs.arizona.edu]
> Sent: den 25 februari 2003 16:12
> To: Lars-Erik Jonsson (EAB); Rohc (E-mail)
> Subject: Re: [rohc] Updating the ROHC impl. guide
> 
> > 4) Section 4.11 seems to be totally incorrect, as RTP SN is only
> >    16 bits long, and feedback packets can carry 16 bits. I do 
> >    not know where this section comes from, but it should be removed.
> 
> I don't have the document in front of me, but the ESP SN is 32 bits, 
> so that may explain it. 

The current text says:

"4.11.  Multiple SN options in one feedback packet

  The length of the sequence number field in the original RTP
  header is 32 bits.  A decompressor can't send back all the
  32 bits in a feedback packet unless it uses multiple SN options
  in one feedback packet.  Section 5.7.6.1 declares that a
  FEEDABCK-2 packet can contain variable number of feedback
  options and the options can appear in any order.

  A compressor - that wants to be conform to the specification -
  should be able to process multiple SN options in one feedback
  packet."

You might be correct that it should have said ESP instead of RTP.
Any opinions about what to do with this section?

/L-E
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Subject: [rohc] ROHC interoperability test case draft updated
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Hi all,

The 3095 test case draft has been updated:
draft-ietf-rohc-rtp-rfc3095-interoperability-01.txt

A few test cases have been added for the RTP profile,
based on observations from collecting interoperability
status. Further, applicable tests have been listed also
for the UDP and ESP profiles. Please let me know if
there is something that is not covered by current tests.

The updated draft marks tests (*) for which
interoperability has been reported, but it should be
noted that the current status documentation is neither
complete nor guaranteed correct. The intension was to
indicate how status is planned to be documented, and I
am currently starting to collect interoperability reports.
I will ask for interoperability reports from those I know
have done interoperability testing of 3095, and all reports
will be cross-verified. The detailed information on who 
have reported what will only be known to me and Carsten,
while the draft will just show the status summary.

/L-E

--------------------------------------
Lars-Erik Jonsson
IETF ROHC WG Chair
E-mail: lars-erik.jonsson@ericsson.com
Phone: +46 920 20 21 07
Home: +46 920 999 57

My opinions are my personal opinions and should not be considered
as the opinions of my employer, if not explicitly stated. 
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> > > 4) Section 4.11 seems to be totally incorrect, as RTP SN is
> > >    only 16 bits long, and feedback packets can carry 16 bits.
> > >    I do not know where this section comes from, but it should
> > >    be removed.
> > 
> > I don't have the document in front of me, but the ESP SN is 
> > 32 bits, so that may explain it. 
> 
> The current text says:
> 
> "4.11.  Multiple SN options in one feedback packet
> 
>   The length of the sequence number field in the original RTP
>   header is 32 bits.  A decompressor can't send back all the
>   32 bits in a feedback packet unless it uses multiple SN options
>   in one feedback packet.  Section 5.7.6.1 declares that a
>   FEEDABCK-2 packet can contain variable number of feedback
>   options and the options can appear in any order.
> 
>   A compressor - that wants to be conform to the specification -
>   should be able to process multiple SN options in one feedback
>   packet."
> 
> You might be correct that it should have said ESP instead of RTP.
> Any opinions about what to do with this section?

What about rewriting the section to apply only to profile 3, ESP?

/L-E
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: DCCP Profile for EPIC
	Author(s)	: J. Ozegovic, M. Mornar
	Filename	: draft-ozegovic-mornar-dccp-epic-00.txt
	Pages		: 20
	Date		: 2003-2-25
	
This draft describes a profile for compressing Datagram Congestion 
Control Protocol DCCP/IP using the Efficient Protocol Independent 
Compression (EPIC) [2] scheme.
The RObust Header Compression [1] scheme is designed to compress
packet headers over error prone channels.  It is built around an
extensible core framework that can be tailored to compress new
protocol stacks by adding additional ROHC profiles.
DCCP [3] is a relatively new protocol and has not yet entered the 
process of wide implementation. It is intended to provide a unified
congestion control interface to the end user applications. It should
offer the same unreliable transport as UDP does, and on the other 
hand, include mechanisms for congestion avoidance and control.
This profile describes a new profile for ROHC which will allow DCCP/
IP headers to be compressed.

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http://www.ietf.org/internet-drafts/draft-ozegovic-mornar-dccp-epic-00.txt

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From mailnull@www1.ietf.org  Wed Feb 26 09:31:57 2003
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From: "Price, Richard" <richard.price@roke.co.uk>
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Hi,

In the interests of kick-starting the discussion on TCP
compression, we'd like to make our current implementation of a
TCP profile available to the WG.

We've based the implementation on the latest published TCP
compression drafts, filling in the gaps where necessary to get a
working solution.  The main piece currently missing from the drafts
is a description of how to encode each field in the TCP/IP header,
so we've added a very simple proposal for this based on the current
behaviour draft.

We're hoping that the profile will evolve based on further input
from the WG, so any comments would be gratefully received ;-)

You can download the complete implementation from:

http://sigcomp.srmr.co.uk/~rjp/RohcFn_Tcp.zip

The zip file contains the following 5 modules, described below:

http://sigcomp.srmr.co.uk/~rjp/RohcFn_TcpEncoding.hs
http://sigcomp.srmr.co.uk/~rjp/RohcFn_StandardLibrary.hs
http://sigcomp.srmr.co.uk/~rjp/RohcFn_Primitives.hs
http://sigcomp.srmr.co.uk/~rjp/RohcFn_Harness.hs
http://sigcomp.srmr.co.uk/~rjp/RohcFn_Tests.hs

TCP encoding:  Defines how to compress each field in a TCP/IP
header, using encoding methods from the standard library (see
below).  The choice of encoding methods is based on the behaviour
draft <draft-ietf-rohc-tcp-field-behavior-01.txt>, but greatly
simplified - we'd appreciate suggestions on how to make the profile
more efficient, robust and future-proof.

Standard library:  The library of encoding methods from Chapter 7
of <draft-ietf-rohc-formal-notation-00.txt>.

Primitives:  The basic maths functions and encoding methods from
Chapter 6 of <draft-ietf-rohc-formal-notation-00.txt>.

Test harness:  A simple front-end that can compress and decompress
headers.  It includes a really simple state machine based on
<draft-ietf-rohc-tcp-03.txt> - all it does at present is store the
context between successive headers in a flow.  More complex state
management stuff (feedback, context replication) is on the to-do
list.

Tests:  Some example TCP/IP flows, and a generic test suite to
make sure that the implementation is working correctly.

There's also some documentation, namely:

http://sigcomp.srmr.co.uk/~rjp/RohcFn_Instructions.txt
http://sigcomp.srmr.co.uk/~rjp/RohcFn_To_Do_List.txt
http://sigcomp.srmr.co.uk/~rjp/RohcFn_Proposed_Changes.txt

Let us know if you have any problems running the code, or any
suggestions for improvements etc.

Regards,

Richard
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From: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>
To: "'Mikael Degermark'" <micke@cs.arizona.edu>
Cc: "Rohc (E-mail)" <rohc@ietf.org>
Subject: RE: [rohc] Updating the ROHC impl. guide
Date: Wed, 26 Feb 2003 15:51:59 +0100
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That is ok with me. Then we keep it as it is, just changing
the motivating example to ESP SN (which actually is 32 bits).

OK?
/L-E


> -----Original Message-----
> From: Mikael Degermark [mailto:micke@cs.arizona.edu]
> Sent: den 26 februari 2003 22:27
> To: Lars-Erik Jonsson (EAB)
> Subject: Re: [rohc] Updating the ROHC impl. guide
> 
> 
> Wouldn't it be simplest to have a general rule that multiple 
> SN options
> are allowed? The reason for having this rule is of course 
> ESP, but it seems
> simplest to have an implementation allow it for any CID.
> 
> Fits "be conservative in what you send, liberal in what you receive"
> principle.
> 
> Cheers,
> 
> Micke
> ----- Original Message -----
> From: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>
> To: "'Mikael Degermark'" <micke@CS.Arizona.EDU>; "Rohc (E-mail)"
> <rohc@ietf.org>
> Sent: Wednesday, February 26, 2003 6:39 AM
> Subject: RE: [rohc] Updating the ROHC impl. guide
> 
> 
> > > > > 4) Section 4.11 seems to be totally incorrect, as RTP SN is
> > > > >    only 16 bits long, and feedback packets can carry 16 bits.
> > > > >    I do not know where this section comes from, but it should
> > > > >    be removed.
> > > >
> > > > I don't have the document in front of me, but the ESP SN is
> > > > 32 bits, so that may explain it.
> > >
> > > The current text says:
> > >
> > > "4.11.  Multiple SN options in one feedback packet
> > >
> > >   The length of the sequence number field in the original RTP
> > >   header is 32 bits.  A decompressor can't send back all the
> > >   32 bits in a feedback packet unless it uses multiple SN options
> > >   in one feedback packet.  Section 5.7.6.1 declares that a
> > >   FEEDABCK-2 packet can contain variable number of feedback
> > >   options and the options can appear in any order.
> > >
> > >   A compressor - that wants to be conform to the specification -
> > >   should be able to process multiple SN options in one feedback
> > >   packet."
> > >
> > > You might be correct that it should have said ESP instead of RTP.
> > > Any opinions about what to do with this section?
> >
> > What about rewriting the section to apply only to profile 3, ESP?
> >
> > /L-E
> > _______________________________________________
> > Rohc mailing list
> > Rohc@ietf.org
> > https://www1.ietf.org/mailman/listinfo/rohc
> >
> 
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From mailnull@www1.ietf.org  Wed Feb 26 12:31:35 2003
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From: "Price, Richard" <richard.price@roke.co.uk>
To: "'rohc@ietf.org'" <rohc@ietf.org>
Cc: "'Lars-Erik Jonsson (EAB)'" <Lars-Erik.Jonsson@epl.ericsson.se>
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Hi,

Now that our TCP compression code is available, I'd like to
think about updating the Formal Notation draft.

As usual I've left this till the last minute, so there's no
time left to discuss any major technical changes - fortunately
I don't think that any are required at this stage.  Based on
our implementation experience, the only technical changes I'd
like to propose are the following:

1. We should add "Self-describing variable-length values" from
   Section 4.5.6 of RFC 3095 to the library of encoding methods.

2. A function for calculating an n-bit CRC should be added to
   the maths library.

Obviously, both of the above would be nice to have when
generating the final TCP profile.

I'd also like to propose some terminology changes to make the
notation draft more readable, and to hopefully fix the
inconsistencies between the notation draft and other TCP-related
drafts.  A list of these terminology changes is available from
http://sigcomp.srmr.co.uk/~rjp/RohcFn_Proposed_Changes.txt

Comments on the above proposed changes would be much appreciated!

Regards,

Richard

-----Original Message-----
From: Price, Richard 
Sent: Wednesday, February 26, 2003 2:35 PM
To: 'rohc@ietf.org'
Subject: [rohc] TCP compression profile


Hi,

In the interests of kick-starting the discussion on TCP
compression, we'd like to make our current implementation of a
TCP profile available to the WG.

We've based the implementation on the latest published TCP
compression drafts, filling in the gaps where necessary to get a
working solution.  The main piece currently missing from the drafts
is a description of how to encode each field in the TCP/IP header,
so we've added a very simple proposal for this based on the current
behaviour draft.

We're hoping that the profile will evolve based on further input
from the WG, so any comments would be gratefully received ;-)

You can download the complete implementation from:

http://sigcomp.srmr.co.uk/~rjp/RohcFn_Tcp.zip

The zip file contains the following 5 modules, described below:

http://sigcomp.srmr.co.uk/~rjp/RohcFn_TcpEncoding.hs
http://sigcomp.srmr.co.uk/~rjp/RohcFn_StandardLibrary.hs
http://sigcomp.srmr.co.uk/~rjp/RohcFn_Primitives.hs
http://sigcomp.srmr.co.uk/~rjp/RohcFn_Harness.hs
http://sigcomp.srmr.co.uk/~rjp/RohcFn_Tests.hs

TCP encoding:  Defines how to compress each field in a TCP/IP
header, using encoding methods from the standard library (see
below).  The choice of encoding methods is based on the behaviour
draft <draft-ietf-rohc-tcp-field-behavior-01.txt>, but greatly
simplified - we'd appreciate suggestions on how to make the profile
more efficient, robust and future-proof.

Standard library:  The library of encoding methods from Chapter 7
of <draft-ietf-rohc-formal-notation-00.txt>.

Primitives:  The basic maths functions and encoding methods from
Chapter 6 of <draft-ietf-rohc-formal-notation-00.txt>.

Test harness:  A simple front-end that can compress and decompress
headers.  It includes a really simple state machine based on
<draft-ietf-rohc-tcp-03.txt> - all it does at present is store the
context between successive headers in a flow.  More complex state
management stuff (feedback, context replication) is on the to-do
list.

Tests:  Some example TCP/IP flows, and a generic test suite to
make sure that the implementation is working correctly.

There's also some documentation, namely:

http://sigcomp.srmr.co.uk/~rjp/RohcFn_Instructions.txt
http://sigcomp.srmr.co.uk/~rjp/RohcFn_To_Do_List.txt
http://sigcomp.srmr.co.uk/~rjp/RohcFn_Proposed_Changes.txt

Let us know if you have any problems running the code, or any
suggestions for improvements etc.

Regards,

Richard
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Subject: RE: [rohc] Updating the ROHC impl. guide
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Lars-Erik,

Some comments below.

Zhigang
 
> 1) Section 4.9 should clarify that a UO-1-ID packet with an
>    Extension 3 must never be sent with RND flags that changes
>    the packet interpretation, i.e. that would violate the base
>    condition for UO-1-ID (at least one RND value must be 0).
>    The last line of the first paragraph is also redundant and
>    unclear.
> 
>  * Proposed modified text for the first paragraph of section 4.9:
> 
>   "Extension-3 is applied to give values and indicate changes to
>    fields other than SN, TS and IP-ID in IP header(s) and RTP
>    header.  In case of UO-1-ID packets, it should be noted that
>    values provided in extensions do not update the context, with
>    an exception for SN, TS and IP-ID fields, which always update
>    the context (Section 5.7.3.). It should also be noted that a
>    UO-1-ID packet with Extension 3 must never be sent with RND
>    flags that changes the packet interpretation, i.e. that would
>    violate the base condition for UO-1-ID (at least one RND value
>    must be 0)."

Sorry, I probably missed the discussions in the past, but what is the 
reason for this section? Why Extension-3 needs special clarification,
but not other extensions? I thought the following texts from RFC 3095 
are already clear.

Section 5.7: 
(The same text is repeated in section 5.7.5.1. I remember we intentionally
make it redundantly clear, as it is a crucial point for correct parsing.)

   "When no IPv4 header is present in the static context, or the RND
   flags for all IPv4 headers in the context have been established to be
   1, the packet types R-1-ID, R-1-TS, UO-1-ID, and UO-1-TS MUST NOT be
   used.

   While in the transient state in which an RND flag is being
   established, the packet types R-1-ID, R-1-TS, UO-1-ID, and UO-1-TS
   MUST NOT be used.  This implies that the RND flag(s) of the Extension
   3 may have to be inspected before the format of a base header
   carrying an Extension 3 can be determined."

Section 5.7.3:

      "Note: UO-1-ID cannot be used if there is no IPv4 header in the
      context or if value(RND) and value(RND2) are both 1."

Section 5.7.3:

      UO-1-ID packets update context(IP-ID).  Values provided in
      extensions, except those in other SN, TS, or IP-ID fields, do not
      update the context.
 
> 6) As suggested by Zhigang and agreed on the list, we should give
>    an additional clarification in section 4.7.
> 
>  * Proposed additional paragraph in 4.7:
>   "An updating packet thus updates the reference values of all
>    header fields, either explicitly or implicitly. In UO-mode,
>    all packets are updating packets (except for UO-1-ID, which
>    only updates TS, SN and IP-ID, see 5.7.3), while in R-mode
>    all packets with a CRC are updating packets."

The above text has some self-contradiction. For example, the text
seems to say UO-1-ID is not an updating packet, which is untrue.
It looks to me that we are clarifying two things here: a) what are 
updating packets, and b) what do they update. So, can we re-arrange
above text as follows?

"In UO-mode, all packets are updating packets, while in R-mode
all packets with a CRC are updating packets. An updating packet
in R-mode updates the reference values of all header fields, either 
explicitly or implicitly. Same is true for updating packets in
UO-mode, except for UO-1-ID, which only updates TS, SN and IP-ID
(see 5.7.3)."
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From: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>
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Subject: RE: [rohc] Updating the ROHC impl. guide
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Zhigang,

The issues addressed in the implemeter's guide are mainly issues
identified during implementation (interoperability test discussions).
Some of the issues are clear from 3095, but repeated questions still
indicate that additional reminders to implementers are useful.


> > 1) Section 4.9 should clarify that a UO-1-ID packet with an
> >    Extension 3 must never be sent with RND flags that changes
> >    the packet interpretation, i.e. that would violate the base
> >    condition for UO-1-ID (at least one RND value must be 0).
> >    The last line of the first paragraph is also redundant and
> >    unclear.
> > 
> >  * Proposed modified text for the first paragraph of section 4.9:
> > 
> >   "Extension-3 is applied to give values and indicate changes to
> >    fields other than SN, TS and IP-ID in IP header(s) and RTP
> >    header.  In case of UO-1-ID packets, it should be noted that
> >    values provided in extensions do not update the context, with
> >    an exception for SN, TS and IP-ID fields, which always update
> >    the context (Section 5.7.3.). It should also be noted that a
> >    UO-1-ID packet with Extension 3 must never be sent with RND
> >    flags that changes the packet interpretation, i.e. that would
> >    violate the base condition for UO-1-ID (at least one RND value
> >    must be 0)."
> 
> Sorry, I probably missed the discussions in the past, but what is the 
> reason for this section? Why Extension-3 needs special clarification,
> but not other extensions? I thought the following texts from RFC 3095 
> are already clear.

Section 4.9 of the implementer's guide was originally written for the
purpose of reminding implementers of something that is actually clear
in 3095, as there seemed to be confusion about UO-1-ID and its updating
properties for fields in Extension 3. This should be clear from 3095,
but all questions motivated an additional reminder. The point about RND
in such packets is however something that is not clear from 3095. If
RND is changed in an UOR-2 with Extension 3, the decompressor must
re-parse the packet (as explained in 4.10), but if this happens for 
UO-1-ID, something is obviously wrong.

 
> > 6) As suggested by Zhigang and agreed on the list, we should give
> >    an additional clarification in section 4.7.
> > 
> >  * Proposed additional paragraph in 4.7:
> >   "An updating packet thus updates the reference values of all
> >    header fields, either explicitly or implicitly. In UO-mode,
> >    all packets are updating packets (except for UO-1-ID, which
> >    only updates TS, SN and IP-ID, see 5.7.3), while in R-mode
> >    all packets with a CRC are updating packets."
> 
> The above text has some self-contradiction. For example, the text
> seems to say UO-1-ID is not an updating packet, which is untrue.

OK, let's put it this way then:

   "An updating packet thus updates the reference values of all
    header fields, either explicitly or implicitly. In UO-mode,
    all packets are updating packets (note that UO-1-ID packets
    only update TS, SN and IP-ID, see 5.7.3), while in R-mode
    all packets with a CRC are updating packets."


Thanks!
/L-E
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From: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>
To: "'Price, Richard'" <richard.price@roke.co.uk>,
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Subject: RE: [rohc] TCP compression profile
Date: Thu, 27 Feb 2003 11:13:08 +0100
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Richard,

I hope this input will be useful for the further TCP profile work,
i.e. for defining the compressed header formats, but it is a rather
substantial material to read and understand (especially considering
previous problems of getting people involved in these notation
discussions) so we will have to give this some time.

Anyway, I have to provide some comments and guidelines on procedural
matters, considering this input. This is an IETF WG, and input is
expected to be provided either:
1) in internet-drafts, or
2) in messages to the WG mail list 

Some material, like implementations, might of course be difficult
to provide through those channels, and then exceptions may have to
be made. However, such exceptions should only be made when really
necessary, both because we have common IETF principles to simplify
for all involved parties and be consistent, and because of archival
reasons. Except for implementations, material for the WG to consider
must be posted through our official channels (drafts or mail list).

/L-E

--------------------------------------
Lars-Erik Jonsson
IETF ROHC WG Chair
E-mail: lars-erik.jonsson@ericsson.com
Phone: +46 920 20 21 07
Home: +46 920 999 57

My opinions are my personal opinions and should not be considered
as the opinions of my employer, if not explicitly stated. 


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From: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>
To: "'rohc@ietf.org'" <rohc@ietf.org>
Subject: RE: [rohc] TCP compression profile
Date: Thu, 27 Feb 2003 11:15:35 +0100
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> The main piece currently missing from the drafts is a description
> of how to encode each field in the TCP/IP header, so we've added a
> very simple proposal for this based on the current behavior draft.

How to encode the original fields are rather well "suggested" in the
behavior draft, but what we are still missing in the profile draft
is (1) a description and summary of what "control information" is
needed (e.g. context replication), (2) a summary of how we then have
decided to compress each piece of information (original fields+control),
and (3, most concrete) defining the actual compressed header formats,
i.e. the bits on the wire. 1) is partially already done in the draft,
but some pieces are still missing. The notation was intended to either
be used as a descriptive language to do 2) in the draft, or be used as
a tool to by-pass 2) and together with some encoding method produce 3)
directly for inclusion in the draft. It is not clear to me at this
point whether people are considering the former or the latter approach.
If we do put 2) in the profile draft, we have taken one step and may
then decide how to go further to define the actual compressed formats.
On the other hand, if we decide to go directly to 3), we need to agree
on how to use the notation as a tool to generate compressed header
formats, and we must find a good way of documenting these in the
profile specification. A third approach is of course not to use the
notation at all, but just define the compressed formats as we did in
RFC 3095.

Assuming that we do not take this third approach, the first step would
in any case be to agree on the notation. Any opinions on if the
notation would be about the same independent of how it will be used.

BR
/L-E
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From: "Price, Richard" <richard.price@roke.co.uk>
To: "'Lars-Erik Jonsson (EAB)'" <Lars-Erik.Jonsson@epl.ericsson.se>,
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Subject: RE: [rohc] TCP compression profile
Date: Thu, 27 Feb 2003 12:49:09 -0000
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Hi,

So, just to check, is there any part of our implementation that
would be useful to document separately (either in internet-drafts
or on the mailing list)?

The implementation has two new encoding methods not currently
described in the notation draft (see my previous post on updating
the formal notation).  I believe it would be good to add these to
the next version of the draft - if no-one objects, I can make this
update before the deadline next week.

The other new material that we've provided is a set of test
vectors for the implementation (including some TCP flows, as
well as some generic tests to make sure that the implementation
is working correctly).  Might it be worth putting these in a
draft at some stage, like we did for the SigComp torture tests?

Regards,

Richard

> -----Original Message-----
> From: Lars-Erik Jonsson (EAB) 
> [mailto:Lars-Erik.Jonsson@epl.ericsson.se]
> Sent: Thursday, February 27, 2003 10:13 AM
> To: Price, Richard; 'rohc@ietf.org'
> Subject: RE: [rohc] TCP compression profile
> 
> 
> Richard,
> 
> I hope this input will be useful for the further TCP profile work,
> i.e. for defining the compressed header formats, but it is a rather
> substantial material to read and understand (especially considering
> previous problems of getting people involved in these notation
> discussions) so we will have to give this some time.
> 
> Anyway, I have to provide some comments and guidelines on procedural
> matters, considering this input. This is an IETF WG, and input is
> expected to be provided either:
> 1) in internet-drafts, or
> 2) in messages to the WG mail list 
> 
> Some material, like implementations, might of course be difficult
> to provide through those channels, and then exceptions may have to
> be made. However, such exceptions should only be made when really
> necessary, both because we have common IETF principles to simplify
> for all involved parties and be consistent, and because of archival
> reasons. Except for implementations, material for the WG to consider
> must be posted through our official channels (drafts or mail list).
> 
> /L-E
> 
> --------------------------------------
> Lars-Erik Jonsson
> IETF ROHC WG Chair
> E-mail: lars-erik.jonsson@ericsson.com
> Phone: +46 920 20 21 07
> Home: +46 920 999 57
> 
> My opinions are my personal opinions and should not be considered
> as the opinions of my employer, if not explicitly stated. 
> 
> 
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From mailnull@www1.ietf.org  Thu Feb 27 10:54:25 2003
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From: "Price, Richard" <richard.price@roke.co.uk>
To: "'Lars-Erik Jonsson (EAB)'" <Lars-Erik.Jonsson@epl.ericsson.se>,
        "'rohc@ietf.org'" <rohc@ietf.org>
Subject: RE: [rohc] TCP compression profile
Date: Thu, 27 Feb 2003 15:55:11 -0000
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> > The main piece currently missing from the drafts is a description
> > of how to encode each field in the TCP/IP header, so we've added a
> > very simple proposal for this based on the current behavior draft.
> 
> How to encode the original fields are rather well "suggested" in the
> behavior draft

This certainly seems to be the case based on our initial test TCP flows.
One interesting open issue, however, is the level of complexity needed
in the final TCP encoding.  Our current implementation only uses a very
basic set of packet formats, designed specifically to be human-readable.
We could improve the efficiency, robustness and future-proofing by
following the behaviour draft more exactly, at a cost of making the
resulting packet formats harder to understand.

> but what we are still missing in the profile draft
> is (1) a description and summary of what "control information" is
> needed (e.g. context replication), (2) a summary of how we then have
> decided to compress each piece of information (original fields+control),
> and (3, most concrete) defining the actual compressed header formats,
> i.e. the bits on the wire. 1) is partially already done in the draft,
> but some pieces are still missing. The notation was intended to either
> be used as a descriptive language to do 2) in the draft, or be used as
> a tool to by-pass 2) and together with some encoding method produce 3)
> directly for inclusion in the draft. It is not clear to me at this
> point whether people are considering the former or the latter approach.
> If we do put 2) in the profile draft, we have taken one step and may
> then decide how to go further to define the actual compressed formats.

If we choose the former approach then the notation defines which
encoding methods are applied to which fields (including the "control"
fields not present in the original TCP/IP header).  To get from this to
"bits on the wire", we just need to define a library of 4 or 5 basic
encoding methods for the notation to use.

> On the other hand, if we decide to go directly to 3), we need to agree
> on how to use the notation as a tool to generate compressed header
> formats, and we must find a good way of documenting these in the
> profile specification. A third approach is of course not to use the
> notation at all, but just define the compressed formats as we did in
> RFC 3095.

The above is a good summary of the two ways the formal notation could be
used to provide the "bits on the wire" for the TCP profile.  In both cases
we need to somehow define a basic library of encoding methods - in
RFC 3095 this is just done by hand, and in the current notation draft we
do exactly the same.

The difference between the two approaches is therefore quite simple: we
need to define "packet formats" that apply one or more encoding methods
to each field, and we can do this either using the formal notation, or
in the "RFC 3095 style" using a combination of diagrams and accompanying
text.

Which of the above approaches to use is a matter of opinion (we could
even provide the packet formats using both, although I'm not keen on
this as we then have the additional worry of synchronising the two
approaches and making sure that they define the same bits on the wire).

My own preference is that the packet formats should be defined using the
formal notation (not that I'm biased of course!).  The main advantage of
this is that, as Carsten pointed out in Atlanta, several programming
languages have a compiler for the formal notation already built-in.  So
all of the packet formats can be implemented for free, just by pasting
the description of TCP/IP into the compiler.  As a bonus, this will help
to eliminate misunderstandings about how the packet formats behave (e.g.
which fields update the context, which offsets to use for LSB encoding
etc.)

> Assuming that we do not take this third approach, the first step would
> in any case be to agree on the notation. Any opinions on if the
> notation would be about the same independent of how it will be used.

I don't believe that the notation will need to change depending on which
of the above approaches is used.

Regards,

Richard
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From mailnull@www1.ietf.org  Thu Feb 27 12:22:16 2003
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From: "Price, Richard" <richard.price@roke.co.uk>
To: "'rohc@ietf.org'" <rohc@ietf.org>
Subject: FW: [rohc] Updating the Formal Notation draft
Date: Thu, 27 Feb 2003 17:24:24 -0000
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Hi,

I have a couple of editorial proposals for making the notation
draft a bit simpler and more readable.  Comments welcome!

Section 3.1 talks about ordinary BNF, which is irrelevant to
the definition of the formal notation.  For clarity I propose
to delete this section from the draft.

Section 4.6 on "control fields" is redundant given the
information contained in <draft-ietf-rohc-tcp-03.txt>.  Since
Ghyslain explains how control fields work much better than I
do, I propose to leave the ROHC-TCP draft as it is and delete
Section 4.6 from the formal notation.

Conversely, one issue that the current draft doesn't explain
very well is the scope of the formal notation - what it does,
what it doesn't do, how it relates to the ROHC framework and
to ROHC profiles etc.  I'd like to add a new section to
clarify this issue:

3.X. Scope of ROHC-FN

   The following section describes the scope of ROHC-FN, and
   explains how it relates to the overall ROHC framework and to
   specific ROHC profiles.

   The ROHC framework is common to all profiles, and provides
   generic features such as padding and segmentation of the
   compressed packets (useful if the link layer can only handle
   a limited range of packet sizes).

   A ROHC profile is a description of how to compress a certain
   protocol stack over a certain type of link.  For example, ROHC
   profiles are available for RTP/UDP/IP and many other protocol
   stacks.

   Each ROHC profile can be further subdivided into the following
   two components:

   a)  Packet formats for compressing and decompressing headers
   b)  State machine

   The job of the packet formats is to define how to compress and
   decompress headers.  The packet formats must define the
   compressed version of each uncompressed header (and vice versa).

   The packet formats will typically compress headers relative to
   a "context" of field values from previous headers in a flow.
   This improves the overall compression ratio, due to taking into
   account redundancies between successive headers.

   The job of the state machine is to ensure that the profile is
   robust against bit errors and dropped packets.  The state
   machine manages the context, providing feedback and other
   mechanisms to ensure that the compressor and decompressor
   contexts are kept in sync.

   ROHC-FN is designed to help provide the packet formats for use
   in new ROHC profiles.  It offers a library of encoding methods
   for compressing fields, and a mechanism for combining these
   encoding methods to create new packet formats tailored to a
   specific protocol stack.  Note however that the state machine
   for the new profiles is beyond the scope of ROHC-FN, and must
   be provided separately.

Comments on the above text would be much appreciated!

Regards,

Richard

-----Original Message-----
From: Price, Richard 
Sent: Wednesday, February 26, 2003 5:33 PM
To: 'rohc@ietf.org'
Cc: 'Lars-Erik Jonsson (EAB)'
Subject: [rohc] Updating the Formal Notation draft


Hi,

Now that our TCP compression code is available, I'd like to
think about updating the Formal Notation draft.

As usual I've left this till the last minute, so there's no
time left to discuss any major technical changes - fortunately
I don't think that any are required at this stage.  Based on
our implementation experience, the only technical changes I'd
like to propose are the following:

1. We should add "Self-describing variable-length values" from
   Section 4.5.6 of RFC 3095 to the library of encoding methods.

2. A function for calculating an n-bit CRC should be added to
   the maths library.

Obviously, both of the above would be nice to have when
generating the final TCP profile.

I'd also like to propose some terminology changes to make the
notation draft more readable, and to hopefully fix the
inconsistencies between the notation draft and other TCP-related
drafts.  A list of these terminology changes is available from
http://sigcomp.srmr.co.uk/~rjp/RohcFn_Proposed_Changes.txt

Comments on the above proposed changes would be much appreciated!

Regards,

Richard

-----Original Message-----
From: Price, Richard 
Sent: Wednesday, February 26, 2003 2:35 PM
To: 'rohc@ietf.org'
Subject: [rohc] TCP compression profile


Hi,

In the interests of kick-starting the discussion on TCP
compression, we'd like to make our current implementation of a
TCP profile available to the WG.

We've based the implementation on the latest published TCP
compression drafts, filling in the gaps where necessary to get a
working solution.  The main piece currently missing from the drafts
is a description of how to encode each field in the TCP/IP header,
so we've added a very simple proposal for this based on the current
behaviour draft.

We're hoping that the profile will evolve based on further input
from the WG, so any comments would be gratefully received ;-)

You can download the complete implementation from:

http://sigcomp.srmr.co.uk/~rjp/RohcFn_Tcp.zip

The zip file contains the following 5 modules, described below:

http://sigcomp.srmr.co.uk/~rjp/RohcFn_TcpEncoding.hs
http://sigcomp.srmr.co.uk/~rjp/RohcFn_StandardLibrary.hs
http://sigcomp.srmr.co.uk/~rjp/RohcFn_Primitives.hs
http://sigcomp.srmr.co.uk/~rjp/RohcFn_Harness.hs
http://sigcomp.srmr.co.uk/~rjp/RohcFn_Tests.hs

TCP encoding:  Defines how to compress each field in a TCP/IP
header, using encoding methods from the standard library (see
below).  The choice of encoding methods is based on the behaviour
draft <draft-ietf-rohc-tcp-field-behavior-01.txt>, but greatly
simplified - we'd appreciate suggestions on how to make the profile
more efficient, robust and future-proof.

Standard library:  The library of encoding methods from Chapter 7
of <draft-ietf-rohc-formal-notation-00.txt>.

Primitives:  The basic maths functions and encoding methods from
Chapter 6 of <draft-ietf-rohc-formal-notation-00.txt>.

Test harness:  A simple front-end that can compress and decompress
headers.  It includes a really simple state machine based on
<draft-ietf-rohc-tcp-03.txt> - all it does at present is store the
context between successive headers in a flow.  More complex state
management stuff (feedback, context replication) is on the to-do
list.

Tests:  Some example TCP/IP flows, and a generic test suite to
make sure that the implementation is working correctly.

There's also some documentation, namely:

http://sigcomp.srmr.co.uk/~rjp/RohcFn_Instructions.txt
http://sigcomp.srmr.co.uk/~rjp/RohcFn_To_Do_List.txt
http://sigcomp.srmr.co.uk/~rjp/RohcFn_Proposed_Changes.txt

Let us know if you have any problems running the code, or any
suggestions for improvements etc.

Regards,

Richard
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Subject: [rohc] Updating the Formal Notation draft
From: cabo@Informatik.Uni-Bremen.DE (Dr. Carsten Bormann)
Date: 27 Feb 2003 20:50:54 +0100
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ROHCers,

some comments (WG chair hat off) on
http://sigcomp.srmr.co.uk/~rjp/RohcFn_Proposed_Changes.txt below.

Gruesse, Carsten

    Based on our implementation experience we'd like to propose a
    few changes to the current formal notation draft (see
    <draft-ietf-rohc-formal-notation-00.txt>):

    Terminology changes:

    The following changes have no technical effect on the notation,
    but will hopefully make the draft easier to understand:

    1. We should give the notation a better name, e.g. ROHC-FN
       (Formal Notation for Robust Header Compression).

If, as you explain in a later message, it is just about packet
formats, it probably should be named ROHC-FNPF or some such?

    2. The "rules" should be called "encoding methods" to bring the 
       terminology in line with RFC 3095.

d/c (don't care).

    3. The "labels" should be renamed "temporary variables" to
       make it clearer what they actually do!

Hmm, this creates the impression the notation has imperative
semantics.  I thought we were trying to avoid this.  Bindings?

    4. The names of encoding methods and variables should be
       cleanly separated, e.g. by using lowercase for the names
       of encoding methods and uppercase for variable names.

This might conflict with the choice of Haskell (uppercase is reserved
for specific things in that language).  (It also would conflict with
Prolog.)

    Parsing the notation using Haskell:

    A drawback with the current notation draft is that even though
    it's based on a well-known language (namely BNF from RFC 2234),
    some proprietary extensions are needed to make the notation
    powerful enough to describe the headers we're interested in.
    This means that off-the-shelf BNF parsers like yacc can't handle
    the notation in its current form.

    A simpler approach would be to base the notation on a more
    powerful language, which already offers all of the features we
    need as standard.  A number of sufficiently powerful languages
    are already available (Prolog, Haskell, Mercury etc.) - we propose
    to use Haskell because its syntax is closest to the notation
    currently documented in <draft-ietf-rohc-formal-notation-00.txt>.

This is a quite weak argument for the choice of notational basis.
Obviously, we need to have a better one.  (I'm making this argument
not only because I did my first prototype in Prolog...)

    Only a handful of changes are needed to convert the notation
    into executable Haskell code:

    1. We should create new encoding methods using the symbol "="
       rather than "::=".

d/c (in Prolog, it's something different again).

    2. In a list expression (e.g. [2 * j for j in 0..n]) the reserved
       words "for" and "in" should be replaced by the symbols "|"
       and "<-".

The base language wins.

    Miscellaneous changes:

    The following changes aren't required to make the notation
    Haskell-compatible, but we think that they're a good idea anyway:

    1. We should add "Self-describing variable-length values" from
       Section 4.5.6 of RFC 3095 to the library of encoding methods.

Sure. The Library will grow with our use of the notation.

    2. A function for calculating an n-bit CRC should be added to
       the maths library.

Sure.

    3. We should make the notation more readable by removing the
       unnecessary parentheses and commas, and just delimiting the
       parameters of an encoding method using whitespace instead.

Readability is in the eye of the beholder...
In contrast to the lambda crowd, I prefer the assignment of roles to
things to be clearly visible.
In the end, of course, the base language we choose wins...

    4. Lists of elements should be clearly marked using a well-known
       symbol (e.g. square braces).

    5. Currently the lower bound for the number of elements in a
       list is inclusive but the upper bound is exclusive.  This is
       confusing - for consistency, both bounds should be inclusive.

    6. The "floor" function should be renamed "div" (because "floor"
       is a well-known term that means something completely different).

    7. The "Null" keyword should be renamed "Nil" (because "Null" is
       a well-known term that means something completely different).

    8. When we need the integer value of a field, we currently just
       quote the field name and assume that readers know that we want
       the integer value rather than the bits.  This might get
       confusing, so for clarity we propose to explicitly indicate
       whenever we need the integer value of a field (e.g. by writing
       "val" before the field name).

This is obviously required.

    9. In the current notation, when we define new parameters we
       need to explicitly state the parameter type (integer, text
       string etc.).  It isn't actually necessary to do this, because
       a compiler can infer the parameter types just by looking at the
       expressions in which the parameters are used.  For example, if
       we encounter the expression "m + n" then we know that m and n
       must both be integers.  So for simplicity, we should make
       parameter types optional (or not bother with them at all).

Stating the types may increase understandability.
Again, of course, the base language wins.

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Subject: RE: [rohc] Updating the ROHC impl. guide
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Lars-Erik,

> Some of the issues are clear from 3095, but repeated questions still
> indicate that additional reminders to implementers are useful.

The reason I asked was that my proposal to add some texts back in 
April 2002 were rejected on the basis that they were clear from 3095,
even though they addressed questions that have been raised in ROHC 
mailing list. So, I guess above is the new criteria for content
of the user guide? 

> The point about RND
> in such packets is however something that is not clear from 3095. If
> RND is changed in an UOR-2 with Extension 3, the decompressor must
> re-parse the packet (as explained in 4.10), but if this happens for 
> UO-1-ID, something is obviously wrong.

I thought it's clear from the following text in 3095.

   "While in the transient state in which an RND flag is being
   established, the packet types R-1-ID, R-1-TS, UO-1-ID, and UO-1-TS
   MUST NOT be used."

> > The above text has some self-contradiction. For example, the text
> > seems to say UO-1-ID is not an updating packet, which is untrue.
> 
> OK, let's put it this way then:
> 
>    "An updating packet thus updates the reference values of all
>     header fields, either explicitly or implicitly. In UO-mode,
>     all packets are updating packets (note that UO-1-ID packets
>     only update TS, SN and IP-ID, see 5.7.3), while in R-mode
>     all packets with a CRC are updating packets."

But there is still a controdiction between the first sentence
and the second one. In fact, the first sentence is correct only
for R-mode. For UO-mode, what an updating packet updates depends on
packet type.

My proposed text in earlier email (copied below again) solves the 
above problem. I'm wondering do you disagree with any particular
point in the text?

"In UO-mode, all packets are updating packets, while in R-mode
all packets with a CRC are updating packets. An updating packet
in R-mode updates the reference values of all header fields, either 
explicitly or implicitly. Same is true for updating packets in
UO-mode, except for UO-1-ID, which only updates TS, SN and IP-ID
(see 5.7.3)."

BR,
Zhigang
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From: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>
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Subject: RE: [rohc] Updating the ROHC impl. guide
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> > Some of the issues are clear from 3095, but repeated questions still
> > indicate that additional reminders to implementers are useful.
> 
> So, I guess above is the new criteria for content of the user guide? 

Well, the section we talk about has been there for quite a while.
Anyway, there is no single criteria, we have to look at each case.
We do not want this document to grow too much, but when questions are
repeatedly asked, we have to consider adding something. When things
have been added during the last year, it has been a result of repeated
questions on the interops, and/or the mailing list. The additions we
have made have lately been related to very special details, which
probably (hopefully) indicates that we should not have to add much more
to this document (but it will stay as an I-D, as previously agreed).

Cheers,
/L-E
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From: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>
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Richard,

In general, I think we should not make any changes to the
actual notation at this point (although the draft might be
updated for other reasons), as we do not have a common view
either of how to proceed with this, or for example if we 
should do this "Haskell"-step, or something similar using
another functional language. We need to discuss this more,
then we can get a stable ground for the content before we
do the changes.

However, we might want to update it just to change the title,
and maybe some of the "environmental texts". I've provided
some comments on this below.

BR
/L-E


> Section 4.6 on "control fields" is redundant given the
> information contained in <draft-ietf-rohc-tcp-03.txt>.  

Well, it might be worth mentioning that information that
can be noted is either original header fields, or additional
control fields introduced by the compression scheme.

> I propose to leave the ROHC-TCP draft as it is and delete
> Section 4.6 from the formal notation.

Of course, the TCP control field can not be discussed as
part of the general notation.


> Conversely, one issue that the current draft doesn't explain
> very well is the scope of the formal notation - what it does,
> what it doesn't do, how it relates to the ROHC framework and
> to ROHC profiles etc. 

This is indeed very important, and should be done before we
go much further. However, I do not think we do have a common
understanding about that, as it has never been settled.


> I'd like to add a new section to clarify this issue:
> 
> 3.X. Scope of ROHC-FN
> 
>    The following section describes the scope of ROHC-FN, and
>    explains how it relates to the overall ROHC framework and to
>    specific ROHC profiles.
> 
>    The ROHC framework is common to all profiles, and provides
>    generic features such as padding and segmentation of the
>    compressed packets (useful if the link layer can only handle
>    a limited range of packet sizes).

Well, the framework provides the base for all ROHC compression.
It defines the profile concept, which makes ROHC a general platform
for compression schemes. It sets link layer requirements, and 
especially negotiation requirements for all ROHC profiles. It 
defines a set of common functions, as segmentation, CID's, and
padding. It also defines common packet formats (IR, IR-DYN, Feedback,
Short-CID expander, etc), and it defines a generic, profile
independent, handling of feedback. One can also say that it defines
the general principles for doing ROHC compression.


>    A ROHC profile is a description of how to compress a certain
>    protocol stack over a certain type of link.  For example, ROHC
>    profiles are available for RTP/UDP/IP and many other protocol
>    stacks.
> 
>    Each ROHC profile can be further subdivided into the following
>    two components:
> 
>    a)  Packet formats for compressing and decompressing headers

I would say "compressed packet formats"

>    b)  State machine

I am not sure "state machine" captures all the rest that is there,
but I'll think more about this.


>    ROHC-FN is designed to help provide the packet formats for use
>    in new ROHC profiles.

It is a notation form for the information to compress, it is not clear
to me at this point how far you really get with the notation.


> Comments on the above text would be much appreciated!

This is a good start, as I think it is essential that we try to define
what the notation is, what it will be used for, and how it fits in
the "big ROHC picture".

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> Anyway, there is no single criteria, we have to look at each case.

Since this is a WG document, it would help to have some kind of
criteria/rules for what should be included and what should not.
Also, if the criteria change, it should be announced/documented.

This can help discussions and save time, especially when there
is disagreement.

Zhigang

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Lars-Erik,

Any thoughts on my proposed text for section 4.7? In particular,
if you have think it is correct? Note that I added UO-0 in the
last sentence. Also, what about UO-1 and UO-1-TS? Do they update
all the header fields or just a subset?

------------------
"In UO-mode, all packets are updating packets, while in R-mode
all packets with a CRC are updating packets. An updating packet
in R-mode updates the reference values of all header fields, either 
explicitly or implicitly. Same is true for updating packets in
UO-mode, except for UO-1-ID and UO-0, which only update TS, SN 
and IP-ID (see 5.7.3)."
------------------

Zhigang
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