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Subject: [rohc] SigComp States
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I'm studying now SigComp and wondering about some issues:

(1) Quoting from draft-ietf-roch-sigcomp-sip-00.txt (section 3): =93Usual=
ly,=20
any states created during the lifetime of a compartment will be "logicall=
y"=20
deleted when the compartment is closed. A logical deletion becomes a=20
physical one when all the compartments that created the (same) state are=20
closed.=94  When do compartments share the same state? When do different=20
compartments become related to the same state?

(2) Quoting from RFC 3320 (section 6.2): =93Each instance of the UDVM can=
 pass=20
up to four state creation requests to the state handler=85=94 Why does a =
UDVM=20
need more than a single allocated state item?

(3) Continuing the quote from the previous question: =93When the state ha=
ndler=20
receives a state creation request from the UDVM it takes the following=20
steps: =85  3. If the state creation request contains a state_identifier =
that=20
already exists=85=94 Is there any clear example to this kind of situation=
?

I would appreciate any piece of information.

Thanks,
John.

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Hello John,

Thanks for comments on sigcomp-sip. Please see my reply inline.

BR,
Zhigang

> (1) [snip] When do compartments share the same state? When do=20
> different compartments become related to the same state?

There are two possible cases: a) two compartments accidentally
creates the same state without knowing it, and b) they intentionally
create the same state. The chance of a) may be small in practice,
but a sigcomp implementation must be able to handle it correctly.
Case b) is more like a state "sharing". An example could be
multiple compartments (from the same sigcomp implementation) share
a state of bytecode or dictionary.

> (2) [snip] Why does a UDVM need more than a single allocated state =
item?

The compressor may want to create more than one state in a sigcomp
message, e.g., one state containing bytecode and one containing=20
dictionary. Of course, this is up to implementation. It may just
create one state. The purpose is to make SigComp reasonably flexible
without compromise security.

> (3) [snip] 3. If the state creation request contains a=20
> state_identifier that already exists..." Is there any clear=20
> example to this kind of situation?

An example could be for sigcomp over UDP. A compressor may try to
create the same state in two sigcomp messages before it knows whether
or not the first sigcomp message is received at the other endpoint.

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From: "West, Mark" <mark.a.west@roke.co.uk>
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Hi all,

This seems like as good an opportunity as any to summarise the changes
we've made in the notation since last time around...

... and they're pretty substantial!

There is a fundamental change to the philosophy of the notation:
previously we have tried to describe the packet formats and how to
compress each field -- the compressed packet formats were derived in some
(standardised way) from this description.

After some intense discussion, and given that this seemed to involve
standardising some non-trivial algorithms, we have taken an different
approach.  Now, the aim is to describe the uncompressed packet format; the
compressed packet formats; and the mapping between these two.

That is, the notation is now focussed on describing the compressed packet
formats (rather than a description that can be used to generate packet
formats).

It is perhaps easiest to explain the changes by comparison...

Previous form:

	foo ::= bar
	        baz

	bar ::= static / irregular(8)
	baz ::= value(8, 0) / value(8, 1)

would have given rise to the 4 packet formats:

	bar[static] + baz[value(8,0)]
	bar[static] + baz[value(8,1)]
	bar[irregular(8)] + baz[value(8,0)]
	bar[irregular(8)] + baz[value(8,1)]

and probabilities would be used to assign indicators to these formats in
some standardised form (e.g. EPIC or EPIC-lite).

All change!

The new form captures formats: we can define the uncompressed format and
capture (say) 3 packet formats as:

	foo ::= compressed_packet_formats,

	{ co_formats ::= constant(3),

	  % this is the first packet format
	  % note that the 8-bit discriminator is chosen
	  % simply to byte-align the packets...

	  co_format_0 ::= discriminator, % 8 bits

          { discriminator ::= '00000000',
            bar           ::= static,
            baz           ::= value(8,0)
          },

          co_format_1 ::= discriminator, % 8 bits

	  { discriminator ::= '00000001',
	    bar           ::= static,
	    baz           ::= value(8,1)
	  }

	  co_format_2 ::= discriminator, % 7 bits
                          baz,           % 1 bit
                          bar,           % 8 bits

	  { discriminator ::= '0000001',
	     bar          ::= irregular(8),
	     baz          ::= index(8, [0, 1])
                              % 1-bit choice between 0/1
	  }

	  uncompressed_format ::= foo,

	  foo ::= bar, % 8 bits
	          baz  % 8 bits
	}

	[with a fair degree of elision...]

You can see from the above (hopefully!) that there is no need to run any
algorithm to generate packet formats: they are specified in full within
the notation.

In order to allow 'reasoning' about the correctness of the profile, and
for test purposes, there is a prolog implementation to help things
along...

For more details, see the draft!

Comments, as ever, welcome...

... and feel free to collar me for discussions at Minneapolis!

Cheers,

Mark.

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


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To: John Smith <johnyjsmith1970@hotmail.com>
Cc: rohc@ietf.org
Date: Sun, 2 Nov 2003 22:12:10 +0000 (GMT Standard Time)
Subject: Re: [rohc] SigComp States
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Hi John,

>
> I'm studying now SigComp and wondering about some issues:
>
> (1) Quoting from draft-ietf-roch-sigcomp-sip-00.txt (section 3): =
=93Usually,
> any states created during the lifetime of a compartment will be =
"logically"
> deleted when the compartment is closed. A logical deletion becomes a
> physical one when all the compartments that created the (same) state =
are
> closed.=94  When do compartments share the same state? When do =
different
> compartments become related to the same state?
>

Think about bytecode or an implementation-specific dictionary -- all
instances of that particular implementation could create the same piece =
of
state or states.  (My bytecode might save itself; all end-points =
running
this code will do the same thing).

> (2) Quoting from RFC 3320 (section 6.2): =93Each instance of the UDVM =
can pass
> up to four state creation requests to the state handler=85=94 Why =
does a UDVM
> need more than a single allocated state item?
>

Again, think about saving bytecode or some non-standard (but
cross-compartment) dictionary as well as the circular buffer.  It is =
easy
to consider examples where 2 or more states are created in response to =
a
single message.

You should also never lose sight of the fact that 3320 defines the
decompressor.  We neither know, nor particularly want to constrain, =
what
compression algorithms can be implemented.  4 state creation requests =
seem
to allow enough latitude...

> (3) Continuing the quote from the previous question: =93When the =
state handler
> receives a state creation request from the UDVM it takes the =
following
> steps: =85  3. If the state creation request contains a =
state_identifier that
> already exists=85=94 Is there any clear example to this kind of =
situation?
>

If one instance of an endpoint has created a piece of state (e.g.
bytecode) the state will have a specific state identifier.  If another
instance creates the same piece of state it will have the same state
identifier (obviously).  This should be allowed to succeed (and will
create the 'multiple compartments referencing the same state' of your
first question).  However, if some endpoint created a different piece =
of
state that happened to have the same state identifier (i.e. a hash
collision), the state creation must fail (otherwise we would have an
ambiguity in accessing state).

Cheers,

Mark.

--=20
Mark A. West, Senior 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: "Surtees, Abigail" <abigail.surtees@roke.co.uk>
To: "'rohc@ietf.org'" <rohc@ietf.org>
Date: Wed, 5 Nov 2003 10:12:49 -0000 
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Hi All,

The SigComp implementer's guide has not been updated for Minneapolis so
still stands at version 01 (June 2003).  There has been only one issue
discussed on the list since then that should go into the next version.  This
is the issue of state retention priority (see
http://www1.ietf.org/mail-archive/working-groups/rohc/current/msg01748.html
and the responses).

I'll propose some text to cover this issue (and any others that arise in the
mean time) after Minneapolis.  

Best regards,

Abbie


Abbie Surtees
Senior Engineer
Networks
Roke Manor Research Ltd
Old Salisbury Lane
Romsey, SO51 0ZN
Tel:   +44 (0) 1794 833131


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Subject: [rohc] Question about ROHC FN
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Hi,

I am reading ROHC FN draft 02 and get confused about the following definitions:

The "from_list" encoding method defined in 5.3.3 is used to extact the 
presence and order flags from a list of items encoded with the 
list_of_known_length encoding method. The order_flag and and presence_flag 
are all control fields defined in 5.4.3 to indicate whether there are 
changes of order and presence in the list_of_known_length. These are all clear.

However, I am confused about their usage in the example shown at the end of 
5.4.3. The relevant part of the example is copied as follows.



    list_eg               ::= single_packet_format,
    {
         ...
      order_flag          ::= control_field,
      {
        base_field        ::= same_as(list_eg.list_of_fields.order_flag),
        compressed_method ::= irregular(1)
      },

      presence_flag       ::= control_field,
      {
        base_field        ::= same_as
                                  (list_eg.list_of_fields.presence_flag),
        compressed_method ::= irregular(1)
      },

      list_of_fields      ::= list_of_known_length,
      {
        list_length       ::= constant(8),
        order_flag        ::= from_list(list_eg.order_flag),
        presence_flag     ::= from_list(list_eg.presence_flag),
           ...
      }
    }


This seems to be a circular definition. For example, list_eg.order_flag is 
same_as(list_eg.list_of_fields.order_flag), while 
list_eg.list_of_fields.order_flag is from_list(list_eg.order_flag). Could 
anyone please explain it to me? Have I misunderstood anything here?

Thank you very much.
Haipeng





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Hi Folks,

The sigcomp-sip-00 has received few comments on the list (my thanks
to those who have kindly provided their feedbacks, particularly Abbie,
Jan, Pekka, and Robert).=20

The comments so far focus on the following issues:
- Is 8K minimum DMS too large for SIP/SigComp?
- Is 2K minimum SMS necessary?
- Need for multiplexing SigComp messages and SIP messages on=20
the same TCP connection?

Also, there has been little progress made on the compartment and state
management.

Since it seems to me that more discussions are needed on those issues,=20
the sigcomp-sip-00 was not updated for the Minneapolis meeting. I plan
to update the I-D in December before it expires, hopefully with less
issues open.=20

I would like to use this opportunity to encourage everyone to send your
comments on this list.=20

BR,
Zhigang


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Subject: [rohc] Questions about SigComp over SIP
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Hi,

I would like to develop SigComp over SIP, and find some issues unclear:

1.	Quoting from RFC3486 (Compressing the Session Initiation Protocol): =93=
The=20
presence of this parameter (comp=3Dsigcomp) in a URI indicates that the=20
request has to be compressed using SigComp, as defined in [2]=94 (section=
 no.=20
2).
Then, below in section no.7 it continues with: =93If a SIP client sends a=
=20
compressed request and the client transaction times out without having=20
received any response, the client SHOULD retry the same request without=20
using compression.=94  This quotation instructs us to send the same messa=
ge=20
without specifying the following details:
=B7	Is the parameter =91comp=3Dsigcomp=92 has to be removed from the orig=
inal=20
request URI before re-sending the uncompressed SIP request message?
=B7	In case that a SIP message has to be re-sent uncompressed, should it=20
contain different Via branch value (as it is done after transmission=20
failure, when using DNS) or different CSeq?


2.	Is there any example of SigComp state that is created dynamically and =
is=20
used by more than a single compartment? I=92m looking for an example of s=
tate=20
that is not =91global=92 or created during initialization process (such a=
s state=20
that keeps the dictionary or state that keeps compression/decompression=20
algorithm).


3.	Do the compression/decompression algorithms instruct us which states a=
nd=20
how many of them have to be created in order to support the SigComp featu=
re?=20
or Does the states creation depend on the way that SigComp is implemented=
?

I would appreciate answers to any of my questions.

Thank You,
John.

_________________________________________________________________
Protect your PC - get McAfee.com VirusScan Online=20
http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3D3963


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Subject: [rohc] Last Call: 'RObust Header Compression (ROHC): A Compression Profile
 for IP' to Proposed Standard
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The IESG has received a request from the Robust Header Compression WG to 
consider the following document:

- 'RObust Header Compression (ROHC): A Compression Profile for IP '
   <draft-ietf-rohc-ip-only-05.txt> as a Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by 2003-11-28.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-rohc-ip-only-05.txt


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Subject: [rohc] WiMAX Summit
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C'est un message de format MIME en plusieurs parties.

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Are WiMAX and WiFi complementary (local vs. metropolitan area =
networking)?=20
When and how will WiMAX evolve towards mobility?=20
Can WiMAX be considered as a migration path to 4G?=20
How is addressed the interoperability challenge?=20

These questions, among others, will be addressed during the WiMAX =
Summit, to be held in Paris on May 25 to 28, 2004.=20

A call for papers has been launched on-line at:
http://www.upperside.fr/wimax04/wimax04intro.htm

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<DIV>
<DIV><FONT size=3D2>Are WiMAX and WiFi complementary (local vs. =
metropolitan=20
area&nbsp;networking)? <BR>When and how will WiMAX evolve towards =
mobility?=20
<BR>Can WiMAX be considered as a migration path to 4G? <BR>How is =
addressed the=20
interoperability challenge? <BR><BR>These questions, among others, will =
be=20
addressed during the <FONT class=3Dtextebold color=3D#2d795f>WiMAX =
Summit</FONT>, to=20
be held in <SPAN class=3Dunderline>Paris on May 25 to 28, </SPAN>2004. =
<BR><BR>A=20
call for papers has been launched on-line at:</FONT></DIV>
<DIV><FONT size=3D2><A=20
href=3D"http://www.upperside.fr/wimax04/wimax04intro.htm">http://www.uppe=
rside.fr/wimax04/wimax04intro.htm</A></FONT></DIV><FONT=20
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From: "Finking, Robert" <robert.finking@roke.co.uk>
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Subject: [rohc] RE: Question about ROHC FN
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Jin,

Thanks for your question about "from_list" encoding. The whole of list encoding is certainly the most complicated aspect of the ROHC-FN proposal - thank you for taking the time to absorb it.

The answer is fairly straightforward; the notation covers both the information needed for encoding and that needed for decoding. The "from_list" encoding method looks at both the compressed and uncompressed packet to see which it knows. If  the uncompressed packet value (i.e. at the compressor), then the value is calculated from the uncompressed packet and associated context, as described in the FN draft. If the compressed packet value is known then the value is extracted directly from the compressed packet. This was not made explicit in the draft, thanks for highlighting the need for clarification on this point.

As an aside, this technique is also used elsewhere. If you look at the contemporary TCP profile (http://www.ietf.org/internet-drafts/draft-ietf-rohc-tcp-05.txt), in section 6.5.2. Default encoding methods, you will see that the notation for encoding a sequence number uses a similar technique. The definition appears to be circular:

       tcp_seq_number             ::=   derived_value,
       { field_length             ::=   constant(32),
         field_value              ::=
                              expression(uncomp(tcp_seq_number_scaled) *
                              uncomp(tcp_payload_size)                 +
                              uncomp(tcp_seq_number_residue))
       },

	 ...

       tcp_seq_number_scaled        ::=   control_field,

       { base_field                 ::=
                           expression(uncomp(tcp_ip.tcp_seq_number) //
                           uncomp(tcp_ip.tcp_payload_size)),
         compressed_method          ::=   static
       },

       tcp_seq_number_residue       ::=   control_field,

       { base_field                 ::=
                           expression(uncomp(tcp_ip.tcp_seq_number) mod
                           uncomp(tcp_ip.tcp_payload_size)),
         compressed_method          ::=   static
       },
       ...

Here we have sequence number defined in terms of scaled value and residue. We also have scaled value and residue defined in terms of sequence number. This looks circular, however one is use for compression and the other for decompression. If the value of a field is known a priori, then there is no need to calculate it.

The obvious question is why this doesn't happen with all encoding techniques. The fact is that most of them are sufficiently simple that the two definitions of encoding and decoding behaviour are encapsulated within the encoding method. For example static encoding implicitly checks whether the field value in the uncompressed packet is known, if so, it doesn't need to do anything (since static encoding takes up zero bits in the compressed packet). If it doesn't know the uncompressed value of the field, then it is extracted from the context. The two behaviours are still there, they're just hidden to keep the notation simple.

I hope this answers your question.

Regards

Robert Finking

-----Original Message-----
From: Haipeng Jin [mailto:haipengj@qualcomm.com] 
Sent: Wednesday, November 05, 2003 4:35 PM
To: Rohc (E-mail)
Subject: [rohc] Question about ROHC FN


Hi,

I am reading ROHC FN draft 02 and get confused about the following definitions:

The "from_list" encoding method defined in 5.3.3 is used to extact the 
presence and order flags from a list of items encoded with the 
list_of_known_length encoding method. The order_flag and and presence_flag 
are all control fields defined in 5.4.3 to indicate whether there are 
changes of order and presence in the list_of_known_length. These are all clear.

However, I am confused about their usage in the example shown at the end of 
5.4.3. The relevant part of the example is copied as follows.



    list_eg               ::= single_packet_format,
    {
         ...
      order_flag          ::= control_field,
      {
        base_field        ::= same_as(list_eg.list_of_fields.order_flag),
        compressed_method ::= irregular(1)
      },

      presence_flag       ::= control_field,
      {
        base_field        ::= same_as
                                  (list_eg.list_of_fields.presence_flag),
        compressed_method ::= irregular(1)
      },

      list_of_fields      ::= list_of_known_length,
      {
        list_length       ::= constant(8),
        order_flag        ::= from_list(list_eg.order_flag),
        presence_flag     ::= from_list(list_eg.presence_flag),
           ...
      }
    }


This seems to be a circular definition. For example, list_eg.order_flag is 
same_as(list_eg.list_of_fields.order_flag), while 
list_eg.list_of_fields.order_flag is from_list(list_eg.order_flag). Could 
anyone please explain it to me? Have I misunderstood anything here?

Thank you very much.
Haipeng





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To: "Finking, Robert" <robert.finking@roke.co.uk>,
        "'rohc@ietf.org'" <rohc@ietf.org>
From: Haipeng Jin <haipengj@qualcomm.com>
Subject: Re: [rohc] RE: Question about ROHC FN
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Robert,

Thank you for your explanation, that is very clear.

I have another question about the encoding method "uncompressible".

In ROHC FN 5.5.1, the example shows


      test_uncompressible ::= single_packet_format,
      {
        uncompressed_data ::=   length,
                                data,

        compressed_data   ::=   data,

        length            ::=   value(4, 1),

        data              ::=   uncompressible
                                  (test_uncompressible.length, 2, 2, 4)
      }


I am wondering whether it actually should be

         data ::= uncompressible (test_uncompressible.length, 2, 2, -4).

The uncompressible size of data should be 
(uncomp(test_uncompressible.length)/2)*2 - 4 instead 
of  (uncomp(test_uncompressible.length)/2)*2 + 4, right?

Also since you are mentioning ROHC-TCP, there seems  to be an inconsistent 
in using "uncompressible" in the ROHC FN draft and ROHC TCP draft. Again, 
ROHC FN 5.5.1 says


    The encoding method takes four parameters. The first parameter gives
    the base value for calculating the length of the data, whilst the
    next three parameters specify a divisor, multiplier and offset which
    scale the variable so that it specifies the size of the
    uncompressible field in bits.

So the uncompressible method is defined as uncompressible (base, divisor, 
multiplier, offset). While in ROHC-TCP 6.5.3, the generic format for 
compressing TCP option is given as:

        generic                   ::=   single_packet_format,

        { uncompressed_data       ::=   kind,                   %  8 bits
                                        length,                 %  8 bits
                                        data_item,     %  data_size bits

          compressed_data         ::=   data_item,

          kind                    ::=   static,
          length                  ::=   static,
          data_item               ::=
             uncompressible(tcp_ip.tcp_options.generic.length, 8, 1, -16)
        }
      }
    }
    }.

If we use the definition for uncompressible in ROHC-FN, the data_item size 
will be (tcp_ip.tcp_options.generic.length)/8 - 16, but the actual size in 
bits should be (tcp_ip. ... .length)*8 - 16. Right?

Thanks,

Haipeng


At 01:42 PM 11/7/2003 +0000, Finking, Robert wrote:
>Jin,
>
>Thanks for your question about "from_list" encoding. The whole of list 
>encoding is certainly the most complicated aspect of the ROHC-FN proposal 
>- thank you for taking the time to absorb it.
>
>The answer is fairly straightforward; the notation covers both the 
>information needed for encoding and that needed for decoding. The 
>"from_list" encoding method looks at both the compressed and uncompressed 
>packet to see which it knows. If  the uncompressed packet value (i.e. at 
>the compressor), then the value is calculated from the uncompressed packet 
>and associated context, as described in the FN draft. If the compressed 
>packet value is known then the value is extracted directly from the 
>compressed packet. This was not made explicit in the draft, thanks for 
>highlighting the need for clarification on this point.
>
>As an aside, this technique is also used elsewhere. If you look at the 
>contemporary TCP profile 
>(http://www.ietf.org/internet-drafts/draft-ietf-rohc-tcp-05.txt), in 
>section 6.5.2. Default encoding methods, you will see that the notation 
>for encoding a sequence number uses a similar technique. The definition 
>appears to be circular:
>
>        tcp_seq_number             ::=   derived_value,
>        { field_length             ::=   constant(32),
>          field_value              ::=
>                               expression(uncomp(tcp_seq_number_scaled) *
>                               uncomp(tcp_payload_size)                 +
>                               uncomp(tcp_seq_number_residue))
>        },
>
>         ...
>
>        tcp_seq_number_scaled        ::=   control_field,
>
>        { base_field                 ::=
>                            expression(uncomp(tcp_ip.tcp_seq_number) //
>                            uncomp(tcp_ip.tcp_payload_size)),
>          compressed_method          ::=   static
>        },
>
>        tcp_seq_number_residue       ::=   control_field,
>
>        { base_field                 ::=
>                            expression(uncomp(tcp_ip.tcp_seq_number) mod
>                            uncomp(tcp_ip.tcp_payload_size)),
>          compressed_method          ::=   static
>        },
>        ...
>
>Here we have sequence number defined in terms of scaled value and residue. 
>We also have scaled value and residue defined in terms of sequence number. 
>This looks circular, however one is use for compression and the other for 
>decompression. If the value of a field is known a priori, then there is no 
>need to calculate it.
>
>The obvious question is why this doesn't happen with all encoding 
>techniques. The fact is that most of them are sufficiently simple that the 
>two definitions of encoding and decoding behaviour are encapsulated within 
>the encoding method. For example static encoding implicitly checks whether 
>the field value in the uncompressed packet is known, if so, it doesn't 
>need to do anything (since static encoding takes up zero bits in the 
>compressed packet). If it doesn't know the uncompressed value of the 
>field, then it is extracted from the context. The two behaviours are still 
>there, they're just hidden to keep the notation simple.
>
>I hope this answers your question.
>
>Regards
>
>Robert Finking
>
>-----Original Message-----
>From: Haipeng Jin [mailto:haipengj@qualcomm.com]
>Sent: Wednesday, November 05, 2003 4:35 PM
>To: Rohc (E-mail)
>Subject: [rohc] Question about ROHC FN
>
>
>Hi,
>
>I am reading ROHC FN draft 02 and get confused about the following 
>definitions:
>
>The "from_list" encoding method defined in 5.3.3 is used to extact the
>presence and order flags from a list of items encoded with the
>list_of_known_length encoding method. The order_flag and and presence_flag
>are all control fields defined in 5.4.3 to indicate whether there are
>changes of order and presence in the list_of_known_length. These are all 
>clear.
>
>However, I am confused about their usage in the example shown at the end of
>5.4.3. The relevant part of the example is copied as follows.
>
>
>
>     list_eg               ::= single_packet_format,
>     {
>          ...
>       order_flag          ::= control_field,
>       {
>         base_field        ::= same_as(list_eg.list_of_fields.order_flag),
>         compressed_method ::= irregular(1)
>       },
>
>       presence_flag       ::= control_field,
>       {
>         base_field        ::= same_as
>                                   (list_eg.list_of_fields.presence_flag),
>         compressed_method ::= irregular(1)
>       },
>
>       list_of_fields      ::= list_of_known_length,
>       {
>         list_length       ::= constant(8),
>         order_flag        ::= from_list(list_eg.order_flag),
>         presence_flag     ::= from_list(list_eg.presence_flag),
>            ...
>       }
>     }
>
>
>This seems to be a circular definition. For example, list_eg.order_flag is
>same_as(list_eg.list_of_fields.order_flag), while
>list_eg.list_of_fields.order_flag is from_list(list_eg.order_flag). Could
>anyone please explain it to me? Have I misunderstood anything here?
>
>Thank you very much.
>Haipeng
>
>
>
>
>
>_______________________________________________
>Rohc mailing list
>Rohc@ietf.org
>https://www1.ietf.org/mailman/listinfo/rohc
>
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Date: Sun, 09 Nov 2003 19:17:17 +0100
From: Juergen Quittek <quittek@ccrle.nec.de>
To: "C. M. Heard" <heard@pobox.com>
cc: ROHC mailing list <rohc@ietf.org>,
        "Wijnen, Bert (Bert)" <bwijnen@lucent.com>
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Mike,

Thanks again for your careful check and the very good comments.
Please see replies inline:

--On 31.10.2003 0:15 h -0800 C. M. Heard wrote:

> On Tue, 21 Oct 2003, Juergen Quittek wrote:
>> > Below please find the detailed responses on all of your comments
>> > with two exceptions:
>> >
>> >   - 4.) IPR Notices
>> >         We are still checking the patent issues.
>> >         We will post the results on the mailing list
>> >         (with CC to you and Bert).
>>
>> Together with Lars-Erik, we checked the IPR issues.
>> They do not apply to the MIB or to MIB implementations.
>
> OK
>
>> >   - 11.) (e) ordering of the index components
>> >         This is discussed in a separate email thread on the list.
>>
>> We had long discussions on this issue.  So far indexing has looked like:
> [ ... details snipped, but each table used ifIndex ... ]
>> We want to change this such that the rohcChannelID becomes unique
>> for a ROHC MIB implementation independent of the interface or
>> instance to which the channel belongs.
>>
>> This simplifies the profile table, all context tables and the
>> RTP packet size table:
>>
>>   rohcProfileTable:       rohcChannelID, rohcProfile
>>   rohcContextTable:       rohcChannelID, rohcContextCID
>>   rohcUncmprContextTable: rohcChannelID, rohcContextCID
>>   rohcRtpContextTable:    rohcChannelID, rohcContextCID
>>   rohcRtpPacketSizeTable: rohcChannelID, rohcContextCID, rohcRtpPacketSize
>>
>> For the rohc channel table we still want to keep the old indexing
>> scheme, because we believe it is useful to be able to access channels
>> per interface and to access Instances by the rohc InstanceType:
>>
>>   rohcChannelTable:       ifIndex, rohcChannelID
>>   rohcInstanceTable:      ifIndex, rohcInstanceType, rohcChannelID
>>
>> Addressing a good suggestion from Abbie
>>
>> >>>>> On Tue, 8 Jul 2003, Surtees, Abigail wrote:
>> Abbie> However, looking at it from a MIB point of view I think
>> Abbie> it is confusing to have a seemingly redundant index.
>> Abbie>
>> Abbie> Hence, I suggest that rohcInstanceType continues to be
>> Abbie> an index but that there is some text added to the draft
>> Abbie> explaining why it is there and that if you want to
>> Abbie> uniquely identify one rohc instance (whether compressor
>> Abbie> or decompressor) then ifIndex and rohcChannelID are
>> Abbie> sufficient to do this.
>>
>> we suggest to modify the DESCRIPTION clauses [ ... details snipped ... ]
>>
>> A consequence of these changes would be that modules
>> ROHC-UNCMPRESSED-MIB and ROHC-RTP-MIB do not use anymore the object
>> ifIndex (or other objects from the IF-MIB and the IF-MIB can be
>> removed from the IMPORT clause and the DESCRIPTION clause of the
>> compliance statements.
>
> Allow me to point out that all three MIB modules have counters
> and that each counter points to ifCounterDiscontinuityTime as
> its discontinuity indicator.  So it seems to me that there are
> two problems here:
>
> - there is still a dependence on the IF-MIB

OK. I will fix this in the conformance section after we got feedback
from the IESG.

> - the absence of ifIndex from the index list makes it hard for
> a management application to find out which counters have
> sufferred a discontinuity when an instance if ifConcerDiscontinuityTime
> changes.
>
> So, I think you would be better off to re-instate ifIndex to the
> INDEX clauses and mention of IF-MIB in the compliance statements.

I'm hesitating to go back.  After many discussions, I am convinced
that we found a well balanced compromise.

The management application can read the rohcChannelTable (that is
still indexed by the ifIndex) and find a list of all channels that
are affected with their respective rohcChannelId.  Then it can
use the rohcChannelId as index of all counters in other tables,
which suffered from discontinuity.

> Another point that I noticed is that the OID assignments for the
> columnar objects are not contiguous, doubtless as a result of
> removing columns from the tables.  Here is the result of a quick
> grep:
>        ::= { rohcChannelEntry 2 }
>        ::= { rohcChannelEntry 3 }
>        ::= { rohcChannelEntry 4 }
>        ::= { rohcChannelEntry 5 }
>        ::= { rohcChannelEntry 13 }

Yes, this needs to be fixed.

>        ::= { rohcInstanceEntry 2 }
>        ::= { rohcInstanceEntry 4 }

And this also.

>        ::= { rohcInstanceEntry 5 }
>        ::= { rohcInstanceEntry 6 }
>        ::= { rohcInstanceEntry 7 }
>        ::= { rohcInstanceEntry 8 }
>        ::= { rohcInstanceEntry 9 }
>        ::= { rohcInstanceEntry 10 }
>        ::= { rohcInstanceEntry 11 }
>        ::= { rohcInstanceEntry 12 }
>        ::= { rohcInstanceEntry 13 }
>        ::= { rohcInstanceEntry 14 }
>        ::= { rohcInstanceEntry 15 }
>        ::= { rohcInstanceEntry 16 }
>        ::= { rohcInstanceEntry 17 }
>        ::= { rohcInstanceEntry 18 }
>        ::= { rohcInstanceEntry 19 }
>        ::= { rohcInstanceEntry 20 }
>        ::= { rohcProfileEntry 2 }
>        ::= { rohcProfileEntry 3 }
>        ::= { rohcProfileEntry 4 }
>        ::= { rohcProfileEntry 5 }
>        ::= { rohcProfileEntry 6 }
>        ::= { rohcContextEntry 2 }
>        ::= { rohcContextEntry 3 }
>        ::= { rohcContextEntry 4 }
>        ::= { rohcContextEntry 5 }
>        ::= { rohcContextEntry 6 }
>        ::= { rohcContextEntry 7 }
>        ::= { rohcContextEntry 8 }
>        ::= { rohcContextEntry 9 }
>        ::= { rohcContextEntry 10 }
>        ::= { rohcContextEntry 11 }
>        ::= { rohcContextEntry 12 }
>        ::= { rohcContextEntry 13 }
>        ::= { rohcContextEntry 14 }
>        ::= { rohcContextEntry 15 }
>        ::= { rohcContextEntry 16 }
>        ::= { rohcContextEntry 17 }
>        ::= { rohcContextEntry 18 }
>        ::= { rohcContextEntry 19 }
>        ::= { rohcContextEntry 20 }
>        ::= { rohcContextEntry 21 }
>        ::= { rohcContextEntry 22 }
>        ::= { rohcRtpContextEntry 3 }
>        ::= { rohcRtpContextEntry 4 }
>        ::= { rohcRtpContextEntry 5 }
>        ::= { rohcRtpContextEntry 6 }
>        ::= { rohcRtpContextEntry 7 }
>        ::= { rohcRtpContextEntry 8 }
>        ::= { rohcRtpContextEntry 9 }
>        ::= { rohcRtpContextEntry 10 }
>        ::= { rohcRtpContextEntry 11 }
>        ::= { rohcRtpContextEntry 12 }
>        ::= { rohcRtpContextEntry 13 }
>        ::= { rohcRtpContextEntry 14 }
>        ::= { rohcRtpContextEntry 15 }
>        ::= { rohcRtpContextEntry 16 }
>        ::= { rohcRtpContextEntry 17 }
>        ::= { rohcRtpPacketSizeEntry 3 }
>        ::= { rohcRtpPacketSizeEntry 4 }
>        ::= { rohcRtpPacketSizeEntry 5 }
>        ::= { rohcRtpPacketSizeEntry 6 }
>        ::= { rohcUncmprContextEntry 3 }
>        ::= { rohcUncmprContextEntry 4 }
>        ::= { rohcUncmprContextEntry 5 }
>
> You may want to consider fixing this if you decide to
> re-spin the MIB before publication.

Certainly. If we just change the compliancy statements and
the OID numbering, we can fix this at any stage in the
document process until the 48 hours review by the authors.

If we change the indexing again, we might have to discuss
it with more people.

> Regards,
>
> Mike

Thanks,

    Juergen








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Subject: RE: [rohc] Questions about SigComp over SIP
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Hi John,

Just to be accurate, the email subject would be better to read
"SIP over SigComp".

Please see comments in line.

BR,
Zhigang

> 1.	Quoting from RFC3486 (Compressing the Session=20
> Initiation Protocol): "The=20
> presence of this parameter (comp=3Dsigcomp) in a URI indicates that =
the=20
> request has to be compressed using SigComp, as defined in=20
> [2]" (section no.=20
> 2).
> Then, below in section no.7 it continues with: "If a SIP=20
> client sends a=20
> compressed request and the client transaction times out=20
> without having=20
> received any response, the client SHOULD retry the same=20
> request without=20
> using compression."  This quotation instructs us to send the=20
> same message=20
> without specifying the following details:
> =B7	Is the parameter 'comp=3Dsigcomp' has to be removed from=20
> the original=20
> request URI before re-sending the uncompressed SIP request message?

My understanding of RFC3486 (if correct) is that it's not needed.
The parameter "comp=3Dsigcomp" is mainly to indicate the *receiver's*=20
support of SigComp, not a mandate for a sender to send it. See quoted
text from section 2 below. (I agree the "hast to" wording mentioned =
above
is confusing. I don't know how it ended up there.)

   Therefore, the presence of comp=3Dsigcomp indicates that the SIP =
entity
   identified by the URI or by the Via header field supports SigComp and
   is willing to receive compressed messages.  Having comp=3Dsigcomp =
mean
   "willingness" as well as "support" allows the receiver of a SIP
   message to influence the decision of whether or not to use SigComp at
   a given time.


> =B7	In case that a SIP message has to be re-sent=20
> uncompressed, should it=20
> contain different Via branch value (as it is done after transmission=20
> failure, when using DNS) or different CSeq?

The values should be set according to SIP, regardless whether or not
SigComp is used. SigComp is just a transport.

> 2.	Is there any example of SigComp state that is created=20
> dynamically and is=20
> used by more than a single compartment? I'm looking for an=20
> example of state=20
> that is not 'global' or created during initialization process=20
> (such as state=20
> that keeps the dictionary or state that keeps=20
> compression/decompression=20
> algorithm).

I don't have one from the top of my head.=20
=20
> 3.	Do the compression/decompression algorithms instruct us=20
> which states and=20
> how many of them have to be created in order to support the=20
> SigComp feature?=20
> or Does the states creation depend on the way that SigComp is=20
> implemented?

Those depend on the compressor (sender). SigComp essentially=20
specifies the expected behaviour/constraints of the decompressor
(receiver). It's up to the compressor implementation on how
to use the SigComp framework and drive the operations.
Think about Java VM and applet.=20

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From: "West, Mark" <mark.a.west@roke.co.uk>
To: zhigang.c.liu@nokia.com
Cc: johnyjsmith1970@hotmail.com, rohc@ietf.org
Date: Mon, 10 Nov 2003 16:58:53 +0000 (GMT Standard Time)
Subject: RE: [rohc] Questions about SigComp over SIP
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Hi John, Zhigang,

A very small additional comment to one of Zhigang's answers...


> > 2.	Is there any example of SigComp state that is created
> > dynamically and is
> > used by more than a single compartment? I'm looking for an
> > example of state
> > that is not 'global' or created during initialization process
> > (such as state
> > that keeps the dictionary or state that keeps
> > compression/decompression
> > algorithm).
>
> I don't have one from the top of my head.

Well, isn't the bytecode an example of this?

It depends what you mean by 'initialisation', but the bytecode is not
'universal'.  Whereas, the 'static dictionary' is, I would imagine, always
present -- it's hard to imagine the dictionary (for example) being *sent*
by an endpoint... (I can only really see it being use if its presence is
advertised / mandated for the particular environment).

In any case, it's not really clear why it's important to have an
example... it can happen and is explicitly supported in the definition of
state management in RFC 3320.  So an implementation has to handle it, even
if you don't think that it's going to occur very often...

Cheers,

Mark.


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Haipeng,

The "uncompressible" encoding method is something of an
anachronism. We included it as a quick hack for handling
variable-length uncompressible fields, way back in the very
first version of the formal notation, and it has somehow
managed to survive unchanged ever since.

As your email makes clear, trying to use the "uncompressible"
encoding method can result in a serious headache. The
source of the confusion is the large selection of mysterious
parameters that follow the encoding method - it's difficult
to remember what the parameters are for, and the order
in which they're supposed to occur.

In more recent versions of the formal notation, we've tried
to reduce the reliance on magic numbers by converting the
potentially confusing parameters into proper, named fields
in their own right. We could easily do this for the
uncompressible encoding method as well, e.g:

foo ::= uncompressible
{
   base_length ::= same_as(some_other_field)
   multiplier ::= 8
   divisor ::= 1
   offset ::= 16
}

Even better, why not put the expression for deriving the
length of the uncompressible field straight into the formal
notation? This would look something like:

foo ::= uncompressible
{
   field_length ::= expression(uncomp(some_other_field) * 8 + 16)
}

Either of the above would help to avoid confusion about
what the parameters following the "uncompressible"
encoding method are actually for.

Regards,

Richard Price

P.S. Regarding the specific points raised below, I think that
the example in the ROHC-FN draft is correct - the length of
the uncompressible data is intended to be:

(uncomp(test_uncompressible.length)/2)*2 + 4

However, the use of "uncompressible" encoding in the
TCP profile is wrong - the parameters for the multiplier
and the divisor are the wrong way round. Giving each
parameter an explicit name would help to avoid this ;-)

>>----------
>>From: 	Haipeng Jin[SMTP:HAIPENGJ@QUALCOMM.COM]
>>Sent: 	Friday, November 07, 2003 6:25:58 PM
>>To: 	Finking, Robert; 'rohc@ietf.org'
>>Subject: 	Re: [rohc] RE: Question about ROHC FN
>>Auto forwarded by a Rule
>>
>Robert,
>
>Thank you for your explanation, that is very clear.
>
>I have another question about the encoding method "uncompressible".
>
>In ROHC FN 5.5.1, the example shows
>
>
>      test_uncompressible ::= single_packet_format,
>      {
>        uncompressed_data ::=   length,
>                                data,
>
>        compressed_data   ::=   data,
>
>        length            ::=   value(4, 1),
>
>        data              ::=   uncompressible
>                                  (test_uncompressible.length, 2, 2, 4)
>      }
>
>
>I am wondering whether it actually should be
>
>         data ::= uncompressible (test_uncompressible.length, 2, 2, -4).
>
>The uncompressible size of data should be 
>(uncomp(test_uncompressible.length)/2)*2 - 4 instead of  
>(uncomp(test_uncompressible.length)/2)*2 + 4, right?
>
>Also since you are mentioning ROHC-TCP, there seems  to be an inconsistent 
>in using "uncompressible" in the ROHC FN draft and ROHC TCP draft. Again, 
>ROHC FN 5.5.1 says
>
>
>    The encoding method takes four parameters. The first parameter gives
>    the base value for calculating the length of the data, whilst the
>    next three parameters specify a divisor, multiplier and offset which
>    scale the variable so that it specifies the size of the
>    uncompressible field in bits.
>
>So the uncompressible method is defined as uncompressible (base, divisor, 
>multiplier, offset). While in ROHC-TCP 6.5.3, the generic format for 
>compressing TCP option is given as:
>
>        generic                   ::=   single_packet_format,
>
>        { uncompressed_data       ::=   kind,                   %  8 bits
>                                        length,                 %  8 bits
>                                        data_item,     %  data_size bits
>
>          compressed_data         ::=   data_item,
>
>          kind                    ::=   static,
>          length                  ::=   static,
>          data_item               ::=
>             uncompressible(tcp_ip.tcp_options.generic.length, 8, 1, -16)
>        }
>      }
>    }
>    }.
>
>If we use the definition for uncompressible in ROHC-FN, the data_item size 
>will be (tcp_ip.tcp_options.generic.length)/8 - 16, but the actual size in 
>bits should be (tcp_ip. ... .length)*8 - 16. Right?
>
>Thanks,
>
>Haipeng

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From: "Lars-Erik Jonsson (LU/EAB)" <lars-erik.jonsson@ericsson.com>
To: rohc@ietf.org
Date: Tue, 11 Nov 2003 18:07:41 +0100
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Subject: [rohc] ROHC TCP requirements, compression if IPv4 Options are present?
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While reviewing the ROHC TCP requirements, we have found that it
suggests compression should work also with IPv4 Options. We believe
this requirement does not really make sense, and suggest to replace
the current text:
"2.2.  Supported Headers and Kinds of TCP Streams

    1. IPv4 and IPv6: Must support both IPv4 and IPv6. This means that
       all possible changes in the IP header fields must be handled by
       the compression scheme, and commonly changing fields should be
       compressed efficiently. Compression must not be disabled if IPv4
       Options or IPv6 Extensions are present. The compression scheme
       must further consider as normal operation the scenario where
       Explicit Congestion Notification (ECN) [RFC-3168] is applied and
       support efficient compression also in the case when the ECN bits
       are used."

With:
"2.2.  Supported Headers and Kinds of TCP Streams

    1. IPv4 and IPv6: Must support both IPv4 and IPv6. This means that
       all possible changes in the IP header fields must be handled by
       the compression scheme, and commonly changing fields should be
       compressed efficiently. Compression must not be disabled if IPv6
       Extensions are present. The compression scheme must further
       consider as normal operation the scenario where Explicit
       Congestion Notification (ECN) [RFC-3168] is applied, and support =

       efficient compression also in the case when the ECN bits are =
used.

/Lars-Erik et al.


-----------------------------------
Lars-Erik Jonsson, M.Sc
Wireless IP Optimizations
AWARE - Advanced Wireless Algorithm Research
Ericsson Research, Corporate Unit
Ericsson AB
Box 920, S-971 28 Lule=E5, 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.=20

_______________________________________________
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Subject: RE: [rohc] Old issue - multiplexing compressed and uncompressed messages in the same TCP connection -  long
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Hello All,

Thanks for Robert to raise this old issue again. My personal=20
view has been that multiplexing is really good (and easy) to=20
have. Below Robert has given a good in-depth description of the
reasons. I'd like also to hear opinions from other people,
especially if they are different. Otherwise, I plan to update
sigcomp-sip I-D accordingly.

BR,
Zhigang

> -----Original Message-----
> From: rohc-admin@ietf.org [mailto:rohc-admin@ietf.org]On Behalf Of ext
> Robert.Sugar@nokia.com
> Sent: October 20, 2003 03:18 AM
> To: rohc@ietf.org
> Subject: [rohc] Old issue - multiplexing compressed and uncompressed
> messages in the same TCP connection - long
>=20
>=20
> Hi all,
>=20
>  You probably remember that discussion over $subject. As far=20
> as I can remember, conclusion was that although it could be=20
> implemented with a common transport layer, there is really no=20
> need to add the extra complexity.=20
>=20
> RFC3486 has the following, reflecting that:
> "If a SIP client sends a compressed request and the client=20
> transaction times out without having received any response,=20
> the client SHOULD retry the same request without using=20
> compression.  If the compressed request was sent over a TCP=20
> connection, the client SHOULD close that connection and open=20
> a new one to send the uncompressed request. Otherwise the=20
> server would not be able to detect the beginning of the new message."
>=20
> So, why do we need compressed and uncompressed messages=20
> multiplexed in the first place?
>=20
> 1. Fallback to uncompressed mode (also suggested in RFC3486)
> 2. Large messages: OUTPUT instruction stops after 64Kbytes of=20
> data, so SIP messages larger than 64K are to be sent=20
> uncomperssed. This is not an impossible case, since SIP=20
> messages may carry arbitrary MIME attachments, such as XML=20
> (->presence info), WAV (->ringing tone), JPG (-> caller's photo), etc.
> 3. If there is no comp=3Dsigcomp URI to the proxy (proxy=20
> discovery usually yields only the IP address), initial=20
> messages are to be sent uncompressed, but subsequent messages=20
> could be sent compressed if comp=3Dsigcomp is inserted into the=20
> Record-Route by the proxy. If no comp=3Dsigcomp contact is=20
> given later, all initial requests will be sent uncompressed.
> 4. RFC3486 allows inserting comp=3Dsigcomp into SIP headers=20
> anytime. Preference for insertion might be for example the=20
> load of the proxy. If there are parallel sessions in the same=20
> compartment, at least one compressed and one uncompressed,=20
> TCP connections should be closed and re-opened anytime=20
> messages interleave. This would cause unnecessary extra=20
> delays and roundtrips due to the TCP handshake.
>=20
> Why is it hard to close and re-open the TCP connections?
>=20
>  Problem is that if we use lower-layer security, such as=20
> IPSec (as in 3GPP for one example), we cannot open and close=20
> ports at will, because security associations have fixed=20
> selectors (source IP, source port, destination IP,=20
> destination port, protocol). The only way we could open a new=20
> TCP connection is to use exactly the same 5 selectors as the=20
> original one.
> According to TCP: "Another effect of this 2MSL wait is that=20
> while the TCP connection is in the 2MSL wait, the socket pair=20
> defining the connection (client IP address, client port=20
> number, server IP address, server port number) cannot be=20
> reused.  That connection can only be reused when the 2MSl=20
> wait is over."
> MSL is the maximum segment lifetime, which is by RFC=20
> recommendation 2 minutes, but many implementations use 30=20
> seconds. Effectively, it means that the connection cannot be=20
> reused for 1-4 minutes after closure.
>=20
> How it could be implemented?
>=20
> A possible implementation could be:
> -	Transport looks at the first byte of the message, and=20
> cheks whether it starts with 11111.
> -	If yes, the stream is passed to SigComp that reads it=20
> till 0xFFFF and returns
> -	If no, the stream is passed to SIP that reads it as=20
> long as indicated by Content-Length
> -	Rinse, Repeat
>=20
>  Of course if the Content-Length is improperly filled,=20
> missing or SigComp messages contain additional 0xFFFF or do=20
> not end with 0xFFFF, it might happen that a parser receives=20
> messages that are erroneous, partial or belong to the other=20
> protocol. In that case the next message or all messages=20
> inside that stream could be lost. But similar problems also=20
> arise in a stream containing SIP or SigComp messages only.
>=20
> What do you think?
>=20
> BR,
> Robert
>=20
> _______________________________________________
> Rohc mailing list
> Rohc@ietf.org
> https://www1.ietf.org/mailman/listinfo/rohc
>=20

_______________________________________________
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From: "West, Mark" <mark.a.west@roke.co.uk>
To: zhigang.c.liu@nokia.com
Cc: Robert.Sugar@nokia.com, rohc@ietf.org
Date: Wed, 12 Nov 2003 22:39:43 +0000 (GMT Standard Time)
Subject: RE: [rohc] Old issue - multiplexing compressed and uncompressed
 messages in the same TCP connection -  long
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Hi all,

I still go along with what I still believe was the majority view from S-F,
which was: 'yeah, you could do this, but why would you want to..?'

But first of all, can I be *really* pedantic about terminology..?  We
should differentiate between:
- SigComp message; and
- non-SigComp message

With respect to the former, these can be compressed or uncompressed.
(Further, the compression used can make use of states at the remote
end-point, or be state-less).

The main justification given for multiplexing sigcomp and non-sigcomp
messages seems to be that it's too hard to re-open a new TCP connection.
Now, there may be transport-friendliness issues with continually closing &
opening TCP connections, but the idea of having a fixed rather than
ephemeral source port seems slightly weird...  not least because that
seems to break recovery if you close a session on detection of wrong
content-length, for example?

I agree that there are many reasons for needing to be able to send a mix
of SigComp & non-SigComp messages between 2 SIP peers.  These are not
necessarily reasons for multiplexing these messages on the same TCP flow,
though.

If you were allowed to have multiple TCP connections or to re-open a new
connection easily, why would you want to do multiplexing on a single
connection..?

This implementation suggestion:

> > A possible implementation could be:
> > -	Transport looks at the first byte of the message, and
> > cheks whether it starts with 11111.
> > -	If yes, the stream is passed to SigComp that reads it
> > till 0xFFFF and returns
> > -	If no, the stream is passed to SIP that reads it as
> > long as indicated by Content-Length
> > -	Rinse, Repeat

makes it sound easy, but the throw-away 'is passed to' seems to me to
conceal all of the complexity / layering problems / tricky stuff...

So, no, I don't like the idea of putting SigComp & non-SigComp messages on
the same connection...

... but feel free to try and convince me otherwise..!

Cheers,

Mark.



On Wed, 12 Nov 2003 zhigang.c.liu@nokia.com wrote:

> Date: Wed, 12 Nov 2003 16:25:26 -0500
> From: zhigang.c.liu@nokia.com
> To: Robert.Sugar@nokia.com, rohc@ietf.org
> Subject: RE: [rohc] Old issue - multiplexing compressed and uncompressed
>     messages in the same TCP connection -  long
>
> Hello All,
>
> Thanks for Robert to raise this old issue again. My personal
> view has been that multiplexing is really good (and easy) to
> have. Below Robert has given a good in-depth description of the
> reasons. I'd like also to hear opinions from other people,
> especially if they are different. Otherwise, I plan to update
> sigcomp-sip I-D accordingly.
>
> BR,
> Zhigang
>

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Subject: multiplexing SigComp and non-SigComp messages on the same TCP connection (was RE: [rohc] Old issue - multiplexing compressed and uncompressed messages in the same TCP connection -  long)
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Hi Mark,

I'm glad that my "threat" in my last sentence achieved its purpose. :-)
And thanks for the commetns. I feel discussions are really needed to=20
clear this subject.

Please see my comments in line.

Zhigang

> But first of all, can I be *really* pedantic about terminology..?  We
> should differentiate between:
> - SigComp message; and
> - non-SigComp message

You're absolutely right. I was too busy to correct this, but this
is important. I changed the subject line to reflect this.

> If you were allowed to have multiple TCP connections or to=20
> re-open a new connection easily, why would you want to do=20
> multiplexing on a single connection..?

This is the key question. Actually that is one of the questions=20
I've asked Robert during our offline discussion. The answers were=20
(if I understood correctly):

- Re-opening a new TCP connection means extra delay. I tend
to agree, since the establishment and tear-down of a TCP connection=20
require quite some procedures in the state machine.=20

- In certain cases, we are not allowed to have multiple TCP connections.
I learned that this is the case for 3GPP, in particular, due to security
policy as Robert described earlier. However, it is unclear to
me if it is simply a bad choice made by 3GPP, or it is due to some=20
technical reasons that are more generic than 3GPP. This is something
I would like to figure out during this discussion.

> This implementation suggestion:
>=20
> > > A possible implementation could be:
> > > -	Transport looks at the first byte of the message, and
> > > cheks whether it starts with 11111.
> > > -	If yes, the stream is passed to SigComp that reads it
> > > till 0xFFFF and returns
> > > -	If no, the stream is passed to SIP that reads it as
> > > long as indicated by Content-Length
> > > -	Rinse, Repeat
>=20
> makes it sound easy, but the throw-away 'is passed to' seems to me to
> conceal all of the complexity / layering problems / tricky stuff...

I do agree this doesn't look nice in the layering diagram. However,
I don't think it is complex or tricky to implement. SigComp already=20
requires quite interactions with SIP layer anyway. Or, perhaps I
missed something here?

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From: "West, Mark" <mark.a.west@roke.co.uk>
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Hi Zhigang,

I agree -- you can't beat a good threat... ;-)

Comments, as ever, in-line.

Cheers,

Mark.


> > If you were allowed to have multiple TCP connections or to
> > re-open a new connection easily, why would you want to do
> > multiplexing on a single connection..?
>
> This is the key question. Actually that is one of the questions
> I've asked Robert during our offline discussion. The answers were
> (if I understood correctly):
>
> - Re-opening a new TCP connection means extra delay. I tend
> to agree, since the establishment and tear-down of a TCP connection
> require quite some procedures in the state machine.
>

This is true: there's a 3-way handshake to establish a new TCP connection.
I'm not sure how much of a big deal this is, since most of the conditions
for sending a non-SigComp message involve significant latency
(retransmission; message > 64kbyte; ...)

However, from my reading of RFC 3261 (I am not now, nor have ever claimed
to be any kind of SIP expert; so if anyone wishes to correct this, feel
free..!) you could have a SigComp & non-SigComp TCP connection open...

> - In certain cases, we are not allowed to have multiple TCP connections.
> I learned that this is the case for 3GPP, in particular, due to security
> policy as Robert described earlier. However, it is unclear to
> me if it is simply a bad choice made by 3GPP, or it is due to some
> technical reasons that are more generic than 3GPP. This is something
> I would like to figure out during this discussion.
>

My personal opinion is that this security model is broken.  It is usual to
have ephemeral source ports and I can't see how it helps to nail down the
source port.  RFC 3261 section 18 says that the source port is 'often
ephemeral'.  Moreoever, 3261 describes having a 'dwell time' for a
connection to allow a transaction to complete within a single connection.
After this time, the connection is/can be closed.  The 'fixed port' model
means that you can't re-open a connection within the time-wait time...

... also all the text in section 18 about error conditions and re-opening
connections seems to be rendered unusable in this scenario...

> > This implementation suggestion:
> >
> > > > A possible implementation could be:
> > > > -	Transport looks at the first byte of the message, and
> > > > cheks whether it starts with 11111.
> > > > -	If yes, the stream is passed to SigComp that reads it
> > > > till 0xFFFF and returns
> > > > -	If no, the stream is passed to SIP that reads it as
> > > > long as indicated by Content-Length
> > > > -	Rinse, Repeat
> >
> > makes it sound easy, but the throw-away 'is passed to' seems to me to
> > conceal all of the complexity / layering problems / tricky stuff...
>
> I do agree this doesn't look nice in the layering diagram. However,
> I don't think it is complex or tricky to implement. SigComp already
> requires quite interactions with SIP layer anyway. Or, perhaps I
> missed something here?
>

I agree that there are significant interactions between SIP and SigComp.
However, these are all defined in RFC 3320.  The point is that SigComp
should (from a transport perspective) be able to look (mostly) like an
alternative transport for SIP.  Multiplexing really breaks that model.

I think my point is that *not* allowing multiplexing means that you can
separate the implementation details.  (You don't have to if you don't want
to, but you *can*).  If you insist on allowing multiplexing, then a
receiving endpoint *must* support it, and that's going to put a bunch of
constraints and requirements on the implementation.  I don't think that
those constraints need to be there, and so I'd prefer not to have them...


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Subject: Re: [rohc] sigcomp-sip status
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Here is a braindump of a breakfast meeting Mark and I just had.
All great ideas are Mark's, all errors are mine...
Please excuse the terseness, this is mainly to preserve the many 
threads we had.

Gruesse, Carsten

2003-11-13 Mark West/cabo

Re sigcomp-sip draft:

8 K DMS is a no-brainer

2 K SMS means
-- can't compress to stateless proxies (i.e., "really stateless"
proxies can't advertise sigcomp)
0 K SMS means
-- can't establish state in initial message and rely on it
-- can still assume it's there -- need to wait for ACK to find out
for real
Mark will find a number for what 0 K SMS "costs".

Multiplexing:
Please use minimal UDVM programs (INPUT-BYTE, OUTPUT, JUMP) to carry
uncompressed messages in SigComp streams.  This is seven bytes plus a
four-byte SigComp header.

Compartment:
For INVITE/dialogs, the draft seems fine (but see below).
Note that, when a dialog ends, state may need to be kept around for
the odd retransmission (cf. TIME_WAIT).

Re: Maintaining state across dialogs: I propose to create a new SIP
header like:
SigComp-State-Promise: 1230402134098913284a729f 20h
(or absolute time, but relative seems more conservative --
implementations will need to compensate for RTTs and some time skew)
As with any state promise, this is just an optimization, so need not
be reliable.

REGISTER transactions?
There is no dialog to attach the compartment to.
BTW.  It would be nice to be able to keep user state (e.g., for a
re-register) for the duration of the registration.

What other pieces of state (in addition to dialog and registration)
are there that we may want to attach a compartment to?

The transaction/dialog dilemma could be solved by installing the
compartment for a transaction and later binding that to a dialog.
Some implementations may not even know the compartment at 100 Trying
time, so state installation may have to wait for 180 Ringing.
This would require the state to remain in limbo until SIP stack is
ready to acknowledge, which is what we are trying to minimize.


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From: "Jan Christoffersson (LU/EAB)" <jan.christoffersson@ericsson.com>
To: zhigang.c.liu@nokia.com, "'West, Mark'" <mark.a.west@roke.co.uk>,
        Robert.Sugar@nokia.com, rohc@ietf.org
Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp messages on t
	he same TCP connection
Date: Fri, 14 Nov 2003 11:17:53 +0100
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Hi all,

it seems to be different opinions on how to handle SigComp and SIP over TCP. I think that multiplexing over the same TCP connection is a good idea. Of course, there can be situations where it is not wanted, but then, there is nothing stopping you from opening a separate connection. To me, the main arguments for allowing multiplexing over the same TCP connection are

1. The extra delay caused by opening and closing TCP connections when switching between SIP and SigComp should not be neglected. In some cases it may be of little concern but in others it may be important. Reducing delay has been the most important driving factor for SigComp. Multiplexing over the same connection reduces the delay.

2.  We have the tools needed for multiplexing SigComp and SIP on the same TCP connection. Why not do it? It's faster than opening a new connection.

3. Mark argues that if multiplexing is allowed, the receiver needs to support it. But doesn't it already need to do this? As Zhigang pointed out, RFC3486 states that

"If a SIP client sends a compressed request and the client transaction
   times out without having received any response, the client SHOULD
   retry the same request without using compression.  If the compressed
   request was sent over a TCP connection, the client SHOULD close that
   connection and open a new one to send the uncompressed request."

In this text, it says SHOULD and not MUST, i.e. there could be implementations that send SIP and SigComp on the same TCP connection, so all implementations should be able to handle it. In any case, we have delimiters to determine where SigComp messages start and end, and if there is something other than SigComp in between an end and a start, we should be able to handle it.

I feel that we should allow multiplexing of SigComp and SIP on the same TCP connection.

Best Regards
Jan
-----Original Message-----
From: rohc-admin@ietf.org [mailto:rohc-admin@ietf.org]On Behalf Of West,
Mark
Sent: den 13 november 2003 14:57
To: zhigang.c.liu@nokia.com
Cc: Robert.Sugar@nokia.com; rohc@ietf.org
Subject: [rohc] Re: multiplexing SigComp and non-SigComp messages on the
same TCP connection



Hi Zhigang,

I agree -- you can't beat a good threat... ;-)

Comments, as ever, in-line.

Cheers,

Mark.


> > If you were allowed to have multiple TCP connections or to
> > re-open a new connection easily, why would you want to do
> > multiplexing on a single connection..?
>
> This is the key question. Actually that is one of the questions
> I've asked Robert during our offline discussion. The answers were
> (if I understood correctly):
>
> - Re-opening a new TCP connection means extra delay. I tend
> to agree, since the establishment and tear-down of a TCP connection
> require quite some procedures in the state machine.
>

This is true: there's a 3-way handshake to establish a new TCP connection.
I'm not sure how much of a big deal this is, since most of the conditions
for sending a non-SigComp message involve significant latency
(retransmission; message > 64kbyte; ...)

However, from my reading of RFC 3261 (I am not now, nor have ever claimed
to be any kind of SIP expert; so if anyone wishes to correct this, feel
free..!) you could have a SigComp & non-SigComp TCP connection open...

> - In certain cases, we are not allowed to have multiple TCP connections.
> I learned that this is the case for 3GPP, in particular, due to security
> policy as Robert described earlier. However, it is unclear to
> me if it is simply a bad choice made by 3GPP, or it is due to some
> technical reasons that are more generic than 3GPP. This is something
> I would like to figure out during this discussion.
>

My personal opinion is that this security model is broken.  It is usual to
have ephemeral source ports and I can't see how it helps to nail down the
source port.  RFC 3261 section 18 says that the source port is 'often
ephemeral'.  Moreoever, 3261 describes having a 'dwell time' for a
connection to allow a transaction to complete within a single connection.
After this time, the connection is/can be closed.  The 'fixed port' model
means that you can't re-open a connection within the time-wait time...

... also all the text in section 18 about error conditions and re-opening
connections seems to be rendered unusable in this scenario...

> > This implementation suggestion:
> >
> > > > A possible implementation could be:
> > > > -	Transport looks at the first byte of the message, and
> > > > cheks whether it starts with 11111.
> > > > -	If yes, the stream is passed to SigComp that reads it
> > > > till 0xFFFF and returns
> > > > -	If no, the stream is passed to SIP that reads it as
> > > > long as indicated by Content-Length
> > > > -	Rinse, Repeat
> >
> > makes it sound easy, but the throw-away 'is passed to' seems to me to
> > conceal all of the complexity / layering problems / tricky stuff...
>
> I do agree this doesn't look nice in the layering diagram. However,
> I don't think it is complex or tricky to implement. SigComp already
> requires quite interactions with SIP layer anyway. Or, perhaps I
> missed something here?
>

I agree that there are significant interactions between SIP and SigComp.
However, these are all defined in RFC 3320.  The point is that SigComp
should (from a transport perspective) be able to look (mostly) like an
alternative transport for SIP.  Multiplexing really breaks that model.

I think my point is that *not* allowing multiplexing means that you can
separate the implementation details.  (You don't have to if you don't want
to, but you *can*).  If you insist on allowing multiplexing, then a
receiving endpoint *must* support it, and that's going to put a bunch of
constraints and requirements on the implementation.  I don't think that
those constraints need to be there, and so I'd prefer not to have them...


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From: "West, Mark" <mark.a.west@roke.co.uk>
To: "Jan Christoffersson (LU/EAB)" <jan.christoffersson@ericsson.com>
Cc: zhigang.c.liu@nokia.com, Robert.Sugar@nokia.com, rohc@ietf.org
Date: Fri, 14 Nov 2003 14:09:16 +0000 (GMT Standard Time)
Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp messages on t
 he same TCP connection
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Hi Jan,

Comments added!

Cheers,

Mark.

>
> 1. The extra delay caused by opening and closing TCP connections when
> switching between SIP and SigComp should not be neglected. In some cases
> it may be of little concern but in others it may be important. Reducing
> delay has been the most important driving factor for SigComp.
> Multiplexing over the same connection reduces the delay.
>

But most of the cases where you want to switch between SigComp /
non-SigComp are inherently cases where there is inherent latency.

And RFC 3261 (if I read it correctly, but I'm no SIP guru), talks about
only try to keep a transaction within in a single connection...

> 2.  We have the tools needed for multiplexing SigComp and SIP on the
> same TCP connection. Why not do it? It's faster than opening a new
> connection.
>

But the 'tools' are horrible!  Reiterating the sentiments from S-F: 'yeah,
you could, but why would you want to..?'

> 3. Mark argues that if multiplexing is allowed, the receiver needs to
> support it. But doesn't it already need to do this? As Zhigang pointed
> out, RFC3486 states that
>
> "If a SIP client sends a compressed request and the client transaction
>    times out without having received any response, the client SHOULD
>    retry the same request without using compression.  If the compressed
>    request was sent over a TCP connection, the client SHOULD close that
>    connection and open a new one to send the uncompressed request."
>
> In this text, it says SHOULD and not MUST, i.e. there could be
> implementations that send SIP and SigComp on the same TCP connection, so
> all implementations should be able to handle it. In any case, we have
> delimiters to determine where SigComp messages start and end, and if
> there is something other than SigComp in between an end and a start, we
> should be able to handle it.
>

That is slicing a semantic a shade too thin for me!  But, if you want to
play word games, then I'll point out that 'uncompressed' does not mean 'do
not use SigComp'.

Carsten & I have whipped up a nice little 13-byte standard header:

  0xf8, 0x00, 0xa1, 0x1c, 0x01, 0x86, 0x09, 0x22,
  0x86, 0x01, 0x16, 0xf9, 0x23, ...

that will simply output the following bytes!

It's a SigComp message, but there's no compression...

And let's not forget that the purpose of draft-ietf-rohc-sip-sigcomp is to
fill in the missing pieces of the mapping jigsaw (i.e. the bits that
aren't in RFC 3486).  And that's what we're arguing about.  (Although I
was fairly sure that we'd already decided this.  *sigh*)

> I feel that we should allow multiplexing of SigComp and SIP on the same
> TCP connection.
>


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From: "West, Mark" <mark.a.west@roke.co.uk>
To: Carsten Bormann <cabo@tzi.org>
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Subject: Re: [rohc] sigcomp-sip status
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Hi all,

A bit of a follow-up, as promised/threatened...

... and I reserve the right to include errors of my own..! ;-)

>
> 2 K SMS means
> -- can't compress to stateless proxies (i.e., "really stateless"
> proxies can't advertise sigcomp)
> 0 K SMS means
> -- can't establish state in initial message and rely on it
> -- can still assume it's there -- need to wait for ACK to find out
> for real
> Mark will find a number for what 0 K SMS "costs".
>

I'm assuming the following simple sequence of 3 messages in a one-way
flow: REGISTER, INVITE, and ACK.  These have been run through a script
that I use for assessing compression efficiency.

For these results, the compression algorithm is EPIC (from the SigComp
users' guide), 'cause that's what I happen to have to hand...

The uncompressed message sizes are 322, 489 and 288 bytes, respectively.

The EPIC bytecode that I'm currently playing with is ~400 bytes.

So:

Per-message compression with static dictionary gives:
126, 201 & 131 bytes = 458 bytes in total (not including SigComp message
overhead or bytecode)
Bytecode adds 400 bytes to each message.  This, it can be seen, is not
terribly good value!  If bytecode were known to be available (advertised
by remote endpoint), some of this overhead could be avoided.

Dynamic compression (also with static dictionary) gives:
126, 157 & 23 bytes = 306 bytes total (not including overheads)
Bytecode adds 400 bytes to first message.

If you deferred the creation of state until receiving the advertisement of
2k SMS, then it would look (as expected), like:
126, 201 & 23 bytes = 350 bytes total (not including overheads)
Bytecode adds 400 bytes to the first 2 messages (excepting advertisements
or well-known bytecode)

As an observation, I can't see any reason why you would use the last case!
If it is *possible* that SMS will be available, and you know that it will
be at least 2k, then why not make an 'optimistic' state-creation request
on the first message?  If no SMS is advertised / no ack is received in a
response, then you have to use per-message / stateless compression.  If,
however, SMS is advertised / an ack is received, then everything should
proceed exactly as if you had known that there was 2k SMS all along..!

For the per-message / boot-strap case, we've played around a bit with
compact byte-code for simple compression such that the compressed message
+ bytecode is still smaller than the original message.

The 'best-case' though, would clearly be to have a 'well-known' algorithm
that could be used for the per-message / stateless scenario.

Hopefully that more or less answers the question..?

Cheers,

Mark.

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Subject: [rohc] Re: multiplexing SigComp and non-SigComp messages on the same TCP
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"zhigang.c.liu@nokia.com" <zhigang.c.liu@nokia.com> writes:
>- In certain cases, we are not allowed to have multiple TCP connections.
>I learned that this is the case for 3GPP, in particular, due to security
>policy as Robert described earlier. However, it is unclear to
>me if it is simply a bad choice made by 3GPP, or it is due to some 
>technical reasons that are more generic than 3GPP. This is something
>I would like to figure out during this discussion.

	I'd hazard to say that this is simply a series of bad choices
	made by MMUSIC, IPSEC, 3GPP and this working group. ;)

	In any case, SigComp relies on external protocol to provide it
	with security. The SIP does not provide such protection, so it
	is up to lower layers. 3GPP chose IPSEC because SIP uses UDP, so
	they could not use TLS. IPSEC does not naturally provide
	security at application layer like that SIP requires here, so
	3GPP made an IPSEC hack that secures only one UDP and TCP
	connection at a time.

	Personally I'd like to blame ITU-T, because they specified
	horribly complex H.323 so that it simply had to be destroyed by
	SIP, and only selling argument SIP had back then was that it was
	simple and used UDP.

	In any case, I don't think 3GPP is going to change their rel5
	security solution any more (it has been tried many time before
	and now they are almost complete with next release).

>> This implementation suggestion:

>> > > A possible implementation could be:
>> > > -	Transport looks at the first byte of the message, and
>> > > cheks whether it starts with 11111.
>> > > -	If yes, the stream is passed to SigComp that reads it
>> > > till 0xFFFF and returns
>> > > -	If no, the stream is passed to SIP that reads it as
>> > > long as indicated by Content-Length
>> > > -	Rinse, Repeat

>> makes it sound easy, but the throw-away 'is passed to' seems to me to
>> conceal all of the complexity / layering problems / tricky stuff...

>I do agree this doesn't look nice in the layering diagram. However,
>I don't think it is complex or tricky to implement. SigComp already 
>requires quite interactions with SIP layer anyway. Or, perhaps I
>missed something here?

	Speaking from SIP point-of-view: SigComp sits on one layer above
	SIP parser (at transport layer), so this kind of solution fits
	nicely in the SIP layering. ;-)
<
					Pekka


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From: Pekka Pessi <Pekka.Pessi@nokia.com>
Date: Fri, 14 Nov 2003 20:55:00 +0200
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Subject: [rohc] Byte copying rules with STATE-ACCESS and END-MESSAGE
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Hello all SigComp lawyers,
	
The RFC 3320 does not state whether the byte copying rules should be
used when retrieving the partial state identifier given to the
STATE-ACCESS operation. Also spec is silent with the requested feedback
or returned parameters to the END-MESSAGE operation.

The partial state identifier given to STATE-FREE is byte copied, so I'd
expect similar parameter to STATE-ACCESS to be byte copied, too.

If byte copying rules should *not* be used, it might be a good idea to
mention that somewhere, or even add some torture tests to test correct
operation.

--Pekka

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Subject: Re: [rohc] Re: multiplexing SigComp and non-SigComp messages on the
 same TCP connection
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 Mark's message of "Thu, 13 Nov 2003 13:57:24 +0000 (GMT Standard Time)")
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West, Mark <mark.a.west@roke.co.uk> writes:
>...Moreoever, 3261 describes having a 'dwell time' for a
>connection to allow a transaction to complete within a single connection.
>After this time, the connection is/can be closed.  

	In general, the connection should be kept around for 64*T1 or 32
	seconds (or longer, if the client has longer T1 than 500 ms). 
	Because only the client knows its T1 value, it should close the
	connection, not the server.

	If the client is a SIP UA behind a NAT box, it wants to keep the
	connection open for the duration of its registration. If it
	wants to open another connection, in most cases it has to do a
	REGISTER request in order to get the new connection
	authenticated. Because the TCP connections are expensive for the
	server, it wants to ditch the old one after the new connection
	has been authenticated.

	The same happens with a 3GPP UE: if it wants to open a new TCP
	connection, it has to first re-authenticate itself. The IPSEC
	protection is given to the new connection using the keys derived
	from the authentication procedure. The old connection is closed.

>The 'fixed port' model
>means that you can't re-open a connection within the time-wait time...

>... also all the text in section 18 about error conditions and re-opening
>connections seems to be rendered unusable in this scenario...

	They are not unusable, they just have the added penalty of
	re-running the authentication procedures.

	BTW, how is the SigComp for TLS? If people are not happy with
	3GPP handling security with IPSEC (I'm not saying that they are
	alone here), they should provide 3GPP with an alternative.

>I agree that there are significant interactions between SIP and SigComp.
>However, these are all defined in RFC 3320.  The point is that SigComp
>should (from a transport perspective) be able to look (mostly) like an
>alternative transport for SIP.  Multiplexing really breaks that model.

	Well, from SIP transport point of view SigComp looks like just
	alternative parser for SIP. Transport takes packets (in case of
	UDP) or streams of bytes (in case of TCP), stuffs them into the
	parser and gets SIP messages out of it. The combined SIP/SigComp
	parser can look inside SIP message and give compartment
	identifier back to SigComp.

>I think my point is that *not* allowing multiplexing means that you can
>separate the implementation details. (You don't have to if you don't
>want to, but you *can*). If you insist on allowing multiplexing, then a
>receiving endpoint *must* support it, and that's going to put a bunch
>of constraints and requirements on the implementation. I don't think
>that those constraints need to be there, and so I'd prefer not to have
>them...

	SigComp has lot of things that break assumptions about
	transport, especially TCP transport. TCP is a stream of bytes,
	SigComp provides messages, etc.

						Pekka

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> As an observation, I can't see any reason why you would use=20
> the last case!
> If it is *possible* that SMS will be available, and you know=20
> that it will be at least 2k, then why not make an 'optimistic'=20
> state-creation request on the first message?  If no SMS is=20
> advertised / no ack is received in a response, then you have=20
> to use per-message / stateless compression.  If, however,=20
> SMS is advertised / an ack is received, then everything should
> proceed exactly as if you had known that there was 2k SMS all along..!

This is true in this example. However, the example has two assumptions:

a) The state is only the bytecode --> you have to upload it in the=20
first message anyway.

b) Each message always triggers a message on the reverse direction
--> you'll know the truth when you send the second message, just in
time before you have to guess.

If *either* of the assumptions is untrue, we can no longer say 0KB SMS
costs nothing.

- For example, if assumption b) is untrue, we may have endpoint A
sends two messages (message 1 and 2) in a row before endpoint B returns
a response. Then, should A send bytecode state in message 2? For SigComp
over TCP, the bytecode in message 2 can be avoided if A knows the SMS
of B > 0.

- RE assumption a), A may want to upload dictionary state in the first=20
message, which can be used to compressed the second or even subsequent=20
messages it is going to send. This is particularly useful if the delay=20
of the first message is not important (which is the case for REGISTER).=20
However, without knowing whether B has SMS > 0, how can A decide whether
to upload the dictionary state and enjoy the reward latter? Unlike =
uploading
bytecode state in the first message (which you have to anyway), your
investment may or may not lead to any benefit depending on SMS of B.

BR,
Zhigang

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Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp messages on t he same TCP connection
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Thread-Topic: [rohc] Re: multiplexing SigComp and non-SigComp messages on t he same TCP connection
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Hi Mark,

> That is slicing a semantic a shade too thin for me!  But, if=20
> you want to play word games, then I'll point out that=20
> 'uncompressed' does not mean 'do not use SigComp'.

I don't think Jan is playing word game here. Few people really=20
enjoy it. SHOULD and MUST are well defined words in IETF. This
is no point to argue that further.=20

RE "uncompressed". Evidently and unfortunately, people used it with=20
different meanings in mind. And in fact, it remains arguable if a =
sigcomp
message encapsulating an uncompressed SIP message can be really called =
an=20
"uncompressed message". One may say it is still a compressed message,=20
but with NULL compression algorithm (like the NULL encryption algorithms
for IPsec). This starts looking like a word game, so, I'd better just=20
stop here. Anyway, we already agreed to non-sigcomp messages when we
do mean non-sigcomp message. This is not my point.

My point is that when RFC 3261 states: "... the client SHOULD close that
connection and open a new one to send the uncompressed request.",
it *should* mean to send non-SigComp messages on the new TCP connection. =

Logically, if the sender suspects a SigComp failure (see futher below)
has happened, it should avoid using SigComp.

> Carsten & I have whipped up a nice little 13-byte standard header:
>=20
>   0xf8, 0x00, 0xa1, 0x1c, 0x01, 0x86, 0x09, 0x22,
>   0x86, 0x01, 0x16, 0xf9, 0x23, ...
>=20
> that will simply output the following bytes!

Nice code indeed. For those who are wondering, the first 3 bytes
are SigComp header. And the remaining 10 bytes are the decompression
byte code: INPUT-BYTES(1, 64, +9), OUTPUT(64, 1), Jump(-7),=20
END-MESSAGE(with zeros operands initialized by UDVM).

This should work fine to resolve the SigComp failures due to state
desynchronization. But what if the UDVM somehow malfunctions so that=20
it cannot execute the bytecode correctly? Of course, this should=20
rarely happen, but is still a possibility as we're talking about
SigComp failures.

(Also, as a less important issue, it takes time for the UDVM execute
the byte-by-byte I/O instructions. Of course, a smart implementation
may analyze/compile the code instead of interpreting it. But can we
assume that?)

BR,
Zhigang


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From: Juergen Quittek <quittek@ccrle.nec.de>
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        "Wijnen, Bert (Bert)" <bwijnen@lucent.com>
Subject: Re: [rohc] MIB Doctor review of draft-ietf-rohc-mib-rtp-06.txt
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Mike,

While committing the changes I read again my replies on your comments
and there is one point where I am not that sure anymore that it is
wrong in the last version. Please see below.

--On 09.11.2003 19:17 h +0100 Juergen Quittek wrote:

 [...]
Mike> Another point that I noticed is that the OID assignments for the
Mike> columnar objects are not contiguous, doubtless as a result of
Mike> removing columns from the tables.  Here is the result of a quick
Mike> grep:
Mike>        ::= { rohcChannelEntry 2 }
Mike>        ::= { rohcChannelEntry 3 }
Mike>        ::= { rohcChannelEntry 4 }
Mike>        ::= { rohcChannelEntry 5 }
Mike>        ::= { rohcChannelEntry 13 }
Juergen>
Juergen> Yes, this needs to be fixed.
Juergen>
Mike>        ::= { rohcInstanceEntry 2 }
Mike>        ::= { rohcInstanceEntry 4 }
Juergen>
Juergen> And this also.

Wait ... isn't this correct?

Considering the indexing this looks at least not completely
wrong to me because object rohcChannelID is already defined
as element of the rohcChannelTable:

  |  |  +--rohcInstanceTable(2)
  |  |  |  |
  |  |  |  +--rohcInstanceEntry(1) [ifIndex,rohcInstanceType,rohcChannelID]
  |  |  |     |
  |  |  |     +-- --- Enumeration           rohcInstanceType(2)
  |  |  |     +-- r-n RohcChannelIdentifier rohcInstanceFBChannelID(4)
  |  |  |     +-- r-n ObjectIdentifier      rohcInstanceVendor(5)
  |  |  |     +-- r-n SnmpAdminString       rohcInstanceVersion(6)
  |  |  |     +--  . . .

Mike, can we leave this as it is?
If not, what is the correct number assignment?

Thanks,

    Juergen




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On Sat, 15 Nov 2003, Juergen Quittek wrote:
> While committing the changes I read again my replies on your comments
> and there is one point where I am not that sure anymore that it is
> wrong in the last version. Please see below.
> 
> --On 09.11.2003 19:17 h +0100 Juergen Quittek wrote:
> 
>  [...]
> Mike> Another point that I noticed is that the OID assignments for the
> Mike> columnar objects are not contiguous, doubtless as a result of
> Mike> removing columns from the tables.  Here is the result of a quick
> Mike> grep:
....
> Mike>        ::= { rohcInstanceEntry 2 }
> Mike>        ::= { rohcInstanceEntry 4 }
...
> Wait ... isn't this correct?
> 
> Considering the indexing this looks at least not completely
> wrong to me because object rohcChannelID is already defined
> as element of the rohcChannelTable:
> 
>   |  |  +--rohcInstanceTable(2)
>   |  |  |  |
>   |  |  |  +--rohcInstanceEntry(1) [ifIndex,rohcInstanceType,rohcChannelID]
>   |  |  |     |
>   |  |  |     +-- --- Enumeration           rohcInstanceType(2)
>   |  |  |     +-- r-n RohcChannelIdentifier rohcInstanceFBChannelID(4)
>   |  |  |     +-- r-n ObjectIdentifier      rohcInstanceVendor(5)
>   |  |  |     +-- r-n SnmpAdminString       rohcInstanceVersion(6)
>   |  |  |     +--  . . .
> 
> Mike, can we leave this as it is?
> If not, what is the correct number assignment?

Normally we don't leave gaps in the OID assignments for columnar
objects even if the row object has index components in other tables.
However, that is really just a question of style;  it is not an
error to have gaps in OID assignments.  So if you feel strongly
about this case I won't object if you choose to leave it as is.
Bert may feel differently and I will let him speak for himself.

Regards,

Mike


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From: "Jan Christoffersson (LU/EAB)" <jan.christoffersson@ericsson.com>
To: "'West, Mark'" <mark.a.west@roke.co.uk>, Carsten Bormann
	 <cabo@tzi.org>
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Subject: RE: [rohc] sigcomp-sip status
Date: Mon, 17 Nov 2003 14:00:33 +0100
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Hi,

I think allowing 0 byte SMS is reasonable. The extra cost of allowing this (if the true size is >0)if the optimistic approach is used is mostly zero.

Even if we expect and hope that all client and servers have a non zero SMS, I think we should still allow a zero byte SMS. It could perhaps be useful for temporarily overloaded proxies.

Best Regards
Jan

-----Original Message-----
From: rohc-admin@ietf.org [mailto:rohc-admin@ietf.org]On Behalf Of West,
Mark
Sent: den 14 november 2003 16:45
To: Carsten Bormann
Cc: rohc@ietf.org
Subject: Re: [rohc] sigcomp-sip status



Hi all,

A bit of a follow-up, as promised/threatened...

... and I reserve the right to include errors of my own..! ;-)

>
> 2 K SMS means
> -- can't compress to stateless proxies (i.e., "really stateless"
> proxies can't advertise sigcomp)
> 0 K SMS means
> -- can't establish state in initial message and rely on it
> -- can still assume it's there -- need to wait for ACK to find out
> for real
> Mark will find a number for what 0 K SMS "costs".
>

I'm assuming the following simple sequence of 3 messages in a one-way
flow: REGISTER, INVITE, and ACK.  These have been run through a script
that I use for assessing compression efficiency.

For these results, the compression algorithm is EPIC (from the SigComp
users' guide), 'cause that's what I happen to have to hand...

The uncompressed message sizes are 322, 489 and 288 bytes, respectively.

The EPIC bytecode that I'm currently playing with is ~400 bytes.

So:

Per-message compression with static dictionary gives:
126, 201 & 131 bytes = 458 bytes in total (not including SigComp message
overhead or bytecode)
Bytecode adds 400 bytes to each message.  This, it can be seen, is not
terribly good value!  If bytecode were known to be available (advertised
by remote endpoint), some of this overhead could be avoided.

Dynamic compression (also with static dictionary) gives:
126, 157 & 23 bytes = 306 bytes total (not including overheads)
Bytecode adds 400 bytes to first message.

If you deferred the creation of state until receiving the advertisement of
2k SMS, then it would look (as expected), like:
126, 201 & 23 bytes = 350 bytes total (not including overheads)
Bytecode adds 400 bytes to the first 2 messages (excepting advertisements
or well-known bytecode)

As an observation, I can't see any reason why you would use the last case!
If it is *possible* that SMS will be available, and you know that it will
be at least 2k, then why not make an 'optimistic' state-creation request
on the first message?  If no SMS is advertised / no ack is received in a
response, then you have to use per-message / stateless compression.  If,
however, SMS is advertised / an ack is received, then everything should
proceed exactly as if you had known that there was 2k SMS all along..!

For the per-message / boot-strap case, we've played around a bit with
compact byte-code for simple compression such that the compressed message
+ bytecode is still smaller than the original message.

The 'best-case' though, would clearly be to have a 'well-known' algorithm
that could be used for the per-message / stateless scenario.

Hopefully that more or less answers the question..?

Cheers,

Mark.

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From: "Jan Christoffersson (LU/EAB)" <jan.christoffersson@ericsson.com>
To: "'West, Mark'" <mark.a.west@roke.co.uk>
Cc: zhigang.c.liu@nokia.com, Robert.Sugar@nokia.com, rohc@ietf.org
Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp messages on t
	 he same TCP connection
Date: Mon, 17 Nov 2003 14:46:39 +0100
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Hi Mark,

Comments inline.

-----Original Message-----
From: West, Mark [mailto:mark.a.west@roke.co.uk]
Sent: den 14 november 2003 15:09
To: Jan Christoffersson (LU/EAB)
Cc: zhigang.c.liu@nokia.com; Robert.Sugar@nokia.com; rohc@ietf.org
Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp messages on
t he same TCP connection



Hi Jan,

Comments added!

Cheers,

Mark.

>
> 1. The extra delay caused by opening and closing TCP connections when
> switching between SIP and SigComp should not be neglected. In some cases
> it may be of little concern but in others it may be important. Reducing
> delay has been the most important driving factor for SigComp.
> Multiplexing over the same connection reduces the delay.
>

But most of the cases where you want to switch between SigComp /
non-SigComp are inherently cases where there is inherent latency.

[JC]I agree that these cases will probably have a lot of latency, so this argument isn't very heavy. Still, less latency is better.

And RFC 3261 (if I read it correctly, but I'm no SIP guru), talks about
only try to keep a transaction within in a single connection...

> 2.  We have the tools needed for multiplexing SigComp and SIP on the
> same TCP connection. Why not do it? It's faster than opening a new
> connection.
>

But the 'tools' are horrible!  Reiterating the sentiments from S-F: 'yeah,
you could, but why would you want to..?'

[JC]I don't know why this is so horrible? Could you please elaborate...

> 3. Mark argues that if multiplexing is allowed, the receiver needs to
> support it. But doesn't it already need to do this? As Zhigang pointed
> out, RFC3486 states that
>
> "If a SIP client sends a compressed request and the client transaction
>    times out without having received any response, the client SHOULD
>    retry the same request without using compression.  If the compressed
>    request was sent over a TCP connection, the client SHOULD close that
>    connection and open a new one to send the uncompressed request."
>
> In this text, it says SHOULD and not MUST, i.e. there could be
> implementations that send SIP and SigComp on the same TCP connection, so
> all implementations should be able to handle it. In any case, we have
> delimiters to determine where SigComp messages start and end, and if
> there is something other than SigComp in between an end and a start, we
> should be able to handle it.
>

That is slicing a semantic a shade too thin for me!  But, if you want to
play word games, then I'll point out that 'uncompressed' does not mean 'do
not use SigComp'.
[JC] I didn't intend this to be a word game. To me this is a recommendation to open a new TCP connection. I'm perfectly happy with that. But I don't think we should turn the recommendation into a must and forbid sending SigComp and non-SigComp on the same TCP connection.

Carsten & I have whipped up a nice little 13-byte standard header:

  0xf8, 0x00, 0xa1, 0x1c, 0x01, 0x86, 0x09, 0x22,
  0x86, 0x01, 0x16, 0xf9, 0x23, ...

that will simply output the following bytes!

It's a SigComp message, but there's no compression...

[JC]Yes, this is perfectly fine, unless the message is larger than 64 K. 

And let's not forget that the purpose of draft-ietf-rohc-sip-sigcomp is to
fill in the missing pieces of the mapping jigsaw (i.e. the bits that
aren't in RFC 3486).  And that's what we're arguing about.  (Although I
was fairly sure that we'd already decided this.  *sigh*)

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The 2004 WiFi Voice Conference will stand in Paris on May 25 to 28, =
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The call for papers dead line has been extended until December 15, 2003. =

Get more details at:
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Hi Zhigang,

Regarding a), I agree with you.  This is a counter-example that I had not
considered.  So, yes, if you don't know that the endpoint has SMS available,
you may decide not to upload user-specific dictionary state, and there may
well be a cost associated with that.

I'm not in a position to quantify that at the moment, but I accept the
point.

So far as b) is concerned, I don't think that this is (necessarily) just an
SMS issue (although it is a valid point).  If I send 2 messages to the
remote endpoint without receiving feedback, how do I know that the message
has successfully arrived; been correctly decompressed; and that the SIP
application has authorised the state creation?  If *any* of those are false,
then I can't make use of the state, even if remote SMS is available.  (Use
of TCP makes the first likely, but doesn't impact the others...)

Cheers,

Mark.


> 
> This is true in this example. However, the example has two 
> assumptions:
> 
> a) The state is only the bytecode --> you have to upload it in the 
> first message anyway.
> 
> b) Each message always triggers a message on the reverse direction
> --> you'll know the truth when you send the second message, just in
> time before you have to guess.
> 
> If *either* of the assumptions is untrue, we can no longer say 0KB SMS
> costs nothing.
> 
> - For example, if assumption b) is untrue, we may have endpoint A
> sends two messages (message 1 and 2) in a row before endpoint 
> B returns
> a response. Then, should A send bytecode state in message 2? 
> For SigComp
> over TCP, the bytecode in message 2 can be avoided if A knows the SMS
> of B > 0.
> 
> - RE assumption a), A may want to upload dictionary state in 
> the first 
> message, which can be used to compressed the second or even 
> subsequent 
> messages it is going to send. This is particularly useful if 
> the delay 
> of the first message is not important (which is the case for 
> REGISTER). 
> However, without knowing whether B has SMS > 0, how can A 
> decide whether
> to upload the dictionary state and enjoy the reward latter? 
> Unlike uploading
> bytecode state in the first message (which you have to anyway), your
> investment may or may not lead to any benefit depending on SMS of B.
> 
> BR,
> Zhigang
> 

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From: "West, Mark" <mark.a.west@roke.co.uk>
To: "'Jan Christoffersson (LU/EAB)'" <jan.christoffersson@ericsson.com>
Cc: zhigang.c.liu@nokia.com, Robert.Sugar@nokia.com, rohc@ietf.org
Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp messages on t
	 he same TCP connection
Date: Wed, 19 Nov 2003 15:31:10 -0000
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Hi Jan,

Yes, less latency is good (that has always been a goal of SigComp).  And I agree that messages > 64kbyte are a problem with SigComp!  Still, you can't win 'em all...

Multiplexing is horrible (or, at the very least messy) because you inherently bind the SigComp and SIP transport handling for detecting the end of a message.  If you don't allow multiplexing, you are free to implement it this way (if you want), but if you *do* allow multiplexing then this becomes a requirement...

But, like I say, I thought that the 'non-multiplexing' issue had been agreed.  And it's still not clear to me why it's important to be able to do it!

Cheers,

Mark.




> -----Original Message-----
> From: Jan Christoffersson (LU/EAB)
> [mailto:jan.christoffersson@ericsson.com]
> Sent: 17 November 2003 13:47
> To: West, Mark
> Cc: zhigang.c.liu@nokia.com; Robert.Sugar@nokia.com; rohc@ietf.org
> Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp 
> messages on
> t he same TCP connection
> 
> 
> Hi Mark,
> 
> Comments inline.
> 
> -----Original Message-----
> From: West, Mark [mailto:mark.a.west@roke.co.uk]
> Sent: den 14 november 2003 15:09
> To: Jan Christoffersson (LU/EAB)
> Cc: zhigang.c.liu@nokia.com; Robert.Sugar@nokia.com; rohc@ietf.org
> Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp 
> messages on
> t he same TCP connection
> 
> 
> 
> Hi Jan,
> 
> Comments added!
> 
> Cheers,
> 
> Mark.
> 
> >
> > 1. The extra delay caused by opening and closing TCP 
> connections when
> > switching between SIP and SigComp should not be neglected. 
> In some cases
> > it may be of little concern but in others it may be 
> important. Reducing
> > delay has been the most important driving factor for SigComp.
> > Multiplexing over the same connection reduces the delay.
> >
> 
> But most of the cases where you want to switch between SigComp /
> non-SigComp are inherently cases where there is inherent latency.
> 
> [JC]I agree that these cases will probably have a lot of 
> latency, so this argument isn't very heavy. Still, less 
> latency is better.
> 
> And RFC 3261 (if I read it correctly, but I'm no SIP guru), 
> talks about
> only try to keep a transaction within in a single connection...
> 
> > 2.  We have the tools needed for multiplexing SigComp and SIP on the
> > same TCP connection. Why not do it? It's faster than opening a new
> > connection.
> >
> 
> But the 'tools' are horrible!  Reiterating the sentiments 
> from S-F: 'yeah,
> you could, but why would you want to..?'
> 
> [JC]I don't know why this is so horrible? Could you please 
> elaborate...
> 
> > 3. Mark argues that if multiplexing is allowed, the 
> receiver needs to
> > support it. But doesn't it already need to do this? As 
> Zhigang pointed
> > out, RFC3486 states that
> >
> > "If a SIP client sends a compressed request and the client 
> transaction
> >    times out without having received any response, the client SHOULD
> >    retry the same request without using compression.  If 
> the compressed
> >    request was sent over a TCP connection, the client 
> SHOULD close that
> >    connection and open a new one to send the uncompressed request."
> >
> > In this text, it says SHOULD and not MUST, i.e. there could be
> > implementations that send SIP and SigComp on the same TCP 
> connection, so
> > all implementations should be able to handle it. In any 
> case, we have
> > delimiters to determine where SigComp messages start and end, and if
> > there is something other than SigComp in between an end and 
> a start, we
> > should be able to handle it.
> >
> 
> That is slicing a semantic a shade too thin for me!  But, if 
> you want to
> play word games, then I'll point out that 'uncompressed' does 
> not mean 'do
> not use SigComp'.
> [JC] I didn't intend this to be a word game. To me this is a 
> recommendation to open a new TCP connection. I'm perfectly 
> happy with that. But I don't think we should turn the 
> recommendation into a must and forbid sending SigComp and 
> non-SigComp on the same TCP connection.
> 
> Carsten & I have whipped up a nice little 13-byte standard header:
> 
>   0xf8, 0x00, 0xa1, 0x1c, 0x01, 0x86, 0x09, 0x22,
>   0x86, 0x01, 0x16, 0xf9, 0x23, ...
> 
> that will simply output the following bytes!
> 
> It's a SigComp message, but there's no compression...
> 
> [JC]Yes, this is perfectly fine, unless the message is larger 
> than 64 K. 
> 
> And let's not forget that the purpose of 
> draft-ietf-rohc-sip-sigcomp is to
> fill in the missing pieces of the mapping jigsaw (i.e. the bits that
> aren't in RFC 3486).  And that's what we're arguing about.  
> (Although I
> was fairly sure that we'd already decided this.  *sigh*)
> 

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



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From: "Lars-Erik Jonsson (LU/EAB)" <lars-erik.jonsson@ericsson.com>
To: "'West, Mark'" <mark.a.west@roke.co.uk>
Cc: rohc@ietf.org
Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp messages on t
	 he same TCP connection
Date: Wed, 19 Nov 2003 16:53:46 +0100
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I agree with Mark that this is a new and rather controversial
new requirement to introduce at this point, especially since
we have already closed the issue several times.

/L-E


> Multiplexing is horrible (or, at the very least messy) 
> because you inherently bind the SigComp and SIP transport 
> handling for detecting the end of a message.  If you don't 
> allow multiplexing, you are free to implement it this way (if 
> you want), but if you *do* allow multiplexing then this 
> becomes a requirement...
> 
> But, like I say, I thought that the 'non-multiplexing' issue 
> had been agreed.  And it's still not clear to me why it's 
> important to be able to do it!
> 
> Cheers,
> 
> Mark.
> 
> 
> 
> 
> > -----Original Message-----
> > From: Jan Christoffersson (LU/EAB)
> > [mailto:jan.christoffersson@ericsson.com]
> > Sent: 17 November 2003 13:47
> > To: West, Mark
> > Cc: zhigang.c.liu@nokia.com; Robert.Sugar@nokia.com; rohc@ietf.org
> > Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp 
> > messages on
> > t he same TCP connection
> > 
> > 
> > Hi Mark,
> > 
> > Comments inline.
> > 
> > -----Original Message-----
> > From: West, Mark [mailto:mark.a.west@roke.co.uk]
> > Sent: den 14 november 2003 15:09
> > To: Jan Christoffersson (LU/EAB)
> > Cc: zhigang.c.liu@nokia.com; Robert.Sugar@nokia.com; rohc@ietf.org
> > Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp 
> > messages on
> > t he same TCP connection
> > 
> > 
> > 
> > Hi Jan,
> > 
> > Comments added!
> > 
> > Cheers,
> > 
> > Mark.
> > 
> > >
> > > 1. The extra delay caused by opening and closing TCP 
> > connections when
> > > switching between SIP and SigComp should not be neglected. 
> > In some cases
> > > it may be of little concern but in others it may be 
> > important. Reducing
> > > delay has been the most important driving factor for SigComp.
> > > Multiplexing over the same connection reduces the delay.
> > >
> > 
> > But most of the cases where you want to switch between SigComp /
> > non-SigComp are inherently cases where there is inherent latency.
> > 
> > [JC]I agree that these cases will probably have a lot of 
> > latency, so this argument isn't very heavy. Still, less 
> > latency is better.
> > 
> > And RFC 3261 (if I read it correctly, but I'm no SIP guru), 
> > talks about
> > only try to keep a transaction within in a single connection...
> > 
> > > 2.  We have the tools needed for multiplexing SigComp and 
> SIP on the
> > > same TCP connection. Why not do it? It's faster than opening a new
> > > connection.
> > >
> > 
> > But the 'tools' are horrible!  Reiterating the sentiments 
> > from S-F: 'yeah,
> > you could, but why would you want to..?'
> > 
> > [JC]I don't know why this is so horrible? Could you please 
> > elaborate...
> > 
> > > 3. Mark argues that if multiplexing is allowed, the 
> > receiver needs to
> > > support it. But doesn't it already need to do this? As 
> > Zhigang pointed
> > > out, RFC3486 states that
> > >
> > > "If a SIP client sends a compressed request and the client 
> > transaction
> > >    times out without having received any response, the 
> client SHOULD
> > >    retry the same request without using compression.  If 
> > the compressed
> > >    request was sent over a TCP connection, the client 
> > SHOULD close that
> > >    connection and open a new one to send the uncompressed 
> request."
> > >
> > > In this text, it says SHOULD and not MUST, i.e. there could be
> > > implementations that send SIP and SigComp on the same TCP 
> > connection, so
> > > all implementations should be able to handle it. In any 
> > case, we have
> > > delimiters to determine where SigComp messages start and 
> end, and if
> > > there is something other than SigComp in between an end and 
> > a start, we
> > > should be able to handle it.
> > >
> > 
> > That is slicing a semantic a shade too thin for me!  But, if 
> > you want to
> > play word games, then I'll point out that 'uncompressed' does 
> > not mean 'do
> > not use SigComp'.
> > [JC] I didn't intend this to be a word game. To me this is a 
> > recommendation to open a new TCP connection. I'm perfectly 
> > happy with that. But I don't think we should turn the 
> > recommendation into a must and forbid sending SigComp and 
> > non-SigComp on the same TCP connection.
> > 
> > Carsten & I have whipped up a nice little 13-byte standard header:
> > 
> >   0xf8, 0x00, 0xa1, 0x1c, 0x01, 0x86, 0x09, 0x22,
> >   0x86, 0x01, 0x16, 0xf9, 0x23, ...
> > 
> > that will simply output the following bytes!
> > 
> > It's a SigComp message, but there's no compression...
> > 
> > [JC]Yes, this is perfectly fine, unless the message is larger 
> > than 64 K. 
> > 
> > And let's not forget that the purpose of 
> > draft-ietf-rohc-sip-sigcomp is to
> > fill in the missing pieces of the mapping jigsaw (i.e. the bits that
> > aren't in RFC 3486).  And that's what we're arguing about.  
> > (Although I
> > was fairly sure that we'd already decided this.  *sigh*)
> > 
> 
> _______________________________________________
> Rohc mailing list
> Rohc@ietf.org
> https://www1.ietf.org/mailman/listinfo/rohc
> 

_______________________________________________
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Rohc@ietf.org
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Hi Mark,

> So far as b) is concerned, I don't think that this is=20
> (necessarily) just an SMS issue (although it is a valid point).=20
> If I send 2 messages to the remote endpoint without receiving=20
> feedback, how do I know that the message has successfully=20
> arrived; been correctly decompressed; and that the SIP
> application has authorised the state creation?  If *any* of=20
> those are false, then I can't make use of the state, even if=20
> remote SMS is available.  (Use of TCP makes the first likely,=20
> but doesn't impact the others...)

Very valid question. I've actually gone through it too, but
didn't spell it out to make my earlier email short. Now that=20
you correctly pointed them out, my answer is (a bit=20
surprisingly, or may be not): optimistic.

Here is a simple example to show the idea.

A              B
   m1 ------>=20
   m2 ------>
   <------- m3

When A sends the second message m2, it can optimistically assume
m1 has survived the 3 conditions above, and use states uploaded=20
in m1 to compress m2. In most cases (if the compressor is=20
implemented correctly), the assumption should turn out true and=20
thus A enjoys the benefit. In the rare cases where it doesn't,=20
that means this is *already* a SigComp failure for *m1* which=20
requires a recovery procedure for m1. (Here we link to the multiplexing
issue but I wouldn't deviate to that topic at this moment.)=20
So, being optimistic does not add additional penalty.

Note that I've made one assumption in the above argument. That is,
sending m3 is conditioned on the receiving of m1 and m2, in order.=20
This should be a reasonable assumption considering the nature of SIP
signaling. Otherwise, that means=20

1) m3 depends only on one of {m1, m2}. In that case, why would A=20
bother send both messages at all in the first place? We want to
reduce latency.

2) m3 depends on both {m1, m2}, but does not require them to
be received in order. I cannot recall a real-life SIP example of
this. But if it exists, there is indeed a penalty making m2
dependent on m1 *and* when a SigComp failure occurs on m1. Namely,=20
A has to resend m2 (in addition to m1), which is not needed=20
if A sent m1 and m2 independently in the first round. However,
I don't see this is a big concern for two reasons: a) SigComp=20
failure should not be a frequent event, or we are in real trouble;=20
b) the delay caused by resending m2 is second order to the delay=20
caused by tear-down and reestablishment of same TCP connection, or
establishment of a new TCP connection which may require setting
up a new security association.

BR,
Zhigang

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Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp messages on t he same TCP connection
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Thread-Topic: [rohc] Re: multiplexing SigComp and non-SigComp messages on t he same TCP connection
Thread-Index: AcOusjUV8fKCgAL9ShuPtzhdsyJ8gAADe4yQ
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Hi,

> And I agree that messages > 64kbyte are a problem=20
> with SigComp! =20

Is this a valid reason to have multiplexing over TCP?=20

Consider the scenario where the communication between endpoint A
and B involves both delay sensitive short messages and=20
non-delay-sensitive long (> 64KB) messages. Without multiplexing,
two TCP connections are required which leads to issues such as=20
synchronization of messages transmitted on the two TCP connections.
=20
> Multiplexing is horrible (or, at the very least messy)=20
> because you inherently bind the SigComp and SIP transport=20
> handling for detecting the end of a message. =20

For a given TCP stream, multiplexing means the receiver needs
to alternate between two parsing "modes": SigComp mode, in which
SigComp controls the parsing of incoming stream, and transparent
mode, in which the application takes control.=20

This passing of "controls" between SigComp and application does=20
add complexity. But is it really "horrible" or "messy", considering
the binding/interaction between SigComp and application we already
have, such as message authentication, compartment management,=20
feedback, etc?=20

> If you don't allow multiplexing, you are free to implement it=20
> this way (if you want),=20

Not really. The sender has to know if the receiver supports=20
multiplexing. So, it's not just a matter of implementation.

BR,
Zhigang

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Thread-Topic: [rohc] SigComp implementer's guide status
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Subject: [rohc] IP Address Compression for Context Replication
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Hello,
The current Context Replication mechanism suggests to send either whole =
IP address  or no IP address (replication). We have submitted a draft on =
the idea of sending compressed IP address using base IP address. This =
improves header compression efficiency when packet flows carry similar =
(e.g. same prefix) but not exactly same IP addresses. In particular, the =
gain can be significant for IPv6 header compression due to the large =
size (16 bytes) of IPv6 addresses.
   =20
http://www.ietf.org/internet-drafts/draft-saifullah-rohc-ip-addr-comp-con=
text-replication-00.txt

This draft has impacts on section 6.5.5.1 of draft-ietf-rohc-tcp-05.txt, =
where the format of src/dest IP addresses are defined for IR-CR.

Your comments and questions on this draft are more than welcome.

Regards,
Yousuf


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Thread-Topic: [rohc] Re: multiplexing SigComp and non-SigComp messages on t he same TCP connection
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Hi,

If I remember correctly, what has been closed/clarified was that=20
RFC 3320 does not allow multiplexing.=20

However, the issue here is whether or not the multiplexing is needed.=20
Obviously, people already have their own opinions in mind. But until
recently, there haven't been many technical discussions regarding=20
this issue on ROHC mailing list. So, I'm not sure what do you mean
by "we have already closed the issue several times"? Or did I miss
anything?

Either way, what I'd like to see is a serious debate on this issue
from different aspects, so that in the end, we can say with confidence
that we've taken a hard look on the issue and here is our conclusion.

BR,
Zhigang

> -----Original Message-----
> From: rohc-admin@ietf.org [mailto:rohc-admin@ietf.org]On Behalf Of ext
> Lars-Erik Jonsson (LU/EAB)
> Sent: November 19, 2003 09:54 AM
> To: 'West, Mark'
> Cc: rohc@ietf.org
> Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp=20
> messages on
> t he same TCP connection
>=20
>=20
> I agree with Mark that this is a new and rather controversial
> new requirement to introduce at this point, especially since
> we have already closed the issue several times.
>=20
> /L-E
>=20
>=20
> > Multiplexing is horrible (or, at the very least messy)=20
> > because you inherently bind the SigComp and SIP transport=20
> > handling for detecting the end of a message.  If you don't=20
> > allow multiplexing, you are free to implement it this way (if=20
> > you want), but if you *do* allow multiplexing then this=20
> > becomes a requirement...
> >=20
> > But, like I say, I thought that the 'non-multiplexing' issue=20
> > had been agreed.  And it's still not clear to me why it's=20
> > important to be able to do it!
> >=20
> > Cheers,
> >=20
> > Mark.
> >=20
> >=20
> >=20
> >=20
> > > -----Original Message-----
> > > From: Jan Christoffersson (LU/EAB)
> > > [mailto:jan.christoffersson@ericsson.com]
> > > Sent: 17 November 2003 13:47
> > > To: West, Mark
> > > Cc: zhigang.c.liu@nokia.com; Robert.Sugar@nokia.com; rohc@ietf.org
> > > Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp=20
> > > messages on
> > > t he same TCP connection
> > >=20
> > >=20
> > > Hi Mark,
> > >=20
> > > Comments inline.
> > >=20
> > > -----Original Message-----
> > > From: West, Mark [mailto:mark.a.west@roke.co.uk]
> > > Sent: den 14 november 2003 15:09
> > > To: Jan Christoffersson (LU/EAB)
> > > Cc: zhigang.c.liu@nokia.com; Robert.Sugar@nokia.com; rohc@ietf.org
> > > Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp=20
> > > messages on
> > > t he same TCP connection
> > >=20
> > >=20
> > >=20
> > > Hi Jan,
> > >=20
> > > Comments added!
> > >=20
> > > Cheers,
> > >=20
> > > Mark.
> > >=20
> > > >
> > > > 1. The extra delay caused by opening and closing TCP=20
> > > connections when
> > > > switching between SIP and SigComp should not be neglected.=20
> > > In some cases
> > > > it may be of little concern but in others it may be=20
> > > important. Reducing
> > > > delay has been the most important driving factor for SigComp.
> > > > Multiplexing over the same connection reduces the delay.
> > > >
> > >=20
> > > But most of the cases where you want to switch between SigComp /
> > > non-SigComp are inherently cases where there is inherent latency.
> > >=20
> > > [JC]I agree that these cases will probably have a lot of=20
> > > latency, so this argument isn't very heavy. Still, less=20
> > > latency is better.
> > >=20
> > > And RFC 3261 (if I read it correctly, but I'm no SIP guru),=20
> > > talks about
> > > only try to keep a transaction within in a single connection...
> > >=20
> > > > 2.  We have the tools needed for multiplexing SigComp and=20
> > SIP on the
> > > > same TCP connection. Why not do it? It's faster than=20
> opening a new
> > > > connection.
> > > >
> > >=20
> > > But the 'tools' are horrible!  Reiterating the sentiments=20
> > > from S-F: 'yeah,
> > > you could, but why would you want to..?'
> > >=20
> > > [JC]I don't know why this is so horrible? Could you please=20
> > > elaborate...
> > >=20
> > > > 3. Mark argues that if multiplexing is allowed, the=20
> > > receiver needs to
> > > > support it. But doesn't it already need to do this? As=20
> > > Zhigang pointed
> > > > out, RFC3486 states that
> > > >
> > > > "If a SIP client sends a compressed request and the client=20
> > > transaction
> > > >    times out without having received any response, the=20
> > client SHOULD
> > > >    retry the same request without using compression.  If=20
> > > the compressed
> > > >    request was sent over a TCP connection, the client=20
> > > SHOULD close that
> > > >    connection and open a new one to send the uncompressed=20
> > request."
> > > >
> > > > In this text, it says SHOULD and not MUST, i.e. there could be
> > > > implementations that send SIP and SigComp on the same TCP=20
> > > connection, so
> > > > all implementations should be able to handle it. In any=20
> > > case, we have
> > > > delimiters to determine where SigComp messages start and=20
> > end, and if
> > > > there is something other than SigComp in between an end and=20
> > > a start, we
> > > > should be able to handle it.
> > > >
> > >=20
> > > That is slicing a semantic a shade too thin for me!  But, if=20
> > > you want to
> > > play word games, then I'll point out that 'uncompressed' does=20
> > > not mean 'do
> > > not use SigComp'.
> > > [JC] I didn't intend this to be a word game. To me this is a=20
> > > recommendation to open a new TCP connection. I'm perfectly=20
> > > happy with that. But I don't think we should turn the=20
> > > recommendation into a must and forbid sending SigComp and=20
> > > non-SigComp on the same TCP connection.
> > >=20
> > > Carsten & I have whipped up a nice little 13-byte standard header:
> > >=20
> > >   0xf8, 0x00, 0xa1, 0x1c, 0x01, 0x86, 0x09, 0x22,
> > >   0x86, 0x01, 0x16, 0xf9, 0x23, ...
> > >=20
> > > that will simply output the following bytes!
> > >=20
> > > It's a SigComp message, but there's no compression...
> > >=20
> > > [JC]Yes, this is perfectly fine, unless the message is larger=20
> > > than 64 K.=20
> > >=20
> > > And let's not forget that the purpose of=20
> > > draft-ietf-rohc-sip-sigcomp is to
> > > fill in the missing pieces of the mapping jigsaw (i.e.=20
> the bits that
> > > aren't in RFC 3486).  And that's what we're arguing about. =20
> > > (Although I
> > > was fairly sure that we'd already decided this.  *sigh*)
> > >=20
> >=20
> > _______________________________________________
> > Rohc mailing list
> > Rohc@ietf.org
> > https://www1.ietf.org/mailman/listinfo/rohc
> >=20
>=20
> _______________________________________________
> Rohc mailing list
> Rohc@ietf.org
> https://www1.ietf.org/mailman/listinfo/rohc
>=20

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



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Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp messages on t he same TCP connection
Date: Thu, 20 Nov 2003 15:57:30 +0100
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Thread-Topic: [rohc] Re: multiplexing SigComp and non-SigComp messages on t he same TCP connection
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        <mark.a.west@roke.co.uk>
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Hi all,

 Let me give some more facets to the discussion:

 Many times the latency is not just the TCP handshake, but has other =
contributors as well. In case of IPSec, there is no abundance of TCP =
connection possibilities, those are limited to at most the number of SAs =
available. And in order to set up a new SA, the use of an additional =
protocol (e.g. IKE, or RFC3310) is needed. So it is not only the new TCP =
setup delay, but also the SA setup delay. In case of TLS the handshake =
costs quite a few messages too.

 Besides the >64K messages, and general SigComp errors, RFC3486 is the =
other reason to have SigComp and non-SigComp messages multiplexed (at =
least in time). Endpoints have the possibility to indicate willingness =
for receiving SigComp messages on a per transaction basis (for example a =
CPU-challenged endpoint wants to compress delay-sensitive session =
setups, but not instant messages). We should not burden that choice with =
significant additional delay, because that would take away useful =
functionalites from RFC3486.

 In many protocols (HTTP being one of them), TCP connection close has =
also application level meanings, thus opening a new connection may cause =
application level delays as well as a different user experience or =
protocol behavior.

BR,
Robert
=20

> -----Original Message-----
> From: rohc-admin@ietf.org [mailto:rohc-admin@ietf.org]On Behalf Of ext
> zhigang.c.liu@nokia.com
> Sent: November 19,2003 22:01
> To: lars-erik.jonsson@ericsson.com; mark.a.west@roke.co.uk
> Cc: rohc@ietf.org
> Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp=20
> messages on
> t he same TCP connection
>=20
>=20
> Hi,
>=20
> If I remember correctly, what has been closed/clarified was that=20
> RFC 3320 does not allow multiplexing.=20
>=20
> However, the issue here is whether or not the multiplexing is needed.=20
> Obviously, people already have their own opinions in mind. But until
> recently, there haven't been many technical discussions regarding=20
> this issue on ROHC mailing list. So, I'm not sure what do you mean
> by "we have already closed the issue several times"? Or did I miss
> anything?
>=20
> Either way, what I'd like to see is a serious debate on this issue
> from different aspects, so that in the end, we can say with confidence
> that we've taken a hard look on the issue and here is our conclusion.
>=20
> BR,
> Zhigang
>=20
> > -----Original Message-----
> > From: rohc-admin@ietf.org [mailto:rohc-admin@ietf.org]On=20
> Behalf Of ext
> > Lars-Erik Jonsson (LU/EAB)
> > Sent: November 19, 2003 09:54 AM
> > To: 'West, Mark'
> > Cc: rohc@ietf.org
> > Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp=20
> > messages on
> > t he same TCP connection
> >=20
> >=20
> > I agree with Mark that this is a new and rather controversial
> > new requirement to introduce at this point, especially since
> > we have already closed the issue several times.
> >=20
> > /L-E
> >=20
> >=20
> > > Multiplexing is horrible (or, at the very least messy)=20
> > > because you inherently bind the SigComp and SIP transport=20
> > > handling for detecting the end of a message.  If you don't=20
> > > allow multiplexing, you are free to implement it this way (if=20
> > > you want), but if you *do* allow multiplexing then this=20
> > > becomes a requirement...
> > >=20
> > > But, like I say, I thought that the 'non-multiplexing' issue=20
> > > had been agreed.  And it's still not clear to me why it's=20
> > > important to be able to do it!
> > >=20
> > > Cheers,
> > >=20
> > > Mark.
> > >=20
> > >=20
> > >=20
> > >=20
> > > > -----Original Message-----
> > > > From: Jan Christoffersson (LU/EAB)
> > > > [mailto:jan.christoffersson@ericsson.com]
> > > > Sent: 17 November 2003 13:47
> > > > To: West, Mark
> > > > Cc: zhigang.c.liu@nokia.com; Robert.Sugar@nokia.com;=20
> rohc@ietf.org
> > > > Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp=20
> > > > messages on
> > > > t he same TCP connection
> > > >=20
> > > >=20
> > > > Hi Mark,
> > > >=20
> > > > Comments inline.
> > > >=20
> > > > -----Original Message-----
> > > > From: West, Mark [mailto:mark.a.west@roke.co.uk]
> > > > Sent: den 14 november 2003 15:09
> > > > To: Jan Christoffersson (LU/EAB)
> > > > Cc: zhigang.c.liu@nokia.com; Robert.Sugar@nokia.com;=20
> rohc@ietf.org
> > > > Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp=20
> > > > messages on
> > > > t he same TCP connection
> > > >=20
> > > >=20
> > > >=20
> > > > Hi Jan,
> > > >=20
> > > > Comments added!
> > > >=20
> > > > Cheers,
> > > >=20
> > > > Mark.
> > > >=20
> > > > >
> > > > > 1. The extra delay caused by opening and closing TCP=20
> > > > connections when
> > > > > switching between SIP and SigComp should not be neglected.=20
> > > > In some cases
> > > > > it may be of little concern but in others it may be=20
> > > > important. Reducing
> > > > > delay has been the most important driving factor for SigComp.
> > > > > Multiplexing over the same connection reduces the delay.
> > > > >
> > > >=20
> > > > But most of the cases where you want to switch between SigComp /
> > > > non-SigComp are inherently cases where there is=20
> inherent latency.
> > > >=20
> > > > [JC]I agree that these cases will probably have a lot of=20
> > > > latency, so this argument isn't very heavy. Still, less=20
> > > > latency is better.
> > > >=20
> > > > And RFC 3261 (if I read it correctly, but I'm no SIP guru),=20
> > > > talks about
> > > > only try to keep a transaction within in a single connection...
> > > >=20
> > > > > 2.  We have the tools needed for multiplexing SigComp and=20
> > > SIP on the
> > > > > same TCP connection. Why not do it? It's faster than=20
> > opening a new
> > > > > connection.
> > > > >
> > > >=20
> > > > But the 'tools' are horrible!  Reiterating the sentiments=20
> > > > from S-F: 'yeah,
> > > > you could, but why would you want to..?'
> > > >=20
> > > > [JC]I don't know why this is so horrible? Could you please=20
> > > > elaborate...
> > > >=20
> > > > > 3. Mark argues that if multiplexing is allowed, the=20
> > > > receiver needs to
> > > > > support it. But doesn't it already need to do this? As=20
> > > > Zhigang pointed
> > > > > out, RFC3486 states that
> > > > >
> > > > > "If a SIP client sends a compressed request and the client=20
> > > > transaction
> > > > >    times out without having received any response, the=20
> > > client SHOULD
> > > > >    retry the same request without using compression.  If=20
> > > > the compressed
> > > > >    request was sent over a TCP connection, the client=20
> > > > SHOULD close that
> > > > >    connection and open a new one to send the uncompressed=20
> > > request."
> > > > >
> > > > > In this text, it says SHOULD and not MUST, i.e. there could be
> > > > > implementations that send SIP and SigComp on the same TCP=20
> > > > connection, so
> > > > > all implementations should be able to handle it. In any=20
> > > > case, we have
> > > > > delimiters to determine where SigComp messages start and=20
> > > end, and if
> > > > > there is something other than SigComp in between an end and=20
> > > > a start, we
> > > > > should be able to handle it.
> > > > >
> > > >=20
> > > > That is slicing a semantic a shade too thin for me!  But, if=20
> > > > you want to
> > > > play word games, then I'll point out that 'uncompressed' does=20
> > > > not mean 'do
> > > > not use SigComp'.
> > > > [JC] I didn't intend this to be a word game. To me this is a=20
> > > > recommendation to open a new TCP connection. I'm perfectly=20
> > > > happy with that. But I don't think we should turn the=20
> > > > recommendation into a must and forbid sending SigComp and=20
> > > > non-SigComp on the same TCP connection.
> > > >=20
> > > > Carsten & I have whipped up a nice little 13-byte=20
> standard header:
> > > >=20
> > > >   0xf8, 0x00, 0xa1, 0x1c, 0x01, 0x86, 0x09, 0x22,
> > > >   0x86, 0x01, 0x16, 0xf9, 0x23, ...
> > > >=20
> > > > that will simply output the following bytes!
> > > >=20
> > > > It's a SigComp message, but there's no compression...
> > > >=20
> > > > [JC]Yes, this is perfectly fine, unless the message is larger=20
> > > > than 64 K.=20
> > > >=20
> > > > And let's not forget that the purpose of=20
> > > > draft-ietf-rohc-sip-sigcomp is to
> > > > fill in the missing pieces of the mapping jigsaw (i.e.=20
> > the bits that
> > > > aren't in RFC 3486).  And that's what we're arguing about. =20
> > > > (Although I
> > > > was fairly sure that we'd already decided this.  *sigh*)
> > > >=20
> > >=20
> > > _______________________________________________
> > > Rohc mailing list
> > > Rohc@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/rohc
> > >=20
> >=20
> > _______________________________________________
> > Rohc mailing list
> > Rohc@ietf.org
> > https://www1.ietf.org/mailman/listinfo/rohc
> >=20
>=20
> _______________________________________________
> Rohc mailing list
> Rohc@ietf.org
> https://www1.ietf.org/mailman/listinfo/rohc
>=20

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



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From: "West, Mark" <mark.a.west@roke.co.uk>
To: Yousuf.Saifullah@nokia.com
Cc: rohc@ietf.org
Date: Thu, 20 Nov 2003 16:07:56 +0000 (GMT)
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Subject: Re: [rohc] IP Address Compression for Context Replication
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Hi Yousuf,

I'm not sure how often you would set up a temporally close sequence of
short-lived connections to different machines on the same subnet
(web-server load balancing, perhaps?  but I thought that this was often
hidden from the user...)  Do you have any views on this..?

However, more seriously, I note that the draft has an IPR statement :-(

We are trying to make the TCP compression solution as generally useful as
possible and so the inclusion of an encumbered concept is something that
I, personally, would object to.  Can you clarify this position?

As a final comment, I'm not sure how helpful the appendix is, since
classless routing has been used since (at least) RFCs 1517/18/19...
(... and site-local addresses in IPv6 are now deprecated ;-)

Cheers,

Mark.

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



On Wed, 19 Nov 2003 Yousuf.Saifullah@nokia.com wrote:

> Date: Wed, 19 Nov 2003 13:23:33 -0600
> From: Yousuf.Saifullah@nokia.com
> To: rohc@ietf.org
> Subject: [rohc] IP Address Compression for Context Replication
>
> Hello,
> The current Context Replication mechanism suggests to send either whole IP address  or no IP address (replication). We have submitted a draft on the idea of sending compressed IP address using base IP address. This improves header compression efficiency when packet flows carry similar (e.g. same prefix) but not exactly same IP addresses. In particular, the gain can be significant for IPv6 header compression due to the large size (16 bytes) of IPv6 addresses.
>
> http://www.ietf.org/internet-drafts/draft-saifullah-rohc-ip-addr-comp-context-replication-00.txt
>
> This draft has impacts on section 6.5.5.1 of draft-ietf-rohc-tcp-05.txt, where the format of src/dest IP addresses are defined for IR-CR.
>
> Your comments and questions on this draft are more than welcome.
>
> Regards,
> Yousuf
>
>
> _______________________________________________
> Rohc mailing list
> Rohc@ietf.org
> https://www1.ietf.org/mailman/listinfo/rohc
>

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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-03.txt
	Pages		: 13
	Date		: 2003-11-20
	
This document contains requirements for the IP/SCTP header
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From: "West, Mark" <mark.a.west@roke.co.uk>
To: Robert.Sugar@nokia.com
Cc: zhigang.c.liu@nokia.com, lars-erik.jonsson@ericsson.com, rohc@ietf.org
Date: Thu, 20 Nov 2003 22:28:28 +0000 (GMT Standard Time)
Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp messages on t
 he same TCP connection
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Hi all,

If you want a different take on this question, how about this...

We seem to be addressing 2 questions: (1) is multiplexing of SigComp /
non-SigComp messages over TCP *generally* useful; (2) is this useful in a
constrained environment with an IPsec SA between peers..?

Considering (2), something that has not yet been raised is the use of what
is sometimes referred to as 'continuous mode'.  The state model in RFC
3320 exists for 2 reasons: lack of security between end-points; and
(potential) lack of reliability.  The IPsec SA and use of TCP renders both
of these non-issues.  So, why not have a persistent UDVM..?

This is perfectly reasonable within RFC 3320 -- there is no need for
state-storage, as the UDVM is simply kept running.

Since, in this model, there is no need to align SIP messages with SigComp
boundaries, the 64kbyte limit need not apply: SigComp ends up simply
providing a compressed TCP stream...
... so connection close semantics are preserved...

Failures essentially become desynchronisation problems (as could occur
with regular SIP), requiring reset of the TCP connection.

The sending of low processing cost messages can still be supported by
using the header that was suggested earlier (with the decompressor
explicitly looking for this).

And no need for multiplexing..! ;-)

Cheers,

Mark.



On Thu, 20 Nov 2003 Robert.Sugar@nokia.com wrote:

> Date: Thu, 20 Nov 2003 15:57:30 +0100
> From: Robert.Sugar@nokia.com
> To: zhigang.c.liu@nokia.com, lars-erik.jonsson@ericsson.com,
>      mark.a.west@roke.co.uk
> Cc: rohc@ietf.org
> Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp messages on
>     t he same TCP connection
>
> Hi all,
>
>  Let me give some more facets to the discussion:
>
>  Many times the latency is not just the TCP handshake, but has other contributors as well. In case of IPSec, there is no abundance of TCP connection possibilities, those are limited to at most the number of SAs available. And in order to set up a new SA, the use of an additional protocol (e.g. IKE, or RFC3310) is needed. So it is not only the new TCP setup delay, but also the SA setup delay. In case of TLS the handshake costs quite a few messages too.
>
>  Besides the >64K messages, and general SigComp errors, RFC3486 is the other reason to have SigComp and non-SigComp messages multiplexed (at least in time). Endpoints have the possibility to indicate willingness for receiving SigComp messages on a per transaction basis (for example a CPU-challenged endpoint wants to compress delay-sensitive session setups, but not instant messages). We should not burden that choice with significant additional delay, because that would take away useful functionalites from RFC3486.
>
>  In many protocols (HTTP being one of them), TCP connection close has also application level meanings, thus opening a new connection may cause application level delays as well as a different user experience or protocol behavior.
>
> BR,
> Robert
>
>
> > -----Original Message-----
> > From: rohc-admin@ietf.org [mailto:rohc-admin@ietf.org]On Behalf Of ext
> > zhigang.c.liu@nokia.com
> > Sent: November 19,2003 22:01
> > To: lars-erik.jonsson@ericsson.com; mark.a.west@roke.co.uk
> > Cc: rohc@ietf.org
> > Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp
> > messages on
> > t he same TCP connection
> >
> >
> > Hi,
> >
> > If I remember correctly, what has been closed/clarified was that
> > RFC 3320 does not allow multiplexing.
> >
> > However, the issue here is whether or not the multiplexing is needed.
> > Obviously, people already have their own opinions in mind. But until
> > recently, there haven't been many technical discussions regarding
> > this issue on ROHC mailing list. So, I'm not sure what do you mean
> > by "we have already closed the issue several times"? Or did I miss
> > anything?
> >
> > Either way, what I'd like to see is a serious debate on this issue
> > from different aspects, so that in the end, we can say with confidence
> > that we've taken a hard look on the issue and here is our conclusion.
> >
> > BR,
> > Zhigang
> >
> > > -----Original Message-----
> > > From: rohc-admin@ietf.org [mailto:rohc-admin@ietf.org]On
> > Behalf Of ext
> > > Lars-Erik Jonsson (LU/EAB)
> > > Sent: November 19, 2003 09:54 AM
> > > To: 'West, Mark'
> > > Cc: rohc@ietf.org
> > > Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp
> > > messages on
> > > t he same TCP connection
> > >
> > >
> > > I agree with Mark that this is a new and rather controversial
> > > new requirement to introduce at this point, especially since
> > > we have already closed the issue several times.
> > >
> > > /L-E
> > >
> > >
> > > > Multiplexing is horrible (or, at the very least messy)
> > > > because you inherently bind the SigComp and SIP transport
> > > > handling for detecting the end of a message.  If you don't
> > > > allow multiplexing, you are free to implement it this way (if
> > > > you want), but if you *do* allow multiplexing then this
> > > > becomes a requirement...
> > > >
> > > > But, like I say, I thought that the 'non-multiplexing' issue
> > > > had been agreed.  And it's still not clear to me why it's
> > > > important to be able to do it!
> > > >
> > > > Cheers,
> > > >
> > > > Mark.
> > > >
> > > >
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: Jan Christoffersson (LU/EAB)
> > > > > [mailto:jan.christoffersson@ericsson.com]
> > > > > Sent: 17 November 2003 13:47
> > > > > To: West, Mark
> > > > > Cc: zhigang.c.liu@nokia.com; Robert.Sugar@nokia.com;
> > rohc@ietf.org
> > > > > Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp
> > > > > messages on
> > > > > t he same TCP connection
> > > > >
> > > > >
> > > > > Hi Mark,
> > > > >
> > > > > Comments inline.
> > > > >
> > > > > -----Original Message-----
> > > > > From: West, Mark [mailto:mark.a.west@roke.co.uk]
> > > > > Sent: den 14 november 2003 15:09
> > > > > To: Jan Christoffersson (LU/EAB)
> > > > > Cc: zhigang.c.liu@nokia.com; Robert.Sugar@nokia.com;
> > rohc@ietf.org
> > > > > Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp
> > > > > messages on
> > > > > t he same TCP connection
> > > > >
> > > > >
> > > > >
> > > > > Hi Jan,
> > > > >
> > > > > Comments added!
> > > > >
> > > > > Cheers,
> > > > >
> > > > > Mark.
> > > > >
> > > > > >
> > > > > > 1. The extra delay caused by opening and closing TCP
> > > > > connections when
> > > > > > switching between SIP and SigComp should not be neglected.
> > > > > In some cases
> > > > > > it may be of little concern but in others it may be
> > > > > important. Reducing
> > > > > > delay has been the most important driving factor for SigComp.
> > > > > > Multiplexing over the same connection reduces the delay.
> > > > > >
> > > > >
> > > > > But most of the cases where you want to switch between SigComp /
> > > > > non-SigComp are inherently cases where there is
> > inherent latency.
> > > > >
> > > > > [JC]I agree that these cases will probably have a lot of
> > > > > latency, so this argument isn't very heavy. Still, less
> > > > > latency is better.
> > > > >
> > > > > And RFC 3261 (if I read it correctly, but I'm no SIP guru),
> > > > > talks about
> > > > > only try to keep a transaction within in a single connection...
> > > > >
> > > > > > 2.  We have the tools needed for multiplexing SigComp and
> > > > SIP on the
> > > > > > same TCP connection. Why not do it? It's faster than
> > > opening a new
> > > > > > connection.
> > > > > >
> > > > >
> > > > > But the 'tools' are horrible!  Reiterating the sentiments
> > > > > from S-F: 'yeah,
> > > > > you could, but why would you want to..?'
> > > > >
> > > > > [JC]I don't know why this is so horrible? Could you please
> > > > > elaborate...
> > > > >
> > > > > > 3. Mark argues that if multiplexing is allowed, the
> > > > > receiver needs to
> > > > > > support it. But doesn't it already need to do this? As
> > > > > Zhigang pointed
> > > > > > out, RFC3486 states that
> > > > > >
> > > > > > "If a SIP client sends a compressed request and the client
> > > > > transaction
> > > > > >    times out without having received any response, the
> > > > client SHOULD
> > > > > >    retry the same request without using compression.  If
> > > > > the compressed
> > > > > >    request was sent over a TCP connection, the client
> > > > > SHOULD close that
> > > > > >    connection and open a new one to send the uncompressed
> > > > request."
> > > > > >
> > > > > > In this text, it says SHOULD and not MUST, i.e. there could be
> > > > > > implementations that send SIP and SigComp on the same TCP
> > > > > connection, so
> > > > > > all implementations should be able to handle it. In any
> > > > > case, we have
> > > > > > delimiters to determine where SigComp messages start and
> > > > end, and if
> > > > > > there is something other than SigComp in between an end and
> > > > > a start, we
> > > > > > should be able to handle it.
> > > > > >
> > > > >
> > > > > That is slicing a semantic a shade too thin for me!  But, if
> > > > > you want to
> > > > > play word games, then I'll point out that 'uncompressed' does
> > > > > not mean 'do
> > > > > not use SigComp'.
> > > > > [JC] I didn't intend this to be a word game. To me this is a
> > > > > recommendation to open a new TCP connection. I'm perfectly
> > > > > happy with that. But I don't think we should turn the
> > > > > recommendation into a must and forbid sending SigComp and
> > > > > non-SigComp on the same TCP connection.
> > > > >
> > > > > Carsten & I have whipped up a nice little 13-byte
> > standard header:
> > > > >
> > > > >   0xf8, 0x00, 0xa1, 0x1c, 0x01, 0x86, 0x09, 0x22,
> > > > >   0x86, 0x01, 0x16, 0xf9, 0x23, ...
> > > > >
> > > > > that will simply output the following bytes!
> > > > >
> > > > > It's a SigComp message, but there's no compression...
> > > > >
> > > > > [JC]Yes, this is perfectly fine, unless the message is larger
> > > > > than 64 K.
> > > > >
> > > > > And let's not forget that the purpose of
> > > > > draft-ietf-rohc-sip-sigcomp is to
> > > > > fill in the missing pieces of the mapping jigsaw (i.e.
> > > the bits that
> > > > > aren't in RFC 3486).  And that's what we're arguing about.
> > > > > (Although I
> > > > > was fairly sure that we'd already decided this.  *sigh*)
> > > > >
> > > >
> > > > _______________________________________________
> > > > 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
> > >
> >
> > _______________________________________________
> > Rohc mailing list
> > Rohc@ietf.org
> > https://www1.ietf.org/mailman/listinfo/rohc
> >
>

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Hi

Is the compress scheme to compress the RTP/UDP/IP/PPP headers unitedly exist?

qibaojian

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Subject: [rohc] Question about scaled TS encoding in ROHC RTP
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Hi, all,

In ROHC RTP, the scaled TS is encoded using WLSB algorithm. Since the 
wraparound boundary between the TSS_MAX may not correspond to a natural 
boundary between LSBs, a number of unused values are inserted into it after 
TSS_MAX such that their LSBs follow naturally upon each other.  In order to 
ensure the number of valid values in the interpretation interval to be high 
enough to maintain robustness, a scaling rule (RFC3095, 4.5.3) is used to 
increase the number of bits actually sent at wraparound.

Here comes my question: What kind of robustness does the RFC actually refer 
to by maintaining a large number of valid values in the interpretation 
interval? It seems to me that if there are no errors in the bits 
transmitted in the compressed header, then the TS will be decoded correctly 
no matter how many invalid values are in the interval. If there are errors 
in the transmitted bits, then the decompressed TS will always be in error 
no matter how few the number of invalid values in the interval is, an 
erroneous TS is an erroneous TS, no matter it is a valid value or invalid 
value in the interval. Can anyone could please explain the reasoning behind 
the scaling rule?

Actually, I also notice that if the scaling rule is used, there may 
actually be some problem when there are negative wraparounds.
Here is an example, suppose tss_max = 0x0333, reference value vr = 0, the 
value to be coded v = 0x0333, p = 2^(k-2)-1 according to ROHC.

First, LSB encoding:
k       p       interval                        interval offset to vr
2       0           (0, 3)                              (0, 3)
3       1           (0x0337, 6)         (-1, 6)
4       3       (0x033D, 12)            (-3, 12)
5       7       (0x0339, 24)            (-7, 24)
6       15      (0x0331, 48)            (-15, 48)
It seems that 6 bits are needed.

Second, setting the 6 LSBs of vmax to 0 gives us 0x300, according to 
scaling rule, 9 bits will be sent.

Third, at the decompressor, the interpretation interval is given by (0 - 
127, 0+384) = (0x03ff + 1 - 0x7f, 384) = (0x381, 0x180), v = 0x0333 is NOT 
in the interval! Notice that since 9 bits are used in encoding, the 
wraparound boundary is now between 0x03FF and 0 after inserting some unused 
values.

What is the problem here?

Thank you.

Haipeng



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Subject: Re: [rohc] Question about the compress scheme to compress the RTP/UDP/IP/PPP headers
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> Is the compress scheme to compress the RTP/UDP/IP/PPP headers=20
> unitedly=A0exist?

I'm not quite sure about the context for this question, but for PPP=20
over serial, there is very little (one byte) left to compress once you=20=

use all of PPP's compression schemes.

If you are talking about tunnel aggregates, there is a different, much=20=

longer story.

Gruesse, Carsten


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From: "Jan Christoffersson (LU/EAB)" <jan.christoffersson@ericsson.com>
To: "'West, Mark'" <mark.a.west@roke.co.uk>, Robert.Sugar@nokia.com
Cc: zhigang.c.liu@nokia.com,
        "Lars-Erik Jonsson (LU/EAB)"
	 <lars-erik.jonsson@ericsson.com>,
        rohc@ietf.org
Subject: RE: [rohc] Re: multiplexing SigComp and non-SigComp messages on t
	 he same TCP connection
Date: Mon, 24 Nov 2003 11:13:17 +0100
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Hi Mark,

I have some concerns about this too...
Comments inline.

/Jan

> Considering (2), something that has not yet been raised is 
> the use of what
> is sometimes referred to as 'continuous mode'.  The state model in RFC
> 3320 exists for 2 reasons: lack of security between end-points; and
> (potential) lack of reliability.  The IPsec SA and use of TCP 
> renders both
> of these non-issues.  So, why not have a persistent UDVM..?
> 
> This is perfectly reasonable within RFC 3320 -- there is no need for
> state-storage, as the UDVM is simply kept running.

In order to use the continuous mode and compress messages relative to previous messages, the byte codes need to be tailored for this. This means that we need to know that the receiving endpoint is using continuous mode and this means negotiations. How to do this has not been defined.
 
> Since, in this model, there is no need to align SIP messages 
> with SigComp
> boundaries, the 64kbyte limit need not apply: SigComp ends up simply
> providing a compressed TCP stream...
> ... so connection close semantics are preserved...

Quoting from the RFC 3320: 

"Decompression failure occurs if the cumulative number of bytes provided to the dispatcher exceeds 65536 bytes."

I think that the word cumulative means that you cannot break a large message into little pieces and output them one by one. As soon as the total size is 65 K a decompression failure occurs. I even think that continuous mode could worsen the situation: If several SIP messages are decompressed by the same instance of the UDVM (which could be done in continuous mode)it could happen that decompression failure occurs even if no single message is above the individual limit.

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Subject: RE: [rohc] IP Address Compression for Context Replication
Date: Mon, 24 Nov 2003 13:41:25 -0600
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Hi Mark,
Thanks for the comments. Please see my reply below. What is your opinion =
on this draft as a technical solution for enhancing compression =
efficiency?=20

BR,
Yousuf



> -----Original Message-----
> From: ext West, Mark [mailto:mark.a.west@roke.co.uk]
> Sent: Thursday, November 20, 2003 10:08 AM
> To: Saifullah Yousuf (NRC/Dallas)
> Cc: rohc@ietf.org
> Subject: Re: [rohc] IP Address Compression for Context Replication
>=20
>=20
>=20
> Hi Yousuf,
>=20
> I'm not sure how often you would set up a temporally close sequence of
> short-lived connections to different machines on the same subnet
> (web-server load balancing, perhaps?  but I thought that this=20
> was often
> hidden from the user...)  Do you have any views on this..?

I think it would be quite frequent to consider:
- One case is mobile router, e.g. a train connected through a mobile =
router to an external network. In this case the IP addresses can be =
temporally close.
- Another case is Personal Area Network, e.g. a person carrying a group =
of bluetooth devices which are connected via a mobile phone to an =
external network. In this case also the IP addresses will be temporally =
close.=20
- Some networks, e.g. cellular, have IP terminals with same prefix =
indicating their gateways for their external networks.

>=20
> However, more seriously, I note that the draft has an IPR=20
> statement :-(
>=20
> We are trying to make the TCP compression solution as=20
> generally useful as
> possible and so the inclusion of an encumbered concept is=20
> something that
> I, personally, would object to.  Can you clarify this position?
>
Our IPR folks are sending an IPR disclosure related to this draft with =
more information on the following web-page:
http://www.ietf.org/ipr.=20

Once that information is available, the WG can look into the information =
and can see that it satisfy the needs for generally useful.

I would like to say that the proposed IP address compression could be an =
additional optional feature on top of basic context replication. As an =
optional feature, it should not have concern as an encumber concept.


> As a final comment, I'm not sure how helpful the appendix is, since
> classless routing has been used since (at least) RFCs 1517/18/19...
> (... and site-local addresses in IPv6 are now deprecated ;-)
>=20
Thanks. I'll update the appendix.

> Cheers,
>=20
> Mark.
>=20
> --
> Mark A. West, Senior Consultant Engineer
> Roke Manor Research Ltd., Romsey, Hants.  SO51 0ZN
> Phone +44 (0)1794 833311   Fax  +44 (0)1794 833433
>=20
>=20
>=20
> On Wed, 19 Nov 2003 Yousuf.Saifullah@nokia.com wrote:
>=20
> > Date: Wed, 19 Nov 2003 13:23:33 -0600
> > From: Yousuf.Saifullah@nokia.com
> > To: rohc@ietf.org
> > Subject: [rohc] IP Address Compression for Context Replication
> >
> > Hello,
> > The current Context Replication mechanism suggests to send=20
> either whole IP address  or no IP address (replication). We=20
> have submitted a draft on the idea of sending compressed IP=20
> address using base IP address. This improves header=20
> compression efficiency when packet flows carry similar (e.g.=20
> same prefix) but not exactly same IP addresses. In=20
> particular, the gain can be significant for IPv6 header=20
> compression due to the large size (16 bytes) of IPv6 addresses.
> >
> >=20
> http://www.ietf.org/internet-drafts/draft-saifullah-rohc-ip-ad
dr-comp-context-replication-00.txt
>
> This draft has impacts on section 6.5.5.1 of =
draft-ietf-rohc-tcp-05.txt, where the format of src/dest IP addresses =
are defined for IR-CR.
>
> Your comments and questions on this draft are more than welcome.
>
> Regards,
> Yousuf
>
>
> _______________________________________________
> Rohc mailing list
> Rohc@ietf.org
> https://www1.ietf.org/mailman/listinfo/rohc
>

--
Registered Office: Roke Manor Research Ltd, Siemens House, Oldbury, =
Bracknell,
Berkshire. RG12 8FZ

The information contained in this e-mail and any attachments is =
confidential to
Roke Manor Research Ltd and must not be passed to any third party =
without
permission. This communication is for information only and shall not =
create or
change any contractual relationship.


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From: "James Ford" <james_s_ford@hotmail.com>
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Subject: [rohc] SIP-SigComp Questions
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Hi,

I'm implementing now the SigComp feature in my SIP application and wonder=
ing=20
about some undefined issues in the SigComp/SIP RFCs:

(1) A Quotation from the 7th section in RFC 3486 is:=94 If a SIP client s=
ends=20
a compressed request and the client transaction times out without having=20
received any response, the client SHOULD retry the same request without=20
using compression.=94 What is the exact meaning of re-sending the same=20
request? Should we use the same transaction for sending the uncompressed=20
message or maybe we should promote the Cseq for handling the situation th=
at=20
the server-side actually received the compressed message, decompressed it=
=20
and didn=92t manage to reply yet? Moreover, should we change the =91branc=
h=92=20
parameter (Via header) in the uncompressed message?

(2) Regarding the same quotation from the previous question, how should w=
e=20
handle the following scenario: A SIP message should be sent compressed=20
(using SigComp) according to a route list (as a result of DNS request). T=
his=20
route list contains four explicit addresses but only the latter is a=20
reachable address. If we=92ll follow the RFC 3486 instructions, then the =
SIP=20
message will be sent compressed, a timeout will expire,the same message w=
ill=20
be re-sent uncompressed until another timeout will expire and so on. This=
=20
process will be repeated for at least three times and might take too long=
,=20
what can we do in order to avoid this slow mechanism?

(3) According to RFC 3261:=94 It is RECOMMENDED that connections be kept =
open=20
for some implementation-defined duration after the last message was sent =
or=20
received over that connection=85 This is to make it likely that transacti=
ons=20
are completed over the same connection on which they are initiated (for=20
example, request, response, and in the case of INVITE, ACK for non-2xx=20
responses).=94 The standard recommends on sending all the dialog transact=
ions=20
over a single TCP connection. However, RFC 3486 says: =93If a client does=
 not=20
know whether or not the server supports SigComp, but in case the server=20
supported it, it would like to receive compressed responses, this client=20
SHOULD add the parameter comp=3Dsigcomp to the topmost entry of the Via h=
eader=20
field.  The request, however, as stated above, will not be compressed.=94=
 and=20
then <draft-ietf-rohc-sigcomp-sip-00.txt> says in its 4th section:=94=20
Applications MUST NOT mix SIP messages and SigComp messages on a single T=
CP=20
connection.  If the TCP connection is used to carry SigComp messages then=
=20
all messages that are sent over the connection MUST have a SigComp header=
.=94 =20
If we refer to both of the quotations we can conclude that the SigComp=20
standards lead us to use different TCP connections in a single dialog as=20
opposed to the RFC 3261 recommendation, isn=92t it? How can we deal with =
this=20
conflict?


Best Regards,
James.

_________________________________________________________________
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From: "James Ford" <james_s_ford@hotmail.com>
To: rohc@ietf.org
Date: Mon, 24 Nov 2003 13:53:03 +0000
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Subject: [rohc] SIP-SigComp Questions
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Hi,

I'm implementing now the SigComp feature in my SIP application and wonder=
ing=20
about some undefined issues in the SigComp/SIP RFCs:

(1) Can SigComp messages be exchanged intermittently with uncompressed=20
general SIP messages over the same SIP dialog (call)?


(2) Is it possible that two different Transactions in the same dialog (ca=
ll)=20
will use different Compartment Ids?

(3) A Quotation from the 7th section in RFC 3486 is:=94 If a SIP client s=
ends=20
a compressed request and the client transaction times out without having=20
received any response, the client SHOULD retry the same request without=20
using compression.=94 What is the exact meaning of re-sending the same=20
request? Should we use the same transaction for sending the uncompressed=20
message or maybe we should promote the Cseq for handling the situation th=
at=20
the server-side actually received the compressed message, decompressed it=
=20
and didn=92t manage to reply yet? Moreover, should we change the =91branc=
h=92=20
parameter (Via header) in the uncompressed message?

(4) Regarding the same quotation from the previous question, how should w=
e=20
handle the following scenario: A SIP message should be sent compressed=20
(using SigComp) according to a route list (as a result of DNS request). T=
his=20
route list contains four explicit addresses but only the latter is a=20
reachable address. If we=92ll follow the RFC 3486 instructions, then the =
SIP=20
message will be sent compressed, a timeout will expire,the same message w=
ill=20
be re-sent uncompressed until another timeout will expire and so on. This=
=20
process will be repeated for at least three times and might take too long=
,=20
what can we do in order to avoid this slow mechanism?

(5) According to RFC 3261:=94 It is RECOMMENDED that connections be kept =
open=20
for some implementation-defined duration after the last message was sent =
or=20
received over that connection=85 This is to make it likely that transacti=
ons=20
are completed over the same connection on which they are initiated (for=20
example, request, response, and in the case of INVITE, ACK for non-2xx=20
responses).=94 The standard recommends on sending all the dialog transact=
ions=20
over a single TCP connection. However, RFC 3486 says: =93If a client does=
 not=20
know whether or not the server supports SigComp, but in case the server=20
supported it, it would like to receive compressed responses, this client=20
SHOULD add the parameter comp=3Dsigcomp to the topmost entry of the Via h=
eader=20
field.  The request, however, as stated above, will not be compressed.=94=
 and=20
then <draft-ietf-rohc-sigcomp-sip-00.txt> says in its 4th section:=94=20
Applications MUST NOT mix SIP messages and SigComp messages on a single T=
CP=20
connection.  If the TCP connection is used to carry SigComp messages then=
=20
all messages that are sent over the connection MUST have a SigComp header=
.=94 =20
If we refer to both of the quotations we can conclude that the SigComp=20
standards lead us to use different TCP connections in a single dialog as=20
opposed to the RFC 3261 recommendation, isn=92t it? How can we deal with =
this=20
conflict?


Best Regards,
James.

_________________________________________________________________
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From: "Lars-Erik Jonsson (LU/EAB)" <lars-erik.jonsson@ericsson.com>
To: "'Yousuf.Saifullah@nokia.com'" <Yousuf.Saifullah@nokia.com>,
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Cc: rohc@ietf.org
Subject: RE: [rohc] IP Address Compression for Context Replication
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One observation on the non-technical issue...

> > However, more seriously, I note that the draft has an IPR 
> > statement :-(
> > 
> > We are trying to make the TCP compression solution as 
> > generally useful as possible and so the inclusion of an
> > encumbered concept is something that I, personally, would
> > object to.  Can you clarify this position?
> >
> Our IPR folks are sending an IPR disclosure related to this 
> draft with more information on the following web-page:
> http://www.ietf.org/ipr. 
> 
> Once that information is available, the WG can look into the 
> information and can see that it satisfy the needs for 
> generally useful.
> 
> I would like to say that the proposed IP address compression 
> could be an additional optional feature on top of basic 
> context replication. As an optional feature, it should not 
> have concern as an encumber concept.

The only way we can have optional features that would not
have to be implemented would be by having multiple profiles,
as that is our way of providing negotiation of supported 
functionality. 

As a general technical comment, I think there are many small
optimizations we could do with context replication, but that
does not mean it is a good idea to do it.

/L-E 

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From: "Jan Christoffersson (LU/EAB)" <jan.christoffersson@ericsson.com>
To: "'James Ford'" <james_s_ford@hotmail.com>, rohc@ietf.org
Subject: RE: [rohc] SIP-SigComp Questions
Date: Tue, 25 Nov 2003 15:33:01 +0100
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Hi James

Your question number 5 is currently being debated on the ROHC list, see http://www1.ietf.org/mail-archive/working-groups/rohc/current/msg01809.html and follow-ups.

BR
Jan

> (5) According to RFC 3261:" It is RECOMMENDED that 
> connections be kept open 
> for some implementation-defined duration after the last 
> message was sent or 
> received over that connection... This is to make it likely that 
> transactions 
> are completed over the same connection on which they are 
> initiated (for 
> example, request, response, and in the case of INVITE, ACK 
> for non-2xx 
> responses)." The standard recommends on sending all the 
> dialog transactions 
> over a single TCP connection. However, RFC 3486 says: "If a 
> client does not 
> know whether or not the server supports SigComp, but in case 
> the server 
> supported it, it would like to receive compressed responses, 
> this client 
> SHOULD add the parameter comp=sigcomp to the topmost entry of 
> the Via header 
> field.  The request, however, as stated above, will not be 
> compressed." and 
> then <draft-ietf-rohc-sigcomp-sip-00.txt> says in its 4th section:" 
> Applications MUST NOT mix SIP messages and SigComp messages 
> on a single TCP 
> connection.  If the TCP connection is used to carry SigComp 
> messages then 
> all messages that are sent over the connection MUST have a 
> SigComp header."  
> If we refer to both of the quotations we can conclude that 
> the SigComp 
> standards lead us to use different TCP connections in a 
> single dialog as 
> opposed to the RFC 3261 recommendation, isn't it? How can we 
> deal with this 
> conflict?
> 
> 
> Best Regards,
> James.
> 
> _________________________________________________________________
> STOP MORE SPAM with the new MSN 8 and get 2 months FREE* 
> http://join.msn.com/?page=features/junkmail
> 
> 
> _______________________________________________
> Rohc mailing list
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> https://www1.ietf.org/mailman/listinfo/rohc
> 

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Hi James and others,

 Let me start with #5, the hot question:

 It is a very valid point you make here, I was about to raise this as =
well. If multiplexing is not allowed we'd have to open a new TCP =
connection every time the client is not sure whether or not the server =
supports compression, as the first message of the dialog must be =
non-SigComp. This is quite a likely case if the server's address was  =
discovered by external mechanisms (DNS, DHCP, email, etc.). This would =
effectively delay the session setup as opposed to the main goal of RFC =
3320.

 Now to the other questions:

#1=20
Yes, as long as the obey the rules of RFC3486. If there is =
comp=3Dsigcomp in the Via: header in the initial transaction, but not in =
the others, SIP and SigComp messages are interspersed in the same =
dialog.

#2
This is not clearly defined, and depends on how do you handle =
compartments (e.g. is it for one dialog or longer, do you use multiple =
compartments, etc.). I would say a SHOULD (be in the same compartment).

#3=20
In case of UDP it should be part of the SIP retransmission mechanism =
(e.g. first two retransmissions are compressed, third retransmission is =
non-SigComp). In case of TCP you may do nothing and wait for timeout, or =
resend the same message non-SigComp, risking multiple retransmissions if =
there is a general network failure. I favor the second option for the =
sake of robustness, although it is not really recommended by SIP. I =
would not advise to promote CSeq, since this cannot be done with SIP =
responses, only requests.

#4
I do not really see the case here, could you elaborate? Do you refer to =
forking functionality, or why would one send the request to four =
different addresses?

BR,
Robert

> -----Original Message-----
> From: rohc-admin@ietf.org [mailto:rohc-admin@ietf.org]On Behalf Of ext
> James Ford
> Sent: November 24,2003 14:53
> To: rohc@ietf.org
> Subject: [rohc] SIP-SigComp Questions
>=20
>=20
> Hi,
>=20
> I'm implementing now the SigComp feature in my SIP=20
> application and wondering=20
> about some undefined issues in the SigComp/SIP RFCs:
>=20
> (1) Can SigComp messages be exchanged intermittently with=20
> uncompressed=20
> general SIP messages over the same SIP dialog (call)?
>=20
>=20
> 2) Is it possible that two different Transactions in the=20
> same dialog (call)=20
> will use different Compartment Ids?
>=20
> (3) A Quotation from the 7th section in RFC 3486 is:" If a=20
> SIP client sends=20
> a compressed request and the client transaction times out=20
> without having=20
> received any response, the client SHOULD retry the same=20
> request without=20
> using compression." What is the exact meaning of re-sending the same=20
> request? Should we use the same transaction for sending the=20
> uncompressed=20
> message or maybe we should promote the Cseq for handling the=20
> situation that=20
> the server-side actually received the compressed message,=20
> decompressed it=20
> and didn't manage to reply yet? Moreover, should we change=20
> the 'branch'=20
> parameter (Via header) in the uncompressed message?
>=20
> (4) Regarding the same quotation from the previous question,=20
> how should we=20
> handle the following scenario: A SIP message should be sent=20
> compressed=20
> (using SigComp) according to a route list (as a result of DNS=20
> request). This=20
> route list contains four explicit addresses but only the latter is a=20
> reachable address. If we'll follow the RFC 3486 instructions,=20
> then the SIP=20
> message will be sent compressed, a timeout will expire,the=20
> same message will=20
> be re-sent uncompressed until another timeout will expire and=20
> so on. This=20
> process will be repeated for at least three times and might=20
> take too long,=20
> what can we do in order to avoid this slow mechanism?
>=20
> (5) According to RFC 3261:" It is RECOMMENDED that=20
> connections be kept open=20
> for some implementation-defined duration after the last=20
> message was sent or=20
> received over that connection... This is to make it likely that=20
> transactions=20
> are completed over the same connection on which they are=20
> initiated (for=20
> example, request, response, and in the case of INVITE, ACK=20
> for non-2xx=20
> responses)." The standard recommends on sending all the=20
> dialog transactions=20
> over a single TCP connection. However, RFC 3486 says: "If a=20
> client does not=20
> know whether or not the server supports SigComp, but in case=20
> the server=20
> supported it, it would like to receive compressed responses,=20
> this client=20
> SHOULD add the parameter comp=3Dsigcomp to the topmost entry of=20
> the Via header=20
> field.  The request, however, as stated above, will not be=20
> compressed." and=20
> then <draft-ietf-rohc-sigcomp-sip-00.txt> says in its 4th section:"=20
> Applications MUST NOT mix SIP messages and SigComp messages=20
> on a single TCP=20
> connection.  If the TCP connection is used to carry SigComp=20
> messages then=20
> all messages that are sent over the connection MUST have a=20
> SigComp header." =20
> If we refer to both of the quotations we can conclude that=20
> the SigComp=20
> standards lead us to use different TCP connections in a=20
> single dialog as=20
> opposed to the RFC 3261 recommendation, isn't it? How can we=20
> deal with this=20
> conflict?
>=20
>=20
> Best Regards,
> James.
>=20
> _________________________________________________________________
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>=20
>=20
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>=20

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