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To: Pasi.Eronen@nokia.com
Subject: Re: [TLS] I-D ACTION:draft-ietf-tls-psk-03.txt
References: <125EA890549C8641A72F3809CB80DCCD16FE0D@esebe056.ntc.nokia.com>
From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 01 Dec 2004 15:14:53 -0800
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	(Pasi Eronen's message of "Tue, 30 Nov 2004 18:40:46 +0200")
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Peter,

CAn you live with this?

-Ekr

Pasi.Eronen@nokia.com writes:

> Hi,
>
> If we assume the typical scenario would be that a person enters
> the identifier and PSK in the TLS client (and server), then IMHO
> the current approach (UTF-8 encoding) significantly increases
> the chances for interoperability. It also simplifies actual
> implementations, since they don't have to implement encoding/
> decoding operations for all the different types of identifiers
> people might want to use.
>
> If, on the other hand, the identifiers and keys on both
> endpoints come from some key distribution system (or something),
> and this system actually cares about the type of the identifier
> then using e.g. DER for X.500 DNs might make sense... but I
> think the former case will be much more common (and many systems
> will probably just do strcpy/strcmp for something that a person
> entered somewhere, so UTF-8 will work fine).
>
> This way, we also avoid the need to define what exactly is a
> user name and what is not (not exactly simple if the "user" 
> not a person but a device or a software component), and all
> the interoperability issues this would bring up... so IMHO
> we should keep it as it is.
>
> Best regards,
> Pasi
>
>> -----Original Message-----
>> From: Peter Gutmann
>> Sent: Friday, November 26, 2004 1:17 PM
>> To: ietf-tls@lists.certicom.com
>> Subject: Re: [TLS] I-D ACTION:draft-ietf-tls-psk-03.txt
>> 
>> I've just been going through the -03 draft and I'm not sure
>> whether the new requirements for encoding the identity are
>> such a good idea.  The older -02 had "UTF-8 everything" which
>> admittedly was a bit ambiguous, but I think the current
>> requirement to convert things that aren't normally UTF-8 into
>> UTF-8 will cause more interop problems than it solves, because
>> some of the IDs that could potentially be used have ambiguous
>> encodings in text form.  The obvious example is X.500 DNs, not
>> only because (even in a perfect world) there are a pile of
>> sematically equivalent forms that can't be easily compared
>> with a strcmp(), but also because we're in an imperfect world
>> where people put latin-1 into T61Strings that can't be
>> reliably turned into UTF-8 and have multi-AVA RDNs for which
>> some implementations will only see the first AVA and not the
>> others, and a pile of other things.  Having to implement X.500
>> en/decoding and string comparison rules in a TLS
>> implementation is probably asking a bit much.
>> 
>> IPv6 has somewhat similar problems, there are several ways to
>> represent an address, so you'd need to add complex
>> en/decode/comparison logic to your code to handle this.
>> 
>> To resolve this, I think it's necessary to at least provide
>> information on a couple of common ID types and their
>> encodings, e.g:
>> 
>>   User names (where the user may be a human or a device, and 
>>   which usually correspond to account names on the server being 
>>   connected to) are encoded as a UTF-8 character string, without 
>>   any terminating null character.
>> 
>>   IPv4 and IPv6 addresses are encoded as 32-bit and 128-bit 
>>   integers in network byte order.
>> 
>>   URIs are encoded as specified in [RFC2616].
>> 
>>   Domain names are encoded as specified in [whatever the ref is]
>> 
>>   X.500 DNs are encoded as DER-encoded ASN.1 as it appears in 
>>   the certificate or other object from which it's referenced.
>> 
>> Having said that, I'm not sure at the moment why you'd need
>> something other than a straight user name.  For example, I
>> can't see why I'd be connecting to a server and identifying
>> myself as an IP address (I can certainly dream up an imaginary
>> scenario if given a few minutes, but I can't think of any
>> real-world situation where I'd need the ability to do this).
>> So another option would be to leave everything but the user
>> name unspecified and see if anyone ever needs any of the other
>> options, and if so how they want to handle them.
>> 
>> Peter.
>
>
> _______________________________________________
> TLS mailing list
> TLS@lists.ietf.org
> https://www1.ietf.org/mailman/listinfo/tls
>

-- 
[Eric Rescorla                                   ekr@rtfm.com]
                http://www.rtfm.com/

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From: "Hollenbeck, Scott" <shollenbeck@verisign.com>
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Subject: [TLS] Implementations of RFC 3749 (TLS Compression)?
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RFC 3749 has existed as a Proposed Standard since May 2004.  Now that it's
December we've met the six month requirement for progression described in
section 6.2 of RFC 2026:

"A specification shall remain at the Proposed Standard level for at least
six (6) months."

OK, so the TLS spec itself isn't there yet, but that doesn't mean that we
can't document implementation experience if any exists.  Progression to
Draft Standard status requires "at least two independent and interoperable
implementations from different code bases" (RFC 2026, section 4.1.2).  I'm
willing to write the implementation report(s) if there are implementers out
there who are willing to contribute information.  Are there any
implementations?

-Scott-

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From: pgut001@cs.auckland.ac.nz (Peter Gutmann)
To: ekr@rtfm.com
Subject: Re: [TLS] I-D ACTION:draft-ietf-tls-psk-03.txt
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EKR <ekr@rtfm.com> writes:

>CAn you live with this?

It seems to be leaving it as Someone Else's Problem.  I agree that for any
normal/usual scenario the ID is going to be a (text-form) user name, but
requiring that anything that isn't a user name be subject to a non-injective
transformation seems to be asking for trouble.  I could live with having the
handling of user names clearly defined and everything else left open (to be
resolved in future updates if it ever becomes necessary, which I think is
unlikely), but requiring a surjective transformation as the current text does
doesn't seem like a good move.

Peter.

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From: Joo-Won Jung <jwjung@initech.com>
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Hollenbeck, Scott wrote:

>Are there any implementations?
>  
>

As far as I know, there are two implementations, GNUTLS and OpenSSL-CVS,
which may be interoperable. (I tested the connect, send and receive
operations, but I did not test all the cases)

Moreover, GNUTLS implements LZO compression based on RFC-3749 framework.

Hope this helps...

- Joo-Won Jung


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From: "Hollenbeck, Scott" <shollenbeck@verisign.com>
To: "'Joo-Won Jung'" <jwjung@initech.com>
Subject: RE: [TLS] Implementations of RFC 3749 (TLS Compression)?
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> -----Original Message-----
> From: Joo-Won Jung [mailto:jwjung@initech.com]
> Sent: Friday, December 03, 2004 2:16 AM
> To: Hollenbeck, Scott
> Cc: ietf-tls@lists.certicom.com
> Subject: Re: [TLS] Implementations of RFC 3749 (TLS Compression)?
>
>
> Hollenbeck, Scott wrote:
>
> >Are there any implementations?
> > 
> >
>
> As far as I know, there are two implementations, GNUTLS and
> OpenSSL-CVS,
> which may be interoperable. (I tested the connect, send and receive
> operations, but I did not test all the cases)
>
> Moreover, GNUTLS implements LZO compression based on RFC-3749
> framework.
>
> Hope this helps...

Greatly, thanks.  Could I ask anyone else that's familiar with those
implementations to contact me off-list?

Thanks,
-Scott-

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Please use the new TLS mailing list address, tls@lists.ietf.org. The old
one @lists.certicom.com is forwarding now, but it won't be for long.

Thanks,

Win Treese

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From: "Hollenbeck, Scott" <shollenbeck@verisign.com>
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Win,

You might want to ask the Secretariat to update the working group charter
web page.  It still lists the "ietf-tls@lists.certicom.com" address.

-Scott-

> -----Original Message-----
> From: tls-bounces@lists.ietf.org [mailto:tls-bounces@lists.ietf.org] 
> Sent: Friday, December 03, 2004 4:28 PM
> To: tls@ietf.org
> Subject: [TLS] please use the new list address
> 
> 
> 
> Please use the new TLS mailing list address, 
> tls@lists.ietf.org. The old
> one @lists.certicom.com is forwarding now, but it won't be for long.
> 
> Thanks,
> 
> Win Treese
> 
> _______________________________________________
> TLS mailing list
> TLS@lists.ietf.org
> https://www1.ietf.org/mailman/listinfo/tls
> 

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From: pgut001@cs.auckland.ac.nz (Peter Gutmann)
To: tls@ietf.org
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Subject: [TLS] PSK question
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I was reading through the draft again an noticed:

  To help the client in selecting which identity to use, the server can
  provide a "PSK identity hint" in the ServerKeyExchange message (note that if
  no hint is provided, a ServerKeyExchange message is still sent)

I have no idea what an "identity hint" could possibly be (the spec says it's
optional, but I'd have no idea what to put in there in any case).  If the
field is used for something, what's it supposed to be used for?  If not,
perhaps it should be dropped.

(Hmm, and I'll repeat my earlier gripe that apart from its use to convey this
 unknown quantity, there's no need to send the server keyex at all.  In other
 words in 99.99% of cases it's just going to be code and message bloat).

Peter.

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From: pgut001@cs.auckland.ac.nz (Peter Gutmann)
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Subject: [TLS] Re: PSK question
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Another thing I just noticed:

  The premaster secret is formed as follows: if the PSK is N octets long,
  concatenate an uint16 with the value N, N zero octets, a second uint16 with
  the value N, and the PSK itself.

What's the point of the N zero octets?  If other_secret is empty (which it is
for the simplest case where there's no DH, RSA, or other keying material
present) then shouldn't the:

  opaque other_secret<0..2^16-1>;

just be a 16-bit length of zero and no data octets?  Why the unnecessary zero
octets?

Peter.

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Subject: RE: [TLS] PSK question
Date: Wed, 8 Dec 2004 19:27:38 +0200
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Peter Gutmann wrote:
>=20
> I was reading through the draft again an noticed:
>=20
>   To help the client in selecting which identity to use,=20
>   the server can provide a "PSK identity hint" in the=20
>   ServerKeyExchange message (note that if no hint is=20
>   provided, a ServerKeyExchange message is still sent)
>=20
> I have no idea what an "identity hint" could possibly be (the=20
> spec says it's optional, but I'd have no idea what to put in=20
> there in any case).  If the field is used for something, what's=20
> it supposed to be used for?  If not, perhaps it should be dropped.

This depends on how you manage the keys for TLS and what you're
using TLS for (the application). But the basic idea is the same
one as in listing certificate authorities in the CertificateRequest=20
message, giving the client something more to work on in the
(probably rare) case that it has some choice.

(E.g., in the 3GPP GAA/GBA system, the hint tells the client
which "bootstrapping function" it should use to bootstrap a
shared key between the client and the server, if it hasn't
done that already.)

> (Hmm, and I'll repeat my earlier gripe that apart from its use
> to convey this unknown quantity, there's no need to send the
> server keyex at all.  In other words in 99.99% of cases it's
> just going to be code and message bloat).

I agree that in many cases it's 6 bytes of message bloat, but
I'm not so sure about the code bloat; it might even be simpler
not to have any special code paths for zero-length identity
hint. (And DH ciphersuites work this way, too; the
ClientKeyExchange message is sent even if it's empty.)

Best regards,
Pasi

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Subject: RE: [TLS] Re: PSK question
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Peter Gutmann wrote:
>=20
> Another thing I just noticed:
>=20
>   The premaster secret is formed as follows: if the PSK=20
>   is N octets long, concatenate an uint16 with the value N,=20
>   N zero octets, a second uint16 with the value N, and the=20
>   PSK itself.
>=20
> What's the point of the N zero octets?  If other_secret is=20
> empty (which it is for the simplest case where there's no DH,=20
> RSA, or other keying material present) then shouldn't the:
>=20
>   opaque other_secret<0..2^16-1>;
>=20
> just be a 16-bit length of zero and no data octets?  Why the=20
> unnecessary zero octets?

This was already discussed earlier, see e.g. issue #5 in=20
http://www.imc.org/ietf-tls/mail-archive/msg04389.html and
http://www.imc.org/ietf-tls/mail-archive/msg04098.html.=20

BR,
Pasi

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From: pgut001@cs.auckland.ac.nz (Peter Gutmann)
To: Pasi.Eronen@nokia.com, pgut001@cs.auckland.ac.nz, tls@ietf.org
Subject: RE: [TLS] PSK question
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<Pasi.Eronen@nokia.com> writes:

>(E.g., in the 3GPP GAA/GBA system, the hint tells the client
>which "bootstrapping function" it should use to bootstrap a
>shared key between the client and the server, if it hasn't
>done that already.)

Having something like this in the text for guidance would be useful.
At the moment I know of no authentication protocol that needs or uses
an "identity hint", which makes it rather hard to see the utility of 
this field.

>I agree that in many cases it's 6 bytes of message bloat, but
>I'm not so sure about the code bloat; it might even be simpler
>not to have any special code paths for zero-length identity
>hint.

I can't speak for anyone else, but in my case there's extra code needed
on both the server (to memcpy() in a fixed zero-contents message) and
client (to skip the same).

Peter.

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<Pasi.Eronen@nokia.com> writes:

>This was already discussed earlier, see e.g. issue #5 in
>http://www.imc.org/ietf-tls/mail-archive/msg04389.html and
>http://www.imc.org/ietf-tls/mail-archive/msg04098.html.

That was a somewhat different discussion, my concern there was that the MD5
half was being fed all zeroes.  Whether you feed it explicit all-zeroes
(required by the current "zeroes || PSK" construct) or implicit all-zeroes
(from "PSK" with the LHS set to zeroes) shouldn't make any difference...
hmm, OK maybe the explicit-zeroes is more explicit, there just seemed to be a 
lot of unnecessary extra messing with the input required to format the data
this way, whereas simply feeding the PSK straight to the PRF without extra
processing seemed cleaner.

Peter.

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[ This announcement was cc'ed to ietf-tls@lists.certicom.com. 
  Someone should probably change this to tls@ietf.org ]

I looked at some of the changes in draft-ietf-tls-rfc2246-bis-09.txt, 
and my comments are below.

Section 6.2.1:
> Note: Data of different TLS Record layer content types MAY be
>       interleaved. Application data is generally of higher precedence
>       for transmission than other content types and therefore handshake

I thought application data had _lower_ precedencs than other
content types. SSL 3.0, TLS 1.0 and draft 8 says so.
Is this a change to TLS in draft 9?

Section 6.3:
>   Implementation note:
>       The currently defined which requires the most material is
>       AES_256_CBC_SHA, defined in [TLSAES]. It requires 2 x 32 byte

I think "cipher spec" is missing from this sentence.

Section A.5:
>    CipherSuite      TLS_KRB5_EXPORT_WITH_DES_CBC_40_SHA  = { 0x00,0x26
>   };
>    CipherSuite      TLS_KRB5_EXPORT_WITH_RC2_CBC_40_SHA  = { 0x00,0x27
>   };
>    CipherSuite      TLS_KRB5_EXPORT_WITH_RC4_40_SHA      = { 0x00,0x28
>   };
>    CipherSuite      TLS_KRB5_EXPORT_WITH_DES_CBC_40_MD5  = { 0x00,0x29
>   };
>    CipherSuite      TLS_KRB5_EXPORT_WITH_RC2_CBC_40_MD5  = { 0x00,0x2A
>   };
>    CipherSuite      TLS_KRB5_EXPORT_WITH_RC4_40_MD5      = { 0x00,0x2B
>   };

Strange formatting here.

-- 
Jostein Tveit <josteitv@pvv.ntnu.no>

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Peter, do you have a strong objection to this proceeding as-is
or do you just find it mildly unaesthetic?

If the second, I'd like to call it soup.

-Ekr

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From: pgut001@cs.auckland.ac.nz (Peter Gutmann)
To: ekr@rtfm.com
Subject: Re: [TLS] Re: PSK question
In-Reply-To: <kjacsg34jv.fsf@romeo.rtfm.com>
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EKR <ekr@rtfm.com> writes:

>Peter, do you have a strong objection to this proceeding as-is or do you just
>find it mildly unaesthetic?

I'm not keen on the PSK kludging when it's fed into the PRF but I guess I can
live with that, however I'd really like to have the server keyex/identity hint
made optional.  Except for one extremely obscure situation (the 3GPP thing
that Pasi mentioned), implementors won't even know what they're supposed to do
with this facility, let alone be able to use it.  All it's doing is
unnecessarily complicating the code on both the client and server, and
bloating up the handshake message size.

Peter.


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Subject: RE: [TLS] Re: PSK question
Date: Wed, 15 Dec 2004 11:40:19 +0200
Message-ID: <125EA890549C8641A72F3809CB80DCCD16FE28@esebe056.ntc.nokia.com>
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pgut001@cs.auckland.ac.nz writes:
> EKR <ekr@rtfm.com> writes:
>=20
> > Peter, do you have a strong objection to this proceeding=20
> > as-is or do you just find it mildly unaesthetic?
>=20
> I'm not keen on the PSK kludging when it's fed into the PRF
> but I guess I can live with that, however I'd really like to
> have the server keyex/identity hint made optional.  Except for
> one extremely obscure situation (the 3GPP thing that Pasi
> mentioned), implementors won't even know what they're supposed
> to do with this facility, let alone be able to use it.  All
> it's doing is unnecessarily complicating the code on both the
> client and server, and bloating up the handshake message size.

It should not complicate the client code at all, since even if
the client does not use the hint for anything, it has to be able
to skip it. And the "bloat" we're talking on server side is maybe=20
one line of code.

So IMHO it really does not matter either way (reducing message
bloat by 6 bytes increases specification bloat by a couple of
sentences), so if this is keeping the draft from going forward,
let's change it.

In Section 2, change

   "To help the client in selecting which identity to use,
   the server can provide a "PSK identity hint" in the
   ServerKeyExchange message (note that if no hint is=20
   provided, a ServerKeyExchange message is still sent)."

to

   "To help the client in selecting which identity to use,
   the server can provide a "PSK identity hint" in the
   ServerKeyExchange message. If no hint is provided,=20
   the ServerKeyExchange message is omitted."

In Section 2, change

   "The elements in parenthesis are not included when PSK=20
   key exchange algorithm is used."

to

   "The elements in parenthesis are not included when PSK
   key exchange algorithm is used, and "*" indicates a
   situation-dependent message that is not always sent."

In the figure in Section 2, change=20

  "ServerKeyExchange"

to
 =20
  "ServerKeyExchange*"

In Section 3, change

   "When these ciphersuites are used, the ServerKeyExchange and
   ClientKeyExchange also include the Diffie-Hellman parameters.
   The PSK identity and identity hint fields have the same
   meaning as in the previous section."

to

   "When these ciphersuites are used, the ServerKeyExchange and
   ClientKeyExchange messages also include the Diffie-Hellman
   parameters.  The PSK identity and identity hint fields have
   the same meaning as in the previous section (note that
   the ServerKeyExchange message is always sent even if no
   PSK identity hint is provided)."

In Section 4, change

   "As in normal RSA ciphersuites, the server must send a
   Certificate message.  The format of the ServerKeyExchange
   and ClientKeyExchange messages is shown below."

to

   "As in normal RSA ciphersuites, the server must send a
   Certificate message.  The format of the ServerKeyExchange=20
   and ClientKeyExchange messages is shown below.  If no PSK=20
   identity hint is provided, the ServerKeyExchange message=20
   is omitted."

Would this be better?

Best regards,
Pasi



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Subject: Re: [TLS] Re: PSK question
References: <125EA890549C8641A72F3809CB80DCCD16FE28@esebe056.ntc.nokia.com>
From: Eric Rescorla <ekr@rtfm.com>
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<Pasi.Eronen@nokia.com> writes:

> pgut001@cs.auckland.ac.nz writes:
>> EKR <ekr@rtfm.com> writes:
>> 
>> > Peter, do you have a strong objection to this proceeding 
>> > as-is or do you just find it mildly unaesthetic?
>> 
>> I'm not keen on the PSK kludging when it's fed into the PRF
>> but I guess I can live with that, however I'd really like to
>> have the server keyex/identity hint made optional.  Except for
>> one extremely obscure situation (the 3GPP thing that Pasi
>> mentioned), implementors won't even know what they're supposed
>> to do with this facility, let alone be able to use it.  All
>> it's doing is unnecessarily complicating the code on both the
>> client and server, and bloating up the handshake message size.
>
> It should not complicate the client code at all, since even if
> the client does not use the hint for anything, it has to be able
> to skip it. And the "bloat" we're talking on server side is maybe 
> one line of code.
>
> So IMHO it really does not matter either way (reducing message
> bloat by 6 bytes increases specification bloat by a couple of
> sentences), so if this is keeping the draft from going forward,
> let's change it.
This looks good to me. Peter, can we call this soup?

-Ekr

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To: Jostein Tveit <josteitv@pvv.ntnu.no>
Subject: Re: [TLS] I-D ACTION:draft-ietf-tls-rfc2246-bis-09.txt
References: <200412082120.QAA04428@ietf.org>
	<ayhpt1imew4.fsf@bacchus.pvv.ntnu.no>
From: Eric Rescorla <ekr@rtfm.com>
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Jostein Tveit <josteitv@pvv.ntnu.no> writes:

> [ This announcement was cc'ed to ietf-tls@lists.certicom.com. 
>   Someone should probably change this to tls@ietf.org ]
>
> I looked at some of the changes in draft-ietf-tls-rfc2246-bis-09.txt, 
> and my comments are below.
>
> Section 6.2.1:
>> Note: Data of different TLS Record layer content types MAY be
>>       interleaved. Application data is generally of higher precedence
>>       for transmission than other content types and therefore handshake
>
> I thought application data had _lower_ precedencs than other
> content types. SSL 3.0, TLS 1.0 and draft 8 says so.
> Is this a change to TLS in draft 9?
It's a thinko. I propose:

Note: Data of different TLS Record layer content types MAY be
interleaved. Application data is generally of lower precedence for
transmission than other content types and therefore pending
application data may be pre-empted by handshake traffic.  However,
records MUST be delivered to the network in the same order as they are
protected by the record layer. Recipients MUST receive and process
interleaved application layer traffic during handshakes subsequent to
the first one on a connection.

> Section 6.3:
>>   Implementation note:
>>       The currently defined which requires the most material is
>>       AES_256_CBC_SHA, defined in [TLSAES]. It requires 2 x 32 byte
>
> I think "cipher spec" is missing from this sentence.
Thanks.

> Section A.5:
>>    CipherSuite      TLS_KRB5_EXPORT_WITH_DES_CBC_40_SHA  = { 0x00,0x26
>>   };
>>    CipherSuite      TLS_KRB5_EXPORT_WITH_RC2_CBC_40_SHA  = { 0x00,0x27
>>   };
>>    CipherSuite      TLS_KRB5_EXPORT_WITH_RC4_40_SHA      = { 0x00,0x28
>>   };
>>    CipherSuite      TLS_KRB5_EXPORT_WITH_DES_CBC_40_MD5  = { 0x00,0x29
>>   };
>>    CipherSuite      TLS_KRB5_EXPORT_WITH_RC2_CBC_40_MD5  = { 0x00,0x2A
>>   };
>>    CipherSuite      TLS_KRB5_EXPORT_WITH_RC4_40_MD5      = { 0x00,0x2B
>>   };
Fixed.

-Ekr

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--NextPart

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

	Title		: Pre-Shared Key Ciphersuites for Transport Layer Security (TLS)
	Author(s)	: P. Eronen, H. Tschofenig
	Filename	: draft-ietf-tls-psk-05.txt
	Pages		: 16
	Date		: 2004-12-17
	
This document specifies three sets of new ciphersuites for the
   Transport Layer Security (TLS) protocol to support authentication
   based on pre-shared keys.  These pre-shared keys are symmetric keys,
   shared in advance among the communicating parties.  The first set of
   ciphersuites uses only symmetric key operations for authentication.
   The second set uses a Diffie-Hellman exchange authenticated with a
   pre-shared key; and the third set combines public key authentication
   of the server with pre-shared key authentication of the client.

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From tls-bounces@ietf.org  Tue Dec 21 10:53:27 2004
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Subject: [TLS] Implementation Report for RFC 3749
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I have completed an implementation report for RFC 3749 (TLS Compression
Methods) in cooperation with Steve Henson, Simon Josefsson, and Nikos
Mavrogiannopoulos.  I just sent a copy off to the Secretariat.  When they
get around to it the report should be available on the IETF web site:

http://www.ietf.org/IESG/implementation.html

So, the time and implementation hurdles to move 3749 from proposed to draft
have been met.  Now all we have to do is deal with moving the normative
references to draft!

For what it's worth,
-Scott-

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The IESG has received a request from the Transport Layer Security WG to 
consider the following document:

- 'Addition of Camellia Ciphersuites to Transport Layer Security (TLS) '
   <draft-ietf-tls-camellia-06.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 2005-01-10.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-tls-camellia-06.txt


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I would like to submit following inputs as the feedback:
=20
Current draft proposes to add fragmentation, reliability and in-order
delivery to handshake layer while keeping the record layer mostly
unchanged.=20
=20
Draft seems to imply that the handshake layer sends a collection of
handshake messages (referred as flight) and then waits for the other end
to respond. The timeout causes all the messages in flight to be
retransmitted.=20
=20
Here the reliability is being tracked for a complete flight of messages
as a single unit. Given that a flight can have considerable amount of
data, the probability of packet loss seems to be high. And since each
packet loss causes complete flight to be retransmitted, this may be a
large retransmission again. And this may again suffer a packet loss. I
think tracking reliability on per flight and retransmitting complete
flight as a response to packet loss may not be an efficient approach.
=20
Also building the reliability and in-order delivery into handshake layer
makes handshake state machine complicated.
=20
=20
I think another approach worth considering may be to introduce a thin
layer inside record layer which takes care of fragmentation,
reliability, in-order delivery to record layer. In this approach:
=20
1: A Handshake message can never span multiple records. A record can
contain multiple handshake messages though.
=20
This will eliminate the need of handshake message re-assembly.
Assembling a record will always result in at least one complete
handshake message or multiple complete handshake messages.
=20
This is different from the current specification where the handshake can
span multiple records. And will need the change in length field of
record layer (it'll have to be extended to 3 bytes)
=20
=20
2: Records are fragmented and is converted to UDP datagram.=20
=20
These UDP data-grams are then transmitted reliably to other end in the
handshake phase using a very thin layer (built into record layer). It
may be analogous to a simple reliable variant of UDP (something like
TFTP). Here the reliability is tracked for each UDP datagram and only
the missing UDP datagram is re-transmitted.
=20
=20
This may provide following advantages:
=20
1: Handshake layer will be similar to the TLS. There won't be any change
needed in the implementation of state machine and the specification of
handshake messages.
=20
2: A Record can span multiple UDP packets; this will free DTLS from
strict need of having a correct path MTU.
=20
2: Record layer will consist of a thin reliability layer; this layer
will be active during handshake phase and will make sure the reliable
ordered delivery of the handshake messages. During application data
transfer the reliability part of the record layer is switched off.
(Actually this'll mostly depend on the application, and if application
wants to use reliability mode of record layer, that should be possible
too.)
=20
3: Having a thin reliability layer in Record layer will allow greater
flexibility and extensibility. Reliability layer can be replaced with
other reliability mechanism like DCCP in future for providing
reliability/congestion control.
=20
=20
Thanks,
-amit
=20
> Eric Rescorla wrote:

=20

> As we all know, TLS is popular but is only compatible with reliable
> transports. Unfortunately, a number of important protocols (SIP in
> particular) run over datagram transport and therefore cannot use TLS.
In
> response, Nagendra Modadugu and I have designed a variant of TLS which
> we call Datagram TLS (DTLS).  DTLS consists of only the minimal
changes
> required to TLS to make it operate on datagram transport.

=20

> We've submitted DTLS as an I-D which you can find at:
> draft-rescorla-dtls-01.txt

=20

> You can also read our paper on DTLS from ISOC NDSS at:
> http://crypto.stanford.edu/~nagendra/papers/dtls.pdf

=20

> We've already gotten some useful comments from PMTUD people about
> handling PMTU but any comments TSV people have on the protocol would
be
> appreciated.

=20

> Best,
 >-Ekr










=20
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tsvwg at ietf.org
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=20

=20

=20






=20
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<div class=3DSection1><pre><font size=3D2 face=3D"Courier New"><span
style=3D'font-size:10.0pt'>I would like to submit following inputs as =
the feedback:<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>Current =
draft proposes to add fragmentation, reliability and in-order delivery =
to handshake layer while keeping the record layer mostly unchanged. =
<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>Draft =
seems to imply that the handshake layer sends a collection of handshake =
messages (referred as flight) and then waits for the other end to =
respond. The timeout causes all the messages in flight to be =
retransmitted. <o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>Here the =
reliability is being tracked for a complete flight of messages as a =
single unit. Given that a flight can have considerable amount of data, =
the probability of packet loss seems to be high. And since each packet =
loss causes complete flight to be retransmitted, this may be a large =
retransmission again. And this may again suffer a packet loss. I think =
tracking reliability on per flight and retransmitting complete flight as =
a response to packet loss may not be an efficient =
approach.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>Also =
building the reliability and in-order delivery into handshake layer =
makes handshake state machine =
complicated.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>I think =
another approach worth considering may be to introduce a thin layer =
inside record layer which takes care of fragmentation, reliability, =
in-order delivery to record layer. In this =
approach:<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>1: A =
Handshake message can never span multiple records. A record can contain =
multiple handshake messages =
though.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>This will =
eliminate the need of handshake message re-assembly. Assembling a record =
will always result in at least one complete handshake message or =
multiple complete handshake =
messages.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>This is =
different from the current specification where the handshake can span =
multiple records. And will need the change in length field of record =
layer (it&#8217;ll have to be extended to 3 =
bytes)<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>2: =
Records are fragmented and is converted to UDP datagram. =
<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>These UDP =
data-grams are then transmitted reliably to other end in the handshake =
phase using a very thin layer (built into record layer). It may be =
analogous to a simple reliable variant of UDP (something like TFTP). =
Here the reliability is tracked for each UDP datagram and only the =
missing UDP datagram is =
re-transmitted.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>This may =
provide following advantages:<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>1: =
Handshake layer will be similar to the TLS. There won&#8217;t be any =
change needed in the implementation of state machine and the =
specification of handshake =
messages.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>2: A =
Record can span multiple UDP packets; this will free DTLS from strict =
need of having a correct path =
MTU.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>2: Record =
layer will consist of a thin reliability layer; this layer will be =
active during handshake phase and will make sure the reliable ordered =
delivery of the handshake messages. During application data transfer the =
reliability part of the record layer is switched off. (Actually =
this&#8217;ll mostly depend on the application, and if application wants =
to use reliability mode of record layer, that should be possible =
too.)<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>3: Having =
a thin reliability layer in Record layer will allow greater flexibility =
and extensibility. Reliability layer can be replaced with other =
reliability mechanism like DCCP in future for providing =
reliability/congestion control.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>Thanks,<o:p></o:p></span></font></pre><pre><fo=
nt
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>-amit<o:p></o:p></span></font></pre><pre><font=

size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&gt; Eric =
Rescorla wrote:<o:p></o:p></span></font></pre>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&gt; As we all know, TLS is popular but is =
only compatible with reliable<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&gt; =
transports. Unfortunately, a number of important protocols (SIP =
in<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&gt; =
particular) run over datagram transport and therefore cannot use TLS. =
In<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&gt; =
response, Nagendra Modadugu and I have designed a variant of TLS =
which<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&gt; we =
call Datagram TLS (DTLS).&nbsp; DTLS consists of only the minimal =
changes<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&gt; =
required to TLS to make it operate on datagram =
transport.<o:p></o:p></span></font></pre>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&gt; We've submitted DTLS as an I-D which you =
can find at:<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&gt; =
draft-rescorla-dtls-01.txt<o:p></o:p></span></font></pre>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&gt; You can also read our paper on DTLS from =
ISOC NDSS at:<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&gt; <a
href=3D"http://crypto.stanford.edu/~nagendra/papers/dtls.pdf">http://cryp=
to.stanford.edu/~nagendra/papers/dtls.pdf</a><o:p></o:p></span></font></p=
re>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&gt; We've already gotten some useful =
comments from PMTUD people =
about<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&gt; =
handling PMTU but any comments TSV people have on the protocol would =
be<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&gt; =
appreciated.<o:p></o:p></span></font></pre>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&gt; =
Best,<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'> =
&gt;-Ekr<o:p></o:p></span></font></pre>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'><br>
<br>
<o:p></o:p></span></font></p>

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><br>
<br>
<o:p></o:p></span></font></pre><pre><font size=3D2 face=3D"Courier =
New"><span
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>______________________________________________=
_<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>tsvwg =
mailing list<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>tsvwg at =
ietf.org<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'><a
href=3D"https://www1.ietf.org/mailman/listinfo/tsvwg">https://www1.ietf.o=
rg/mailman/listinfo/tsvwg</a><o:p></o:p></span></font></pre>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><br>
<br>
<o:p></o:p></span></font></pre><pre><font size=3D2 face=3D"Courier =
New"><span
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>______________________________________________=
_<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>tsvwg =
mailing list<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>tsvwg at =
ietf.org<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'><a
href=3D"https://www1.ietf.org/mailman/listinfo/tsvwg">https://www1.ietf.o=
rg/mailman/listinfo/tsvwg</a><o:p></o:p></span></font></pre>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><br>
<br>
</span></font><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:
Arial'><o:p></o:p></span></font></p>

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"Amit Jain" <a.jain@f5.com> writes:
> I would like to submit following inputs as the feedback:
Amit,

Thanks for your comments. Responses below.

> Current draft proposes to add fragmentation, reliability and in-order
> delivery to handshake layer while keeping the record layer mostly
> unchanged. 
That's a fair summary, yes.

> Draft seems to imply that the handshake layer sends a collection of
> handshake messages (referred as flight) and then waits for the other end
> to respond. The timeout causes all the messages in flight to be
> retransmitted. 

Yes, that's correct. We considered a cumulative ACK as in TCP
(see Section 4.4 of the DTLS paper) but we ultimately decided that
it wouldn't add that much efficiency improvement and it of
course makes life more complicated. Our initial measurements
seem to support that, but it would of course be very easy to
add ACKs later.

> Here the reliability is being tracked for a complete flight of messages
> as a single unit. Given that a flight can have considerable amount of
> data, the probability of packet loss seems to be high. And since each
> packet loss causes complete flight to be retransmitted, this may be a
> large retransmission again. And this may again suffer a packet loss.

Yes, that's correct, but it's fairly unlikely to be the SAME packet.
You keep going until you've received all the bytes at least once,
not until a given flight comes through intact.

> Also building the reliability and in-order delivery into handshake layer
> makes handshake state machine complicated.
Yes, that's correct. However, the complexity is relatively minimal,
at least in our implementation. 


> I think another approach worth considering may be to introduce a thin
> layer inside record layer which takes care of fragmentation,
> reliability, in-order delivery to record layer. In this approach:
Yes. We considered this approach.  However, we were concerned about
designing a generic reliability layer. Designing reliable transport
protocols is difficult, as evidenced by the fact that we're still
working on TCP after 20+ years. Our feeling was that as the only real
requirement was to make the handshake reliable it was best to tie the
reliability into the handshake process and thus avoid the complexity
and rathole potential of trying to design something more
general. Moreover, there is experience with this kind of state machine
from IKE and other similar protocols.

> 1: A Handshake message can never span multiple records. A record can
> contain multiple handshake messages though.
>  
> This will eliminate the need of handshake message re-assembly.
> Assembling a record will always result in at least one complete
> handshake message or multiple complete handshake messages.

Of course. But it will replace it with record reassembly, which
is basically the same problem.


> This is different from the current specification where the handshake can
> span multiple records. And will need the change in length field of
> record layer (it'll have to be extended to 3 bytes)

Just a clarification: you mean "the current TLS specification" here,
correct? TLS currently has this feature. 


> 2: Records are fragmented and is converted to UDP datagram. 
>  
> These UDP data-grams are then transmitted reliably to other end in the
> handshake phase using a very thin layer (built into record layer). It
> may be analogous to a simple reliable variant of UDP (something like
> TFTP). Here the reliability is tracked for each UDP datagram and only
> the missing UDP datagram is re-transmitted.
>
> This may provide following advantages:
>  
> 1: Handshake layer will be similar to the TLS. There won't be any change
> needed in the implementation of state machine and the specification of
> handshake messages.
That's sort of correct, except that the handshake state machine
will need to do signalling to the record layer about when it wants
reliability. It's also important to note that the DTLS handshake
reliability layer lives mostly at the level of the DTLS handshake
wrapper. The handshake payloads themselves don't change.

> 2: A Record can span multiple UDP packets; this will free DTLS from
> strict need of having a correct path MTU.
Not really. The DTLS handshake doesn't need an exact PMTU. It
just needs an approximate PMTU estimate to make sure that its own
messages get through. Typically it's the applications which need
a PMTU estimate because they actually want to send data in
a single UDP datagram. That's what their non-DTLS behavior
is, after all.

Actually, the approach you suggest makes the PTMU problem
worse in some ways. In the current design, the DTLS implementation
can mostly punt the process of PMTU discovery off to the application.
Because the HS is fairly short, it can use a safe PMTU for its
own traffic and then let the application send large messages until
it becomes clear that it can't. This lets the app use PMTU
discovery mechanisms that are better than those described in
RFC 1191. (Note that this isn't exactly what the DTLS draft
says because I just worked it out recently, but it's a fairly
minor change that doesn't affect interop).

By contrast, if DTLS is in the business of fragmenting data
records, then it has to do so efficiently, which means that
it really needs to know the PMTU. Otherwise, it's at risk
of making fragments that are way too small. I suppose you
could say you'll only fragment handshake records, but that's
kind of a layer violation inside TLS.


> 2: Record layer will consist of a thin reliability layer; this layer
> will be active during handshake phase and will make sure the reliable
> ordered delivery of the handshake messages. During application data
> transfer the reliability part of the record layer is switched off.
This is actually fairly problematic. Remember that in TLS the
handshake and application data can be interlaced during the
rehandshake process. So, it's not a simple matter of reliability
being "on" or "off". You certainly don't want to suddenly
turn applications that expect unreliable semantics into reliable
ones just because they're rehandshaking. (This is a special
problem if the reliability layer provides in-order delivery as
well, which is what's required if you want to not change the
handshake layer at all).

> (Actually this'll mostly depend on the application, and if application
> wants to use reliability mode of record layer, that should be possible
> too.)
I think this illustrates the point I made above about genericity.
The transport requirements for the handshake phase of TLS, which
is short and doesn't move a lot of data, are very different from
the requirements for a generic data transport protocol such as
is provided by the DTLS/TLS record layer. If you're going to
use your reliability layer for data, you're going to want much
better efficiency and congestion control guarantees and suddenly
you're reinventing TCP and SCTP.

> 3: Having a thin reliability layer in Record layer will allow greater
> flexibility and extensibility. Reliability layer can be replaced with
> other reliability mechanism like DCCP in future for providing
> reliability/congestion control.

Well, DCCP wouldn't do the job here since it ONLY provides congestion
control and not reliability. DTLS will of course run fine over DCCP.
It just doesn't make the problem any easier. Michael Tuexen and I
are currently talking about the interaction of DTLS and SCTP, but
I don't think that's a universal solution.
   
-Ekr


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Subject: [TLS] Re: [Tsvwg] Looking for feedback on DTLS
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Murali Bashyam <mbashyam@cisco.com> writes:
> I had a question on the congestion control aspect of the handshake
> protocol in DTLS. The draft doesn't seem to address the congestion
> control requirements of the handshake protocol such as how soon is it
> okay to send an entire flight of packets into the network without
> overloading it.
We suggest a retransmit timer on the order of 500-1000 ms.

> I think there's also a problem in retransmitting the
> entire flight when 1 packet in that flight from the original
> transmission got dropped because it starts burdening an already
> congested network. And it's clear from simulation and experience that
> it's more efficient to use TCP mechanisms such  as SACK/NewReno/Reno
> to recover from scenarios where multiple packets/single packet drop
> occurs in that flight of packets.

Of course, but we're talking extremely small amounts of data here.
It's important to remember that in your average SSL handshake the
total number of bytes exchanged is generally less than 3k and that
the largest single flight (ServerHello ... ServerHelloDone) is
typically less than 2k [1]. In most cases it will fit in 1-2 MTUs.

Obviously, one could optimize this interaction some, but if you're
in a situation where DTLS handshakes represent a significant fraction
of your network congestion, you've got some really serious problems
that marginally better retransmission response is not going to fix.

> Are we really convinced that the amount of code in terms of
> development effort, testing, risk etc to add to handshake protocol to
> bring in in-order delivery, reliability, congestion control etc is
> less than simply addressing the issues that arise as a result of doing
> split channel? It wasn't clear to me that the draft compared the
> details involved in the two approaches.

It's not addressed in the draft, however we do discuss this point in
S 2.2 of our ISOC NDSS paper[2]. The implementation cost of
adding the timeout and retransmission is relatively small and there
are a number of significant difficulties with using multiple channels,
in particular wrt. during-connection rehandshake.

-Ekr


[1] D. Boneh, H. Shacham, E. Rescorla, "Client Side Caching for TLS",
to appear in ACM TISSEC. 
http://crypto.stanford.edu/~dabo/papers/fasttrack.pdf

[2] N. Modadugu and E. Rescorla, "The Design and Implementation of
Datagram TLS", Proc ISOC NDSS 2004.
http://crypto.stanford.edu/~nagendra/papers/dtls.pdf

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Murali Bashyam <mbashyam@cisco.com> writes:

> EKR wrote:
>
>>Murali Bashyam <mbashyam@cisco.com> writes:
>>
>>>I had a question on the congestion control aspect of the handshake
>>>protocol in DTLS. The draft doesn't seem to address the congestion
>>>control requirements of the handshake protocol such as how soon is it
>>>okay to send an entire flight of packets into the network without
>>>overloading it.
>>>
>>We suggest a retransmit timer on the order of 500-1000 ms.
>>
> I was referring more to the TCP capability of slow-start, where TCP
> tries to probe the network
> and expands its flight of packets gradually as opposed to sending the
> entire flight in 1 shot.
That's true, but note that the current TCP initial window
as specified in RFC 3390 is 3 packets with a 1460-byte MSS
and 4 packets with a 512-byte MSS. This looks pretty similar
to what you would see out of a DTLS implementation initially
(though the TCP implementation would back off faster).

> Of course the point which u bring up that the
> flight typically fits in 1-2 packets is true for 1500 MSS probably but
> when MSS is 512 (the common MSS value on the Internet), this
> immediately becomes 6 packets. 

Well, since we're talking about 2k-sized flights here, it's 4 packets,
not 6 for a 512-byte MSS. 512 bytes is actually fairly conservative
in the modern network. According to CAIDA's measurements [1],
over half the TCP traffic by volume uses a 1500-byte packet size:

     The primary features of this distribution originate in the way
     common TCP implementations divide a data stream into
     packets. Approximately 85% of the traffic in these traces is TCP,
     and a large proportion of this TCP traffic is generated by bulk
     transfer applications such as HTTP and FTP. Consequently, the
     majority of the packets seen are one of three sizes: 40 byte
     packets (the minimum packet size for TCP) which carry TCP
     acknowledgments but no payload; 1500 byte packets (the maximum
     Ethernet payload size) from TCP implementations that use path MTU
     discovery; and 552 byte and 576 byte packets from TCP
     implementations that do not use path MTU discovery.


> I just want to highlight that by
> avoiding TCP for handshaking and implementing retransmissions and
> timers solves only the reliability problem but doesn't address
> congestion control including efficient retransmission possibilities
> too.
I don't agree that we're not addressing congestion control.  We
recommend an initial timer value of 500 ms with a typical doubling
exponential backoff with each retransmit. (I know this isn't exactly
what I said earlier, but I had to look it up to refresh my
memory). Given the small amounts of data involved here, I think that's
fairly congestion friendly.

As for efficient retransmission, it's true that DTLS could be smarter
(though with 1500 byte packets and medium-sized (order 1k) server cert
payloads, there's not really much room for improvement) 
but the overall improvement would be fairly modest and would introduce
substantial additional complexity.

>>Of course, but we're talking extremely small amounts of data here.
>>It's important to remember that in your average SSL handshake the
>>total number of bytes exchanged is generally less than 3k and that
>>the largest single flight (ServerHello ... ServerHelloDone) is
>>typically less than 2k [1]. In most cases it will fit in 1-2 MTUs.
>>
> True when the MSS is 1500, not when the MSS is negotiated to be
> 512. What about large certificates being sent by the server, that
> would increase the number of messages?

1k is a fairly large certificate. It's quite uncommon to see a certificate
chain larger than two certificates. That's where I got 2k from.

-Ekr

[1] http://www.caida.org/outreach/papers/2000/AIX0005/AIX0005.html#plen_hist

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Subject: Re: [TLS] Last Call: 'Addition of Camellia Ciphersuites to Transport
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(sorry for the nit-picking)
the ({0x00,0x47}-{0x00,0x4D}) range seems available. Any reason for not choosing
the *_CAMELLIA_256_* numbers contiguously after the *_CAMELLIA_128_* ones?

	Kais.

The IESG wrote:
> The IESG has received a request from the Transport Layer Security WG to 
> consider the following document:
> 
> - 'Addition of Camellia Ciphersuites to Transport Layer Security (TLS) '
>    <draft-ietf-tls-camellia-06.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 2005-01-10.
> 
> The file can be obtained via
> http://www.ietf.org/internet-drafts/draft-ietf-tls-camellia-06.txt
> 
> 
> _______________________________________________
> TLS mailing list
> TLS@lists.ietf.org
> https://www1.ietf.org/mailman/listinfo/tls


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Date: Tue, 28 Dec 2004 12:53:50 -0800
From: Murali Bashyam <mbashyam@cisco.com>
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Subject: [TLS] Re: [Tsvwg] Looking for feedback on DTLS
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Hi

I had a question on the congestion control aspect of the handshake 
protocol in DTLS. The draft doesn't seem to address the congestion 
control requirements of the handshake protocol such as how soon is it 
okay to send an entire flight of packets into the network without 
overloading it. I think there's also a problem in retransmitting the 
entire flight when 1 packet in that flight from the original 
transmission got dropped because it starts burdening an already 
congested network. And it's clear from simulation and experience that 
it's more efficient to use TCP mechanisms such  as SACK/NewReno/Reno  to 
recover from scenarios where multiple packets/single packet drop occurs 
in that flight of packets.

Are we really convinced that the amount of code in terms of development 
effort, testing, risk etc to add to handshake protocol to bring in 
in-order delivery, reliability, congestion control etc is less than 
simply addressing the issues that arise as a result of doing split 
channel? It wasn't clear to me that the draft compared the details 
involved in the two approaches.

Murali

EKR wrote:

>"Amit Jain" <a.jain@f5.com> writes:
>  
>
>>I would like to submit following inputs as the feedback:
>>    
>>
>Amit,
>
>Thanks for your comments. Responses below.
>
>  
>
>>Current draft proposes to add fragmentation, reliability and in-order
>>delivery to handshake layer while keeping the record layer mostly
>>unchanged. 
>>    
>>
>That's a fair summary, yes.
>
>  
>
>>Draft seems to imply that the handshake layer sends a collection of
>>handshake messages (referred as flight) and then waits for the other end
>>to respond. The timeout causes all the messages in flight to be
>>retransmitted. 
>>    
>>
>
>Yes, that's correct. We considered a cumulative ACK as in TCP
>(see Section 4.4 of the DTLS paper) but we ultimately decided that
>it wouldn't add that much efficiency improvement and it of
>course makes life more complicated. Our initial measurements
>seem to support that, but it would of course be very easy to
>add ACKs later.
>
>  
>
>>Here the reliability is being tracked for a complete flight of messages
>>as a single unit. Given that a flight can have considerable amount of
>>data, the probability of packet loss seems to be high. And since each
>>packet loss causes complete flight to be retransmitted, this may be a
>>large retransmission again. And this may again suffer a packet loss.
>>    
>>
>
>Yes, that's correct, but it's fairly unlikely to be the SAME packet.
>You keep going until you've received all the bytes at least once,
>not until a given flight comes through intact.
>
>  
>
>>Also building the reliability and in-order delivery into handshake layer
>>makes handshake state machine complicated.
>>    
>>
>Yes, that's correct. However, the complexity is relatively minimal,
>at least in our implementation. 
>
>
>  
>
>>I think another approach worth considering may be to introduce a thin
>>layer inside record layer which takes care of fragmentation,
>>reliability, in-order delivery to record layer. In this approach:
>>    
>>
>Yes. We considered this approach.  However, we were concerned about
>designing a generic reliability layer. Designing reliable transport
>protocols is difficult, as evidenced by the fact that we're still
>working on TCP after 20+ years. Our feeling was that as the only real
>requirement was to make the handshake reliable it was best to tie the
>reliability into the handshake process and thus avoid the complexity
>and rathole potential of trying to design something more
>general. Moreover, there is experience with this kind of state machine
>from IKE and other similar protocols.
>
>  
>
>>1: A Handshake message can never span multiple records. A record can
>>contain multiple handshake messages though.
>> 
>>This will eliminate the need of handshake message re-assembly.
>>Assembling a record will always result in at least one complete
>>handshake message or multiple complete handshake messages.
>>    
>>
>
>Of course. But it will replace it with record reassembly, which
>is basically the same problem.
>
>
>  
>
>>This is different from the current specification where the handshake can
>>span multiple records. And will need the change in length field of
>>record layer (it'll have to be extended to 3 bytes)
>>    
>>
>
>Just a clarification: you mean "the current TLS specification" here,
>correct? TLS currently has this feature. 
>
>
>  
>
>>2: Records are fragmented and is converted to UDP datagram. 
>> 
>>These UDP data-grams are then transmitted reliably to other end in the
>>handshake phase using a very thin layer (built into record layer). It
>>may be analogous to a simple reliable variant of UDP (something like
>>TFTP). Here the reliability is tracked for each UDP datagram and only
>>the missing UDP datagram is re-transmitted.
>>
>>This may provide following advantages:
>> 
>>1: Handshake layer will be similar to the TLS. There won't be any change
>>needed in the implementation of state machine and the specification of
>>handshake messages.
>>    
>>
>That's sort of correct, except that the handshake state machine
>will need to do signalling to the record layer about when it wants
>reliability. It's also important to note that the DTLS handshake
>reliability layer lives mostly at the level of the DTLS handshake
>wrapper. The handshake payloads themselves don't change.
>
>  
>
>>2: A Record can span multiple UDP packets; this will free DTLS from
>>strict need of having a correct path MTU.
>>    
>>
>Not really. The DTLS handshake doesn't need an exact PMTU. It
>just needs an approximate PMTU estimate to make sure that its own
>messages get through. Typically it's the applications which need
>a PMTU estimate because they actually want to send data in
>a single UDP datagram. That's what their non-DTLS behavior
>is, after all.
>
>Actually, the approach you suggest makes the PTMU problem
>worse in some ways. In the current design, the DTLS implementation
>can mostly punt the process of PMTU discovery off to the application.
>Because the HS is fairly short, it can use a safe PMTU for its
>own traffic and then let the application send large messages until
>it becomes clear that it can't. This lets the app use PMTU
>discovery mechanisms that are better than those described in
>RFC 1191. (Note that this isn't exactly what the DTLS draft
>says because I just worked it out recently, but it's a fairly
>minor change that doesn't affect interop).
>
>By contrast, if DTLS is in the business of fragmenting data
>records, then it has to do so efficiently, which means that
>it really needs to know the PMTU. Otherwise, it's at risk
>of making fragments that are way too small. I suppose you
>could say you'll only fragment handshake records, but that's
>kind of a layer violation inside TLS.
>
>
>  
>
>>2: Record layer will consist of a thin reliability layer; this layer
>>will be active during handshake phase and will make sure the reliable
>>ordered delivery of the handshake messages. During application data
>>transfer the reliability part of the record layer is switched off.
>>    
>>
>This is actually fairly problematic. Remember that in TLS the
>handshake and application data can be interlaced during the
>rehandshake process. So, it's not a simple matter of reliability
>being "on" or "off". You certainly don't want to suddenly
>turn applications that expect unreliable semantics into reliable
>ones just because they're rehandshaking. (This is a special
>problem if the reliability layer provides in-order delivery as
>well, which is what's required if you want to not change the
>handshake layer at all).
>
>  
>
>>(Actually this'll mostly depend on the application, and if application
>>wants to use reliability mode of record layer, that should be possible
>>too.)
>>    
>>
>I think this illustrates the point I made above about genericity.
>The transport requirements for the handshake phase of TLS, which
>is short and doesn't move a lot of data, are very different from
>the requirements for a generic data transport protocol such as
>is provided by the DTLS/TLS record layer. If you're going to
>use your reliability layer for data, you're going to want much
>better efficiency and congestion control guarantees and suddenly
>you're reinventing TCP and SCTP.
>
>  
>
>>3: Having a thin reliability layer in Record layer will allow greater
>>flexibility and extensibility. Reliability layer can be replaced with
>>other reliability mechanism like DCCP in future for providing
>>reliability/congestion control.
>>    
>>
>
>Well, DCCP wouldn't do the job here since it ONLY provides congestion
>control and not reliability. DTLS will of course run fine over DCCP.
>It just doesn't make the problem any easier. Michael Tuexen and I
>are currently talking about the interaction of DTLS and SCTP, but
>I don't think that's a universal solution.
>   
>-Ekr
>
>
>_______________________________________________
>tsvwg mailing list
>tsvwg@ietf.org
>https://www1.ietf.org/mailman/listinfo/tsvwg
>  
>


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Hi<br>
<br>
I had a question on the congestion control aspect of the handshake protocol
in DTLS. The draft doesn't seem to address the congestion control requirements
of the handshake protocol such as how soon is it okay to send an entire flight
of packets into the network without overloading it. I think there's also
a problem in retransmitting the entire flight when 1 packet in that flight
from the original transmission got dropped because it starts burdening an
already congested network. And it's clear from simulation and experience
that it's more efficient to use TCP mechanisms such &nbsp;as SACK/NewReno/Reno
&nbsp;to recover from scenarios where multiple packets/single packet drop occurs
in that flight of packets.<br>
<br>
Are we really convinced that the amount of code in terms of development effort,
testing, risk etc to add to handshake protocol to bring in in-order delivery,
reliability, congestion control etc is less than simply addressing the issues
that arise as a result of doing split channel? It wasn't clear to me that
the draft compared the details involved in the two approaches.<br>
<br>
Murali <br>
<br>
EKR wrote:<br>
<blockquote type="cite" cite="mid86d5wugwin.fsf@romeo.rtfm.com">
  <pre wrap="">"Amit Jain" <a class="moz-txt-link-rfc2396E" href="mailto:a.jain@f5.com">&lt;a.jain@f5.com&gt;</a> writes:
  </pre>
  <blockquote type="cite">
    <pre wrap="">I would like to submit following inputs as the feedback:
    </pre>
  </blockquote>
  <pre wrap=""><!---->Amit,

Thanks for your comments. Responses below.

  </pre>
  <blockquote type="cite">
    <pre wrap="">Current draft proposes to add fragmentation, reliability and in-order
delivery to handshake layer while keeping the record layer mostly
unchanged. 
    </pre>
  </blockquote>
  <pre wrap=""><!---->That's a fair summary, yes.

  </pre>
  <blockquote type="cite">
    <pre wrap="">Draft seems to imply that the handshake layer sends a collection of
handshake messages (referred as flight) and then waits for the other end
to respond. The timeout causes all the messages in flight to be
retransmitted. 
    </pre>
  </blockquote>
  <pre wrap=""><!---->
Yes, that's correct. We considered a cumulative ACK as in TCP
(see Section 4.4 of the DTLS paper) but we ultimately decided that
it wouldn't add that much efficiency improvement and it of
course makes life more complicated. Our initial measurements
seem to support that, but it would of course be very easy to
add ACKs later.

  </pre>
  <blockquote type="cite">
    <pre wrap="">Here the reliability is being tracked for a complete flight of messages
as a single unit. Given that a flight can have considerable amount of
data, the probability of packet loss seems to be high. And since each
packet loss causes complete flight to be retransmitted, this may be a
large retransmission again. And this may again suffer a packet loss.
    </pre>
  </blockquote>
  <pre wrap=""><!---->
Yes, that's correct, but it's fairly unlikely to be the SAME packet.
You keep going until you've received all the bytes at least once,
not until a given flight comes through intact.

  </pre>
  <blockquote type="cite">
    <pre wrap="">Also building the reliability and in-order delivery into handshake layer
makes handshake state machine complicated.
    </pre>
  </blockquote>
  <pre wrap=""><!---->Yes, that's correct. However, the complexity is relatively minimal,
at least in our implementation. 


  </pre>
  <blockquote type="cite">
    <pre wrap="">I think another approach worth considering may be to introduce a thin
layer inside record layer which takes care of fragmentation,
reliability, in-order delivery to record layer. In this approach:
    </pre>
  </blockquote>
  <pre wrap=""><!---->Yes. We considered this approach.  However, we were concerned about
designing a generic reliability layer. Designing reliable transport
protocols is difficult, as evidenced by the fact that we're still
working on TCP after 20+ years. Our feeling was that as the only real
requirement was to make the handshake reliable it was best to tie the
reliability into the handshake process and thus avoid the complexity
and rathole potential of trying to design something more
general. Moreover, there is experience with this kind of state machine
from IKE and other similar protocols.

  </pre>
  <blockquote type="cite">
    <pre wrap="">1: A Handshake message can never span multiple records. A record can
contain multiple handshake messages though.
 
This will eliminate the need of handshake message re-assembly.
Assembling a record will always result in at least one complete
handshake message or multiple complete handshake messages.
    </pre>
  </blockquote>
  <pre wrap=""><!---->
Of course. But it will replace it with record reassembly, which
is basically the same problem.


  </pre>
  <blockquote type="cite">
    <pre wrap="">This is different from the current specification where the handshake can
span multiple records. And will need the change in length field of
record layer (it'll have to be extended to 3 bytes)
    </pre>
  </blockquote>
  <pre wrap=""><!---->
Just a clarification: you mean "the current TLS specification" here,
correct? TLS currently has this feature. 


  </pre>
  <blockquote type="cite">
    <pre wrap="">2: Records are fragmented and is converted to UDP datagram. 
 
These UDP data-grams are then transmitted reliably to other end in the
handshake phase using a very thin layer (built into record layer). It
may be analogous to a simple reliable variant of UDP (something like
TFTP). Here the reliability is tracked for each UDP datagram and only
the missing UDP datagram is re-transmitted.

This may provide following advantages:
 
1: Handshake layer will be similar to the TLS. There won't be any change
needed in the implementation of state machine and the specification of
handshake messages.
    </pre>
  </blockquote>
  <pre wrap=""><!---->That's sort of correct, except that the handshake state machine
will need to do signalling to the record layer about when it wants
reliability. It's also important to note that the DTLS handshake
reliability layer lives mostly at the level of the DTLS handshake
wrapper. The handshake payloads themselves don't change.

  </pre>
  <blockquote type="cite">
    <pre wrap="">2: A Record can span multiple UDP packets; this will free DTLS from
strict need of having a correct path MTU.
    </pre>
  </blockquote>
  <pre wrap=""><!---->Not really. The DTLS handshake doesn't need an exact PMTU. It
just needs an approximate PMTU estimate to make sure that its own
messages get through. Typically it's the applications which need
a PMTU estimate because they actually want to send data in
a single UDP datagram. That's what their non-DTLS behavior
is, after all.

Actually, the approach you suggest makes the PTMU problem
worse in some ways. In the current design, the DTLS implementation
can mostly punt the process of PMTU discovery off to the application.
Because the HS is fairly short, it can use a safe PMTU for its
own traffic and then let the application send large messages until
it becomes clear that it can't. This lets the app use PMTU
discovery mechanisms that are better than those described in
RFC 1191. (Note that this isn't exactly what the DTLS draft
says because I just worked it out recently, but it's a fairly
minor change that doesn't affect interop).

By contrast, if DTLS is in the business of fragmenting data
records, then it has to do so efficiently, which means that
it really needs to know the PMTU. Otherwise, it's at risk
of making fragments that are way too small. I suppose you
could say you'll only fragment handshake records, but that's
kind of a layer violation inside TLS.


  </pre>
  <blockquote type="cite">
    <pre wrap="">2: Record layer will consist of a thin reliability layer; this layer
will be active during handshake phase and will make sure the reliable
ordered delivery of the handshake messages. During application data
transfer the reliability part of the record layer is switched off.
    </pre>
  </blockquote>
  <pre wrap=""><!---->This is actually fairly problematic. Remember that in TLS the
handshake and application data can be interlaced during the
rehandshake process. So, it's not a simple matter of reliability
being "on" or "off". You certainly don't want to suddenly
turn applications that expect unreliable semantics into reliable
ones just because they're rehandshaking. (This is a special
problem if the reliability layer provides in-order delivery as
well, which is what's required if you want to not change the
handshake layer at all).

  </pre>
  <blockquote type="cite">
    <pre wrap="">(Actually this'll mostly depend on the application, and if application
wants to use reliability mode of record layer, that should be possible
too.)
    </pre>
  </blockquote>
  <pre wrap=""><!---->I think this illustrates the point I made above about genericity.
The transport requirements for the handshake phase of TLS, which
is short and doesn't move a lot of data, are very different from
the requirements for a generic data transport protocol such as
is provided by the DTLS/TLS record layer. If you're going to
use your reliability layer for data, you're going to want much
better efficiency and congestion control guarantees and suddenly
you're reinventing TCP and SCTP.

  </pre>
  <blockquote type="cite">
    <pre wrap="">3: Having a thin reliability layer in Record layer will allow greater
flexibility and extensibility. Reliability layer can be replaced with
other reliability mechanism like DCCP in future for providing
reliability/congestion control.
    </pre>
  </blockquote>
  <pre wrap=""><!---->
Well, DCCP wouldn't do the job here since it ONLY provides congestion
control and not reliability. DTLS will of course run fine over DCCP.
It just doesn't make the problem any easier. Michael Tuexen and I
are currently talking about the interaction of DTLS and SCTP, but
I don't think that's a universal solution.
   
-Ekr


_______________________________________________
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EKR wrote:

>Murali Bashyam <mbashyam@cisco.com> writes:
>  
>
>>I had a question on the congestion control aspect of the handshake
>>protocol in DTLS. The draft doesn't seem to address the congestion
>>control requirements of the handshake protocol such as how soon is it
>>okay to send an entire flight of packets into the network without
>>overloading it.
>>    
>>
>We suggest a retransmit timer on the order of 500-1000 ms.
>
I was referring more to the TCP capability of slow-start, where TCP 
tries to probe the network
and expands its flight of packets gradually as opposed to sending the 
entire flight in 1 shot. Of course the point which u bring up that the 
flight typically fits in 1-2 packets is true for 1500 MSS probably but 
when MSS is 512 (the common MSS value on the Internet), this immediately 
becomes 6 packets.  I just want to highlight that by avoiding TCP for 
handshaking and implementing retransmissions and timers solves only the 
reliability problem but doesn't address congestion control including 
efficient retransmission possibilities too.

>
>  
>
>>I think there's also a problem in retransmitting the
>>entire flight when 1 packet in that flight from the original
>>transmission got dropped because it starts burdening an already
>>congested network. And it's clear from simulation and experience that
>>it's more efficient to use TCP mechanisms such  as SACK/NewReno/Reno
>>to recover from scenarios where multiple packets/single packet drop
>>occurs in that flight of packets.
>>    
>>
>
>Of course, but we're talking extremely small amounts of data here.
>It's important to remember that in your average SSL handshake the
>total number of bytes exchanged is generally less than 3k and that
>the largest single flight (ServerHello ... ServerHelloDone) is
>typically less than 2k [1]. In most cases it will fit in 1-2 MTUs.
>
True when the MSS is 1500, not when the MSS is negotiated to be 512. 
What about large certificates being sent by the server, that would 
increase the number of messages?

Murali

>
>Obviously, one could optimize this interaction some, but if you're
>in a situation where DTLS handshakes represent a significant fraction
>of your network congestion, you've got some really serious problems
>that marginally better retransmission response is not going to fix.
>
>  
>
>>Are we really convinced that the amount of code in terms of
>>development effort, testing, risk etc to add to handshake protocol to
>>bring in in-order delivery, reliability, congestion control etc is
>>less than simply addressing the issues that arise as a result of doing
>>split channel? It wasn't clear to me that the draft compared the
>>details involved in the two approaches.
>>    
>>
>
>It's not addressed in the draft, however we do discuss this point in
>S 2.2 of our ISOC NDSS paper[2]. The implementation cost of
>adding the timeout and retransmission is relatively small and there
>are a number of significant difficulties with using multiple channels,
>in particular wrt. during-connection rehandshake.
>
>-Ekr
>
>
>[1] D. Boneh, H. Shacham, E. Rescorla, "Client Side Caching for TLS",
>to appear in ACM TISSEC. 
>http://crypto.stanford.edu/~dabo/papers/fasttrack.pdf
>
>[2] N. Modadugu and E. Rescorla, "The Design and Implementation of
>Datagram TLS", Proc ISOC NDSS 2004.
>http://crypto.stanford.edu/~nagendra/papers/dtls.pdf
>  
>


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<br>
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EKR wrote:<br>
<blockquote type="cite" cite="mid86k6r2f488.fsf@romeo.rtfm.com">
  <pre wrap="">Murali Bashyam <a class="moz-txt-link-rfc2396E" href="mailto:mbashyam@cisco.com">&lt;mbashyam@cisco.com&gt;</a> writes:
  </pre>
  <blockquote type="cite">
    <pre wrap="">I had a question on the congestion control aspect of the handshake
protocol in DTLS. The draft doesn't seem to address the congestion
control requirements of the handshake protocol such as how soon is it
okay to send an entire flight of packets into the network without
overloading it.
    </pre>
  </blockquote>
  <pre wrap=""><!---->We suggest a retransmit timer on the order of 500-1000 ms.</pre>
</blockquote>
I was referring more to the TCP capability of slow-start, where TCP tries
to probe the network<br>
and expands its flight of packets gradually as opposed to sending the entire
flight in 1 shot. Of course the point which u bring up that the flight typically
fits in 1-2 packets is true for 1500 MSS probably but when MSS is 512 (the
common MSS value on the Internet), this immediately becomes 6 packets. &nbsp;I
just want to highlight that by avoiding TCP for handshaking and implementing
retransmissions and timers solves only the reliability problem but doesn't
address congestion control including efficient retransmission possibilities
too.<br>
<blockquote type="cite" cite="mid86k6r2f488.fsf@romeo.rtfm.com">
  <pre wrap="">

  </pre>
  <blockquote type="cite">
    <pre wrap="">I think there's also a problem in retransmitting the
entire flight when 1 packet in that flight from the original
transmission got dropped because it starts burdening an already
congested network. And it's clear from simulation and experience that
it's more efficient to use TCP mechanisms such  as SACK/NewReno/Reno
to recover from scenarios where multiple packets/single packet drop
occurs in that flight of packets.
    </pre>
  </blockquote>
  <pre wrap=""><!---->
Of course, but we're talking extremely small amounts of data here.
It's important to remember that in your average SSL handshake the
total number of bytes exchanged is generally less than 3k and that
the largest single flight (ServerHello ... ServerHelloDone) is
typically less than 2k [1]. In most cases it will fit in 1-2 MTUs.</pre>
</blockquote>
True when the MSS is 1500, not when the MSS is negotiated to be 512. What
about large certificates being sent by the server, that would increase the
number of messages?<br>
<br>
Murali<br>
<blockquote type="cite" cite="mid86k6r2f488.fsf@romeo.rtfm.com">
  <pre wrap="">

Obviously, one could optimize this interaction some, but if you're
in a situation where DTLS handshakes represent a significant fraction
of your network congestion, you've got some really serious problems
that marginally better retransmission response is not going to fix.

  </pre>
  <blockquote type="cite">
    <pre wrap="">Are we really convinced that the amount of code in terms of
development effort, testing, risk etc to add to handshake protocol to
bring in in-order delivery, reliability, congestion control etc is
less than simply addressing the issues that arise as a result of doing
split channel? It wasn't clear to me that the draft compared the
details involved in the two approaches.
    </pre>
  </blockquote>
  <pre wrap=""><!---->
It's not addressed in the draft, however we do discuss this point in
S 2.2 of our ISOC NDSS paper[2]. The implementation cost of
adding the timeout and retransmission is relatively small and there
are a number of significant difficulties with using multiple channels,
in particular wrt. during-connection rehandshake.

-Ekr


[1] D. Boneh, H. Shacham, E. Rescorla, "Client Side Caching for TLS",
to appear in ACM TISSEC. 
<a class="moz-txt-link-freetext" href="http://crypto.stanford.edu/~dabo/papers/fasttrack.pdf">http://crypto.stanford.edu/~dabo/papers/fasttrack.pdf</a>

[2] N. Modadugu and E. Rescorla, "The Design and Implementation of
Datagram TLS", Proc ISOC NDSS 2004.
<a class="moz-txt-link-freetext" href="http://crypto.stanford.edu/~nagendra/papers/dtls.pdf">http://crypto.stanford.edu/~nagendra/papers/dtls.pdf</a>
  </pre>
</blockquote>
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Hi folks,

We just posted an internet-draft updating the extensions RFC to deal =
with
the issue of which documents can define new extensions - the current RFC
allows only standards track RFCs while this draft allows any RFC based =
on
IETF consensus (which basically means IESG review).

I've summarized the detailed list of changes below.

Eric/Win ... How do you want to handle this? Given the minor nature of =
the
updates, can we proceed with working group last call on this (and the =
ECC in
TLS draft)?

Best regards. Simon=20

Here's a summary of the changes:

- Assignment of ExtensionType values handed over to IANA. New values can =
be
assigned only via the IETF Consensus process. (This means that an RFC is
required and the RFC must have undergone IESG review.) This affects =
Section
2.3, Section 5, and Section 8.

- The text concerning handling of extensions during session resumption
requests has been clarified since the previous text was ambiguous, and =
it
would be bad to have to update the RFC again to deal with any RFCs that =
come
along and need clarity on this. This affects the last paragraph of =
Section
2.3 and the third paragraph of Section 3.

There are also a few minor editorial changes - updating references, =
author
addresses and the like.

> -----Original Message-----
> From: tls-bounces@lists.ietf.org=20
> [mailto:tls-bounces@lists.ietf.org] On Behalf Of=20
> Internet-Drafts@ietf.org
> Sent: Wednesday, December 29, 2004 3:43 PM
> To: i-d-announce@ietf.org
> Cc: ietf-tls@lists.certicom.com
> Subject: [TLS] I-D ACTION:draft-ietf-tls-rfc3546bis-00.txt
>=20
>=20
> A New Internet-Draft is available from the on-line=20
> Internet-Drafts directories. This draft is a work item of the=20
> Transport Layer Security Working Group of the IETF.
>=20
> 	Title		: Transport Layer Security (TLS) Extensions
> 	Author(s)	: S. Blake-Wilson, et al.
> 	Filename	: draft-ietf-tls-rfc3546bis-00.txt
> 	Pages		: 29
> 	Date		: 2004-12-29
> =09
> This document describes extensions that may be used to add
>    functionality to Transport Layer Security (TLS).  It provides both
>    generic extension mechanisms for the TLS handshake client=20
> and server
>    hellos, and specific extensions using these generic mechanisms.
>=20
> A URL for this Internet-Draft is:=20
> http://www.ietf.org/internet-drafts/draft-ietf-tls-rfc3546bis-00.txt
>=20
> To remove yourself from the I-D Announcement list, send a message to=20
> i-d-announce-request@ietf.org with the word unsubscribe in=20
> the body of the message. =20
> You can also visit=20
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> to change your subscription settings.
>=20
>=20
> Internet-Drafts are also available by anonymous FTP. Login=20
> with the username "anonymous" and a password of your e-mail=20
> address. After logging in, type "cd internet-drafts" and then
> 	"get draft-ietf-tls-rfc3546bis-00.txt".
>=20
> A list of Internet-Drafts directories can be found in=20
> http://www.ietf.org/shadow.html=20
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>=20
>=20
> Internet-Drafts can also be obtained by e-mail.
>=20
> Send a message to:
> 	mailserv@ietf.org.
> In the body type:
> 	"FILE /internet-drafts/draft-ietf-tls-rfc3546bis-00.txt".
> =09
> NOTE:	The mail server at ietf.org can return the document in
> 	MIME-encoded form by using the "mpack" utility.  To use this
> 	feature, insert the command "ENCODING mime" before the "FILE"
> 	command.  To decode the response(s), you will need "munpack" or
> 	a MIME-compliant mail reader.  Different MIME-compliant=20
> mail readers
> 	exhibit different behavior, especially when dealing with
> 	"multipart" MIME messages (i.e. documents which have been split
> 	up into multiple messages), so check your local documentation on
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> 	=09
> 	=09
> Below is the data which will enable a MIME compliant mail=20
> reader implementation to automatically retrieve the ASCII=20
> version of the Internet-Draft.
>=20



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From tls-bounces@ietf.org  Thu Dec 30 11:02:44 2004
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"Simon Blake-Wilson" <sblakewilson@bcisse.com> writes:
> We just posted an internet-draft updating the extensions RFC to deal with
> the issue of which documents can define new extensions - the current RFC
> allows only standards track RFCs while this draft allows any RFC based on
> IETF consensus (which basically means IESG review).
>
> I've summarized the detailed list of changes below.
>
> Eric/Win ... How do you want to handle this? Given the minor nature of the
> updates, can we proceed with working group last call on this (and the ECC in
> TLS draft)?
Given the holidays, etc. let's see what people have to say over the next
few weeks. If there are no substantive comments I'll initiate WGLC.

-Ekr

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From: Bernard Aboba <aboba@internaut.com>
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Subject: [TLS] Re: PSK question
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I am also concerned about the interoperability of the
client/server identity exchange specified in draft-ietf-tls-psk-05.txt.
In reading the posts by Peter Gutmann, I find the arguments persuasive:

http://www.imc.org/ietf-tls/mail-archive/msg04402.html
http://www.imc.org/ietf-tls/mail-archive/msg04418.html

I'd suggest that this issue needs to be addressed before this
specification can progress.

The following potential problems come to mind:

a. Generality.  On reading the discussion,  it would appear that this
facility was designed for use within 3GPP, where the "hint" refers to
the "bootstrapping" function used to bootstrap the shared key between
client and server.  However, I believe that IETF specifications should
also be able to interoperate widely so that they can be used on the
Internet.  I'm not clear that the current specification will interoperate.
For example, if the spec is going to support multiple ID formats, then
support for various formats should be negotiated.

b. Compatibility with existing specifications. One of the major uses of
this specification is in WLAN authentication.  However, mandating at least
one form for the client identifier, it is not clear how wecan guarantee
conformance to RFC 2486, which standardizes the format of the Network
Access Identifier (NAI).

c. Difficulties in operation with AAA.  Without mandating at least one
form for the server identifier, it is not clear to me how a client can
unambiguously identify what server it is talking to.  It is
quite possible that AAA servers will not be synchronized with respect to
their session caches, or even the pre-shared keys to be utilized (e.g.
LDAP directory synchronization can lag behind), so that this could be a
problem.

d. IPR disclosure. IPR has been filed relating to "hint" mechanisms:
http://www.ietf.org/ietf/IPR/telecom-italia-ipr-wlan-access.txt
I am therefore concerned that making "hints" an integral part of this
specification will create problems with encumbrance.  This problem can be
avoided by clearly mandating support for certain client and serverid
formats.



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Subject: RE: [TLS] Re: PSK question
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Bernard Aboba wrote:
>=20
> I am also concerned about the interoperability of the
> client/server identity exchange specified in=20
> draft-ietf-tls-psk-05.txt.

Thanks for your comments, Bernard!

> In reading the posts by Peter Gutmann, I find the arguments=20
> persuasive:
>=20
> http://www.imc.org/ietf-tls/mail-archive/msg04402.html
> http://www.imc.org/ietf-tls/mail-archive/msg04418.html
>=20
> I'd suggest that this issue needs to be addressed before this
> specification can progress.
>=20
> The following potential problems come to mind:
>=20
> a. Generality.  On reading the discussion, it would appear
> that this facility was designed for use within 3GPP, where the
> "hint" refers to the "bootstrapping" function used to
> bootstrap the shared key between client and server.  However,
> I believe that IETF specifications should also be able to
> interoperate widely so that they can be used on the Internet.
> I'm not clear that the current specification will
> interoperate.  For example, if the spec is going to support
> multiple ID formats, then support for various formats should
> be negotiated.

TLS-PSK alone does not do anything useful, and cannot be used by
itself: it's always a component in a larger system. It is these
systems that either interoperate or don't. So if e.g. 3GPP runs
protocol FOO over TLS (using PSK ciphersuites), there's no need
(or possibility) to interoperate with someone running a
different protocol BAR over TLS.=20

The approach so far has been to put only the common parts in=20
TLS specs (RFC 2246, 2712, etc.) and application-specific parts=20
elsewhere. For some uses of TLS, we have separate documents=20
(like RFC 2595 for IMAP/POP-over-TLS, RFC 2818 for HTTP-over-TLS);
others specify TLS details together with the rest of the protocol=20
(e.g. RFC 2716 for EAP-TLS, RFC 3788 for SIGTRAN). These TLS=20
details include, among other things, what ciphersuites and what=20
kind of identifiers (e.g. dNSName SubjectAltName) should be be=20
used (and we don't have negotiation for these either).

I think this approach has worked so far with other TLS=20
ciphersuites, and I don't see any special reason why it would=20
fail especially with PSK ciphersuites...?

> b. Compatibility with existing specifications. One of the
> major uses of this specification is in WLAN authentication.
> However, mandating at least one form for the client
> identifier, it is not clear how we can guarantee conformance to
> RFC 2486, which standardizes the format of the Network Access
> Identifier (NAI).

Agreed; I just think the right place to mandate this is in the
"upper layer protocol" using TLS (in this case, EAP-TLS),
not the TLS specs.

> c. Difficulties in operation with AAA.  Without mandating at
> least one form for the server identifier, it is not clear to
> me how a client can unambiguously identify what server it is
> talking to.  It is quite possible that AAA servers will not be
> synchronized with respect to their session caches, or even the
> pre-shared keys to be utilized (e.g.  LDAP directory
> synchronization can lag behind), so that this could be a
> problem.

This problem also seems related to one specific application
(EAP-TLS) that uses TLS, not TLS itself.

> d. IPR disclosure. IPR has been filed relating to "hint"
> mechanisms:
> http://www.ietf.org/ietf/IPR/telecom-italia-ipr-wlan-access.txt
> I am therefore concerned that making "hints" an integral part
> of this specification will create problems with encumbrance.
> This problem can be avoided by clearly mandating support for
> certain client and serverid formats.

The IPR disclosure is related to network discovery and
selection, and does not mention the word "hint" anywhere.

Best regards & Happy New Year,
Pasi

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