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From: Takeshi Yoshino <tyoshino@google.com>
Date: Thu, 7 Jul 2011 21:35:43 +0900
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Subject: Re: [hybi] Payload only compression extension, again
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Here's new draft.
http://tools.ietf.org/html/draft-tyoshino-hybi-websocket-perframe-deflate-01
http://tools.ietf.org/tools/rfcdiff/rfcdiff.pyht?url1=http://tools.ietf.org/id/draft-tyoshino-hybi-websocket-perframe-deflate-00.txt&url2=http://tools.ietf.org/id/draft-tyoshino-hybi-websocket-perframe-deflate-01.txt

I've applied the change below, added some examples and changed requirement
on negotiation. Extension parameters (e.g. LZ77 window size negotiation) can
be added in future versions. Old server implementation just ignores received
extension parameters and reply with only extension token to ask the client
to fall back to old version.

Thanks,
Takeshi


On Tue, May 10, 2011 at 16:30, Takeshi Yoshino <tyoshino@google.com> wrote:

> I can change my proposal to accept last blocks (blocks with BFINAL=1) to
> allow use of java.util.zip. In my proposal, receiver expects 3 header bits
> 0b000 (and padding to byte boundary) on the tail of the application data
> portion. Not to complicate receiver logic much, I'd make change like this
> - sender: send(<compressed data flushed with Z_FINISH> + "\x00")
> - receiver: inflate(received_data + "\x00\x00\xff\xff"). every time
> inflater encounters end of stream (the last octet of a last block), renew
> inflater and keep decompressing the remainder.
> or
> - sender: send(<compressed data flushed with Z_FINISH>)
> - receiver: inflate(received_data). every time inflater encounters end of
> stream, renew inflater and keep decompressing the remainder. if the end of
> received_data is not end of stream, inflate("\x00\x00\xff\xff").
>
> Either algorithm works for Z_SYNC_FLUSH-ed and "\x00\x00\xff\xff" stripped
> frame, too.
>
> Takeshi
>
>
>
> On Tue, May 3, 2011 at 07:52, Greg Wilkins <gregw@intalio.com> wrote:
>
>> On 3 May 2011 08:02, John Tamplin <jat@google.com> wrote:
>> > My point was I see little value in having a compression extension which
>> > starts the compression state afresh with every frame, as on real-world
>> > applications this gives poor performance.
>>
>> +1
>>
>> My extension is more about prototyping plugable extension that work
>> well in a chain of extension, plus handling the negotiation and per
>> frame issues.    I'll follow the lead of the others/experience on the
>> exact algorithms to use (even though I think that is going to cause me
>> endless headaches with mismatched licenses and getting IP audits done
>> at eclipse :(
>>
>> cheers
>> _______________________________________________
>> hybi mailing list
>> hybi@ietf.org
>> https://www.ietf.org/mailman/listinfo/hybi
>>
>
>

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<div>Here&#39;s new draft.</div><div><a href=3D"http://tools.ietf.org/html/=
draft-tyoshino-hybi-websocket-perframe-deflate-01">http://tools.ietf.org/ht=
ml/draft-tyoshino-hybi-websocket-perframe-deflate-01</a></div><div><a href=
=3D"http://tools.ietf.org/tools/rfcdiff/rfcdiff.pyht?url1=3Dhttp://tools.ie=
tf.org/id/draft-tyoshino-hybi-websocket-perframe-deflate-00.txt&amp;url2=3D=
http://tools.ietf.org/id/draft-tyoshino-hybi-websocket-perframe-deflate-01.=
txt">http://tools.ietf.org/tools/rfcdiff/rfcdiff.pyht?url1=3Dhttp://tools.i=
etf.org/id/draft-tyoshino-hybi-websocket-perframe-deflate-00.txt&amp;url2=
=3Dhttp://tools.ietf.org/id/draft-tyoshino-hybi-websocket-perframe-deflate-=
01.txt</a></div>

<div><div><br></div><div><div>I&#39;ve applied the change below, added some=
 examples and changed requirement on negotiation. Extension parameters (e.g=
. LZ77 window size negotiation) can be added in future versions. Old server=
 implementation just ignores received extension parameters and reply with o=
nly extension token to ask the client to fall back to old version.</div>

</div><div><br></div><div>Thanks,</div>Takeshi<br>
<br><br><div class=3D"gmail_quote">On Tue, May 10, 2011 at 16:30, Takeshi Y=
oshino <span dir=3D"ltr">&lt;<a href=3D"mailto:tyoshino@google.com">tyoshin=
o@google.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

I can change my proposal to accept last blocks (blocks with=A0BFINAL=3D1) t=
o allow use of java.util.zip. In my proposal, receiver expects 3 header bit=
s 0b000 (and padding to byte boundary) on the tail of the application data =
portion.=A0Not to complicate receiver logic much, I&#39;d make change like =
this<div>


- sender: send(&lt;compressed data flushed with Z_FINISH&gt; + &quot;\x00&q=
uot;)</div><div>- receiver: inflate(received_data + &quot;\x00\x00\xff\xff&=
quot;). every time inflater encounters end of stream (the last octet of a l=
ast block), renew inflater and keep decompressing the remainder.<br>


<div><div><div>or</div><div>- sender: send(&lt;compressed data flushed with=
 Z_FINISH&gt;)</div><div>- receiver: inflate(received_data).=A0every time i=
nflater encounters end of stream, renew inflater and keep decompressing the=
 remainder. if the end of received_data is not end of stream, inflate(&quot=
;\x00\x00\xff\xff&quot;).</div>


<div><br></div><div>Either algorithm works for Z_SYNC_FLUSH-ed and &quot;\x=
00\x00\xff\xff&quot; stripped frame, too.</div><div><br></div><div><font co=
lor=3D"#888888">Takeshi</font><div><div></div><div class=3D"h5"><br>
<br><br><div class=3D"gmail_quote">On Tue, May 3, 2011 at 07:52, Greg Wilki=
ns <span dir=3D"ltr">&lt;<a href=3D"mailto:gregw@intalio.com" target=3D"_bl=
ank">gregw@intalio.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex">


<div>On 3 May 2011 08:02, John Tamplin &lt;<a href=3D"mailto:jat@google.com=
" target=3D"_blank">jat@google.com</a>&gt; wrote:<br>
&gt; My point was I see little value in having a compression extension whic=
h<br>
&gt; starts the compression state afresh with every frame, as on real-world=
<br>
&gt; applications this gives poor performance.<br>
<br>
</div>+1<br>
<br>
My extension is more about prototyping plugable extension that work<br>
well in a chain of extension, plus handling the negotiation and per<br>
frame issues. =A0 =A0I&#39;ll follow the lead of the others/experience on t=
he<br>
exact algorithms to use (even though I think that is going to cause me<br>
endless headaches with mismatched licenses and getting IP audits done<br>
at eclipse :(<br>
<div><div></div><div><br>
cheers<br>
_______________________________________________<br>
hybi mailing list<br>
<a href=3D"mailto:hybi@ietf.org" target=3D"_blank">hybi@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/hybi</a><br>
</div></div></blockquote></div><br></div></div></div></div></div></div>
</blockquote></div><br></div>

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From ibc@aliax.net  Thu Jul  7 05:41:39 2011
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Date: Thu, 7 Jul 2011 14:41:38 +0200
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
To: Takeshi Yoshino <tyoshino@google.com>
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Subject: Re: [hybi] Payload only compression extension, again
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2011/7/7 Takeshi Yoshino <tyoshino@google.com>:
> Old server implementation just ignores received extension parameters and
> reply with only extension token to ask the client to fall back to old
> version.

Hi, WS is still a draft, IMHO the specs should not include backawards
compatibility mechanisms with other revisions of the spec (while still
a draft). Maybe I miss something?

Regards.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: John Tamplin <jat@google.com>
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On Thu, Jul 7, 2011 at 8:41 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wro=
te:

> Hi, WS is still a draft, IMHO the specs should not include backawards
> compatibility mechanisms with other revisions of the spec (while still
> a draft). Maybe I miss something?
>

The problem is that there are deployed implementations, and breaking them
would be bad for adoption of the spec.  I agree, you don't want to clutter
up the spec with too much support for old versions, but in this case there
is very little cost.

In an ideal world, the spec would be finished before it was deployed, but
that also means not getting implementation experience back into the spec, s=
o
there are tradeoffs both ways.

--=20
John A. Tamplin
Software Engineer (GWT), Google

--000e0cd51b405c730c04a77b61b5
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<div class=3D"gmail_quote">On Thu, Jul 7, 2011 at 8:41 AM, I=C3=B1aki Baz C=
astillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.ne=
t</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

Hi, WS is still a draft, IMHO the specs should not include backawards<br>
compatibility mechanisms with other revisions of the spec (while still<br>
a draft). Maybe I miss something?<br></blockquote></div><div><br></div>The =
problem is that there are deployed implementations, and breaking them would=
 be bad for adoption of the spec. =C2=A0I agree, you don&#39;t want to clut=
ter up the spec with too much support for old versions, but in this case th=
ere is very little cost.<div>

<br></div><div>In an ideal world, the spec would be finished before it was =
deployed, but that also means not getting implementation experience back in=
to the spec, so there are tradeoffs both ways.<br><div><br>-- <br>John A. T=
amplin<br>

Software Engineer (GWT), Google<br>
</div></div>

--000e0cd51b405c730c04a77b61b5--

From alexey.melnikov@isode.com  Thu Jul  7 11:42:00 2011
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Subject: Re: [hybi] -09: IANA considerations
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Peter Saint-Andre wrote:
 [...]

>Sections 11. and 11.2 both say:
>
>   Interoperability considerations.
>      None.
>
>However, the template in RFC 4395 says:
>
>   Interoperability considerations.
>      If you are aware of any details regarding your scheme that might
>      impact interoperability, please identify them here.  For example:
>      proprietary or uncommon encoding method; inability to support
>      multibyte character sets; incompatibility with types or versions
>      of any underlying protocol.
>
>Now, Section 5.1 of the WebSocket spec says:
>
>   2.   The Method of the request MUST be GET and the HTTP version MUST
>        be at least 1.1.
>
>So it seems that the interoperability considerations of our URI
>registration requests might need to say something about incompability
>with HTTP 1.0.
>  
>
I've replaced "None" with "Use of WebSocket requires use of HTTP version 
1.1 or higher.". I hope this is sufficient.

>Section 11.6 mentions private use tokens beginning with "x-". Ick. :)
>  
>
I think this is still a debate to have in the Apps Area meeting in 
Quebec ;-).

>Typo in Section 11.10: "Paragraph 10 of Paragraph 4 of Section 5.1 "
>  
>
I've removed "of Paragraph 4", as I think the Paragraph 10 of Section 
5.1 is the correct reference.

>Section 11.12 says that assignment of WebSocket Version Numbers shall be
>"RFC Required", but then requests assignment of version numbers 0-8 to
>prior submissions of this Internet-Draft. The requested assignments are
>at odds with the stated policy.
>
I am not entirely sure that RFC Required prevents what the document 
does, but in order to clarify I've added:

      In order to improve interoperability with intermediate versions
      published in Internet Drafts, version numbers associated with such 
drafts
      might be registered in this registry.


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Peter Saint-Andre wrote:

>Here are some comments on Section 4 (I had comments on Section 3 of -08
>but they were fixed in -09).
>
>In 4.2...
>
>   frame-rsv1              = %x0 ; 1 bit, MUST be 0
>
>   frame-rsv2              = %x0 ; 1 bit, MUST be 0
>
>   frame-rsv3              = %x0 ; 1 bit, MUST be 0
>
>I think we mean "MUST be 0 unless negotiated otherwise".
>  
>
Yes, changed.

>In 4.3...
>
>At the beginning of Section 4.4 there is a nice paragraph starting with
>"The primary purpose of fragmentation is...", but we don't have
>something similar about masking. What is masking intended to accomplish?
>Is it supposed to have security properties? Etc.
>
>   The client MUST mask all frames sent to the server.  A server MUST
>   close the connection upon receiving a frame with the MASK bit set to
>   0.
>
>If framing is truly mandatory for all frames (not "if the parties
>negotiate masking, then the client MUST mask all frames..."), why have
>the MASK bit in the first place? Would frame-masked be set to 0 only for
>control frames (or control frames not containing a body)?
>  
>
I think this would require a bit of discussion and some text (not 
necessarily from you).

>In 4.5.2 and 4.5.3, it might be helpful to say explicitly whether Ping
>frames and Pong frames are allowed to contain a body.
>  
>
I think this is already clear, e.g.
    A Pong frame sent in response to a Ping frame must have identical
    Application Data as found in the message body of the Ping frame
    being replied to.

>In 4.8, I think we mean "prescriptive" instead of "proscriptive" (from
>"proscribe" = "forbid").
>
Fixed.

>o  Extension data may be placed in the payload data before the
>      application data.
>
>Do we mean "MAY"?  
>
I don't feel particularly strongly about this, but this sentence is 
included in the list of examples, so use of MAY seems inapropriate.


From alexey.melnikov@isode.com  Thu Jul  7 11:59:54 2011
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Subject: Re: [hybi] -09: SHA-1
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Peter Saint-Andre wrote:

>In finishing my review of -09, I noticed the normative reference to
>[FIPS.180-2.2002] for SHA-1. The WG might want to think about whether
>the existing attacks against SHA-1 might raise concerns about a lack of
>collision resistance for the Sec-WebSocket-Key header (see RFC 4270 and
>RFC 6194 for details). It's probably worth saying something about this
>in the security considerations.
>
There is no requirement for collision resistence, as far as I 
understand. IMHO, we can even use MD5.
SHA-1 is used to protect against replays (different requests will have 
different client key and thus different hashes) and intermediaries that 
would try to mess with websocket traffic without understanding it.

Do you think that needs to be said in the Security Considerations?


From alexey.melnikov@isode.com  Thu Jul  7 12:05:39 2011
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Hi,

Willy Tarreau wrote:

>Hi,
>
>On Wed, Jun 15, 2011 at 01:39:41PM -0700, Dan Adkins wrote:  
>
>>Comment:
>>
>>Sec 1.3. Opening Handshake
>>
>>   Headers in the handshake are sent by the client in a random order;
>>   the order is not meaningful.
>>
>>Saying "random order" here is misleading; it implies that the client
>>must shuffle the headers (I understand old versions of the protocol
>>required this.)  If the order is not meaningful, then it is perfectly
>>fine for an implementation to always send them in the same order.    
>>
>
>Good point.
>  
>
>>I think the wording from RFC 2616 (HTTP/1.1) is clearer:
>>
>>   The order in which header fields with differing field names are
>>   received is not significant.
>>    
>>
>
>Or maybe a mix of both, something around this ?
>  
>
>     Headers in the handshake may be sent by the client in any order,
>     so the order in which header fields with differing field names
>     are received is not significant.
>
I've used a slightly modified version of this text:

    In compliance with RFC 2616, header fields in the handshake may be 
sent by
    the client in any order, so the order in which different header fields
    are received is not significant.

>The same point is valid for the response format BTW.
>  
>
Right.

>  
>


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From: Alexey Melnikov <alexey.melnikov@isode.com>
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Subject: Re: [hybi] -09: abstract and introduction
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Peter Saint-Andre wrote:

>I've started reviewing -09. For several reasons (mostly because my
>review time keeps getting interrupted), I'll provide feedback by
>section. This set of comments is mostly editorial.
>
>The abstract seems over-specific. I suggest deleting the parentheticals
>
I've done that in my copy ...

>and putting them in a slightly-longer first paragraph in the intro
>(before Section 1.1).
>
... but I haven't done that part yet.

>Section 1.1 has always struck me as strange. It sounds as if we're
>developing an IM protocol here! I suggest:
>
>   Historically, creating Web applications that need bidirectional
>   communication between a client and a server (e.g., instant messaging
>   and gaming applications) has required an abuse of HTTP to poll the
>   server for updates while sending upstream notifications as distinct
>   HTTP calls. [RFC6202]
>
I've used your text, thanks!

>I think it would also be helpful to provide an additional paragraph at
>the end of Section 1.1 with a bit of justification for why people who
>want to write bidirectional Web applications can't simply use existing
>application protocols such as IRC and XMPP. Yes, port 80 is the one true
>port, and developers complain that it would be a pain to build protocols
>like IRC and XMPP directly into a browser, but nothing says it's impossible.
>  
>
I am afraid I will need some help with such text.

>BTW [I-D.ietf-httpstate-cookie] is now [RFC6265] (thanks, Adam!).
>
Updated.

>Section 1.3 says:
>
>   If the
>   server does not wish to accept connections from this origin, it can
>   choose to reject the connection by sending an appropriate HTTP error
>   code.
>
>Which code?
>  
>
The section is informative. Do you really want the specific error 
code(s) listed?

>Section 1.3 says:
>
>   A JavaScript application cannot set a header starting with "Sec-" via
>   XHR.
>
>Why not? What is the nature of the restriction?
>
I think you got an answer to this on the mailing list.

>Section 1.3 says:
>
>   If the |Sec-WebSocket-Accept|
>   value does not match the expected value, or if the header is missing,
>   or if the HTTP status code is not 101, the connection will not be
>   established and WebSocket frames will not be sent.
>
>Do we really mean the following?
>
>   If the |Sec-WebSocket-Accept|
>   value does not match the expected value, or if the header is missing,
>   or if the HTTP status code is not 101, the browser MUST NOT
>   establish the connection and MUST NOT send WebSocket frames.
>
I think you got an answer to this on the mailing list: the section is 
informative, so it would be better to avoid using RFC 2119 language here.

>Section 1.4 uses the term "man-in-the-middle proxies". The term "man in
>the middle" might lead people to think of man-in-the-middle attacks as
>in RFC 4949 (BTW, an informational reference to RFC 4949 would be good
>in Section 2.1). I suggest "middlebox proxies" or "intermediate proxies"
>or some more neutral term.
>
I've changed that to "intercepting proxies", I think that is the correct 
term.

>In Section 1.5, I suggest...
>
>OLD:
>   It is expected
>   that metadata would be layered on top of WebSocket by the application
>   layer, in the same way that metadata is layered on top of TCP by the
>   application layer (HTTP).
>
>NEW:
>   It is expected
>   that metadata would be layered on top of WebSocket by the application
>   layer, in the same way that metadata is layered on top of TCP by the
>   application layer (e.g., HTTP).
>
Changed, thanks.

>This paragraph is hard to read:
>
>   Conceptually, WebSocket is really just a layer on top of TCP that
>   adds a Web "origin"-based security model for browsers; adds an
>   addressing and protocol naming mechanism to support multiple services
>   on one port and multiple host names on one IP address; layers a
>   framing mechanism on top of TCP to get back to the IP packet
>   mechanism that TCP is built on, but without length limits; and
>   includes an additional closing handshake in-band that is designed to
>   work in the presence of proxies and other intermediaries.  Other than
>   that, it adds nothing.  Basically it is intended to be as close to
>   just exposing raw TCP to script as possible given the constraints of
>   the Web. It's also designed in such a way that its servers can share
>   a port with HTTP servers, by having its handshake be a valid HTTP
>   Upgrade request mechanism also.
>
>I suggest breaking it up into bullet points:
>
>   Conceptually, WebSocket is really just a layer on top of TCP that
>   does the following:
>
>   * adds a Web "origin"-based security model for browsers
>
>   * adds an addressing and protocol naming mechanism to support
>   multiple services on one port and multiple host names on one IP
>   address;
>
>   * layers a framing mechanism on top of TCP to get back to the IP
>   packet mechanism that TCP is built on, but without length limits
>
>   * includes an additional closing handshake in-band that is designed
>   to work in the presence of proxies and other intermediaries
>
>   Other than that, WebSocket adds nothing.  Basically it is intended
>   to be as close to just exposing raw TCP to script as possible given
>   the constraints of the Web. It's also designed in such a way that
>   its servers can share a port with HTTP servers, by having its
>   handshake be a valid HTTP Upgrade request mechanism also.
>
Changed. I will review later on if the text is actually correctly 
describe the protocol.

>Section 1.7 says:
>
>   By default the WebSocket protocol uses port 80 for regular WebSocket
>   connections and port 443 for WebSocket connections tunneled over TLS
>   [RFC2818].
>
>I know this has been discussed on the list, but how can an entity
>discover if WebSocket is offered on ports other than 80/443? Would the
>web server use a redirect to do that? Could it define SRV records?
>  
>
I think that was also extensively discussed on the mailing list.

>Section 1.9 says:
>
>   These subprotocol names should be registered as per Section 11.10.
>   To avoid potential collisions, it is recommended to use names that
>   contain the domain name of the subprotocol's originator.
>
>Following the trail, we find this in Section 11.10:
>
>   Subprotocol Identifier
>      The identifier of the subprotocol, as will be used in the Sec-
>      WebSocket-Protocol header registered in Section 11.9 of this
>      specification.  The value must conform to the requirements given
>      in Paragraph 10 of Paragraph 4 of Section 5.1 of this
>      specification, namely the value must be a token as defined by RFC
>      2616 [RFC2616].
>
>In RFC 2616 we find:
>
>       token          = 1*<any CHAR except CTLs or separators>
>
>And in RFC 5234 we find:
>
>         CHAR           =  %x01-7F
>                                ; any 7-bit US-ASCII character,
>                                ;  excluding NUL
>
>So a Subprotocol Identifier is limited to 7-bit ASCII. Given that the
>domain name of the subprotocol's originator might be an IDNA, how will
>the originator be able to include their domain name in the identifier?
>This needs to be specified.
>
I think just saying "ASCII domain name" in Section 1.9 should be 
sufficient. If you think the document should have a reference to the 
most recent IDNA specs, such reference can be added.


From w@1wt.eu  Thu Jul  7 12:15:53 2011
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Hi Alexey,

On Thu, Jul 07, 2011 at 06:40:47PM +0100, Alexey Melnikov wrote:
> I've used a slightly modified version of this text:
> 
>    In compliance with RFC 2616, header fields in the handshake may be 
> sent by
>    the client in any order, so the order in which different header fields
>    are received is not significant.

This looks perfect to me.

Thanks,
Willy


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Peter Saint-Andre wrote:

>More comments...
>
>Section 6.1 states:
>
>   5.  If the data is being sent by the client, the frame(s) MUST be
>       masked as defined in Section 4.3.
>
>Section 6.2 states:
>
>   Data frames received by a server from a client MUST be unmasked as
>   described in Section 4.3.
>
>The word "unmasked" makes it sound like this contradicts the text in
>Section 6.1 -- as in, "must not be masked" as opposed to "the server
>shall remove the masking applied by the client".
>
Changed. I've read "unmask" as "remove masking", but I see why this can 
be confusing.

>In Section 7, the text about the close /reason/ makes it sound as if an
>application might choose to show UTF-8 encoded data to an end user. That
>might lead the reader to think that language tagging might be necessary.
>Is it?
>
I think I've complained about this privately as well :-)

>In Section 8.2, there is no deterministic server behavior upon receiving
>data that is not valid UTF-8. Why? What use cases would motivate
>accepting such data instead of just closing the connection?
>
>Section 9.1 says:
>
>   Any extension-token used MUST either be a registered token
>   (registration TBD), or have a prefix of "x-" to indicate a private-
>   use token.
>
>It's probably not a good idea to have "registration TBD" in a document
>that is going for IETF Last Call. :) Presumably a forward pointer to
>Section 11.6 would suffice.
>
Fixed.

>Do we really want to encourage use of "x-"? See here for relevant
>considerations (I plan to submit an updated version soon):
>
>http://tools.ietf.org/id/draft-saintandre-xdash-considered-harmful-01.txt
>


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(as chairs)

status:
------
Gabriel and I started the WG last call process for version -07 on April 
25th, 2011 (until May 9th),
since then two new version -08 and -09 have been produced
to address all the comments/discussion received during the WG LC time 
and after

the IESG process requires that once the WG LC is over and the document 
is ready
it has to be submitted to IESG for publication...
so we have asked the IESG to publish it,
I will be the Document Shepherd for it.


(if you are not familiar with the IETF process see: 
http://tools.ietf.org/html/rfc4858#section-3 )

Now we are on the AD evaluation phase:
Peter Saint-Andre is the Application AD responsible for HyBi wg;
(you can check all the information from the tracker:
https://datatracker.ietf.org/doc/draft-ietf-hybi-thewebsocketprotocol/ )

In order to address the last, mostly editorials comments/fix we have 
asked the editors to produce
a new version -10 for July 11th

as soon the version -10 is produced we will ask
to start the two weeks broader IESG/IETF last call that involve all the 
IETF community (not only the HyBi wg):
the discussion will happen on the IETF discussion list (ietf@ietf.org):
http://www.ietf.org/list/discussion.html


The idea is to take advantage of the IETF81 meeting in Quebec to discuss 
all the comments received
on version -10



what next:
---------
now that the WebSocket Protocol is stable, it is time to discuss the 
extension that we have put on hold for a while
and decide on what we want to work.
among the others I have in my notes:
     - Multiplexing
     - per Frame deflate extension 
(draft-tyoshino-hybi-websocket-perframe-deflate)
     - Timeout (draft-thomson-hybi-http-timeout)
     - DNS SRV for WebSocket (draft-ibc-websocket-dns-srv)


forgive me and let me know if I forgot something... or there is other 
stuff you think is important to discuss


cheers
/Sal

-- 
Salvatore Loreto
www.sloreto.com


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<html>
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    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#ffffff" text="#000000">
    (as chairs)<br>
    <br>
    status:<br>
    ------<br>
    Gabriel and I started the WG last call process for version -07 on
    April 25th, 2011 (until May 9th),<br>
    since then two new version -08 and -09 have been produced <br>
    to address all the comments/discussion received during the WG LC
    time and after <br>
    <br>
    the IESG process requires that once the WG LC is over and the
    document is ready <br>
    it has to be submitted to IESG for publication...<br>
    so we have asked the IESG to publish it, <br>
    I will be the Document Shepherd for it.<br>
    <br>
    <br>
    (if you are not familiar with the IETF process see:
    <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/rfc4858#section-3">http://tools.ietf.org/html/rfc4858#section-3</a> )<br>
    <br>
    Now we are on the AD evaluation phase:<br>
    Peter Saint-Andre is the Application AD responsible for HyBi wg;<br>
    (you can check all the information from the tracker:<br>
    <a class="moz-txt-link-freetext" href="https://datatracker.ietf.org/doc/draft-ietf-hybi-thewebsocketprotocol/">https://datatracker.ietf.org/doc/draft-ietf-hybi-thewebsocketprotocol/</a>
    )<br>
    <br>
    In order to address the last, mostly editorials comments/fix we have
    asked the editors to produce<br>
    a new version -10 for July 11th<br>
    <br>
    as soon the version -10 is produced we will ask<br>
    to start the two weeks broader IESG/IETF last call that involve all
    the IETF community (not only the HyBi wg):<br>
    the discussion will happen on the IETF discussion list
    (<a class="moz-txt-link-abbreviated" href="mailto:ietf@ietf.org">ietf@ietf.org</a>):<br>
    <a class="moz-txt-link-freetext" href="http://www.ietf.org/list/discussion.html">http://www.ietf.org/list/discussion.html</a><br>
    <br>
    <br>
    The idea is to take advantage of the IETF81 meeting in Quebec to
    discuss all the comments received<br>
    on version -10<br>
    <br>
    <br>
    <br>
    what next:<br>
    ---------<br>
    now that the WebSocket Protocol is stable, it is time to discuss the
    extension that we have put on hold for a while<br>
    and decide on what we want to work.<br>
    among the others I have in my notes:<br>
    &nbsp;&nbsp;&nbsp; - Multiplexing<br>
    &nbsp;&nbsp;&nbsp; - per Frame deflate extension
    (draft-tyoshino-hybi-websocket-perframe-deflate)<br>
    &nbsp;&nbsp;&nbsp; - Timeout (draft-thomson-hybi-http-timeout)<br>
    &nbsp;&nbsp;&nbsp; - DNS SRV for WebSocket (draft-ibc-websocket-dns-srv)<br>
    <br>
    <br>
    forgive me and let me know if I forgot something... or there is
    other stuff you think is important to discuss<br>
    <br>
    <br>
    cheers<br>
    /Sal<br>
    <span style="font-size: 11pt; font-family:
      &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73,
      125);"></span>
    <pre class="moz-signature" cols="72">-- 
Salvatore Loreto
<a class="moz-txt-link-abbreviated" href="http://www.sloreto.com">www.sloreto.com</a></pre>
  </body>
</html>

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From djc.ochtman@gmail.com  Fri Jul  8 01:48:59 2011
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On Fri, Jul 8, 2011 at 10:33, Salvatore Loreto
<salvatore.loreto@ericsson.com> wrote:
> what next:
> ---------
> now that the WebSocket Protocol is stable, it is time to discuss the
> extension that we have put on hold for a while
> and decide on what we want to work.
> among the others I have in my notes:
> =C2=A0=C2=A0=C2=A0 - Multiplexing
> =C2=A0=C2=A0=C2=A0 - per Frame deflate extension
> (draft-tyoshino-hybi-websocket-perframe-deflate)
> =C2=A0=C2=A0=C2=A0 - Timeout (draft-thomson-hybi-http-timeout)
> =C2=A0=C2=A0=C2=A0 - DNS SRV for WebSocket (draft-ibc-websocket-dns-srv)

For the deflate extension, it seems to me that there's a fair amount
of consensus on this list (in "deflate-stream and masking") that the
deflate-stream extension as specified in draft-09 is suboptimal. Given
that, it seems prudent to exclude it from the WebSocket Protocol spec
for now and await the outcome of the perframe-deflate efforts.

As for DNS SRV, I briefly looked it over and think it would certainly
be nice to specify. I just wonder if it wouldn't make more sense to
put it inside the WebSocket Protocol spec, given the mandatory
machinations required from WebSocket client applications.

Cheers,

Dirkjan

From salvatore.loreto@ericsson.com  Fri Jul  8 02:12:54 2011
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time for HyBi session in the final version of the agenda:

HYBI Session 1 (2 hours)
Thursday, Afternoon Session III 1740-1940
Room Name: 202




cheers
/Sal

-- 
Salvatore Loreto
www.sloreto.com



From ibc@aliax.net  Fri Jul  8 03:52:35 2011
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2011/7/8 Dirkjan Ochtman <dirkjan@ochtman.nl>:
> As for DNS SRV, I briefly looked it over and think it would certainly
> be nice to specify. I just wonder if it wouldn't make more sense to
> put it inside the WebSocket Protocol spec, given the mandatory
> machinations required from WebSocket client applications.

Hi, the problem with the extensions is that we are speaking mostly
about webbrowsers. So if just a widely used webbrowser in the world
decides not to implement an extension (ie: DNS SRV if approbed) then
WebSocket service providers could not rely on such mechanism.

I suppose the same is true for other extensions (but I'm not sure
about their backwards-compatibility and so). In the case of DNS SRV,
all the clients must implement it. If not, it's like if it does not
exist because no one service provider will want to use DNS SRV knowing
that some known clients cannot use them.

Just wondering. Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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On Fri, Jul 8, 2011 at 12:52, I=C3=B1aki Baz Castillo <ibc@aliax.net> wrote=
:
> Hi, the problem with the extensions is that we are speaking mostly
> about webbrowsers. So if just a widely used webbrowser in the world
> decides not to implement an extension (ie: DNS SRV if approbed) then
> WebSocket service providers could not rely on such mechanism.

That was kind of my point (but admittedly poorly expressed): we should
put it in the spec (i.e. not as an extension) instead of specifying it
separately as an extension.

Cheers,

Dirkjan

From alexey.melnikov@isode.com  Fri Jul  8 09:02:44 2011
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Peter Saint-Andre wrote:

>Here are some comments on Section 5, again mostly editorial...
>
>Section 5.1 uses the terms "/resource name/" whereas Section 3 uses the
>term "resource-name".
>
I will try to clarify.

>Section 5.1 talks about a "/secure/ flag" but that term is not used in
>Section 3. Perhaps modify the latter to say:
>
>   The URI is called "secure" (and it said that "the secure flag is
>   set") if the scheme component matches "wss" case-insensitively.
>
>Change "may" to "MAY" in this sentence...
>
Changed.

>   Clients running in controlled environments, e.g. browsers on mobile
>   handsets tied to specific carriers, may offload the management of the
>   connection to another agent on the network.
>
>It's not clear to me what we mean by "another agent". Is this, for
>example, separate software that runs on the same handset as the
>WebSocket, an application provided by the handset manufacturer or
>service provider that the handset client interacts with via an API, some
>kind of third-party service, or any of the above?
>  
>
All of the above. I am not convinced that that needs clarifying, I think 
the text is clear as is.

>There are a few confusing phrases and long sentences here:
>
>       If the client cannot determine the IP address of the remote host
>       (for example because all communication is being done through a
>       proxy server that performs DNS queries itself), then the client
>       MUST assume for the purposes of this step that each host name
>       refers to a distinct remote host, and should instead limit the
>       total number of simultaneous connections that are not established
>       to a reasonably low number (e.g., in a Web browser, simultaneous
>       pending connections to a.example.com and b.example.com would be
>       allowed, but if thirty connections are requested, that may not be
>       allowed.  The limit should consider the number of tabs the user
>       has open.
>
>I suggest:
>
>       If the client cannot determine the IP address of the remote host
>       (for example because all communication is being done through a
>       proxy server that performs DNS queries itself), then the client
>       MUST assume for the purposes of this step that each host name
>       refers to a distinct remote host.  Instead, the client SHOULD
>       limit the total number of simultaneous pending connections to a
>       reasonably low number (e.g., the client might allow simultaneous
>       pending connections to a.example.com and b.example.com, but
>       might not allow thirty simultaneous pending connections to
>       various hosts).  In a Web browser context, the client SHOULD
>       consider the number of tabs the user has open in setting a limit
>       to the number of simultaneous pending connections.
>  
>
I've used the following variation of your text, which I think is more 
correct:

          If the client cannot determine the IP address of the remote host
          (for example because all communication is being done through a 
proxy
          server that performs DNS queries itself), then the client MUST
          assume for the purposes of this step that each host name 
refers to a
          distinct remote host, and instead the client SHOULD limit the 
total number of
          simultaneous pending connections to a reasonably
          low number (e.g., the client might allow simultaneous pending 
connections
          to a.example.com and b.example.com, but if thirty simultaneous 
connections
          to a single host are requested, that may not be allowed).
          In a Web browser context, the client SHOULD consider the 
number of tabs the user
          has open in setting a limit to the number of simultaneous 
pending connections.

>I have one nit and one concern with this text:
>
>       NOTE: This makes it harder for a script to perform a denial of
>       service attack by just opening a large number of WebSocket
>       connections to a remote host.  A server can further reduce the
>       load on itself when attacked by making use of this by pausing
>       before closing the connection, as that will reduce the rate at
>       which the client reconnects.
>
>Nit: change "by making use of this by pausing before closing the
>connection" to "by pausing before closing the connection".
>
>Concern: this specification says nothing about the reconnection logic.
>  
>
While your suggestion might be a good idea, I would like to hear more 
feedback from the WG.

>You might adapt some of the text from Section 3.3 of RFC 6120:
>
>###
>
>3.3.  Reconnection
>
>   It can happen that an XMPP server goes offline unexpectedly while
>   servicing TCP connections from connected clients and remote servers.
>   Because the number of such connections can be quite large, the
>   reconnection algorithm employed by entities that seek to reconnect
>   can have a significant impact on software performance and network
>   congestion.  If an entity chooses to reconnect, it:
>
>   o  SHOULD set the number of seconds that expire before reconnecting
>      to an unpredictable number between 0 and 60 (this helps to ensure
>      that not all entities attempt to reconnect at exactly the same
>      number of seconds after being disconnected).
>
>   o  SHOULD back off increasingly on the time between subsequent
>      reconnection attempts (e.g., in accordance with "truncated binary
>      exponential backoff" as described in [ETHERNET]) if the first
>      reconnection attempt does not succeed.
>
>###
>
>A question about this text:
>
>       Servers
>       can refuse to accept connections from hosts with an excessive
>       number of existing connections
>
>Do we mean "hosts" here or instead "clients" or "IP addresses"?
>  
>
I've changed "hosts" to "host/IP addresses" (I assume the former comes 
from some reverse DNS lookup). I don't think "client" is going to be 
useful, because in a general case the server can't tell if it is talking 
to the same client or not. (Hmm, unless the client is using the same TLS 
certificate or something like this.)

But anyway, this text is informative and I think leaving it as is or 
adding "IP addresses" is good enough.

>   3.  _Proxy Usage_: If the client is configured to use a proxy when
>       using the WebSocket protocol to connect to host /host/ and/or
>       port /port/, then the client SHOULD connect to that proxy and ask
>       it to open a TCP connection to the host given by /host/ and the
>       port given by /port/.
>
>Do we really mean "and/or"? I don't think you can connect to just a port. :)
>  
>
Changed.

>s/SOCKS/SOCKS5 [RFC1928]/
>
>s/[RFC3548]/[RFC4648]/
>
Updated.

>Please also specify which flavor of base64 (Section 4 or Section 5 of
>RFC 4648) and specify whether padding bits are included at the end of
>the block.
>
>It looks as if [I-D.ietf-websec-origin] is a normative reference, since
>we're taking the ABNF for the "Sec-WebSocket-Origin" from that spec.
>
Agreed.

>I don't think that should hold up publication of the WebSocket spec, but
>if you're worried about that then please provide reviews of the same
>origin spec on the WebSec list. :)
>
>https://www.ietf.org/mailman/listinfo/websec
>
>   10.  The request MAY include a header with the name "Sec-WebSocket-
>        Protocol".  If present, this value indicates the subprotocol(s)
>        the client wishes to speak, ordered by preference.
>
>Please specify that the subprotocols are provided in a comma-separated
>list. (And again in Section 5.2.1.)
>
>   Once the client's opening handshake has been sent, the client MUST
>   wait for a response from the server before sending any further data.
>
>I don't see anywhere a definition of the server's behavior (e.g., it
>MUST close the WebSocket if the client tries to send data before the
>server sends a response).
>
I don't think mandating that is a good idea. The client might pipeline 
its data in anticipation of successful outcome of the handshake. That 
was discussed on the mailing list.

>In Section 5.2.2...
>
>   At this point, the server may begin
>   sending (and receiving) data.
>
>I think at this point both parties (client and server) are allowed to
>send data, right?
>
Yes, but in general case the client doesn't know that until it reads the 
response to the handshake. So I suggest no change is needed.


From gregw@intalio.com  Sun Jul 10 18:26:53 2011
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On 8 July 2011 18:48, Dirkjan Ochtman <dirkjan@ochtman.nl> wrote:
> On Fri, Jul 8, 2011 at 10:33, Salvatore Loreto
> For the deflate extension, it seems to me that there's a fair amount
> of consensus on this list (in "deflate-stream and masking") that the
> deflate-stream extension as specified in draft-09 is suboptimal. Given
> that, it seems prudent to exclude it from the WebSocket Protocol spec
> for now and await the outcome of the perframe-deflate efforts.

+1

From internet-drafts@ietf.org  Mon Jul 11 05:33:18 2011
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the BiDirectional or Server-Initiated HTT=
P Working Group of the IETF.

	Title           : The WebSocket protocol
	Author(s)       : Ian Fette
                          Alexey Melnikov
	Filename        : draft-ietf-hybi-thewebsocketprotocol-10.txt
	Pages           : 68
	Date            : 2011-07-11

   The WebSocket protocol enables two-way communication between a client
   running untrusted code running in a controlled environment to a
   remote host that has opted-in to communications from that code.  The
   security model used for this is the Origin-based security model
   commonly used by Web browsers.  The protocol consists of an opening
   handshake followed by basic message framing, layered over TCP.  The
   goal of this technology is to provide a mechanism for browser-based
   applications that need two-way communication with servers that does
   not rely on opening multiple HTTP connections (e.g. using
   XMLHttpRequest or &lt;iframe&gt;s and long polling).

   Please send feedback to the hybi@ietf.org mailing list.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-hybi-thewebsocketprotocol-10=
.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-hybi-thewebsocketprotocol-10.=
txt

From djc.ochtman@gmail.com  Mon Jul 11 06:01:17 2011
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On Mon, Jul 11, 2011 at 14:33,  <internet-drafts@ietf.org> wrote:
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-hybi-thewebsocketprotocol-10.txt

Tools version: http://tools.ietf.org/html/draft-ietf-hybi-thewebsocketprotocol-10

Editorial nit, page 15, line 22:
"(and it said that "the secure flag is set")" -> "it is said that"?

Technical differences (quick overview):

- Clarified that ASCII-encoded domain names should be sent
- Connection should be failed on unknown opcode (instead of ignoring frame)
- reserved bits maybe non-0 when negotiated
- Sec-WebSocket-Accept value leading/trailing whitespace should be ignored
- client must fail the connection when server-sent frame is non-conformant
- new status code 1007
- UTF-8 encoding failure -> must fail the connection

Cheers,

Dirkjan

From julian.reschke@gmx.de  Mon Jul 11 06:28:48 2011
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On 2011-06-18 12:12, Julian Reschke wrote:
> ...
> REC-wsc-ui-20100812: [REC] ok
> WSAPI: not checked
>
> Note:
>
> RFC3490: [PROPOSED STANDARD] obsoleted by RFC5890 RFC5891
> RFC3548: [INFORMATIONAL] obsoleted by RFC4648
> draft-ietf-httpstate-cookie-20: Alternate version available: 23, later
> published as: RFC6265
> WSAPI: not checked
>
>
> The first three should be updated (with IDNA probably being tricky :-().
>
> The last one is
>
> [WSAPI] Hickson, I., "The Web Sockets API", August 2010,
> <http://dev.w3.org/html5/websockets/>.
>
> and needs to be updated as well. To enable automated checking with
> <http://greenbytes.de/tech/webdav/rfc2629xslt/rfc2629xslt.html#checking-references>,
> I'd make it:
>
> <reference anchor='WD-websockets'
> target='http://www.w3.org/TR/2011/WD-websockets-20110419/'>
> <front>
> <title>The WebSocket API</title>
> <author fullname='Ian Hickson' surname='Hickson' initials='I.'/>
> <date year='2011' month='April' day='19'/>
> </front>
> <seriesInfo name='W3C Working Draft' value='WD-websockets-20110419'/>
> <annotation>
> Latest version available at
> <eref target='http://www.w3.org/TR/websockets/'/>.
> </annotation>
> </reference>

Apparently this wasn't fixed (the spec still references a draft dated 
August 2010).

Also, new:

	[I-D.ietf-websec-origin]	
			Barth, A., "The Web Origin Concept",	
			draft-ietf-websec-origin-00 (work in progress),	
			December 2010.

I believe this has been replaced by

   <http://tools.ietf.org/html/draft-ietf-websec-origin-02>

(Adam, you should tell the IETF secretariat about replacements like 
these so that the pub database is correct).

Best regards, Julian

From julian.reschke@gmx.de  Mon Jul 11 06:31:21 2011
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On 2011-07-11 15:28, Julian Reschke wrote:
> ...
> (Adam, you should tell the IETF secretariat about replacements like
> these so that the pub database is correct).
> ...

Sorry, my confusion. ID base name is indeed is the same.

From stpeter@stpeter.im  Mon Jul 11 06:43:45 2011
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Subject: [hybi] IETF Last Call
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<hat type='AD'/>

I've requested IETF Last Call for draft-ietf-hybi-thewebsocketprotocol,
which means the document will begin to experience wider review. Some of
the upcoming feedback will be provided on the ietf@ietf.org discussion
list, so you might want to temporarily subscribe to that list:

https://www.ietf.org/mailman/listinfo/ietf

Also, please note that we have a normative reference to this document:

http://tools.ietf.org/html/draft-ietf-websec-origin-02

It would be very helpful for HyBi folks to review that document and post
any comments to the WEBSEC WG:

https://www.ietf.org/mailman/listinfo/websec

Thanks!

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From iesg-secretary@ietf.org  Mon Jul 11 07:02:29 2011
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Subject: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The	WebSocket protocol) to Proposed Standard
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The IESG has received a request from the BiDirectional or
Server-Initiated HTTP WG (hybi) to consider the following document:
- 'The WebSocket protocol'
  <draft-ietf-hybi-thewebsocketprotocol-10.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 substantive comments to the
ietf@ietf.org mailing lists by 2011-07-25. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


   The WebSocket protocol enables two-way communication between a client
   running untrusted code running in a controlled environment to a
   remote host that has opted-in to communications from that code.  The
   security model used for this is the Origin-based security model
   commonly used by Web browsers.  The protocol consists of an opening
   handshake followed by basic message framing, layered over TCP.  The
   goal of this technology is to provide a mechanism for browser-based
   applications that need two-way communication with servers that does
   not rely on opening multiple HTTP connections (e.g. using
   XMLHttpRequest or <iframe>s and long polling).

   Please send feedback to the hybi@ietf.org mailing list.




The file can be obtained via
http://datatracker.ietf.org/doc/draft-ietf-hybi-thewebsocketprotocol/

IESG discussion can be tracked via
http://datatracker.ietf.org/doc/draft-ietf-hybi-thewebsocketprotocol/


No IPR declarations have been submitted directly on this I-D.



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--000e0cd5919604005304a7d785b1
Content-Type: text/plain; charset=ISO-8859-1

Side by side:
http://tools.ietf.org/rfcdiff?url2=draft-ietf-hybi-thewebsocketprotocol-10

Some typos

4.5.3.
A Pong frame sent in response to a Ping frame must have
must -> MUST

Client's handshake step 5 in 5.1
present in the client' handshake
client' -> client's

7.1.7.
remote endpoint after being instruted
instruted -> instructed

--000e0cd5919604005304a7d785b1
Content-Type: text/html; charset=ISO-8859-1
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<div>Side by side:=A0<a href=3D"http://tools.ietf.org/rfcdiff?url2=3Ddraft-=
ietf-hybi-thewebsocketprotocol-10">http://tools.ietf.org/rfcdiff?url2=3Ddra=
ft-ietf-hybi-thewebsocketprotocol-10</a></div><div><br></div><div>Some typo=
s</div>

<div><br></div><div>4.5.3.</div><div>A Pong frame sent in response to a Pin=
g frame must have</div><div>must -&gt; MUST</div><div><br></div><div>Client=
&#39;s handshake step 5 in 5.1</div><div>present in the client&#39; handsha=
ke</div>


<div>client&#39; -&gt; client&#39;s</div><div><br></div><div>7.1.7.</div><d=
iv>remote endpoint after being instruted</div><div>instruted -&gt; instruct=
ed</div>

--000e0cd5919604005304a7d785b1--

From Martin.Thomson@commscope.com  Mon Jul 11 22:38:15 2011
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From: "Thomson, Martin" <Martin.Thomson@commscope.com>
To: Mykyta Yevstifeyev <evnikita2@gmail.com>
Date: Tue, 12 Jul 2011 13:39:08 +0800
Thread-Topic: Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The	WebSocket protocol) to Proposed Standard
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On 2011-07-12 at 14:40:52, Mykyta Yevstifeyev wrote:
> Section 4.2.  I have a concern with usage of ABNF here.  Let's see:
>=20
> >     frame-fin               =3D %x0 ; more frames of this message
> follow
> >                             / %x1 ; final frame of this message
>=20
> Which means that ABNF represents characters only; no bits may be=20
> represented by it. =20

That's not true.  RFC 5234 merely uses ASCII in its example.  The terminal =
values for this particular application of ABNF could be a single bit in pla=
ce of a character.  5234 doesn't prohibit this, though it would make most o=
f the default rules meaningless and it would require a little bit of care.

B0 =3D %x0
B1 =3D %x1
BIT =3D B0 / B1
frame =3D fin reserved opcode (unmasked-payload / masked-payload) ; ...
fin =3D BIT
reserved =3D 3BIT
opcode =3D 4BIT ; or continuation (4B0) / text (3B0 B1) / ...
unmasked-payload =3D B0 payload-length payload
masked-payload =3D B1 payload-length 32BIT payload
payload-length =3D (B0 6BIT) / (6B1 ( B0 16BIT / B1 B0 63BIT ))
payload =3D 8<payload-length in bytes>

Other sections can use a character as a terminal value, since that's easier=
; ASCII, ISO-8859-1 or UTF-8, it matters not.


I notice that the picture in 4.2 says only 63 bits if the first length is 1=
27, which is an error, sort of.

--Martin



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Date: Tue, 12 Jul 2011 07:40:52 +0300
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Cc: hybi@ietf.org, draft-ietf-hybi-thewebsocketprotocol@tools.ietf.org
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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Hello,

I'd like to comment the draft-ietf-hybi-thewebsocketprotocol-10, which 
is currently in IETF Last Call.

Throughout the document:

>     _This section is non-normative._

are quite unusual.  Such statements occur at the beginning of 
Introduction, which is meant to be nob-normative a priori, its 
subsections, and Section 4.7, Examples.  These sections, IMO, doesn't 
need to be additionally marked as non-normative.

Section 2.1 makes use of a bit unusual notation; explicitly, it often 
excludes possibility to reference ABNF productions in angle brackets in 
the text.  I don't object to the existing notation (but see below), but 
ABNF rules should be referenced in other way.

Section 3.  I propose to rewrite the first paragraph as follows:

>     This specification defines two URI schemes for WebSocket protocol -
>     'ws' and 'wss'.  Their syntax is defined below using ABNF [RFC5234]
>     in the<ws-uri>  and<wss-uri>, respectively:
>
>       ws-uri  = "ws:" "//" host [ ":" port ] path-abempty [ "?" query ]
>       wss-uri = "wss:" "//" host [ ":" port ] path-abempty [ "?" query ]
>
>     where the<host>,<port>,<path-abempty>  and<query>  rules are
>     defined in RFC 3986 [RFC3986].

Rationale: (1) The first paragraph gets clearer. (2) ABNF is changed not 
ot use pros-vals (RFC 5234) (3) s/path/path-abempty/ to directly import 
it from RFC 3986 (4) Several editorial issues fixed.

I see a number of times the term "WebSocket URI" is used in the draft.  
I would be useful to clarify that the "WebSocket URI" is either a valid 
'ws' or 'wss' URI.

Section 4.2.  I have a concern with usage of ABNF here.  Let's see:

>     frame-fin               = %x0 ; more frames of this message follow
>                             / %x1 ; final frame of this message

Looking at RFC 5234, Section 2.3 we find:

>     Rules resolve into a string of terminal values, sometimes called
>     characters.

Which means that ABNF represents characters only; no bits may be 
represented by it.  Creating the normative and formal definition of a 
header is just impossible.  The above rule will be correctly interpreted as:

>     frame-fin               =<ASCII NUL character 0x00>
>                             /<ASCII SOH character 0x01>

which isn't what you mean.  Changing "x" to "d" to represent binary 
values won't help, since all other bits will be considered to be zeros 
and the result will be the same.  Conclusion: there is no other way 
other that to remove the ABNF definition of WS message here.

Sections 4.5.2. and 4.5.3.  I think the names "ping" and "pong" can 
better be replaced with "echo request" and "echo response".

Section 5.2.2, bullet 3, sub-bullet 4.  When defining the ABNF for a 
header, the header name should be included in it as well.  So the first 
line should be:

> sec-websocket-accept-header = "Sec-WebSocket-Accept:" base64-value

This will also deal with the issue regarding the name of ABNF production 
for the header.

Section 9.1:

>           extension-token = registered-token / private-use-token

If you want RFC 2616 ABNF, this should be changed to:

>           extension-token = registered-token | private-use-token

Ibid:

>           Sec-WebSocket-Extensions: bar; baz=2
>
>     is exactly equivalent to
>
>           Sec-WebSocket-Extensions: foo, bar; baz=2

These two examples don't match the aforementioned ABNF; the space before 
"baz=2" should be removed.

Section 11.1:

> 11.1.  Registration of "ws:" Scheme

The ":" colon isn't included in the scheme name.  So replacing "ws:" 
with 'ws' should be OK.

>     A |ws:| URI identifies a WebSocket server and resource name.

In Section 2.1 you mentioned that |foo| is used when "foo" is a header.  
Please use simple quotes.

>        In ABNF terms using the terminals from the URI specifications:
>        [RFC5234] [RFC3986]
>
>             "ws" ":" hier-part [ "?" query ]

This isn't what you described in Section 3. <hier-part> includes the 
<userinfo> part, which shouldn't be present in WebSocket URIs.  This 
ABNF should be fixed to match one in Section 3.

>        The<path>  [RFC3986] and<query>  components form the resource name
>        sent to the server to identify the kind of service desired.  Other
>        components have the meanings described in RFC3986.

If you adopt my proposal to Section 3, this should be fixed in the same way.

>     References.
>        RFC XXXX

References here mean (from RFC 4395):

>     References.
>        Include full citations for all referenced documents.  Registration
>        templates for provisional registration may be included in an
>        Internet Draft; when the documents expire or are approved for
>        publication as an RFC, the registration will be updated.

So this field should refer to Section 14 of the draft.

Section 11.2: the same applies.

Section 11.12:

>        Version Number  | Reference
>      -+----------------+-----------------------------------------+-
>       | 0              + draft-ietf-hybi-thewebsocketprotocol-00 |
>      -+----------------+-----------------------------------------+-
>         [ . . . ]
>      -+----------------+-----------------------------------------+-
>       | 9              + draft-ietf-hybi-thewebsocketprotocol-09 |
>      -+----------------+-----------------------------------------+-

This looks very strange, since all revisions of the draft will become a 
single RFC.  I propose:

>        Version Number  | Reference
>      -+----------------+-----------------------------------------+-
>       | 1              + This document                           |
>      -+----------------+-----------------------------------------+-

Moreover, what is the allowed range of version numbers (and all other 
values for which the registries are created)?  What entries are 
Unassigned and what are Reserved (I suppose 0 might be Reserved, others 
- Unassigned).

I find the only 2 definitions of syntax of header field described in 
your document - Sec-WebSocket-Accept and Sec-WebSocket-Extensions.  
Where are the other syntax definitions for those header fields 
registered in Section 11?

References.  RFC 3490 is a downward reference here; RFC 3967 procedure 
should have been used.  All other of them seem OK.

I hope my comments were useful.

Thanks,
Mykyta Yevstifeyev

11.07.2011 17:02, The IESG wrote:
> The IESG has received a request from the BiDirectional or
> Server-Initiated HTTP WG (hybi) to consider the following document:
> - 'The WebSocket protocol'
>    <draft-ietf-hybi-thewebsocketprotocol-10.txt>  as a Proposed Standard

From julian.reschke@gmx.de  Mon Jul 11 23:59:53 2011
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Cc: hybi@ietf.org, ietf@ietf.org, draft-ietf-hybi-thewebsocketprotocol@tools.ietf.org
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On 2011-07-12 06:40, Mykyta Yevstifeyev wrote:
> ...
> Throughout the document:
>
>> _This section is non-normative._
>
> are quite unusual. Such statements occur at the beginning of
> Introduction, which is meant to be nob-normative a priori, its
> subsections, and Section 4.7, Examples. These sections, IMO, doesn't
> need to be additionally marked as non-normative.
> ...

+1

> Section 3. I propose to rewrite the first paragraph as follows:
>
>> This specification defines two URI schemes for WebSocket protocol -
>> 'ws' and 'wss'. Their syntax is defined below using ABNF [RFC5234]
>> in the<ws-uri> and<wss-uri>, respectively:
>>
>> ws-uri = "ws:" "//" host [ ":" port ] path-abempty [ "?" query ]
>> wss-uri = "wss:" "//" host [ ":" port ] path-abempty [ "?" query ]
>>
>> where the<host>,<port>,<path-abempty> and<query> rules are
>> defined in RFC 3986 [RFC3986].
>
> Rationale: (1) The first paragraph gets clearer. (2) ABNF is changed not
> ot use pros-vals (RFC 5234) (3) s/path/path-abempty/ to directly import
> it from RFC 3986 (4) Several editorial issues fixed.

-10

Granted, it doesn't use prose values anymore, but then it get's 
incomplete. I believe putting references to ABNF productions from other 
specs into prose values is absolutely the right thing to do (as opposed 
to just mention them in prose).

> Section 5.2.2, bullet 3, sub-bullet 4. When defining the ABNF for a
> header, the header name should be included in it as well. So the first
> line should be:
> ...

Why?

> Section 9.1:
>
>> extension-token = registered-token / private-use-token
>
> If you want RFC 2616 ABNF, this should be changed to:
> ...

yes.

>> extension-token = registered-token | private-use-token
>
> Ibid:
>
>> Sec-WebSocket-Extensions: bar; baz=2
>>
>> is exactly equivalent to
>>
>> Sec-WebSocket-Extensions: foo, bar; baz=2
>
> These two examples don't match the aforementioned ABNF; the space before
> "baz=2" should be removed.
 > ...

They do, as the RFC 2616 ABNF allows implied Linear Whitespace.

That being said, it might be a good idea to revisit the choice of 
syntax, or at least to clarify the LWS situation.

>> In ABNF terms using the terminals from the URI specifications:
>> [RFC5234] [RFC3986]
>>
>> "ws" ":" hier-part [ "?" query ]
>
> This isn't what you described in Section 3. <hier-part> includes the
> <userinfo> part, which shouldn't be present in WebSocket URIs. This ABNF
> should be fixed to match one in Section 3.

...or removed. Why are there two?

>> The<path> [RFC3986] and<query> components form the resource name
>> sent to the server to identify the kind of service desired. Other
>> components have the meanings described in RFC3986.
>
> If you adopt my proposal to Section 3, this should be fixed in the same
> way.
>
>> References.
>> RFC XXXX
>
> References here mean (from RFC 4395):
>
>> References.
>> Include full citations for all referenced documents. Registration
>> templates for provisional registration may be included in an
>> Internet Draft; when the documents expire or are approved for
>> publication as an RFC, the registration will be updated.
>
> So this field should refer to Section 14 of the draft.

"Section 14 of this document".

> Section 11.2: the same applies.
>
> Section 11.12:
>
>> Version Number | Reference
>> -+----------------+-----------------------------------------+-
>> | 0 + draft-ietf-hybi-thewebsocketprotocol-00 |
>> -+----------------+-----------------------------------------+-
>> [ . . . ]
>> -+----------------+-----------------------------------------+-
>> | 9 + draft-ietf-hybi-thewebsocketprotocol-09 |
>> -+----------------+-----------------------------------------+-
> ...


This is indeed fishy and I would be really surprised if IESG and RFC 
Editor let this pass.

If 0..9 can't be reassigned then let's just state they are reserved.

> ...

Best regards, Julian

From ibc@aliax.net  Tue Jul 12 00:48:51 2011
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/12 Mykyta Yevstifeyev <evnikita2@gmail.com>:
>> =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0Sec-WebSocket-Extensions: bar; baz=3D2
>>
>> =C2=A0 =C2=A0is exactly equivalent to
>>
>> =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0Sec-WebSocket-Extensions: foo, bar; ba=
z=3D2
>
> These two examples don't match the aforementioned ABNF; the space before
> "baz=3D2" should be removed.

Hi, are you sure of that? In SIP protocol (which inherits from HTTP
grammar) a header parameter (;foo=3Dlalala) can contain spaces anywhere
(before/after the ";", before/after the "=3D"). So something like:

   Sec-WebSocket-Extensions: foo  , bar  ; baz =3D 2

is valid.

However it's not clear for me wheter in this example "baz=3D2" is a
header param or a param just for the value "bar". In the last case it
would mean a specific header syntax, so spaces could be not allowed.
Could you please point to the ABNF grammar you meant?

Thanks.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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On 2011-07-12 09:48, IÃ±aki Baz Castillo wrote:
> 2011/7/12 Mykyta Yevstifeyev<evnikita2@gmail.com>:
>>>           Sec-WebSocket-Extensions: bar; baz=2
>>>
>>>     is exactly equivalent to
>>>
>>>           Sec-WebSocket-Extensions: foo, bar; baz=2
>>
>> These two examples don't match the aforementioned ABNF; the space before
>> "baz=2" should be removed.
>
> Hi, are you sure of that? In SIP protocol (which inherits from HTTP
> grammar) a header parameter (;foo=lalala) can contain spaces anywhere
> (before/after the ";", before/after the "="). So something like:
>
>     Sec-WebSocket-Extensions: foo  , bar  ; baz = 2
>
> is valid.
>
> However it's not clear for me wheter in this example "baz=2" is a
> header param or a param just for the value "bar". In the last case it
> would mean a specific header syntax, so spaces could be not allowed.
> Could you please point to the ABNF grammar you meant?

According to 
<http://tools.ietf.org/html/draft-ietf-hybi-thewebsocketprotocol-10#section-9.1> 
it's a parameter of "bar".

That being said, I'm not happy with

          extension-param = token [ "=" token ]

In HTTP header fields, parameters usually support both token and 
quoted-string form.

Making this special means that existing header field parser code can't 
be easily re-used.

Best regards, Julian


From ibc@aliax.net  Tue Jul 12 01:23:09 2011
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2011/7/12 Julian Reschke <julian.reschke@gmx.de>:
> That being said, I'm not happy with
>
> =C2=A0 =C2=A0 =C2=A0 =C2=A0 extension-param =3D token [ "=3D" token ]
>
> In HTTP header fields, parameters usually support both token and
> quoted-string form.

Right.


> Making this special means that existing header field parser code can't be
> easily re-used.

Well, not exaclty as any parser supporting tokens and quoted-strings
would also parse "just" tokens :)
However I agree that making new/custom grammar for a simple header
makes no sense.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From julian.reschke@gmx.de  Tue Jul 12 01:51:53 2011
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On 2011-07-12 10:23, IÃ±aki Baz Castillo wrote:
> 2011/7/12 Julian Reschke<julian.reschke@gmx.de>:
>> That being said, I'm not happy with
>>
>>          extension-param = token [ "=" token ]
>>
>> In HTTP header fields, parameters usually support both token and
>> quoted-string form.
>
> Right.
>
>
>> Making this special means that existing header field parser code can't be
>> easily re-used.
>
> Well, not exaclty as any parser supporting tokens and quoted-strings
> would also parse "just" tokens :)

But if they *are* re-used, those recipients will accept quoted-strings 
as well (thus accepting invalid header fields). This could be harmless, 
or it could cause them to be used in practice, forcing other recipients 
to accept them as well.

> ...

Best regards, Julian

From evnikita2@gmail.com  Tue Jul 12 01:58:39 2011
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12.07.2011 8:39, Thomson, Martin wrote:
> On 2011-07-12 at 14:40:52, Mykyta Yevstifeyev wrote:
>> Section 4.2.  I have a concern with usage of ABNF here.  Let's see:
>>
>>>      frame-fin               = %x0 ; more frames of this message
>> follow
>>>                              / %x1 ; final frame of this message
>> Which means that ABNF represents characters only; no bits may be
>> represented by it.
> That's not true.  RFC 5234 merely uses ASCII in its example.  The terminal values for this particular application of ABNF could be a single bit in place of a character.  5234 doesn't prohibit this
ABNF is designed to deal with characters represented in octets.  The 
terminal value %xHH stands for ASCII character HH, so it's true that %x0 
stands for NUL; so does %b0.  The <BIT> production doesn't denote actual 
bits, but rather representation of bits.

> BIT            =  "0" / "1"

implies that <BIT> may be either ASCII 0x31 - "1" - or ASCII 0x30 - 
"0".  Once more, ABNF is for chars/octets, not bits.

Mykyta Yevstifeyev
> , though it would make most of the default rules meaningless and it would require a little bit of care.
>
> [ . . . ]
>
>
> I notice that the picture in 4.2 says only 63 bits if the first length is 127, which is an error, sort of.
>
> --Martin
>
>
>


From julian.reschke@gmx.de  Tue Jul 12 02:15:07 2011
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On 2011-07-12 11:09, Mykyta Yevstifeyev wrote:
> ...
>>> Section 5.2.2, bullet 3, sub-bullet 4. When defining the ABNF for a
>>> header, the header name should be included in it as well. So the first
>>> line should be:
>>> ...
>>
>> Why?
> There is the following formulation:
>
>> The 'Foo' headers takes the form of <foo-header> ABNF rules below:
>>
>> foo-header = *(APHA/DIGIT)

It should say: "The 'Foo' header field's value takes the form..."

> will result in the message headers like:
>
>> Upgrade: TLS/1.2
>> Connection: Upgrade
>> gfr134
>
> and "gfr134" will be the 'Foo' header. "foo-header = "Foo:"
> *(APHA/DIGIT)" will result in valid:
>
>> Upgrade: TLS/1.2
>> Connection: Upgrade
>> Foo: gfr134
>
> See also eg. RFC 3282, RFC 2616.

Have a look at the HTTPbis drafts.

>
>>
>> [ . . . ]
>>
>> That being said, it might be a good idea to revisit the choice of
>> syntax, or at least to clarify the LWS situation.
> The document may reference the httpbis-p1 where the <n>#<m>rule
> extension will be described for valid ABNF. See
> http://tools.ietf.org/html/draft-ietf-httpbis-p1-messaging-15#section-1.2.1

It could, but my guess is that HyBi doesn't want to wait for HTTPbis.

> ...

Best regards, Julian

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12.07.2011 9:59, Julian Reschke wrote:
> On 2011-07-12 06:40, Mykyta Yevstifeyev wrote:
>
> [ . . . ]
>
>> Section 3. I propose to rewrite the first paragraph as follows:
>>
>>> This specification defines two URI schemes for WebSocket protocol -
>>> 'ws' and 'wss'. Their syntax is defined below using ABNF [RFC5234]
>>> in the<ws-uri> and<wss-uri>, respectively:
>>>
>>> ws-uri = "ws:" "//" host [ ":" port ] path-abempty [ "?" query ]
>>> wss-uri = "wss:" "//" host [ ":" port ] path-abempty [ "?" query ]
>>>
>>> where the<host>,<port>,<path-abempty> and<query> rules are
>>> defined in RFC 3986 [RFC3986].
>>
>> Rationale: (1) The first paragraph gets clearer. (2) ABNF is changed not
>> ot use pros-vals (RFC 5234) (3) s/path/path-abempty/ to directly import
>> it from RFC 3986 (4) Several editorial issues fixed.
>
> -10
>
> Granted, it doesn't use prose values anymore, but then it get's 
> incomplete. I believe putting references to ABNF productions from 
> other specs into prose values is absolutely the right thing to do (as 
> opposed to just mention them in prose).
I don't have any string position in the way of importing the productions 
from other documents.  However, what is above is what I like more.  
However, what we can see, eg. in 
http://tools.ietf.org/html/rfc5538#appendix-A can be fine as well.
>
>> Section 5.2.2, bullet 3, sub-bullet 4. When defining the ABNF for a
>> header, the header name should be included in it as well. So the first
>> line should be:
>> ...
>
> Why?
There is the following formulation:

> The 'Foo' headers takes the form of <foo-header> ABNF rules below:
>
> foo-header = *(APHA/DIGIT)

will result in the message headers like:

>          Upgrade: TLS/1.2
>          Connection: Upgrade
>          gfr134

and "gfr134" will be the 'Foo' header.  "foo-header = "Foo:" 
*(APHA/DIGIT)" will result in valid:

>          Upgrade: TLS/1.2
>          Connection: Upgrade
>          Foo: gfr134

See also eg. RFC 3282, RFC 2616.

>
> [ . . . ]
>
> That being said, it might be a good idea to revisit the choice of 
> syntax, or at least to clarify the LWS situation.
The document may reference the httpbis-p1 where the <n>#<m>rule 
extension will be described for valid ABNF.  See 
http://tools.ietf.org/html/draft-ietf-httpbis-p1-messaging-15#section-1.2.1
>
> [ . . . ]
>> Section 11.2: the same applies.
>>
>> Section 11.12:
>>
>>> Version Number | Reference
>>> -+----------------+-----------------------------------------+-
>>> | 0 + draft-ietf-hybi-thewebsocketprotocol-00 |
>>> -+----------------+-----------------------------------------+-
>>> [ . . . ]
>>> -+----------------+-----------------------------------------+-
>>> | 9 + draft-ietf-hybi-thewebsocketprotocol-09 |
>>> -+----------------+-----------------------------------------+-
>> ...
>
>
> This is indeed fishy and I would be really surprised if IESG and RFC 
> Editor let this pass.
>
> If 0..9 can't be reassigned then let's just state they are reserved.
I believe there is no problems to make the 0..9 spare, except 1, for 
this version of WS.

Mykyta Yevstifeyev
>
>> ...
>
> Best regards, Julian
>


From evnikita2@gmail.com  Tue Jul 12 02:16:02 2011
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Cc: hybi@ietf.org, ietf@ietf.org, draft-ietf-hybi-thewebsocketprotocol@tools.ietf.org
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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12.07.2011 10:48, IÃ±aki Baz Castillo wrote:
> 2011/7/12 Mykyta Yevstifeyev<evnikita2@gmail.com>:
>>>           Sec-WebSocket-Extensions: bar; baz=2
>>>
>>>     is exactly equivalent to
>>>
>>>           Sec-WebSocket-Extensions: foo, bar; baz=2
>> These two examples don't match the aforementioned ABNF; the space before
>> "baz=2" should be removed.
> Hi, are you sure of that? In SIP protocol (which inherits from HTTP
> grammar) a header parameter (;foo=lalala) can contain spaces anywhere
> (before/after the ";", before/after the "="). So something like:
>
>     Sec-WebSocket-Extensions: foo  , bar  ; baz = 2
>
> is valid.
No, its' everything OK with this issue.  LWS is allowed between 
<extension> productions in the header, but not between the parts of this 
productions: <extension-token>s and <extension-param>s.  See 
http://tools.ietf.org/html/rfc2616#section-2.1 and 
http://tools.ietf.org/html/rfc2616#section-2.1.

>
> However it's not clear for me wheter in this example "baz=2" is a
> header param or a param just for the value "bar". In the last case it
> would mean a specific header syntax, so spaces could be not allowed.
> Could you please point to the ABNF grammar you meant?
This was meant to be the parameter for the "bar", as far as I understand.

Mykyta
>
> Thanks.
>


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2011/7/12 Mykyta Yevstifeyev <evnikita2@gmail.com>:
>> That's not true. =C2=A0RFC 5234 merely uses ASCII in its example. =C2=A0=
The terminal
>> values for this particular application of ABNF could be a single bit in
>> place of a character. =C2=A05234 doesn't prohibit this
>
> ABNF is designed to deal with characters represented in octets. =C2=A0The
> terminal value %xHH stands for ASCII character HH, so it's true that %x0
> stands for NUL; so does %b0. =C2=A0The <BIT> production doesn't denote ac=
tual
> bits, but rather representation of bits.
>
>> BIT =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0=3D =C2=A0"0" / "1"
>
> implies that <BIT> may be either ASCII 0x31 - "1" - or ASCII 0x30 - "0".
> =C2=A0Once more, ABNF is for chars/octets, not bits.


Just a comment: RFC 5234 (ABNF) says:

  BIT            =3D  "0" / "1"

   bin-val        =3D  "b" 1*BIT
                        [ 1*("." 1*BIT) / ("-" 1*BIT) ]
                           ; series of concatenated bit values
                           ;  or single ONEOF range


Is it speaking about real bits? or about ones and zeroes (chars)?

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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12.07.2011 12:14, Julian Reschke wrote:
> On 2011-07-12 11:09, Mykyta Yevstifeyev wrote:
>> ...
>>>> Section 5.2.2, bullet 3, sub-bullet 4. When defining the ABNF for a
>>>> header, the header name should be included in it as well. So the first
>>>> line should be:
>>>> ...
>>>
>>> Why?
>> There is the following formulation:
>>
>>> The 'Foo' headers takes the form of <foo-header> ABNF rules below:
>>>
>>> foo-header = *(APHA/DIGIT)
>
> It should say: "The 'Foo' header field's value takes the form..."
This will eliminate the problem.  Currently we have:

> The ABNF of this header is defined as follows:

not its entity.
>
>> will result in the message headers like:
>>
>>> Upgrade: TLS/1.2
>>> Connection: Upgrade
>>> gfr134
>>
>> and "gfr134" will be the 'Foo' header. "foo-header = "Foo:"
>> *(APHA/DIGIT)" will result in valid:
>>
>>> Upgrade: TLS/1.2
>>> Connection: Upgrade
>>> Foo: gfr134
>>
>> See also eg. RFC 3282, RFC 2616.
>
> Have a look at the HTTPbis drafts.
They should also be clear about whether they mean the header field or 
the header field's entity.
>
>>
>>>
>>> [ . . . ]
>>>
>>> That being said, it might be a good idea to revisit the choice of
>>> syntax, or at least to clarify the LWS situation.
>> The document may reference the httpbis-p1 where the <n>#<m>rule
>> extension will be described for valid ABNF. See
>> http://tools.ietf.org/html/draft-ietf-httpbis-p1-messaging-15#section-1.2.1 
>>
>
> It could, but my guess is that HyBi doesn't want to wait for HTTPbis.
That's up to them.

Mykyta
>
>> ...
>
> Best regards, Julian
>


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12.07.2011 12:20, IÃ±aki Baz Castillo wrote:
> 2011/7/12 Mykyta Yevstifeyev<evnikita2@gmail.com>:
>>> That's not true.  RFC 5234 merely uses ASCII in its example.  The terminal
>>> values for this particular application of ABNF could be a single bit in
>>> place of a character.  5234 doesn't prohibit this
>> ABNF is designed to deal with characters represented in octets.  The
>> terminal value %xHH stands for ASCII character HH, so it's true that %x0
>> stands for NUL; so does %b0.  The<BIT>  production doesn't denote actual
>> bits, but rather representation of bits.
>>
>>> BIT            =  "0" / "1"
>> implies that<BIT>  may be either ASCII 0x31 - "1" - or ASCII 0x30 - "0".
>>   Once more, ABNF is for chars/octets, not bits.
>
> Just a comment: RFC 5234 (ABNF) says:
>
>    BIT            =  "0" / "1"
>
>     bin-val        =  "b" 1*BIT
>                          [ 1*("." 1*BIT) / ("-" 1*BIT) ]
>                             ; series of concatenated bit values
>                             ;  or single ONEOF range
>
>
> Is it speaking about real bits? or about ones and zeroes (chars)?
Chars. See Section 2.3 of RFC 5234 on p. 5:

>     ABNF permits the specification of literal text strings directly,
>     enclosed in quotation marks.

<bin-val> denotes the value of ASCII char in binary form.  Eg, 
%b01100011 stands for "c" (see RFC 20).

Mykyta
>


From francis@aspl.es  Tue Jul 12 07:14:01 2011
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard: request for max frame size
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Hi,

Recently, I posted [1] that websocket protocol should include an
indication about max frame size that is willing to accept the connecting
peer. 

Many pointed this is not an issue because you could use a stream
oriented API (like TCP send/recv and others), but that only bypasses the
problem (in some cases) not solves it.

Websocket protocol is frame based and every frame based protocol
designed until now has/need such feature or similar. Even TCP, for those
that proposes to use a stream oriented API as a solution, includes a
more complex mechanism than a simple max frame size (window based ack),
so each party can control/inform the sender. This is critical.

A good example for this would be the next. Let suppose you have a
constrained memory device (either because it is small or because it
accepts a large amount of connections). On that device you deploy a
websocket app that only accepts small messages (< 512 bytes, let's
say). 

In this context, you can hard code at your web app (javascript) that
must not send messages larger than this size (so you control websocket
payload size) and in the case you need to, you must split them
properly. 

However, even with this mechanism, you have no warranty that the browser
or an intermediary will join those messages into a single frame, causing
your device to receive a bigger message/frame than expected.

Assuming this I think we would require either to include a
Max-Frame-Size indication (or a really poor solution: to explicitly
state on the draft that frames must not be coalesced by intermediaries
or browsers). 

Regards,

[1] http://www.ietf.org/mail-archive/web/hybi/current/msg07641.html
-- 
Francis Brosnan Blázquez <francis.brosnan@aspl.es>
ASPL
91 134 14 22 - 91 134 14 45 - 91 116 07 57

AVISO LEGAL

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ASPL - Protección Datos, C/Antonio Suárez 10 A-102, 28802, Alcalá de
Henares (Madrid).


From Martin.Thomson@commscope.com  Tue Jul 12 16:38:26 2011
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To: Mykyta Yevstifeyev <evnikita2@gmail.com>
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Thread-Topic: Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On 2011-07-12 at 18:59:21, Mykyta Yevstifeyev wrote:
> ABNF is designed to deal with characters represented in octets. =20

Not true again.  ABNF has terminal values that are both characters and nume=
ric.  A character mapping (e.g. ASCII, Unicode) can be used to map one to t=
he other.

   Rules resolve into a string of terminal values, sometimes called
   characters.  In ABNF, a character is merely a non-negative integer.

i.e.,
%xHH is a number=20
"a" is a character that is mapped to a number using the selected character =
mapping (Unicode, ASCII)

The number or non-negative integer is the terminal value.  That number migh=
t then be encoded using the encoding (UTF-8, ASCII), but that's not interes=
ting to ABNF.

If you want, you can define the terminal value to be a number of 1-bit size=
.  The string literal can either have no meaning at all, since the strings =
are ASCII [1].  Then you end up with a 1-bit terminal value which is trivia=
lly encoded.

--Martin

[1] Alternatively, a string literal could be redefined to map "0" to 0 and =
"1" to 1 with all other values invalid - ABNF isn't a law or anything.



From gregw@intalio.com  Tue Jul 12 19:53:54 2011
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard: request for max frame size
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Francis et al,

not also that the protocol does support fragmentation and a 1004 frame
too large error.
Even the 1004 error does not carry an indication of what an acceptable
size is, so the client/tool/intermediary that receives a 1004 will
either have to fail or guess a smaller frames size - potentially
binary chopping down to find an acceptable size, which might be sub
optimal.

I simple optional header in the handshake declaring the max frame size
(which intermediaries could adjust) would be very complimentary to the
existing fragmentation and 1004 features and I can't think of any
significant down side.

regards

On 13 July 2011 00:13, Francis Brosnan Blazquez <francis@aspl.es> wrote:
> Hi,
>
> Recently, I posted [1] that websocket protocol should include an
> indication about max frame size that is willing to accept the connecting
> peer.
>
> Many pointed this is not an issue because you could use a stream
> oriented API (like TCP send/recv and others), but that only bypasses the
> problem (in some cases) not solves it.
>
> Websocket protocol is frame based and every frame based protocol
> designed until now has/need such feature or similar. Even TCP, for those
> that proposes to use a stream oriented API as a solution, includes a
> more complex mechanism than a simple max frame size (window based ack),
> so each party can control/inform the sender. This is critical.
>
> A good example for this would be the next. Let suppose you have a
> constrained memory device (either because it is small or because it
> accepts a large amount of connections). On that device you deploy a
> websocket app that only accepts small messages (< 512 bytes, let's
> say).
>
> In this context, you can hard code at your web app (javascript) that
> must not send messages larger than this size (so you control websocket
> payload size) and in the case you need to, you must split them
> properly.
>
> However, even with this mechanism, you have no warranty that the browser
> or an intermediary will join those messages into a single frame, causing
> your device to receive a bigger message/frame than expected.
>
> Assuming this I think we would require either to include a
> Max-Frame-Size indication (or a really poor solution: to explicitly
> state on the draft that frames must not be coalesced by intermediaries
> or browsers).
>
> Regards,
>
> [1] http://www.ietf.org/mail-archive/web/hybi/current/msg07641.html
> --
> Francis Brosnan Bl=E1zquez <francis.brosnan@aspl.es>
> ASPL
> 91 134 14 22 - 91 134 14 45 - 91 116 07 57
>
> AVISO LEGAL
>
> Este mensaje se dirige exclusivamente a su destinatario. Los datos
> incluidos en el presente correo son confidenciales y sometidos a secreto
> profesional, se proh=EDbe divulgarlos, en virtud de las leyes vigentes. S=
i
> usted no lo es y lo ha recibido por error o tiene conocimiento del mismo
> por cualquier motivo, le rogamos que nos lo comunique por este medio y
> proceda a destruirlo o borrarlo.
>
> En virtud de lo dispuesto en la Ley Org=E1nica 15/1999, de 13 de
> diciembre, de Protecci=F3n de Datos de Car=E1cter Personal, le informamos=
 de
> que sus datos de car=E1cter personal, recogidos de fuentes accesibles al
> p=FAblico o datos que usted nos ha facilitado previamente, proceden de
> bases de datos propiedad de Advanced Software Production Line, S.L.
> (ASPL). No obstante, usted puede ejercitar sus derechos de acceso,
> rectificaci=F3n, cancelaci=F3n y oposici=F3n dispuestos en la mencionada =
Ley
> Org=E1nica, notific=E1ndolo por escrito a:
> ASPL - Protecci=F3n Datos, C/Antonio Su=E1rez 10 A-102, 28802, Alcal=E1 d=
e
> Henares (Madrid).
>
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>

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13.07.2011 2:39, Thomson, Martin wrote:
> On 2011-07-12 at 18:59:21, Mykyta Yevstifeyev wrote:
>> ABNF is designed to deal with characters represented in octets.
> Not true again.  ABNF has terminal values that are both characters and numeric.  A character mapping (e.g. ASCII, Unicode) can be used to map one to the other.
No, you're probably misunderstanding RFC 5234.  Everything after 
"%x/d/b" is treated as hexadecimal, decimal or binary representation of 
a particular ASCII/Unicode character, note vice versa.
>
>     Rules resolve into a string of terminal values, sometimes called
>     characters.  In ABNF, a character is merely a non-negative integer.
>
> i.e.,
> %xHH is a number
> "a" is a character that is mapped to a number using the selected character mapping (Unicode, ASCII)
No, a number is mapped to the character, which is implied by the 
aforementioned statement.
>
> The number or non-negative integer is the terminal value.  That number might then be encoded using the encoding (UTF-8, ASCII), but that's not interesting to ABNF.
>
> If you want, you can define the terminal value to be a number of 1-bit size.
How?  ASCII values are hex 00-7F, all 1 octet.  What we currently find 
in the draft like %h0 means the ASCII hexadecimal character 00, NUL, not 
bit 0.  %b0 will mean ASCII binary character 00000000, NUL, too.  So 
it's incorrect that ABNF is suitable for mapping its definitions to bits.

Mykyta
>    The string literal can either have no meaning at all, since the strings are ASCII [1].  Then you end up with a 1-bit terminal value which is trivially encoded.
>
> --Martin
>
> [1] Alternatively, a string literal could be redefined to map "0" to 0 and "1" to 1 with all other values invalid - ABNF isn't a law or anything.
>
>
>


From Brian.Raymor@microsoft.com  Tue Jul 12 21:58:49 2011
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From: Brian Raymor <Brian.Raymor@microsoft.com>
To: "hybi@ietf.org" <hybi@ietf.org>
Thread-Topic: -10: Typo and comment on 5.1 Client Requirements
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Subject: [hybi] -10: Typo and comment on 5.1 Client Requirements
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There is a typo in 7.1.7 - "instruted" intead of "instructed".

    7.1.7.=A0 Fail the WebSocket Connection
    ...
    An endpoint MUST NOT continue to attempt to process data (including a r=
esponding Close frame) from the remote endpoint after being instruted to _F=
ail the WebSocket Connection_.



In 5.1.  Client Requirements, there is a _Fail the WebSocket Connection_ ca=
se for Sec-WebSocket-Extensions but not for Sec-WebSocket-Protocol:

=A0 5.=A0 If the response includes a "Sec-WebSocket-Extensions" header, and
=A0=A0=A0=A0=A0=A0 this header indicates the use of an extension that was n=
ot
=A0=A0=A0=A0=A0=A0 present in the client' handshake (the server has indicat=
ed an
=A0=A0=A0=A0=A0=A0 extension not requested by the client), the client MUST =
_Fail the
=A0=A0=A0=A0=A0=A0 WebSocket Connection_.=A0 (The parsing of this header to=
 determine
=A0=A0=A0=A0=A0=A0 which extensions are requested is discussed in Section 9=
.1.)

For symmetry, should there be a similar paragraph for Sec-WebSocket-Protoco=
l -or- should the Sec-WebSocket-Extensions case be removed, since the corre=
ct behavior is also specified in the paragraph following this one:

   If the server's response does not conform to the requirements for the
   server's handshake as defined in this section and in Section 5.2.2,
   the client MUST _Fail the WebSocket Connection_.

From len.holgate@gmail.com  Wed Jul 13 01:22:09 2011
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From: "Len Holgate" <len.holgate@gmail.com>
To: "'Francis Brosnan Blazquez'" <francis@aspl.es>, "'Hybi'" <hybi@ietf.org>
References: <20110711140229.17432.23519.idtracker@ietfa.amsl.com><4E1BD054.7010103@gmail.com> <1310480036.26452.329.camel@vulcan.aspl.local>
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Cc: ietf@ietf.org, draft-ietf-hybi-thewebsocketprotocol@tools.ietf.org
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard: request for max frame size
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Francis,

I agree with your points and would welcome a max frame size negotiation
header.

However, whilst an intermediary might legitimately change the =
fragmentation
of a frame it cannot merge complete messages. If, in your example, your
client limited itself to messages of a certain size then it doesn't =
matter
what intermediaries do to the frames as the maximum frame size would be
limited to the maximum message size that the client sends. The only time
this wouldn't work is if you were using a 'message of infinite =
fragments'
where you start with a fragmented frame and don't know when you'll send =
the
final fragment.

The communication between peers about maximum message sizes supported =
could
just as easily be in the application level protocol that you're running =
over
the websocket connection.

Len
www.lenholgate.com

> -----Original Message-----
> From: hybi-bounces@ietf.org [mailto:hybi-bounces@ietf.org] On=20
> Behalf Of Francis Brosnan Blazquez
> Sent: 12 July 2011 15:14
> To: Hybi
> Cc: ietf@ietf.org; draft-ietf-hybi-thewebsocketprotocol@tools.ietf.org
> Subject: Re: [hybi] Last Call:=20
> <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket=20
> protocol) to Proposed Standard: request for max frame size
>=20
> Hi,
>=20
> Recently, I posted [1] that websocket protocol should include an
> indication about max frame size that is willing to accept the=20
> connecting
> peer.=20
>=20
> Many pointed this is not an issue because you could use a stream
> oriented API (like TCP send/recv and others), but that only=20
> bypasses the
> problem (in some cases) not solves it.
>=20
> Websocket protocol is frame based and every frame based protocol
> designed until now has/need such feature or similar. Even=20
> TCP, for those
> that proposes to use a stream oriented API as a solution, includes a
> more complex mechanism than a simple max frame size (window=20
> based ack),
> so each party can control/inform the sender. This is critical.
>=20
> A good example for this would be the next. Let suppose you have a
> constrained memory device (either because it is small or because it
> accepts a large amount of connections). On that device you deploy a
> websocket app that only accepts small messages (< 512 bytes, let's
> say).=20
>=20
> In this context, you can hard code at your web app (javascript) that
> must not send messages larger than this size (so you control websocket
> payload size) and in the case you need to, you must split them
> properly.=20
>=20
> However, even with this mechanism, you have no warranty that=20
> the browser
> or an intermediary will join those messages into a single=20
> frame, causing
> your device to receive a bigger message/frame than expected.
>=20
> Assuming this I think we would require either to include a
> Max-Frame-Size indication (or a really poor solution: to explicitly
> state on the draft that frames must not be coalesced by intermediaries
> or browsers).=20
>=20
> Regards,
>=20
> [1] http://www.ietf.org/mail-archive/web/hybi/current/msg07641.html
> --=20
> Francis Brosnan Bl=E1zquez <francis.brosnan@aspl.es>
> ASPL
> 91 134 14 22 - 91 134 14 45 - 91 116 07 57
>=20
> AVISO LEGAL
>=20
> Este mensaje se dirige exclusivamente a su destinatario. Los datos
> incluidos en el presente correo son confidenciales y=20
> sometidos a secreto
> profesional, se proh=EDbe divulgarlos, en virtud de las leyes=20
> vigentes. Si
> usted no lo es y lo ha recibido por error o tiene=20
> conocimiento del mismo
> por cualquier motivo, le rogamos que nos lo comunique por este medio y
> proceda a destruirlo o borrarlo.
>=20
> En virtud de lo dispuesto en la Ley Org=E1nica 15/1999, de 13 de
> diciembre, de Protecci=F3n de Datos de Car=E1cter Personal, le=20
> informamos de
> que sus datos de car=E1cter personal, recogidos de fuentes accesibles =
al
> p=FAblico o datos que usted nos ha facilitado previamente, proceden de
> bases de datos propiedad de Advanced Software Production Line, S.L.
> (ASPL). No obstante, usted puede ejercitar sus derechos de acceso,
> rectificaci=F3n, cancelaci=F3n y oposici=F3n dispuestos en la =
mencionada Ley
> Org=E1nica, notific=E1ndolo por escrito a:
> ASPL - Protecci=F3n Datos, C/Antonio Su=E1rez 10 A-102, 28802, =
Alcal=E1 de
> Henares (Madrid).
>=20
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>=20


From len.holgate@gmail.com  Wed Jul 13 01:22:35 2011
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From: "Len Holgate" <len.holgate@gmail.com>
To: "'Greg Wilkins'" <gregw@intalio.com>, "'Francis Brosnan Blazquez'" <francis@aspl.es>
References: <20110711140229.17432.23519.idtracker@ietfa.amsl.com><4E1BD054.7010103@gmail.com><1310480036.26452.329.camel@vulcan.aspl.local> <CAH_y2NF3pxbUJctg7P1qGNR+mvzaw80+Axh6kCYwx93HiQocug@mail.gmail.com>
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Cc: 'Hybi' <hybi@ietf.org>, ietf@ietf.org, draft-ietf-hybi-thewebsocketprotocol@tools.ietf.org
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard: request for max frame size
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I agree that this would be very useful.=20

Would this be one frame size for both directions, or could it be =
specified
in each direction?=20

I'm a little wary of intermediaries being allowed to adjust this unless
they're only allowed to reduce the amount...

Len
www.lenholgate.com

> -----Original Message-----
> From: hybi-bounces@ietf.org [mailto:hybi-bounces@ietf.org] On=20
> Behalf Of Greg Wilkins
> Sent: 13 July 2011 03:54
> To: Francis Brosnan Blazquez
> Cc: Hybi; ietf@ietf.org;=20
> draft-ietf-hybi-thewebsocketprotocol@tools.ietf.org
> Subject: Re: [hybi] Last Call:=20
> <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket=20
> protocol) to Proposed Standard: request for max frame size
>=20
> Francis et al,
>=20
> not also that the protocol does support fragmentation and a 1004 frame
> too large error.
> Even the 1004 error does not carry an indication of what an acceptable
> size is, so the client/tool/intermediary that receives a 1004 will
> either have to fail or guess a smaller frames size - potentially
> binary chopping down to find an acceptable size, which might be sub
> optimal.
>=20
> I simple optional header in the handshake declaring the max frame size
> (which intermediaries could adjust) would be very complimentary to the
> existing fragmentation and 1004 features and I can't think of any
> significant down side.
>=20
> regards
>=20
> On 13 July 2011 00:13, Francis Brosnan Blazquez=20
> <francis@aspl.es> wrote:
> > Hi,
> >
> > Recently, I posted [1] that websocket protocol should include an
> > indication about max frame size that is willing to accept=20
> the connecting
> > peer.
> >
> > Many pointed this is not an issue because you could use a stream
> > oriented API (like TCP send/recv and others), but that only=20
> bypasses the
> > problem (in some cases) not solves it.
> >
> > Websocket protocol is frame based and every frame based protocol
> > designed until now has/need such feature or similar. Even=20
> TCP, for those
> > that proposes to use a stream oriented API as a solution, includes a
> > more complex mechanism than a simple max frame size (window=20
> based ack),
> > so each party can control/inform the sender. This is critical.
> >
> > A good example for this would be the next. Let suppose you have a
> > constrained memory device (either because it is small or because it
> > accepts a large amount of connections). On that device you deploy a
> > websocket app that only accepts small messages (< 512 bytes, let's
> > say).
> >
> > In this context, you can hard code at your web app (javascript) that
> > must not send messages larger than this size (so you=20
> control websocket
> > payload size) and in the case you need to, you must split them
> > properly.
> >
> > However, even with this mechanism, you have no warranty=20
> that the browser
> > or an intermediary will join those messages into a single=20
> frame, causing
> > your device to receive a bigger message/frame than expected.
> >
> > Assuming this I think we would require either to include a
> > Max-Frame-Size indication (or a really poor solution: to explicitly
> > state on the draft that frames must not be coalesced by=20
> intermediaries
> > or browsers).
> >
> > Regards,
> >
> > [1] http://www.ietf.org/mail-archive/web/hybi/current/msg07641.html
> > --
> > Francis Brosnan Bl=E1zquez <francis.brosnan@aspl.es>
> > ASPL
> > 91 134 14 22 - 91 134 14 45 - 91 116 07 57
> >
> > AVISO LEGAL
> >
> > Este mensaje se dirige exclusivamente a su destinatario. Los datos
> > incluidos en el presente correo son confidenciales y=20
> sometidos a secreto
> > profesional, se proh=EDbe divulgarlos, en virtud de las leyes=20
> vigentes. Si
> > usted no lo es y lo ha recibido por error o tiene=20
> conocimiento del mismo
> > por cualquier motivo, le rogamos que nos lo comunique por=20
> este medio y
> > proceda a destruirlo o borrarlo.
> >
> > En virtud de lo dispuesto en la Ley Org=E1nica 15/1999, de 13 de
> > diciembre, de Protecci=F3n de Datos de Car=E1cter Personal, le=20
> informamos de
> > que sus datos de car=E1cter personal, recogidos de fuentes=20
> accesibles al
> > p=FAblico o datos que usted nos ha facilitado previamente, proceden =
de
> > bases de datos propiedad de Advanced Software Production Line, S.L.
> > (ASPL). No obstante, usted puede ejercitar sus derechos de acceso,
> > rectificaci=F3n, cancelaci=F3n y oposici=F3n dispuestos en la=20
> mencionada Ley
> > Org=E1nica, notific=E1ndolo por escrito a:
> > ASPL - Protecci=F3n Datos, C/Antonio Su=E1rez 10 A-102, 28802, =
Alcal=E1 de
> > Henares (Madrid).
> >
> > _______________________________________________
> > hybi mailing list
> > hybi@ietf.org
> > https://www.ietf.org/mailman/listinfo/hybi
> >
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>=20


From francis@aspl.es  Wed Jul 13 04:39:14 2011
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard: request for max frame size
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Hi Len,

> I agree that this would be very useful. 
> 
> Would this be one frame size for both directions, or could it be specified
> in each direction? 

It should be done in both directions, assuming each party may have
different requirements..

> I'm a little wary of intermediaries being allowed to adjust this unless
> they're only allowed to reduce the amount...

My impression is that this max frame size value would help
intermediaries, and peers, to know how to fragment or coalesce
application level messages. Without such indication they are blind...


From len.holgate@gmail.com  Wed Jul 13 05:13:22 2011
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To: "'Francis Brosnan Blazquez'" <francis@aspl.es>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard: request for max frame size
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> > Would this be one frame size for both directions, or could 
> it be specified
> > in each direction? 
> 
> It should be done in both directions, assuming each party may have
> different requirements..

That was my feeling on it.

> > I'm a little wary of intermediaries being allowed to adjust 
> this unless
> > they're only allowed to reduce the amount...
> 
> My impression is that this max frame size value would help
> intermediaries, and peers, to know how to fragment or coalesce
> application level messages. Without such indication they are blind...

I agree.

Len
www.lenholgate.com


From francis@aspl.es  Wed Jul 13 07:27:53 2011
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard: request for max frame size
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> Francis,

Hi Len,

> I agree with your points and would welcome a max frame size negotiation
> header.

Ok, It shouldn't be a negotiation but an indication to accommodate
communication to each party.

> However, whilst an intermediary might legitimately change the fragmentation
> of a frame it cannot merge complete messages. If, in your example, your
> client limited itself to messages of a certain size

Ok, it is important to note the peer is limiting max frame size not the
message size which may be still infinite in practical terms (63
bits)....and we have fragmentation support to deal with this.

This way intermediaries can split/coalesce fragments according to
receiving peer expectation.

>  then it doesn't matter
> what intermediaries do to the frames as the maximum frame size would be
> limited to the maximum message size that the client sends. The only time
> this wouldn't work is if you were using a 'message of infinite fragments'
> where you start with a fragmented frame and don't know when you'll send the
> final fragment.

Assuming frame size and message size aren't equal, it does matter. 

> The communication between peers about maximum message sizes supported could
> just as easily be in the application level protocol that you're running over
> the websocket connection.

Ok, I see you point and in part you are right, but there is a risk with
this argument:

1) Protocol problems should be solved on its layer, otherwise it will
   cause next layer (application level in this case) to need to solve
   them. It looks reasonable not forcing people to do so.

     Side note: our intention is to layer BEEP on top of websocket, so,
     in practical terms this is not a problem for us because it will
     provide us with all the protection and flow control required.

     However it looks to me this is not a solution for people using
     websocket in other contexts.

2) In general, it has lot of benefits to fragment bigger messages into
   smaller pieces to avoid sick interactions. 

   For example, if you send a really large frame, you won't be able to
   send a control message in the middle until you finish sending the
   frame in place. 

   Having a max frame size indication will help websocket stacks to
   fragment using the right value for each application.

3) No matter API style provided by a websocket stack (frame indication
or stream oriented), having framing running in the background where the
sending peer and intermediaries can coalesce frames without any
indication of the upper level will cause problems that can't be solved
in practical terms by a receiving side. 

In other words, I think having a framing without a max frame size
mechanism (or other mech like ack confirmation, its the same) is like
pretending having a coin with only one side.

> Len
> www.lenholgate.com
> 
> > -----Original Message-----
> > From: hybi-bounces@ietf.org [mailto:hybi-bounces@ietf.org] On 
> > Behalf Of Francis Brosnan Blazquez
> > Sent: 12 July 2011 15:14
> > To: Hybi
> > Cc: ietf@ietf.org; draft-ietf-hybi-thewebsocketprotocol@tools.ietf.org
> > Subject: Re: [hybi] Last Call: 
> > <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket 
> > protocol) to Proposed Standard: request for max frame size
> > 
> > Hi,
> > 
> > Recently, I posted [1] that websocket protocol should include an
> > indication about max frame size that is willing to accept the 
> > connecting
> > peer. 
> > 
> > Many pointed this is not an issue because you could use a stream
> > oriented API (like TCP send/recv and others), but that only 
> > bypasses the
> > problem (in some cases) not solves it.
> > 
> > Websocket protocol is frame based and every frame based protocol
> > designed until now has/need such feature or similar. Even 
> > TCP, for those
> > that proposes to use a stream oriented API as a solution, includes a
> > more complex mechanism than a simple max frame size (window 
> > based ack),
> > so each party can control/inform the sender. This is critical.
> > 
> > A good example for this would be the next. Let suppose you have a
> > constrained memory device (either because it is small or because it
> > accepts a large amount of connections). On that device you deploy a
> > websocket app that only accepts small messages (< 512 bytes, let's
> > say). 
> > 
> > In this context, you can hard code at your web app (javascript) that
> > must not send messages larger than this size (so you control websocket
> > payload size) and in the case you need to, you must split them
> > properly. 
> > 
> > However, even with this mechanism, you have no warranty that 
> > the browser
> > or an intermediary will join those messages into a single 
> > frame, causing
> > your device to receive a bigger message/frame than expected.
> > 
> > Assuming this I think we would require either to include a
> > Max-Frame-Size indication (or a really poor solution: to explicitly
> > state on the draft that frames must not be coalesced by intermediaries
> > or browsers). 
> > 
> > Regards,
> > 
> > [1] http://www.ietf.org/mail-archive/web/hybi/current/msg07641.html
> > -- 
> > Francis Brosnan Blázquez <francis.brosnan@aspl.es>
> > ASPL
> > 91 134 14 22 - 91 134 14 45 - 91 116 07 57
> > 
> > AVISO LEGAL
> > 
> > Este mensaje se dirige exclusivamente a su destinatario. Los datos
> > incluidos en el presente correo son confidenciales y 
> > sometidos a secreto
> > profesional, se prohíbe divulgarlos, en virtud de las leyes 
> > vigentes. Si
> > usted no lo es y lo ha recibido por error o tiene 
> > conocimiento del mismo
> > por cualquier motivo, le rogamos que nos lo comunique por este medio y
> > proceda a destruirlo o borrarlo.
> > 
> > En virtud de lo dispuesto en la Ley Orgánica 15/1999, de 13 de
> > diciembre, de Protección de Datos de Carácter Personal, le 
> > informamos de
> > que sus datos de carácter personal, recogidos de fuentes accesibles al
> > público o datos que usted nos ha facilitado previamente, proceden de
> > bases de datos propiedad de Advanced Software Production Line, S.L.
> > (ASPL). No obstante, usted puede ejercitar sus derechos de acceso,
> > rectificación, cancelación y oposición dispuestos en la mencionada Ley
> > Orgánica, notificándolo por escrito a:
> > ASPL - Protección Datos, C/Antonio Suárez 10 A-102, 28802, Alcalá de
> > Henares (Madrid).
> > 
> > _______________________________________________
> > hybi mailing list
> > hybi@ietf.org
> > https://www.ietf.org/mailman/listinfo/hybi
> > 
> 

-- 
Francis Brosnan Blázquez <francis.brosnan@aspl.es>
ASPL
91 134 14 22 - 91 134 14 45 - 91 116 07 57

AVISO LEGAL

Este mensaje se dirige exclusivamente a su destinatario. Los datos
incluidos en el presente correo son confidenciales y sometidos a secreto
profesional, se prohíbe divulgarlos, en virtud de las leyes vigentes. Si
usted no lo es y lo ha recibido por error o tiene conocimiento del mismo
por cualquier motivo, le rogamos que nos lo comunique por este medio y
proceda a destruirlo o borrarlo.

En virtud de lo dispuesto en la Ley Orgánica 15/1999, de 13 de
diciembre, de Protección de Datos de Carácter Personal, le informamos de
que sus datos de carácter personal, recogidos de fuentes accesibles al
público o datos que usted nos ha facilitado previamente, proceden de
bases de datos propiedad de Advanced Software Production Line, S.L.
(ASPL). No obstante, usted puede ejercitar sus derechos de acceso,
rectificación, cancelación y oposición dispuestos en la mencionada Ley
Orgánica, notificándolo por escrito a:
ASPL - Protección Datos, C/Antonio Suárez 10 A-102, 28802, Alcalá de
Henares (Madrid).


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From: Barry Leiba <barryleiba@computer.org>
To: Mykyta Yevstifeyev <evnikita2@gmail.com>
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Cc: "hybi@ietf.org" <hybi@ietf.org>, "draft-ietf-hybi-thewebsocketprotocol@tools.ietf.org" <draft-ietf-hybi-thewebsocketprotocol@tools.ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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>>>> BIT =A0 =A0 =A0 =A0 =A0 =A0=3D =A0"0" / "1"
>>>
>>> implies that<BIT> =A0may be either ASCII 0x31 - "1" - or ASCII 0x30 - "=
0".
>>> =A0Once more, ABNF is for chars/octets, not bits.

Mykyta, be careful not to confuse the code for generating ABNF (for
which what you say above is correct) with the *use* of ABNF in
protocol specifications (for which you're wrong).

The construct above, << BIT =3D "0" / "1" >>, says that when you're
writing ABNF, you use the characters "0" and "1" (codes 30 and 31) to
represent the bit values.  But when we apply that ABNF in a protocol,
we can get any of these:

the-letter-c =3D %b01100011  ; also %x63

halfbyte-value-3 =3D %b0011  ; also %x3

two-bits-both-set =3D %b11

a-single-one-bit-by-itself =3D %b1

-----
%x3 and %x03 are not the same thing, and %b00000011, %b0011, and %b11
are not the same things.

Barry

From evnikita2@gmail.com  Wed Jul 13 10:08:47 2011
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To: Barry Leiba <barryleiba@computer.org>
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Cc: "hybi@ietf.org" <hybi@ietf.org>, "draft-ietf-hybi-thewebsocketprotocol@tools.ietf.org" <draft-ietf-hybi-thewebsocketprotocol@tools.ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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13.07.2011 17:41, Barry Leiba wrote:
>>>>> BIT            =  "0" / "1"
>>>> implies that<BIT>    may be either ASCII 0x31 - "1" - or ASCII 0x30 - "0".
>>>>   Once more, ABNF is for chars/octets, not bits.
> Mykyta, be careful not to confuse the code for generating ABNF (for
> which what you say above is correct) with the *use* of ABNF in
> protocol specifications (for which you're wrong).
>
> The construct above,<<  BIT = "0" / "1">>, says that when you're
> writing ABNF, you use the characters "0" and "1" (codes 30 and 31) to
> represent the bit values.  But when we apply that ABNF in a protocol,
> we can get any of these:
>
> the-letter-c = %b01100011  ; also %x63
>
> halfbyte-value-3 = %b0011  ; also %x3
>
> two-bits-both-set = %b11
>
> a-single-one-bit-by-itself = %b1
>
> -----
> %x3 and %x03 are not the same thing, and %b00000011, %b0011, and %b11
> are not the same things.

I can't agree with you here.  ABNF treats anything after %<letter> as 
integer, which is stated in Section 2.3 of RFC 5234:

>     Rules resolve into a string of terminal values, sometimes called
>     characters.  In ABNF, a character is merely a non-negative integer.

so leading zeros are ignored, as the have no meaning in integers.  
Moreover, and I'll repeat this for the nth time, ABNF deals with 
characters.  So both %b0011 and %b11, and %b00000011 mean the terminal 
value denoting the character under binary code 11 (octet 00000011b) in 
ASCII, BEL char, as far as I see.  RFC 5234 doesn't imply that ABNF may 
be employed to denote bits, half- or parts of bytes, etc.

Mykyta
>
> Barry
>


From len.holgate@gmail.com  Wed Jul 13 11:36:15 2011
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From: "Len Holgate" <len.holgate@gmail.com>
To: "'Francis Brosnan Blazquez'" <francis@aspl.es>
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Cc: 'Hybi' <hybi@ietf.org>, ietf@ietf.org, draft-ietf-hybi-thewebsocketprotocol@tools.ietf.org
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard: request for max frame size
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Francis,

> Ok, It shouldn't be a negotiation but an indication to accommodate
> communication to each party.

Fair point.

> Ok, it is important to note the peer is limiting max frame 
> size not the
> message size which may be still infinite in practical terms (63
> bits)....and we have fragmentation support to deal with this.

I understand. I was just pointing out that in your example, if the client
instead decided to limit the size of messages that it sent then then nothing
could cause any frames to be larger than that limit when they arrived at the
server.

> 1) Protocol problems should be solved on its layer, otherwise it will
>    cause next layer (application level in this case) to need to solve
>    them. It looks reasonable not forcing people to do so.

Agree.

> 2) In general, it has lot of benefits to fragment bigger messages into
>    smaller pieces to avoid sick interactions. 

Agree.

> 3) No matter API style provided by a websocket stack (frame indication
> or stream oriented), having framing running in the background 
> where the
> sending peer and intermediaries can coalesce frames without any
> indication of the upper level will cause problems that can't be solved
> in practical terms by a receiving side. 

Agree.

Len
www.lenholgate.com


From salvatore.loreto@ericsson.com  Sun Jul 17 14:28:32 2011
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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: [hybi] IETF81 draft agenda
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Hi there,

here just a preliminary draft of the agenda
( http://www.ietf.org/proceedings/81/agenda/hybi.txt )

   Thursday, July 28th, 2011
   17:40 - 19:40 Afternoon Session II
   room 204 B

   -  Administrative/Agenda bash                                (Chairs - 10m)

   -  status and other business about the finalization
      of the main spec                                          (Ian Fette? - 40m)
      (raft-ietf-hybi-thewebsocketprotocol-10)

   -  extensions and other options :			       (? - 60m)
      (e.g. among the others: Multiplex, Frame Compression,
       Timeout etc.)

   -  Open Discussion                                            (All - 10m)



cheers
/Sal

-- 
Salvatore Loreto
www.sloreto.com


From mnot@mnot.net  Mon Jul 18 00:07:12 2011
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From: Mark Nottingham <mnot@mnot.net>
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Subject: [hybi] Questions and comments on draft-hybi-thewebsocketprotocol-10
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Hi,

I've just read through -10, and wanted to ask a few questions. These are =
NOT (yet) Last Call comments, as I haven't kept up with the hybi list =
for a while, and haven't reviewed a draft since around -06.=20

In other words, I'd like to understand things a bit more before making =
LC comments; there may be good reasons for the few things I saw that =
raised my eyebrows.

1) I missed the end of the handshake saga; can someone speak to why GET =
was chosen over, say, OPTIONS? Roy seems to have strong feelings about =
that (see recent discussion on HTTPbis).

2) The Upgrade token has no version; e.g., from the examples in 1.2:
    Upgrade: websocket
Why? The protocol version seems to be carried in the =
Sec-WebSocket-Version header; could it not be moved (or copied) to the =
upgrade token?

3) The last paragraph of 1.2 seems to be a bit of a non-sequiter; all =
protocols need framing, not just those used by event-driven =
implementations.

4) Section 3 defines two new URI schemes for WebSockets, but the =
handshake is entirely HTTP-based, and the default ports are HTTP's. Why =
is a new URI scheme needed here? I understand that client software needs =
something to trigger the upgrade from, but why use the scheme, instead =
of (for example) the API? Is it expected that WS: URIs will be used in =
other places URIs are used, such as in HTML elements like A / IMG / =
SCRIPT, or in browser location bars?

Having two URL schemes use the same default port doesn't feel right.

5) In the diagram at the start of 4.2, the dotted horizontal line in the =
left hand side of the "extended payload length continued" is a bit =
confusing, I think. YMMV.

6) In section 9.1., private-use-token is defined as having an "x-" =
prefix. This goes against emerging BCP, see draft-saintandre-xdash.

7) The contact field for registry entries in 11.13 is listed as =
'hybi@ietf.org'. Is that list going to persist? Common practice is to =
use iesg@ietf.org AIUI.

Cheers,

--
Mark Nottingham   http://www.mnot.net/




From gregw@intalio.com  Mon Jul 18 00:50:30 2011
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Subject: Re: [hybi] Questions and comments on draft-hybi-thewebsocketprotocol-10
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On 18 July 2011 17:06, Mark Nottingham <mnot@mnot.net> wrote:
> In other words, I'd like to understand things a bit more before making LC=
 comments; there may be good reasons for the few things I saw that raised m=
y eyebrows.

_this response is non-normative_

> 1) I missed the end of the handshake saga; can someone speak to why GET w=
as chosen over, say, OPTIONS? Roy seems to have strong feelings about that =
(see recent discussion on HTTPbis).

I think we went with GET because Roy was not able to well communicate
his concerns.   He's done a better job in the recent thread in
httpbis, and I now understand that he is concerned about the
outstanding semantic GET.   However I still don't understand why he
thinks OPTION is any better, as it still is a request that requires a
semantic response.   In either case I think that websockets should be
free to define an implicit response (eg 204 No Content) is represented
by a successful handshake.


> 2) The Upgrade token has no version; e.g., from the examples in 1.2:
> =A0 =A0Upgrade: websocket
> Why? The protocol version seems to be carried in the Sec-WebSocket-Versio=
n header; could it not be moved (or copied) to the upgrade token?

I think we are only just coming around to the acceptance that we
really do need a version beyond the draft stage.  The version was put
in a header as it was originally only considered needed to distinguish
drafts.  If it is going to be there to stay, then perhaps the
protocol/version format would be better.



regards

From w@1wt.eu  Mon Jul 18 01:08:03 2011
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From: Willy Tarreau <w@1wt.eu>
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Subject: Re: [hybi] Questions and comments on draft-hybi-thewebsocketprotocol-10
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On Mon, Jul 18, 2011 at 05:50:28PM +1000, Greg Wilkins wrote:
> On 18 July 2011 17:06, Mark Nottingham <mnot@mnot.net> wrote:
> > In other words, I'd like to understand things a bit more before making LC comments; there may be good reasons for the few things I saw that raised my eyebrows.
> 
> _this response is non-normative_
> 
> > 1) I missed the end of the handshake saga; can someone speak to why GET was chosen over, say, OPTIONS? Roy seems to have strong feelings about that (see recent discussion on HTTPbis).
> 
> I think we went with GET because Roy was not able to well communicate
> his concerns.   He's done a better job in the recent thread in
> httpbis, and I now understand that he is concerned about the
> outstanding semantic GET.

I would add that there was a GET very early in the specification, when
the protocol was not even HTTP compatible yet. It dates back the era
where it was desirable to use an HTTP-like protocol that was easy to
parse by a simple script. Now we're more in line with HTTP and despite
all the discussions on GET vs OPTIONS, it appeared that there was no
real preference for one vs the other, so it was suggested that this
would not change unless there is a valid reason.

> However I still don't understand why he
> thinks OPTION is any better, as it still is a request that requires a
> semantic response.   In either case I think that websockets should be
> free to define an implicit response (eg 204 No Content) is represented
> by a successful handshake.

Alternatively we could consider that the server's hello is the response
to the request.

> > 2) The Upgrade token has no version; e.g., from the examples in 1.2:
> >    Upgrade: websocket
> > Why? The protocol version seems to be carried in the Sec-WebSocket-Version header; could it not be moved (or copied) to the upgrade token?
> 
> I think we are only just coming around to the acceptance that we
> really do need a version beyond the draft stage.  The version was put
> in a header as it was originally only considered needed to distinguish
> drafts.  If it is going to be there to stay, then perhaps the
> protocol/version format would be better.

I too find it more useful to have the version in the Upgrade header than
anywhere else. For intermediaries it can be very useful to know the version
from the Upgrade header because they can decide what processing to start
depending on what they see there.

Regards,
Willy


From ibc@aliax.net  Mon Jul 18 01:54:10 2011
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2011/7/18 Mark Nottingham <mnot@mnot.net>:

> 2) The Upgrade token has no version; e.g., from the examples in 1.2:
> =C2=A0 =C2=A0Upgrade: websocket
> Why? The protocol version seems to be carried in the Sec-WebSocket-Versio=
n header; could it not be moved (or copied) to the upgrade token?

I agree. But such version must be the definitive version (rather than
the draft revision or similar). So for example I suggest:

  Upgrade: websocket/1.0

The version "1.0" should not change in years until a complete new
WebSocket spec borns ("2.0").



> 4) Section 3 defines two new URI schemes for WebSockets, but the handshak=
e is entirely HTTP-based, and the default ports are HTTP's. Why is a new UR=
I scheme needed here? I understand that client software needs something to =
trigger the upgrade from, but why use the scheme, instead of (for example) =
the API? Is it expected that WS: URIs will be used in other places URIs are=
 used, such as in HTML elements like A / IMG / SCRIPT, or in browser locati=
on bars?

An URI is useful as it tells you in a single line the transport to use
("ws" =3D> TCP, "wss" =3D> TLS over TCP), the domain to resolve (or IP)
and optionally the port (or default port 80/443 if not present). Also,
in case of including DNS SRV support [*] an URI is the best way to
extract the data to query.

[*] http://tools.ietf.org/html/draft-ibc-websocket-dns-srv-02



> Having two URL schemes use the same default port doesn't feel right.

That's not true. Current websocket draft states that "ws" schema uses
port 80 by default while "wss" uses 443 (unless explicitly present in
the URI).



> 6) In section 9.1., private-use-token is defined as having an "x-" prefix=
. This goes against emerging BCP, see draft-saintandre-xdash.

I agree. "Those Who Ignore History are Doomed to Repeat It".
I expect authors would take care of it.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From alexey.melnikov@isode.com  Mon Jul 18 07:36:39 2011
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Mark Nottingham wrote:

>Hi,
>  
>
Hi Mark,
Answering to your questions selectively:

>I've just read through -10, and wanted to ask a few questions. These are NOT (yet) Last Call comments, as I haven't kept up with the hybi list for a while, and haven't reviewed a draft since around -06. 
>
>In other words, I'd like to understand things a bit more before making LC comments; there may be good reasons for the few things I saw that raised my eyebrows.
>  
>
 [...]

>2) The Upgrade token has no version; e.g., from the examples in 1.2:
>    Upgrade: websocket
>Why? The protocol version seems to be carried in the Sec-WebSocket-Version header; could it not be moved (or copied) to the upgrade token?
>
What is the value in duplicating the version in the Upgrade token?

>3) The last paragraph of 1.2 seems to be a bit of a non-sequiter; all protocols need framing, not just those used by event-driven implementations.
>
Yes, this can be reworded not to mention event-driven implementations.

 [...]

>7) The contact field for registry entries in 11.13 is listed as 'hybi@ietf.org'. Is that list going to persist?
>
Yes. At least I would be surprised if it wouldn't.

>Common practice is to use iesg@ietf.org AIUI.  
>
I think leave the WG mailing list is Ok. I wouldn't get stuck on this 
and if IESG thinks otherwise, we can change.


From w@1wt.eu  Mon Jul 18 07:46:34 2011
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Hi Alexey,

On Mon, Jul 18, 2011 at 03:35:58PM +0100, Alexey Melnikov wrote:
> >2) The Upgrade token has no version; e.g., from the examples in 1.2:
> >   Upgrade: websocket
> >Why? The protocol version seems to be carried in the Sec-WebSocket-Version 
> >header; could it not be moved (or copied) to the upgrade token?
> >
> What is the value in duplicating the version in the Upgrade token?

I think it can be a easier for some intermediaries to decide what action to
take based on the Upgrade header alone, especially if a client supports 2
versions and the server responds with the selected one.

Regards,
Willy


From stpeter@stpeter.im  Mon Jul 18 12:55:55 2011
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If you have an interest, please review draft-ietf-websec-origin and send 
your feedback to the websec@ietf.org list:

https://www.ietf.org/mailman/listinfo/websec

Thanks!

/psa

-------- Original Message --------
Subject: [websec] WG Last Call on draft-ietf-websec-origin-02 until Aug-15
Date: Mon, 18 Jul 2011 20:38:04 +0100
From: Tobias Gondrom <tobias.gondrom@gondrom.org>
To: websec@ietf.org

Hello dear websec fellows,

after reading the feedback and the update on the origin draft, I have
the impression that the draft is in good shape and like to ask for WG
Last Call for this document:
http://tools.ietf.org/html/draft-ietf-websec-origin-02

As we are close to the IETF meeting in Quebec, this last call will be
extended to four weeks and _*close on August-15.*_ Please make a last
careful review of the draft and submit comments, questions and discuss
items for this draft ASAP. If you perceive any major issues, it might
also make sense to raise them during our meeting in Quebec on July-25.

Kind regards and thank you,

Tobias
chair of websec


Tobias Gondrom
email: tobias.gondrom@gondrom.org
mobile: +447521003005
_______________________________________________
websec mailing list
websec@ietf.org
https://www.ietf.org/mailman/listinfo/websec

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Subject: Re: [hybi] Questions and comments on draft-hybi-thewebsocketprotocol-10
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On Jul 18, 2011, at 12:50 AM, Greg Wilkins wrote:
> On 18 July 2011 17:06, Mark Nottingham <mnot@mnot.net> wrote:
>> 1) I missed the end of the handshake saga; can someone speak to why =
GET was chosen over, say, OPTIONS? Roy seems to have strong feelings =
about that (see recent discussion on HTTPbis).
>=20
> I think we went with GET because Roy was not able to well communicate
> his concerns.   He's done a better job in the recent thread in
> httpbis, and I now understand that he is concerned about the
> outstanding semantic GET.   However I still don't understand why he
> thinks OPTION is any better, as it still is a request that requires a
> semantic response.  =20

OPTIONS is a request for the communication options on the connection
for the target resource.  It is not a request of the resource itself;
it is a meta-request to be processed by the handler of that resource.
It should be obvious why completing the WebSocket handshake is a valid
response to that request if the Upgrade succeeds.  In any case, see

  http://www.ietf.org/mail-archive/web/hybi/current/msg05743.html
  http://www.ietf.org/mail-archive/web/hybi/current/msg05750.html
  http://www.ietf.org/mail-archive/web/hybi/current/msg05751.html

to wonder at how I failed to communicate my concerns.

> In either case I think that websockets should be
> free to define an implicit response (eg 204 No Content) is represented
> by a successful handshake.

GET has specific semantics regarding the state of the resource that is
the target of this request.  If you want to use HTTP over the Web's
standard port, then you are subject to HTTP's definitions.  It is not
that difficult for WebSocket to correctly use GET, as Henrik Frystyk =
Nielsen
just explained on httpbis, if the target resource is meaningful and not
some generic entry point (like those stupid SOAP gateways).

There is no need for an implicit response of "no content" if the result
of making the WS request via HTTP has application-level semantics that
are equivalent to a response to GET after WS takes over the connection.
For example, if the request results in an asynchronous event feed (the
feed named by that URI), joins the user to a running game (named by
that URI), connects the client to the video stream corresponding
to the eyes of a mini robot (named by that URI), etc.  WebSockets
merely needs to state that the client's HTTP request remains
outstanding after the 101 is sent and the WebSockets application is
responsible for handling and responding to that request in accordance
with the semantics of the target resource.

If that's too much to ask, WebSockets is welcome to define its own
unique method, use OPTIONS (and let the framing be the response), or
use a different port and a different protocol for the handshake.

....Roy=

From gregw@intalio.com  Mon Jul 18 23:20:37 2011
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On 19 July 2011 14:49, Roy T. Fielding <fielding@gbiv.com> wrote:
> On Jul 18, 2011, at 12:50 AM, Greg Wilkins wrote:
>> I think we went with GET because Roy was not able to well communicate
>> his concerns.
> [...]
> =A0In any case, see [...] to wonder at how I failed to communicate my con=
cerns.

Roy,
I didn't mean to imply that you didn't try to communicate your
concerns, nor that you were to blame for the failure of that attempt.
It may well be that I/we were not listening correctly.


> If you want to use HTTP over the Web's
> standard port, then you are subject to HTTP's definitions.

and that is something I fully agree with and have advocated long and
hard for in this WG.


>=A0It is not
> that difficult for WebSocket to correctly use GET, as Henrik Frystyk Niel=
sen
> just explained on httpbis, if the target resource is meaningful and not
> some generic entry point (like those stupid SOAP gateways).

I believe that websocket usage will be just the same as other
webapplication usage in this regard.  Some will correctly reference
specific resources, while others will not.  I don't think the spec can
mandate against bad URI space design.


> There is no need for an implicit response of "no content" if the result
> of making the WS request via HTTP has application-level semantics that
> are equivalent to a response to GET after WS takes over the connection.
> For example, if the request results in an asynchronous event feed (the
> feed named by that URI), joins the user to a running game (named by
> that URI), connects the client to the video stream corresponding
> to the eyes of a mini robot (named by that URI), etc. =A0WebSockets
> merely needs to state that the client's HTTP request remains
> outstanding after the 101 is sent and the WebSockets application is
> responsible for handling and responding to that request in accordance
> with the semantics of the target resource.

Ok I think I'm getting you more now.   It is not that you want
WebSockets to give an immediate semantic final response to the GET
request, rather you don't want websocket to consider the 101 to be
that final response because it is by definition a non final response.
So you say that this could be resolved by the spec making it clear
that the event stream accesses by the ws handshake is semantically the
response to the handshake GET?

regards

From ibc@aliax.net  Tue Jul 19 04:51:18 2011
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/11 The IESG <iesg-secretary@ietf.org>:
> The IESG has received a request from the BiDirectional or
> Server-Initiated HTTP WG (hybi) to consider the following document:
> - 'The WebSocket protocol'
> =C2=A0<draft-ietf-hybi-thewebsocketprotocol-10.txt> as a Proposed Standar=
d

Hi, I assume there is no interest in making DNS SRV mechanism exposed
in http://tools.ietf.org/html/draft-ibc-websocket-dns-srv-02 part of
the WebSocket core specification, neither referencing it (in the same
way RFC 3261 "SIP protocol" mandates the usage of RFC 3263 "Locating
SIP servers").

As said before, making such DNS SRV specification an extension (so
present in other document) will mean no success at all, as WebSocket
client implementors (i.e. webbrowser vendors) will not be mandated to
implement it and service providers could not rely on the support of
DNS SRV in web browsers. So nobody will use them (because IE10 decided
not to implement it, for example). IMHO this is sad due the real
advantages DNS SRV provides for a protocol like WebSocket.

Yes, in HTTP there is no special DNS stuff, all the load-balancing and
failover mechanism are done at server side with very complex and
expensive solutions (www.facebook.com resolves to a single IPv4 !!!!).
The question is: should we also inherit every HTTP limitation in
WebSocket?

Thanks.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From francis@aspl.es  Tue Jul 19 05:51:42 2011
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> As said before, making such DNS SRV specification an extension (so
> present in other document) will mean no success at all, as WebSocket
> client implementors (i.e. webbrowser vendors) will not be mandated to
> implement it and service providers could not rely on the support of
> DNS SRV in web browsers. So nobody will use them (because IE10 decided
> not to implement it, for example). IMHO this is sad due the real
> advantages DNS SRV provides for a protocol like WebSocket.
> 
> Yes, in HTTP there is no special DNS stuff, all the load-balancing and
> failover mechanism are done at server side with very complex and
> expensive solutions (www.facebook.com resolves to a single IPv4 !!!!).
> The question is: should we also inherit every HTTP limitation in
> WebSocket?

+1

It is little the effort for web browser implementors and will provide
lot of benefits to end users and developers which would be able to
provide scalable/configurable services easily. I'm with Iñaki: its usage
must be mandatory to really make it available in all browsers...



From dave@cridland.net  Tue Jul 19 06:26:54 2011
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From: Dave Cridland <dave@cridland.net>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Tue Jul 19 12:51:16 2011, Iñaki Baz Castillo wrote:
> 2011/7/11 The IESG <iesg-secretary@ietf.org>:
> > The IESG has received a request from the BiDirectional or
> > Server-Initiated HTTP WG (hybi) to consider the following  
> document:
> > - 'The WebSocket protocol'
> >  <draft-ietf-hybi-thewebsocketprotocol-10.txt> as a Proposed  
> Standard
> 
> Hi, I assume there is no interest in making DNS SRV mechanism  
> exposed
> in http://tools.ietf.org/html/draft-ibc-websocket-dns-srv-02 part of
> the WebSocket core specification, neither referencing it (in the  
> same
> way RFC 3261 "SIP protocol" mandates the usage of RFC 3263 "Locating
> SIP servers").
> 
> As said before, making such DNS SRV specification an extension (so
> present in other document) will mean no success at all, as WebSocket
> client implementors (i.e. webbrowser vendors) will not be mandated  
> to
> implement it and service providers could not rely on the support of
> DNS SRV in web browsers. So nobody will use them (because IE10  
> decided
> not to implement it, for example). IMHO this is sad due the real
> advantages DNS SRV provides for a protocol like WebSocket.
> 
> Yes, in HTTP there is no special DNS stuff, all the load-balancing  
> and
> failover mechanism are done at server side with very complex and
> expensive solutions (www.facebook.com resolves to a single IPv4  
> !!!!).
> The question is: should we also inherit every HTTP limitation in
> WebSocket?

I agree wholeheartedly with this, and strongly recommend that  
mandatory use of SRV is included in the core protocol.

I think with HTTP's very short lived requests, then it's possible to  
work around the lack of SRV support (at a cost), but the benefit is  
markedly higher with the long-lived, stateful sessions we're  
anticipating with WebSocket.

Dave.
-- 
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From gregw@intalio.com  Tue Jul 19 19:34:42 2011
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Subject: Re: [hybi] deflate-stream and masking
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I've just noticed that the w3c is currently intending to make support
for deflate-stream mandatory!

  http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917

This moves this extension from being useless, but mostly harmless, to
being a major impost on servers and intermediaries.
If the browser make this mandatory, then servers will obviously have
to support it at a cost of extra CPU, extra buffers but for no
significant savings in bandwidth.
Intermediaries that wish to act on frame boundaries will also have to
implement it.

This illustrate that having silly options always puts you at risk of
people taking you up on those options.

This extension is demonstrably broken and needs to be either fixed or remov=
ed.

regards



On 20 June 2011 16:33, Greg Wilkins <gregw@intalio.com> wrote:
> As part of my continuing campaign against including deflate-stream in
> the specification as a standard extension, I did a quick test of how
> well it works when applied to masked frames.
>
> I took a days worth of traffic from an IRC channel and wrapped it up
> as JSON messages sent as websocket frames.
> There were 487 message that looked like:
>
> =A0 =A0 {channel:"#webtide", username:"tbecker", text:"joakime: jenkins
> had issues pulling from github a couple of times =A0last week"}
>
> As an unmasked WS stream, it was 50675 bytes, and as a masked stream
> is was 52623 bytes.
> I then compressed both these streams with gzip and got 13306 bytes for
> unmasked and 51704 bytes for the masked!!!!
>
> So for this very typical example, masking was sufficiently random to
> completely negate the benefits of compression.
>
> So the deflate-stream "extension" is:
>
> =A0+ next to useless for inbound traffic
> =A0+ breaks all the rules of what an extension can do
> =A0+ is potentially vulnerable to injection as attackers can send
> repeated patterns that may subvert masking
> =A0+ can be replaced by the in-frame compression extension already propos=
ed.
> =A0+ was inserted in the draft with little or no discussion and without
> clear consensus.
>
> Can I call for a straw poll of who wants to keep this extension in the sp=
ec?
>
>
>
> regards
>

From wsong.cn@gmail.com  Tue Jul 19 23:27:52 2011
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Subject: [hybi] hybi 10 ---- server to client masking
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--000e0cd5743aec838404a87a543f
Content-Type: text/plain; charset=UTF-8

Hi,

A question about server to client masking. In section 4.1 it states
that "Frames
sent from the server to the client are not masked." Does it mean server to
client masking is optional or MUST NOT?

Client to server is clear as "MUST" though.

Thanks
Andy

--000e0cd5743aec838404a87a543f
Content-Type: text/html; charset=UTF-8
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Hi,=C2=A0<div><br></div><div>A question about server to client masking. In =
section 4.1 it states that &quot;<span class=3D"Apple-style-span" style=3D"=
font-family: monospace; font-size: 16px; white-space: pre; ">Frames sent fr=
om the server to the client are not masked.</span>&quot; Does it mean serve=
r to client masking is optional or MUST NOT?=C2=A0</div>

<div><br></div><div>Client to server is clear as &quot;MUST&quot; though.=
=C2=A0</div><div><br></div><div>Thanks</div><div>Andy</div>

--000e0cd5743aec838404a87a543f--

From ibc@aliax.net  Wed Jul 20 02:26:23 2011
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Subject: Re: [hybi] hybi 10 ---- server to client masking
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2011/7/20 Andy W. Song <wsongcn@gmail.com>:
> A question about server to client masking. In section 4.1 it states that
> "Frames sent from the server to the client are not masked." Does it mean
> server to client masking is optional or MUST NOT?
> Client to server is clear as "MUST" though.

"are not masked" means that they MUST NOT be masked.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From theturtle32@gmail.com  Wed Jul 20 02:50:14 2011
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+500.  deflate-stream has always been utterly ridiculous in light of
masking.  It really should get the axe, and with extreme prejudice.  Why is
it still in the spec?  I don't recall anyone citing a reasonable reason for
keeping it, but there have been many very good arguments against it.  It's
not even very clearly specified in the document -- when I implemented it in
my Flash client, I had to read the source code of Andy Green's
implementation to figure out how it was supposed to work -- his
implementation became my specification for the extension.

Brian


On Tue, Jul 19, 2011 at 7:34 PM, Greg Wilkins <gregw@intalio.com> wrote:

> I've just noticed that the w3c is currently intending to make support
> for deflate-stream mandatory!
>
>  http://www.w3.org/Bugs/Public/show_bug.cgi?id=12917
>
> This moves this extension from being useless, but mostly harmless, to
> being a major impost on servers and intermediaries.
> If the browser make this mandatory, then servers will obviously have
> to support it at a cost of extra CPU, extra buffers but for no
> significant savings in bandwidth.
> Intermediaries that wish to act on frame boundaries will also have to
> implement it.
>
> This illustrate that having silly options always puts you at risk of
> people taking you up on those options.
>
> This extension is demonstrably broken and needs to be either fixed or
> removed.
>
> regards
>
>
>
> On 20 June 2011 16:33, Greg Wilkins <gregw@intalio.com> wrote:
> > As part of my continuing campaign against including deflate-stream in
> > the specification as a standard extension, I did a quick test of how
> > well it works when applied to masked frames.
> >
> > I took a days worth of traffic from an IRC channel and wrapped it up
> > as JSON messages sent as websocket frames.
> > There were 487 message that looked like:
> >
> >     {channel:"#webtide", username:"tbecker", text:"joakime: jenkins
> > had issues pulling from github a couple of times  last week"}
> >
> > As an unmasked WS stream, it was 50675 bytes, and as a masked stream
> > is was 52623 bytes.
> > I then compressed both these streams with gzip and got 13306 bytes for
> > unmasked and 51704 bytes for the masked!!!!
> >
> > So for this very typical example, masking was sufficiently random to
> > completely negate the benefits of compression.
> >
> > So the deflate-stream "extension" is:
> >
> >  + next to useless for inbound traffic
> >  + breaks all the rules of what an extension can do
> >  + is potentially vulnerable to injection as attackers can send
> > repeated patterns that may subvert masking
> >  + can be replaced by the in-frame compression extension already
> proposed.
> >  + was inserted in the draft with little or no discussion and without
> > clear consensus.
> >
> > Can I call for a straw poll of who wants to keep this extension in the
> spec?
> >
> >
> >
> > regards
> >
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>

--bcaec52d4c897d58eb04a87d281d
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

+500. =A0deflate-stream has always been utterly ridiculous in light of mask=
ing. =A0It really should get the axe, and with extreme prejudice. =A0Why is=
 it still in the spec? =A0I don&#39;t recall anyone citing a reasonable rea=
son for keeping it, but there have been many very good arguments against it=
. =A0It&#39;s not even very clearly specified in the document -- when I imp=
lemented it in my Flash client, I had to read the source code of Andy Green=
&#39;s implementation to figure out how it was supposed to work -- his impl=
ementation became my specification for the extension.<div>
<br></div><div>Brian<br><br></div><div><br><div class=3D"gmail_quote">On Tu=
e, Jul 19, 2011 at 7:34 PM, Greg Wilkins <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:gregw@intalio.com">gregw@intalio.com</a>&gt;</span> wrote:<br><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex;">
I&#39;ve just noticed that the w3c is currently intending to make support<b=
r>
for deflate-stream mandatory!<br>
<br>
 =A0<a href=3D"http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917" targe=
t=3D"_blank">http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917</a><br>
<br>
This moves this extension from being useless, but mostly harmless, to<br>
being a major impost on servers and intermediaries.<br>
If the browser make this mandatory, then servers will obviously have<br>
to support it at a cost of extra CPU, extra buffers but for no<br>
significant savings in bandwidth.<br>
Intermediaries that wish to act on frame boundaries will also have to<br>
implement it.<br>
<br>
This illustrate that having silly options always puts you at risk of<br>
people taking you up on those options.<br>
<br>
This extension is demonstrably broken and needs to be either fixed or remov=
ed.<br>
<br>
regards<br>
<div><div></div><div class=3D"h5"><br>
<br>
<br>
On 20 June 2011 16:33, Greg Wilkins &lt;<a href=3D"mailto:gregw@intalio.com=
">gregw@intalio.com</a>&gt; wrote:<br>
&gt; As part of my continuing campaign against including deflate-stream in<=
br>
&gt; the specification as a standard extension, I did a quick test of how<b=
r>
&gt; well it works when applied to masked frames.<br>
&gt;<br>
&gt; I took a days worth of traffic from an IRC channel and wrapped it up<b=
r>
&gt; as JSON messages sent as websocket frames.<br>
&gt; There were 487 message that looked like:<br>
&gt;<br>
&gt; =A0 =A0 {channel:&quot;#webtide&quot;, username:&quot;tbecker&quot;, t=
ext:&quot;joakime: jenkins<br>
&gt; had issues pulling from github a couple of times =A0last week&quot;}<b=
r>
&gt;<br>
&gt; As an unmasked WS stream, it was 50675 bytes, and as a masked stream<b=
r>
&gt; is was 52623 bytes.<br>
&gt; I then compressed both these streams with gzip and got 13306 bytes for=
<br>
&gt; unmasked and 51704 bytes for the masked!!!!<br>
&gt;<br>
&gt; So for this very typical example, masking was sufficiently random to<b=
r>
&gt; completely negate the benefits of compression.<br>
&gt;<br>
&gt; So the deflate-stream &quot;extension&quot; is:<br>
&gt;<br>
&gt; =A0+ next to useless for inbound traffic<br>
&gt; =A0+ breaks all the rules of what an extension can do<br>
&gt; =A0+ is potentially vulnerable to injection as attackers can send<br>
&gt; repeated patterns that may subvert masking<br>
&gt; =A0+ can be replaced by the in-frame compression extension already pro=
posed.<br>
&gt; =A0+ was inserted in the draft with little or no discussion and withou=
t<br>
&gt; clear consensus.<br>
&gt;<br>
&gt; Can I call for a straw poll of who wants to keep this extension in the=
 spec?<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; regards<br>
&gt;<br>
_______________________________________________<br>
hybi mailing list<br>
<a href=3D"mailto:hybi@ietf.org">hybi@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/hybi</a><br>
</div></div></blockquote></div><br></div>

--bcaec52d4c897d58eb04a87d281d--

From arman@noemax.com  Wed Jul 20 03:15:36 2011
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To: =?UTF-8?Q?'I=C3=B1aki_Baz_Castillo'?= <ibc@aliax.net>, "'Andy W. Song'" <wsongcn@gmail.com>
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Subject: Re: [hybi] hybi 10 ---- server to client masking
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> > A question about server to client masking. In section 4.1 it states=20
> > that "Frames sent from the server to the client are not masked."=20
> > Does it mean server to client masking is optional or MUST NOT?
> > Client to server is clear as "MUST" though.
>=20
> "are not masked" means that they MUST NOT be masked.

I don't think you are correct. As far as I understand the discussions in =
the group over masking, the server to client masking is optional (server =
MAY MASK). Otherwise we would not need a masking flag in the frame =
header since client-to-server frames would always be masked and =
server-to-client frames would never be masked.

> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>

With best regards,
Arman


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Subject: Re: [hybi] hybi 10 ---- server to client masking
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2011/7/20 Arman Djusupov <arman@noemax.com>:
> I don't think you are correct. As far as I understand the discussions in =
the group over masking, the server to client masking is optional (server MA=
Y MASK). Otherwise we would not need a masking flag in the frame header sin=
ce client-to-server frames would always be masked and server-to-client fram=
es would never be masked.

Does the spec state somewhere that the client MUST be ready to receive
masked frames from the server? or is it just an interpretation due the
existence of such masking flag?

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From arman@noemax.com  Wed Jul 20 04:00:31 2011
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Subject: Re: [hybi] deflate-stream and masking
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> +500.  deflate-stream has always been utterly ridiculous in light of
masking. 

 

Actually when the frame is relatively large, deflate-stream compression does
reduce the size of the payload. For example one of my tests involved sending
a 2MB XML file as a single masked frame which resulted in 90% compression
(i.e. transferred size was a tenth of the original file). So the
"deflate-stream" extension  is not completely useless, it's just very
situational.

 

> Brian

 

With best regards,

Arman

 

 

From: hybi-bounces@ietf.org [mailto:hybi-bounces@ietf.org] On Behalf Of
Brian
Sent: Wednesday, July 20, 2011 12:50 PM
To: Greg Wilkins
Cc: Hybi
Subject: Re: [hybi] deflate-stream and masking

 

+500.  deflate-stream has always been utterly ridiculous in light of
masking.  It really should get the axe, and with extreme prejudice.  Why is
it still in the spec?  I don't recall anyone citing a reasonable reason for
keeping it, but there have been many very good arguments against it.  It's
not even very clearly specified in the document -- when I implemented it in
my Flash client, I had to read the source code of Andy Green's
implementation to figure out how it was supposed to work -- his
implementation became my specification for the extension.

 

Brian

 

On Tue, Jul 19, 2011 at 7:34 PM, Greg Wilkins <gregw@intalio.com> wrote:

I've just noticed that the w3c is currently intending to make support
for deflate-stream mandatory!

 http://www.w3.org/Bugs/Public/show_bug.cgi?id=12917

This moves this extension from being useless, but mostly harmless, to
being a major impost on servers and intermediaries.
If the browser make this mandatory, then servers will obviously have
to support it at a cost of extra CPU, extra buffers but for no
significant savings in bandwidth.
Intermediaries that wish to act on frame boundaries will also have to
implement it.

This illustrate that having silly options always puts you at risk of
people taking you up on those options.

This extension is demonstrably broken and needs to be either fixed or
removed.

regards




On 20 June 2011 16:33, Greg Wilkins <gregw@intalio.com> wrote:
> As part of my continuing campaign against including deflate-stream in
> the specification as a standard extension, I did a quick test of how
> well it works when applied to masked frames.
>
> I took a days worth of traffic from an IRC channel and wrapped it up
> as JSON messages sent as websocket frames.
> There were 487 message that looked like:
>
>     {channel:"#webtide", username:"tbecker", text:"joakime: jenkins
> had issues pulling from github a couple of times  last week"}
>
> As an unmasked WS stream, it was 50675 bytes, and as a masked stream
> is was 52623 bytes.
> I then compressed both these streams with gzip and got 13306 bytes for
> unmasked and 51704 bytes for the masked!!!!
>
> So for this very typical example, masking was sufficiently random to
> completely negate the benefits of compression.
>
> So the deflate-stream "extension" is:
>
>  + next to useless for inbound traffic
>  + breaks all the rules of what an extension can do
>  + is potentially vulnerable to injection as attackers can send
> repeated patterns that may subvert masking
>  + can be replaced by the in-frame compression extension already proposed.
>  + was inserted in the draft with little or no discussion and without
> clear consensus.
>
> Can I call for a straw poll of who wants to keep this extension in the
spec?
>
>
>
> regards
>
_______________________________________________
hybi mailing list
hybi@ietf.org
https://www.ietf.org/mailman/listinfo/hybi

 


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoPlainText>&gt; =
+500.&nbsp; deflate-stream has always been utterly ridiculous in light =
of masking. <o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>Actually when the frame is relatively large, =
deflate-stream compression does reduce the size of the payload. For =
example one of my tests involved sending a 2MB XML file as a single =
masked frame which resulted in 90% compression (i.e. transferred size =
was a tenth of the original file). So the &#8220;deflate-stream&#8221; =
extension&nbsp; is not completely useless, it&#8217;s just very =
situational.<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>&gt; =
Brian<o:p></o:p></p><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>With best regards,<o:p></o:p></p><p =
class=3DMsoPlainText>Arman<o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><div =
style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'><div><div style=3D'border:none;border-top:solid #B5C4DF =
1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
hybi-bounces@ietf.org [mailto:hybi-bounces@ietf.org] <b>On Behalf Of =
</b>Brian<br><b>Sent:</b> Wednesday, July 20, 2011 12:50 =
PM<br><b>To:</b> Greg Wilkins<br><b>Cc:</b> Hybi<br><b>Subject:</b> Re: =
[hybi] deflate-stream and masking<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>+500. =
&nbsp;deflate-stream has always been utterly ridiculous in light of =
masking. &nbsp;It really should get the axe, and with extreme prejudice. =
&nbsp;Why is it still in the spec? &nbsp;I don't recall anyone citing a =
reasonable reason for keeping it, but there have been many very good =
arguments against it. &nbsp;It's not even very clearly specified in the =
document -- when I implemented it in my Flash client, I had to read the =
source code of Andy Green's implementation to figure out how it was =
supposed to work -- his implementation became my specification for the =
extension.<o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>Brian<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal>On Tue, =
Jul 19, 2011 at 7:34 PM, Greg Wilkins &lt;<a =
href=3D"mailto:gregw@intalio.com">gregw@intalio.com</a>&gt; =
wrote:<o:p></o:p></p><p class=3DMsoNormal>I've just noticed that the w3c =
is currently intending to make support<br>for deflate-stream =
mandatory!<br><br>&nbsp;<a =
href=3D"http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917" =
target=3D"_blank">http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917</=
a><br><br>This moves this extension from being useless, but mostly =
harmless, to<br>being a major impost on servers and =
intermediaries.<br>If the browser make this mandatory, then servers will =
obviously have<br>to support it at a cost of extra CPU, extra buffers =
but for no<br>significant savings in bandwidth.<br>Intermediaries that =
wish to act on frame boundaries will also have to<br>implement =
it.<br><br>This illustrate that having silly options always puts you at =
risk of<br>people taking you up on those options.<br><br>This extension =
is demonstrably broken and needs to be either fixed or =
removed.<br><br>regards<o:p></o:p></p><div><div><p =
class=3DMsoNormal><br><br><br>On 20 June 2011 16:33, Greg Wilkins &lt;<a =
href=3D"mailto:gregw@intalio.com">gregw@intalio.com</a>&gt; =
wrote:<br>&gt; As part of my continuing campaign against including =
deflate-stream in<br>&gt; the specification as a standard extension, I =
did a quick test of how<br>&gt; well it works when applied to masked =
frames.<br>&gt;<br>&gt; I took a days worth of traffic from an IRC =
channel and wrapped it up<br>&gt; as JSON messages sent as websocket =
frames.<br>&gt; There were 487 message that looked like:<br>&gt;<br>&gt; =
&nbsp; &nbsp; {channel:&quot;#webtide&quot;, =
username:&quot;tbecker&quot;, text:&quot;joakime: jenkins<br>&gt; had =
issues pulling from github a couple of times &nbsp;last =
week&quot;}<br>&gt;<br>&gt; As an unmasked WS stream, it was 50675 =
bytes, and as a masked stream<br>&gt; is was 52623 bytes.<br>&gt; I then =
compressed both these streams with gzip and got 13306 bytes for<br>&gt; =
unmasked and 51704 bytes for the masked!!!!<br>&gt;<br>&gt; So for this =
very typical example, masking was sufficiently random to<br>&gt; =
completely negate the benefits of compression.<br>&gt;<br>&gt; So the =
deflate-stream &quot;extension&quot; is:<br>&gt;<br>&gt; &nbsp;+ next to =
useless for inbound traffic<br>&gt; &nbsp;+ breaks all the rules of what =
an extension can do<br>&gt; &nbsp;+ is potentially vulnerable to =
injection as attackers can send<br>&gt; repeated patterns that may =
subvert masking<br>&gt; &nbsp;+ can be replaced by the in-frame =
compression extension already proposed.<br>&gt; &nbsp;+ was inserted in =
the draft with little or no discussion and without<br>&gt; clear =
consensus.<br>&gt;<br>&gt; Can I call for a straw poll of who wants to =
keep this extension in the spec?<br>&gt;<br>&gt;<br>&gt;<br>&gt; =
regards<br>&gt;<br>_______________________________________________<br>hyb=
i mailing list<br><a =
href=3D"mailto:hybi@ietf.org">hybi@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/hybi" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/hybi</a><o:p></o:=
p></p></div></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></div></body></html>
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From phil127@gmail.com  Wed Jul 20 05:10:34 2011
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Date: Wed, 20 Jul 2011 14:10:21 +0200
From: Philipp Serafin <phil127@gmail.com>
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Subject: Re: [hybi] deflate-stream and masking
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However, for the same scenario, deflate-frame will have at worst
identical, possibly better results. But unlike deflate-stream,
deflate-frame will archieve the same results in a lot of other
scenarios, too.

Am 20.07.2011 12:58, schrieb Arman Djusupov:
>
> > +500.  deflate-stream has always been utterly ridiculous in light of
> masking.
>
>  
>
> Actually when the frame is relatively large, deflate-stream
> compression does reduce the size of the payload. For example one of my
> tests involved sending a 2MB XML file as a single masked frame which
> resulted in 90% compression (i.e. transferred size was a tenth of the
> original file). So the "deflate-stream" extension  is not completely
> useless, it's just very situational.
>
>  
>
> > Brian
>
>  
>
> With best regards,
>
> Arman
>
>  
>
>  
>

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    However, for the same scenario, deflate-frame will have at worst
    identical, possibly better results. But unlike deflate-stream,
    deflate-frame will archieve the same results in a lot of other
    scenarios, too.<br>
    <br>
    Am 20.07.2011 12:58, schrieb Arman Djusupov:
    <blockquote cite="mid:002701cc46cc$0a04c820$1e0e5860$@noemax.com"
      type="cite">
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        <p class="MsoPlainText">&gt; +500.&nbsp; deflate-stream has always
          been utterly ridiculous in light of masking. <o:p></o:p></p>
        <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
        <p class="MsoPlainText">Actually when the frame is relatively
          large, deflate-stream compression does reduce the size of the
          payload. For example one of my tests involved sending a 2MB
          XML file as a single masked frame which resulted in 90%
          compression (i.e. transferred size was a tenth of the original
          file). So the &#8220;deflate-stream&#8221; extension&nbsp; is not completely
          useless, it&#8217;s just very situational.<o:p></o:p></p>
        <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
        <p class="MsoPlainText">&gt; Brian<o:p></o:p></p>
        <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
        <p class="MsoPlainText">With best regards,<o:p></o:p></p>
        <p class="MsoPlainText">Arman<o:p></o:p></p>
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style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
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From menone7@gmail.com  Wed Jul 20 04:57:33 2011
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Subject: [hybi] <draft-ietf-hybi-thewebsocketprotocol-10> Sec-WebSocket-Key needed?
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>   The WebSocket protocol is an independent TCP-based protocol.  Its
>   only relationship to HTTP is that its handshake is interpreted by
>   HTTP servers as an Upgrade request.

If it independent protocol, then say me why it send messages
with "HTTP/1.1"? If change this part (example "WSOCK/0.11") then
security issues with XmlHttpRequest disappear. Do not supporting
WebSocket servers MUST answer "400 Bad Request". But separate one
from another is very simple. And the need for such a complicated
handshake disappears. No?

From phil127@gmail.com  Wed Jul 20 05:25:01 2011
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Subject: Re: [hybi] <draft-ietf-hybi-thewebsocketprotocol-10> Sec-WebSocket-Key needed?
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A protocol like this would likely be rejected by proxies, too. It's a
design goal for WS that it should be able to pass most existing HTTP
proxies.

Am 20.07.2011 13:57, schrieb Alexander Yastrebov:
>>   The WebSocket protocol is an independent TCP-based protocol.  Its
>>   only relationship to HTTP is that its handshake is interpreted by
>>   HTTP servers as an Upgrade request.
> If it independent protocol, then say me why it send messages
> with "HTTP/1.1"? If change this part (example "WSOCK/0.11") then
> security issues with XmlHttpRequest disappear. Do not supporting
> WebSocket servers MUST answer "400 Bad Request". But separate one
> from another is very simple. And the need for such a complicated
> handshake disappears. No?
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi


From jat@google.com  Wed Jul 20 07:18:48 2011
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From: John Tamplin <jat@google.com>
Date: Wed, 20 Jul 2011 10:18:18 -0400
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On Wed, Jul 20, 2011 at 6:14 AM, Arman Djusupov <arman@noemax.com> wrote:

> I don't think you are correct. As far as I understand the discussions in
> the group over masking, the server to client masking is optional (server MAY
> MASK). Otherwise we would not need a masking flag in the frame header since
> client-to-server frames would always be masked and server-to-client frames
> would never be masked.
>

The flag was requested so intermediaries (including packet sniffers) could
interpret the frame without knowing the context (ie, which direction the
packet is going or negotiated extensions, etc).

-- 
John A. Tamplin
Software Engineer (GWT), Google

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<div class=3D"gmail_quote">On Wed, Jul 20, 2011 at 6:14 AM, Arman Djusupov =
<span dir=3D"ltr">&lt;<a href=3D"mailto:arman@noemax.com">arman@noemax.com<=
/a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

<div class=3D"im">I don&#39;t think you are correct. As far as I understand=
 the discussions in the group over masking, the server to client masking is=
 optional (server MAY MASK). Otherwise we would not need a masking flag in =
the frame header since client-to-server frames would always be masked and s=
erver-to-client frames would never be masked.</div>

</blockquote></div><div><br></div>The flag was requested so intermediaries =
(including packet sniffers) could interpret the frame without knowing the c=
ontext (ie, which direction the packet is going or negotiated extensions, e=
tc).<br clear=3D"all">

<br>-- <br>John A. Tamplin<br>Software Engineer (GWT), Google<br>

--000e0cd6ae5a939a6b04a880e8af--

From alexey.melnikov@isode.com  Wed Jul 20 08:44:35 2011
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Hi group,

I am aware that the Acknowledgments section is not complete. As there is 
way too much traffic on the mailing list, I would appreciate direct 
emails from people who think that they contributed ideas (or 
participated in discussions that lead to some changes) to the document 
or done reviews of it, so that they can be properly acknowledged. If I 
recognize your name, I will simply add you to this section, but if not, 
I might have some followup questions in email.

PLEASE DON'T SEND REPLIES DIRECTLY TO THE MAILING LIST :-).

Best Regards,
Alexey


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Thought I'd replied on list...
---------- Forwarded message ----------
From: "Justin Lee" <jlee@antwerkz.com>
Date: Jul 20, 2011 8:48 AM
Subject: Re: [hybi] deflate-stream and masking
To: "Philipp Serafin" <phil127@gmail.com>

For what it's worth, I haven't bothered implementing deflate-stream on
glassfish because of the concerns here. Removing it would not bother me a
bit. I vote for removing or replacing it.
On Jul 20, 2011 5:10 AM, "Philipp Serafin" <phil127@gmail.com> wrote:
> However, for the same scenario, deflate-frame will have at worst
> identical, possibly better results. But unlike deflate-stream,
> deflate-frame will archieve the same results in a lot of other
> scenarios, too.
>
> Am 20.07.2011 12:58, schrieb Arman Djusupov:
>>
>> > +500. deflate-stream has always been utterly ridiculous in light of
>> masking.
>>
>>
>>
>> Actually when the frame is relatively large, deflate-stream
>> compression does reduce the size of the payload. For example one of my
>> tests involved sending a 2MB XML file as a single masked frame which
>> resulted in 90% compression (i.e. transferred size was a tenth of the
>> original file). So the "deflate-stream" extension is not completely
>> useless, it's just very situational.
>>
>>
>>
>> > Brian
>>
>>
>>
>> With best regards,
>>
>> Arman
>>
>>
>>
>>
>>

--001517447af2c8e56a04a8840777
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<p>Thought I&#39;d replied on list...</p>
<div class=3D"gmail_quote">---------- Forwarded message ----------<br>From:=
 &quot;Justin Lee&quot; &lt;<a href=3D"mailto:jlee@antwerkz.com">jlee@antwe=
rkz.com</a>&gt;<br>Date: Jul 20, 2011 8:48 AM<br>Subject: Re: [hybi] deflat=
e-stream and masking<br>
To: &quot;Philipp Serafin&quot; &lt;<a href=3D"mailto:phil127@gmail.com">ph=
il127@gmail.com</a>&gt;<br><br type=3D"attribution"><p>For what it&#39;s wo=
rth, I haven&#39;t bothered implementing deflate-stream on glassfish becaus=
e of the concerns here. Removing it would not bother me a bit. I vote for r=
emoving or replacing it.</p>
<div class=3D"elided-text">
<div class=3D"gmail_quote">On Jul 20, 2011 5:10 AM, &quot;Philipp Serafin&q=
uot; &lt;<a href=3D"mailto:phil127@gmail.com" target=3D"_blank">phil127@gma=
il.com</a>&gt; wrote:<br type=3D"attribution">&gt; However, for the same sc=
enario, deflate-frame will have at worst<br>

&gt; identical, possibly better results. But unlike deflate-stream,<br>&gt;=
 deflate-frame will archieve the same results in a lot of other<br>&gt; sce=
narios, too.<br>&gt; <br>&gt; Am 20.07.2011 12:58, schrieb Arman Djusupov:<=
br>

&gt;&gt;<br>&gt;&gt; &gt; +500.  deflate-stream has always been utterly rid=
iculous in light of<br>&gt;&gt; masking.<br>&gt;&gt;<br>&gt;&gt;  <br>&gt;&=
gt;<br>&gt;&gt; Actually when the frame is relatively large, deflate-stream=
<br>

&gt;&gt; compression does reduce the size of the payload. For example one o=
f my<br>&gt;&gt; tests involved sending a 2MB XML file as a single masked f=
rame which<br>&gt;&gt; resulted in 90% compression (i.e. transferred size w=
as a tenth of the<br>

&gt;&gt; original file). So the &quot;deflate-stream&quot; extension  is no=
t completely<br>&gt;&gt; useless, it&#39;s just very situational.<br>&gt;&g=
t;<br>&gt;&gt;  <br>&gt;&gt;<br>&gt;&gt; &gt; Brian<br>&gt;&gt;<br>&gt;&gt;=
  <br>

&gt;&gt;<br>&gt;&gt; With best regards,<br>&gt;&gt;<br>&gt;&gt; Arman<br>&g=
t;&gt;<br>&gt;&gt;  <br>&gt;&gt;<br>&gt;&gt;  <br>&gt;&gt;<br></div>
</div></div>

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Unlurking to agree.

When it was optional it was avoidable and non-impairing.   It should not be
mandatory.

+6.02 x10^23 votes for removal from specification (which when deflated is
just +1, I believe)



On Wed, Jul 20, 2011 at 5:50 AM, Brian <theturtle32@gmail.com> wrote:

> +500.  deflate-stream has always been utterly ridiculous in light of
> masking.  It really should get the axe, and with extreme prejudice.  Why is
> it still in the spec?  I don't recall anyone citing a reasonable reason for
> keeping it, but there have been many very good arguments against it.  It's
> not even very clearly specified in the document -- when I implemented it in
> my Flash client, I had to read the source code of Andy Green's
> implementation to figure out how it was supposed to work -- his
> implementation became my specification for the extension.
>
> Brian
>
>
> On Tue, Jul 19, 2011 at 7:34 PM, Greg Wilkins <gregw@intalio.com> wrote:
>
>> I've just noticed that the w3c is currently intending to make support
>> for deflate-stream mandatory!
>>
>>  http://www.w3.org/Bugs/Public/show_bug.cgi?id=12917
>>
>> This moves this extension from being useless, but mostly harmless, to
>> being a major impost on servers and intermediaries.
>> If the browser make this mandatory, then servers will obviously have
>> to support it at a cost of extra CPU, extra buffers but for no
>> significant savings in bandwidth.
>> Intermediaries that wish to act on frame boundaries will also have to
>> implement it.
>>
>> This illustrate that having silly options always puts you at risk of
>> people taking you up on those options.
>>
>> This extension is demonstrably broken and needs to be either fixed or
>> removed.
>>
>> regards
>>
>>
>>
>> On 20 June 2011 16:33, Greg Wilkins <gregw@intalio.com> wrote:
>> > As part of my continuing campaign against including deflate-stream in
>> > the specification as a standard extension, I did a quick test of how
>> > well it works when applied to masked frames.
>> >
>> > I took a days worth of traffic from an IRC channel and wrapped it up
>> > as JSON messages sent as websocket frames.
>> > There were 487 message that looked like:
>> >
>> >     {channel:"#webtide", username:"tbecker", text:"joakime: jenkins
>> > had issues pulling from github a couple of times  last week"}
>> >
>> > As an unmasked WS stream, it was 50675 bytes, and as a masked stream
>> > is was 52623 bytes.
>> > I then compressed both these streams with gzip and got 13306 bytes for
>> > unmasked and 51704 bytes for the masked!!!!
>> >
>> > So for this very typical example, masking was sufficiently random to
>> > completely negate the benefits of compression.
>> >
>> > So the deflate-stream "extension" is:
>> >
>> >  + next to useless for inbound traffic
>> >  + breaks all the rules of what an extension can do
>> >  + is potentially vulnerable to injection as attackers can send
>> > repeated patterns that may subvert masking
>> >  + can be replaced by the in-frame compression extension already
>> proposed.
>> >  + was inserted in the draft with little or no discussion and without
>> > clear consensus.
>> >
>> > Can I call for a straw poll of who wants to keep this extension in the
>> spec?
>> >
>> >
>> >
>> > regards
>> >
>> _______________________________________________
>> hybi mailing list
>> hybi@ietf.org
>> https://www.ietf.org/mailman/listinfo/hybi
>>
>
>
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>
>

--0016364d2c79b962c604a8842229
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Unlurking to agree. =A0 =A0=A0<div><br></div><div>When it was optional it w=
as avoidable and non-impairing. =A0 It should not be mandatory. =A0 =A0<div=
><br></div><div>+6.02 x10^23 votes for removal from specification (which wh=
en deflated is just +1, I believe)</div>
<div><div><br></div><div><br><br><div class=3D"gmail_quote">On Wed, Jul 20,=
 2011 at 5:50 AM, Brian <span dir=3D"ltr">&lt;<a href=3D"mailto:theturtle32=
@gmail.com">theturtle32@gmail.com</a>&gt;</span> wrote:<br><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex;">
+500. =A0deflate-stream has always been utterly ridiculous in light of mask=
ing. =A0It really should get the axe, and with extreme prejudice. =A0Why is=
 it still in the spec? =A0I don&#39;t recall anyone citing a reasonable rea=
son for keeping it, but there have been many very good arguments against it=
. =A0It&#39;s not even very clearly specified in the document -- when I imp=
lemented it in my Flash client, I had to read the source code of Andy Green=
&#39;s implementation to figure out how it was supposed to work -- his impl=
ementation became my specification for the extension.<div>

<br></div><font color=3D"#888888"><div>Brian<br><br></div></font><div><div>=
</div><div class=3D"h5"><div><br><div class=3D"gmail_quote">On Tue, Jul 19,=
 2011 at 7:34 PM, Greg Wilkins <span dir=3D"ltr">&lt;<a href=3D"mailto:greg=
w@intalio.com" target=3D"_blank">gregw@intalio.com</a>&gt;</span> wrote:<br=
>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I&#39;ve just noticed that the w3c is currently intending to make support<b=
r>
for deflate-stream mandatory!<br>
<br>
 =A0<a href=3D"http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917" targe=
t=3D"_blank">http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917</a><br>
<br>
This moves this extension from being useless, but mostly harmless, to<br>
being a major impost on servers and intermediaries.<br>
If the browser make this mandatory, then servers will obviously have<br>
to support it at a cost of extra CPU, extra buffers but for no<br>
significant savings in bandwidth.<br>
Intermediaries that wish to act on frame boundaries will also have to<br>
implement it.<br>
<br>
This illustrate that having silly options always puts you at risk of<br>
people taking you up on those options.<br>
<br>
This extension is demonstrably broken and needs to be either fixed or remov=
ed.<br>
<br>
regards<br>
<div><div></div><div><br>
<br>
<br>
On 20 June 2011 16:33, Greg Wilkins &lt;<a href=3D"mailto:gregw@intalio.com=
" target=3D"_blank">gregw@intalio.com</a>&gt; wrote:<br>
&gt; As part of my continuing campaign against including deflate-stream in<=
br>
&gt; the specification as a standard extension, I did a quick test of how<b=
r>
&gt; well it works when applied to masked frames.<br>
&gt;<br>
&gt; I took a days worth of traffic from an IRC channel and wrapped it up<b=
r>
&gt; as JSON messages sent as websocket frames.<br>
&gt; There were 487 message that looked like:<br>
&gt;<br>
&gt; =A0 =A0 {channel:&quot;#webtide&quot;, username:&quot;tbecker&quot;, t=
ext:&quot;joakime: jenkins<br>
&gt; had issues pulling from github a couple of times =A0last week&quot;}<b=
r>
&gt;<br>
&gt; As an unmasked WS stream, it was 50675 bytes, and as a masked stream<b=
r>
&gt; is was 52623 bytes.<br>
&gt; I then compressed both these streams with gzip and got 13306 bytes for=
<br>
&gt; unmasked and 51704 bytes for the masked!!!!<br>
&gt;<br>
&gt; So for this very typical example, masking was sufficiently random to<b=
r>
&gt; completely negate the benefits of compression.<br>
&gt;<br>
&gt; So the deflate-stream &quot;extension&quot; is:<br>
&gt;<br>
&gt; =A0+ next to useless for inbound traffic<br>
&gt; =A0+ breaks all the rules of what an extension can do<br>
&gt; =A0+ is potentially vulnerable to injection as attackers can send<br>
&gt; repeated patterns that may subvert masking<br>
&gt; =A0+ can be replaced by the in-frame compression extension already pro=
posed.<br>
&gt; =A0+ was inserted in the draft with little or no discussion and withou=
t<br>
&gt; clear consensus.<br>
&gt;<br>
&gt; Can I call for a straw poll of who wants to keep this extension in the=
 spec?<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; regards<br>
&gt;<br>
_______________________________________________<br>
hybi mailing list<br>
<a href=3D"mailto:hybi@ietf.org" target=3D"_blank">hybi@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/hybi</a><br>
</div></div></blockquote></div><br></div>
</div></div><br>_______________________________________________<br>
hybi mailing list<br>
<a href=3D"mailto:hybi@ietf.org">hybi@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/hybi</a><br>
<br></blockquote></div><br></div></div></div>

--0016364d2c79b962c604a8842229--

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Subject: Re: [hybi] hybi 10 ---- server to client masking
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So the flag is the truth, and either end must be prepared to process masking
based on the frame's flag?


On Wed, Jul 20, 2011 at 10:18 AM, John Tamplin <jat@google.com> wrote:

> On Wed, Jul 20, 2011 at 6:14 AM, Arman Djusupov <arman@noemax.com> wrote:
>
>> I don't think you are correct. As far as I understand the discussions in
>> the group over masking, the server to client masking is optional (server MAY
>> MASK). Otherwise we would not need a masking flag in the frame header since
>> client-to-server frames would always be masked and server-to-client frames
>> would never be masked.
>>
>
> The flag was requested so intermediaries (including packet sniffers) could
> interpret the frame without knowing the context (ie, which direction the
> packet is going or negotiated extensions, etc).
>
> --
> John A. Tamplin
> Software Engineer (GWT), Google
>
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>
>

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So the flag is the truth, and either end must be prepared to process maskin=
g based on the frame&#39;s flag?<div><br><br><div class=3D"gmail_quote">On =
Wed, Jul 20, 2011 at 10:18 AM, John Tamplin <span dir=3D"ltr">&lt;<a href=
=3D"mailto:jat@google.com">jat@google.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;"><div class=3D"im"><div class=3D"gmail_quote=
">On Wed, Jul 20, 2011 at 6:14 AM, Arman Djusupov <span dir=3D"ltr">&lt;<a =
href=3D"mailto:arman@noemax.com" target=3D"_blank">arman@noemax.com</a>&gt;=
</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">

<div>I don&#39;t think you are correct. As far as I understand the discussi=
ons in the group over masking, the server to client masking is optional (se=
rver MAY MASK). Otherwise we would not need a masking flag in the frame hea=
der since client-to-server frames would always be masked and server-to-clie=
nt frames would never be masked.</div>


</blockquote></div><div><br></div></div>The flag was requested so intermedi=
aries (including packet sniffers) could interpret the frame without knowing=
 the context (ie, which direction the packet is going or negotiated extensi=
ons, etc).<br clear=3D"all">
<font color=3D"#888888">

<br>-- <br>John A. Tamplin<br>Software Engineer (GWT), Google<br>
</font><br>_______________________________________________<br>
hybi mailing list<br>
<a href=3D"mailto:hybi@ietf.org">hybi@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/hybi</a><br>
<br></blockquote></div><br></div>

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Subject: Re: [hybi] deflate-stream and masking
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But currently it is an extension and thus optional, isn't it?

=20

=20

From: hybi-bounces@ietf.org [mailto:hybi-bounces@ietf.org] On Behalf Of =
David Endicott
Sent: Wednesday 20 July 2011 21:10
To: Brian
Cc: Hybi; Greg Wilkins
Subject: Re: [hybi] deflate-stream and masking

=20

Unlurking to agree.    =20

=20

When it was optional it was avoidable and non-impairing.   It should not =
be mandatory.   =20

=20

+6.02 x10^23 votes for removal from specification (which when deflated =
is just +1, I believe)

=20

=20

On Wed, Jul 20, 2011 at 5:50 AM, Brian <theturtle32@gmail.com> wrote:

+500.  deflate-stream has always been utterly ridiculous in light of =
masking.  It really should get the axe, and with extreme prejudice.  Why =
is it still in the spec?  I don't recall anyone citing a reasonable =
reason for keeping it, but there have been many very good arguments =
against it.  It's not even very clearly specified in the document -- =
when I implemented it in my Flash client, I had to read the source code =
of Andy Green's implementation to figure out how it was supposed to work =
-- his implementation became my specification for the extension.

=20

Brian

=20

On Tue, Jul 19, 2011 at 7:34 PM, Greg Wilkins <gregw@intalio.com> wrote:

I've just noticed that the w3c is currently intending to make support
for deflate-stream mandatory!

 http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917

This moves this extension from being useless, but mostly harmless, to
being a major impost on servers and intermediaries.
If the browser make this mandatory, then servers will obviously have
to support it at a cost of extra CPU, extra buffers but for no
significant savings in bandwidth.
Intermediaries that wish to act on frame boundaries will also have to
implement it.

This illustrate that having silly options always puts you at risk of
people taking you up on those options.

This extension is demonstrably broken and needs to be either fixed or =
removed.

regards




On 20 June 2011 16:33, Greg Wilkins <gregw@intalio.com> wrote:
> As part of my continuing campaign against including deflate-stream in
> the specification as a standard extension, I did a quick test of how
> well it works when applied to masked frames.
>
> I took a days worth of traffic from an IRC channel and wrapped it up
> as JSON messages sent as websocket frames.
> There were 487 message that looked like:
>
>     {channel:"#webtide", username:"tbecker", text:"joakime: jenkins
> had issues pulling from github a couple of times  last week"}
>
> As an unmasked WS stream, it was 50675 bytes, and as a masked stream
> is was 52623 bytes.
> I then compressed both these streams with gzip and got 13306 bytes for
> unmasked and 51704 bytes for the masked!!!!
>
> So for this very typical example, masking was sufficiently random to
> completely negate the benefits of compression.
>
> So the deflate-stream "extension" is:
>
>  + next to useless for inbound traffic
>  + breaks all the rules of what an extension can do
>  + is potentially vulnerable to injection as attackers can send
> repeated patterns that may subvert masking
>  + can be replaced by the in-frame compression extension already =
proposed.
>  + was inserted in the draft with little or no discussion and without
> clear consensus.
>
> Can I call for a straw poll of who wants to keep this extension in the =
spec?
>
>
>
> regards
>
_______________________________________________
hybi mailing list
hybi@ietf.org
https://www.ietf.org/mailman/listinfo/hybi

=20


_______________________________________________
hybi mailing list
hybi@ietf.org
https://www.ietf.org/mailman/listinfo/hybi

=20


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	margin:1.0in 1.25in 1.0in 1.25in;}
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	{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoPlainText>But =
currently it is an extension and thus optional, isn't =
it?<o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><div =
style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'><div><div style=3D'border:none;border-top:solid #B5C4DF =
1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
hybi-bounces@ietf.org [mailto:hybi-bounces@ietf.org] <b>On Behalf Of =
</b>David Endicott<br><b>Sent:</b> Wednesday 20 July 2011 =
21:10<br><b>To:</b> Brian<br><b>Cc:</b> Hybi; Greg =
Wilkins<br><b>Subject:</b> Re: [hybi] deflate-stream and =
masking<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Unlurking to =
agree. &nbsp; &nbsp;&nbsp;<o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>When it was optional it was avoidable and =
non-impairing. &nbsp; It should not be mandatory. &nbsp; =
&nbsp;<o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>+6.02 x10^23 votes for removal from specification =
(which when deflated is just +1, I =
believe)<o:p></o:p></p></div><div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal>On Wed, Jul 20, 2011 at 5:50 AM, Brian &lt;<a =
href=3D"mailto:theturtle32@gmail.com">theturtle32@gmail.com</a>&gt; =
wrote:<o:p></o:p></p><p class=3DMsoNormal>+500. &nbsp;deflate-stream has =
always been utterly ridiculous in light of masking. &nbsp;It really =
should get the axe, and with extreme prejudice. &nbsp;Why is it still in =
the spec? &nbsp;I don't recall anyone citing a reasonable reason for =
keeping it, but there have been many very good arguments against it. =
&nbsp;It's not even very clearly specified in the document -- when I =
implemented it in my Flash client, I had to read the source code of Andy =
Green's implementation to figure out how it was supposed to work -- his =
implementation became my specification for the =
extension.<o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span =
style=3D'color:#888888'>Brian<o:p></o:p></span></p></div><div><div><div><=
p class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal>On =
Tue, Jul 19, 2011 at 7:34 PM, Greg Wilkins &lt;<a =
href=3D"mailto:gregw@intalio.com" =
target=3D"_blank">gregw@intalio.com</a>&gt; wrote:<o:p></o:p></p><p =
class=3DMsoNormal>I've just noticed that the w3c is currently intending =
to make support<br>for deflate-stream mandatory!<br><br>&nbsp;<a =
href=3D"http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917" =
target=3D"_blank">http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917</=
a><br><br>This moves this extension from being useless, but mostly =
harmless, to<br>being a major impost on servers and =
intermediaries.<br>If the browser make this mandatory, then servers will =
obviously have<br>to support it at a cost of extra CPU, extra buffers =
but for no<br>significant savings in bandwidth.<br>Intermediaries that =
wish to act on frame boundaries will also have to<br>implement =
it.<br><br>This illustrate that having silly options always puts you at =
risk of<br>people taking you up on those options.<br><br>This extension =
is demonstrably broken and needs to be either fixed or =
removed.<br><br>regards<o:p></o:p></p><div><div><p =
class=3DMsoNormal><br><br><br>On 20 June 2011 16:33, Greg Wilkins &lt;<a =
href=3D"mailto:gregw@intalio.com" =
target=3D"_blank">gregw@intalio.com</a>&gt; wrote:<br>&gt; As part of my =
continuing campaign against including deflate-stream in<br>&gt; the =
specification as a standard extension, I did a quick test of how<br>&gt; =
well it works when applied to masked frames.<br>&gt;<br>&gt; I took a =
days worth of traffic from an IRC channel and wrapped it up<br>&gt; as =
JSON messages sent as websocket frames.<br>&gt; There were 487 message =
that looked like:<br>&gt;<br>&gt; &nbsp; &nbsp; =
{channel:&quot;#webtide&quot;, username:&quot;tbecker&quot;, =
text:&quot;joakime: jenkins<br>&gt; had issues pulling from github a =
couple of times &nbsp;last week&quot;}<br>&gt;<br>&gt; As an unmasked WS =
stream, it was 50675 bytes, and as a masked stream<br>&gt; is was 52623 =
bytes.<br>&gt; I then compressed both these streams with gzip and got =
13306 bytes for<br>&gt; unmasked and 51704 bytes for the =
masked!!!!<br>&gt;<br>&gt; So for this very typical example, masking was =
sufficiently random to<br>&gt; completely negate the benefits of =
compression.<br>&gt;<br>&gt; So the deflate-stream &quot;extension&quot; =
is:<br>&gt;<br>&gt; &nbsp;+ next to useless for inbound traffic<br>&gt; =
&nbsp;+ breaks all the rules of what an extension can do<br>&gt; &nbsp;+ =
is potentially vulnerable to injection as attackers can send<br>&gt; =
repeated patterns that may subvert masking<br>&gt; &nbsp;+ can be =
replaced by the in-frame compression extension already proposed.<br>&gt; =
&nbsp;+ was inserted in the draft with little or no discussion and =
without<br>&gt; clear consensus.<br>&gt;<br>&gt; Can I call for a straw =
poll of who wants to keep this extension in the =
spec?<br>&gt;<br>&gt;<br>&gt;<br>&gt; =
regards<br>&gt;<br>_______________________________________________<br>hyb=
i mailing list<br><a href=3D"mailto:hybi@ietf.org" =
target=3D"_blank">hybi@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/hybi" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/hybi</a><o:p></o:=
p></p></div></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></div><p =
class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><br>______________________________________=
_________<br>hybi mailing list<br><a =
href=3D"mailto:hybi@ietf.org">hybi@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/hybi" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/hybi</a><o:p></o:=
p></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></div></div></div></bo=
dy></html>
------=_NextPart_000_00B9_01CC4723.F824A3C0--


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Date: Wed, 20 Jul 2011 14:55:00 -0400
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From: David Endicott <dendicott@gmail.com>
To: Alexander Philippou <alex@noemax.com>
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Cc: Hybi <hybi@ietf.org>
Subject: Re: [hybi] deflate-stream and masking
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--20cf301fb94d00d95f04a884c5c8
Content-Type: text/plain; charset=ISO-8859-1

I'm basing my position on Greg's comment above that:

> I've just noticed that the w3c is currently intending to make support
> for deflate-stream mandatory!
> http://www.w3.org/Bugs/Public/show_bug.cgi?id=12917

If I cannot avoid it in my implementations (because browsers make it
mandatory), then I'm strongly opposed.

If it remains an optional extension, then I revert to a Don't Care position.


On Wed, Jul 20, 2011 at 2:28 PM, Alexander Philippou <alex@noemax.com>wrote:

> But currently it is an extension and thus optional, isn't it?****
>
> ** **
>
> ** **
>
> *From:* hybi-bounces@ietf.org [mailto:hybi-bounces@ietf.org] *On Behalf Of
> *David Endicott
> *Sent:* Wednesday 20 July 2011 21:10
> *To:* Brian
> *Cc:* Hybi; Greg Wilkins
>
> *Subject:* Re: [hybi] deflate-stream and masking****
>
> ** **
>
> Unlurking to agree.     ****
>
> ** **
>
> When it was optional it was avoidable and non-impairing.   It should not be
> mandatory.    ****
>
> ** **
>
> +6.02 x10^23 votes for removal from specification (which when deflated is
> just +1, I believe)****
>
> ** **
>
> ** **
>
> On Wed, Jul 20, 2011 at 5:50 AM, Brian <theturtle32@gmail.com> wrote:****
>
> +500.  deflate-stream has always been utterly ridiculous in light of
> masking.  It really should get the axe, and with extreme prejudice.  Why is
> it still in the spec?  I don't recall anyone citing a reasonable reason for
> keeping it, but there have been many very good arguments against it.  It's
> not even very clearly specified in the document -- when I implemented it in
> my Flash client, I had to read the source code of Andy Green's
> implementation to figure out how it was supposed to work -- his
> implementation became my specification for the extension.****
>
> ** **
>
> Brian****
>
> ** **
>
> On Tue, Jul 19, 2011 at 7:34 PM, Greg Wilkins <gregw@intalio.com> wrote:**
> **
>
> I've just noticed that the w3c is currently intending to make support
> for deflate-stream mandatory!
>
>  http://www.w3.org/Bugs/Public/show_bug.cgi?id=12917
>
> This moves this extension from being useless, but mostly harmless, to
> being a major impost on servers and intermediaries.
> If the browser make this mandatory, then servers will obviously have
> to support it at a cost of extra CPU, extra buffers but for no
> significant savings in bandwidth.
> Intermediaries that wish to act on frame boundaries will also have to
> implement it.
>
> This illustrate that having silly options always puts you at risk of
> people taking you up on those options.
>
> This extension is demonstrably broken and needs to be either fixed or
> removed.
>
> regards****
>
>
>
>
> On 20 June 2011 16:33, Greg Wilkins <gregw@intalio.com> wrote:
> > As part of my continuing campaign against including deflate-stream in
> > the specification as a standard extension, I did a quick test of how
> > well it works when applied to masked frames.
> >
> > I took a days worth of traffic from an IRC channel and wrapped it up
> > as JSON messages sent as websocket frames.
> > There were 487 message that looked like:
> >
> >     {channel:"#webtide", username:"tbecker", text:"joakime: jenkins
> > had issues pulling from github a couple of times  last week"}
> >
> > As an unmasked WS stream, it was 50675 bytes, and as a masked stream
> > is was 52623 bytes.
> > I then compressed both these streams with gzip and got 13306 bytes for
> > unmasked and 51704 bytes for the masked!!!!
> >
> > So for this very typical example, masking was sufficiently random to
> > completely negate the benefits of compression.
> >
> > So the deflate-stream "extension" is:
> >
> >  + next to useless for inbound traffic
> >  + breaks all the rules of what an extension can do
> >  + is potentially vulnerable to injection as attackers can send
> > repeated patterns that may subvert masking
> >  + can be replaced by the in-frame compression extension already
> proposed.
> >  + was inserted in the draft with little or no discussion and without
> > clear consensus.
> >
> > Can I call for a straw poll of who wants to keep this extension in the
> spec?
> >
> >
> >
> > regards
> >
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi****
>
> ** **
>
>
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi****
>
> ** **
>

--20cf301fb94d00d95f04a884c5c8
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I&#39;m basing my position on Greg&#39;s comment above that:<div><br></div>=
<div><meta http-equiv=3D"content-type" content=3D"text/html; charset=3Dutf-=
8"><span class=3D"Apple-style-span" style=3D"border-collapse: collapse; fon=
t-family: arial, sans-serif; font-size: 13px; ">&gt; I&#39;ve just noticed =
that the w3c is currently intending to make support<br>
&gt; for deflate-stream mandatory!<br>&gt;=A0<a href=3D"http://www.w3.org/B=
ugs/Public/show_bug.cgi?id=3D12917" target=3D"_blank" style=3D"color: rgb(2=
8, 81, 168); ">http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917</a><br=
></span><br>
</div><div>If I cannot avoid it in my implementations (because browsers mak=
e it mandatory), then I&#39;m strongly opposed.</div><div><br></div><div>If=
 it remains an optional extension, then I revert to a Don&#39;t Care positi=
on.</div>
<div><br></div><div><br><div class=3D"gmail_quote">On Wed, Jul 20, 2011 at =
2:28 PM, Alexander Philippou <span dir=3D"ltr">&lt;<a href=3D"mailto:alex@n=
oemax.com">alex@noemax.com</a>&gt;</span> wrote:<br><blockquote class=3D"gm=
ail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex;">

<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div><p>But currently it=
 is an extension and thus optional, isn&#39;t it?<u></u><u></u></p><p class=
=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D"><u></u>=A0<u>=
</u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D"><u></=
u>=A0<u></u></span></p><div style=3D"border:none;border-left:solid blue 1.5=
pt;padding:0in 0in 0in 4.0pt"><div><div style=3D"border:none;border-top:sol=
id #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt">From:</span></b>=
<span style=3D"font-size:10.0pt"> <a href=3D"mailto:hybi-bounces@ietf.org" =
target=3D"_blank">hybi-bounces@ietf.org</a> [mailto:<a href=3D"mailto:hybi-=
bounces@ietf.org" target=3D"_blank">hybi-bounces@ietf.org</a>] <b>On Behalf=
 Of </b>David Endicott<br>
<b>Sent:</b> Wednesday 20 July 2011 21:10<br><b>To:</b> Brian<br><b>Cc:</b>=
 Hybi; Greg Wilkins</span></p><div class=3D"im"><br><b>Subject:</b> Re: [hy=
bi] deflate-stream and masking<u></u><u></u></div><p></p></div></div><p cla=
ss=3D"MsoNormal">
<u></u>=A0<u></u></p><p class=3D"MsoNormal">Unlurking to agree. =A0 =A0=A0<=
u></u><u></u></p><div><div></div><div class=3D"h5"><div><p class=3D"MsoNorm=
al"><u></u>=A0<u></u></p></div><div><p class=3D"MsoNormal">When it was opti=
onal it was avoidable and non-impairing. =A0 It should not be mandatory. =
=A0 =A0<u></u><u></u></p>
<div><p class=3D"MsoNormal"><u></u>=A0<u></u></p></div><div><p class=3D"Mso=
Normal">+6.02 x10^23 votes for removal from specification (which when defla=
ted is just +1, I believe)<u></u><u></u></p></div><div><div><p class=3D"Mso=
Normal">
<u></u>=A0<u></u></p></div><div><p class=3D"MsoNormal" style=3D"margin-bott=
om:12.0pt"><u></u>=A0<u></u></p><div><p class=3D"MsoNormal">On Wed, Jul 20,=
 2011 at 5:50 AM, Brian &lt;<a href=3D"mailto:theturtle32@gmail.com" target=
=3D"_blank">theturtle32@gmail.com</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">+500. =A0deflate-stream has always been utterly ridi=
culous in light of masking. =A0It really should get the axe, and with extre=
me prejudice. =A0Why is it still in the spec? =A0I don&#39;t recall anyone =
citing a reasonable reason for keeping it, but there have been many very go=
od arguments against it. =A0It&#39;s not even very clearly specified in the=
 document -- when I implemented it in my Flash client, I had to read the so=
urce code of Andy Green&#39;s implementation to figure out how it was suppo=
sed to work -- his implementation became my specification for the extension=
.<u></u><u></u></p>
<div><p class=3D"MsoNormal"><u></u>=A0<u></u></p></div><div><p class=3D"Mso=
Normal" style=3D"margin-bottom:12.0pt"><span style=3D"color:#888888">Brian<=
u></u><u></u></span></p></div><div><div><div><p class=3D"MsoNormal"><u></u>=
=A0<u></u></p>
<div><p class=3D"MsoNormal">On Tue, Jul 19, 2011 at 7:34 PM, Greg Wilkins &=
lt;<a href=3D"mailto:gregw@intalio.com" target=3D"_blank">gregw@intalio.com=
</a>&gt; wrote:<u></u><u></u></p><p class=3D"MsoNormal">I&#39;ve just notic=
ed that the w3c is currently intending to make support<br>
for deflate-stream mandatory!<br><br>=A0<a href=3D"http://www.w3.org/Bugs/P=
ublic/show_bug.cgi?id=3D12917" target=3D"_blank">http://www.w3.org/Bugs/Pub=
lic/show_bug.cgi?id=3D12917</a><br><br>This moves this extension from being=
 useless, but mostly harmless, to<br>
being a major impost on servers and intermediaries.<br>If the browser make =
this mandatory, then servers will obviously have<br>to support it at a cost=
 of extra CPU, extra buffers but for no<br>significant savings in bandwidth=
.<br>
Intermediaries that wish to act on frame boundaries will also have to<br>im=
plement it.<br><br>This illustrate that having silly options always puts yo=
u at risk of<br>people taking you up on those options.<br><br>This extensio=
n is demonstrably broken and needs to be either fixed or removed.<br>
<br>regards<u></u><u></u></p><div><div><p class=3D"MsoNormal"><br><br><br>O=
n 20 June 2011 16:33, Greg Wilkins &lt;<a href=3D"mailto:gregw@intalio.com"=
 target=3D"_blank">gregw@intalio.com</a>&gt; wrote:<br>&gt; As part of my c=
ontinuing campaign against including deflate-stream in<br>
&gt; the specification as a standard extension, I did a quick test of how<b=
r>&gt; well it works when applied to masked frames.<br>&gt;<br>&gt; I took =
a days worth of traffic from an IRC channel and wrapped it up<br>&gt; as JS=
ON messages sent as websocket frames.<br>
&gt; There were 487 message that looked like:<br>&gt;<br>&gt; =A0 =A0 {chan=
nel:&quot;#webtide&quot;, username:&quot;tbecker&quot;, text:&quot;joakime:=
 jenkins<br>&gt; had issues pulling from github a couple of times =A0last w=
eek&quot;}<br>
&gt;<br>&gt; As an unmasked WS stream, it was 50675 bytes, and as a masked =
stream<br>&gt; is was 52623 bytes.<br>&gt; I then compressed both these str=
eams with gzip and got 13306 bytes for<br>&gt; unmasked and 51704 bytes for=
 the masked!!!!<br>
&gt;<br>&gt; So for this very typical example, masking was sufficiently ran=
dom to<br>&gt; completely negate the benefits of compression.<br>&gt;<br>&g=
t; So the deflate-stream &quot;extension&quot; is:<br>&gt;<br>&gt; =A0+ nex=
t to useless for inbound traffic<br>
&gt; =A0+ breaks all the rules of what an extension can do<br>&gt; =A0+ is =
potentially vulnerable to injection as attackers can send<br>&gt; repeated =
patterns that may subvert masking<br>&gt; =A0+ can be replaced by the in-fr=
ame compression extension already proposed.<br>
&gt; =A0+ was inserted in the draft with little or no discussion and withou=
t<br>&gt; clear consensus.<br>&gt;<br>&gt; Can I call for a straw poll of w=
ho wants to keep this extension in the spec?<br>&gt;<br>&gt;<br>&gt;<br>&gt=
; regards<br>
&gt;<br>_______________________________________________<br>hybi mailing lis=
t<br><a href=3D"mailto:hybi@ietf.org" target=3D"_blank">hybi@ietf.org</a><b=
r><a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/hybi</a><u></u><u></u></p>
</div></div></div><p class=3D"MsoNormal"><u></u>=A0<u></u></p></div></div><=
/div><p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>____________=
___________________________________<br>hybi mailing list<br><a href=3D"mail=
to:hybi@ietf.org" target=3D"_blank">hybi@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/hybi</a><u></u><u></u></p></div><p clas=
s=3D"MsoNormal"><u></u>=A0<u></u></p></div></div></div></div></div></div></=
div>
</div></blockquote></div><br></div>

--20cf301fb94d00d95f04a884c5c8--

From pmcmanus@mozilla.com  Wed Jul 20 12:43:21 2011
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Date: Wed, 20 Jul 2011 15:43:13 -0400
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Cc: Hybi <hybi@ietf.org>
Subject: Re: [hybi] deflate-stream and masking
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On Wed, 2011-07-20 at 14:55 -0400, David Endicott wrote:
> I'm basing my position on Greg's comment above that:
> 
> 
> > I've just noticed that the w3c is currently intending to make
> support
> > for deflate-stream mandatory!
> > http://www.w3.org/Bugs/Public/show_bug.cgi?id=12917
> 

The bug references the editor's draft (i.e. a work in progress) of the
API, which I would hope could still be changed.

Specifically what it does is, as I read it, is mandate that clients
implementing the JS API (i.e. browsers) must request exactly one
extension in the handshake ("deflate-stream"). It does not require that
the server accept it (indeed it notes what happens to the js attributes
if the server does not - interoperation continues without compression)
so I don't think we can fairly say that it makes deflate-stream support
mandatory in anything other than browsers and then only actually use it
when servers acquiesce.

More concerning is that it effectively bans any other transport
extensions in browsers.

This obviously doesn't match the expected use of protocol extensions,
which is a much bigger space than just transport compression. Even if
the RFC contained a perfect deflate-frame which was specified by the
API, it still interacts poorly with mux or anything else that
implementations might want to do at the transport layer. I don't
understand why the JS API is even participating in such a direct way.
(specifying that compression applies to the messages which could be used
to make the decision on what extensions to negotiate would be a more
indirect and at least on the face of it sensible way to go).

There are no changes the IETF WG can make to the ietf-draft that can fix
this mismatch.

As for compression, the most productive thing to do seems to me to be to
define in a separate draft deflate-frame and push it through the
adoption/standardization process. No matter what is defined the
extensions mechanism should (and does!) provide a mechanism for
migration for any sensible protocol implementations.

While I don't want to go to the mat for deflate-stream it certainly has
a number of use cases where it does well - most downstream traffic of >
tiny-sized messages, and large upstream messages - so I'm not in the
camp of "just throw it out".





From alex@noemax.com  Wed Jul 20 12:46:56 2011
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Subject: Re: [hybi] deflate-stream and masking
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I vote for keeping it in the spec as it is now =E2=80=93 an optional =
extension. Otherwise we will either remove deflate-stream and ship the =
protocol w/o compression and be subject to understandable criticism, or =
we will start discussing which is the optimal compression and further =
delay standardization until we complete deliberations. Otoh if we =
proceed with the spec as is then compression will be there, it will be =
optional, and we can return to add additional compression methods in the =
immediate future. No harm done to keep it in the spec as is, imho. =
Let=E2=80=99s ship.=20

=20

=20

From: David Endicott [mailto:dendicott@gmail.com]=20
Sent: Wednesday 20 July 2011 21:55
To: Alexander Philippou
Cc: Hybi
Subject: Re: [hybi] deflate-stream and masking

=20

I'm basing my position on Greg's comment above that:

=20

> I've just noticed that the w3c is currently intending to make support
> for deflate-stream mandatory!
>  <http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917> =
http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917

If I cannot avoid it in my implementations (because browsers make it =
mandatory), then I'm strongly opposed.

=20

If it remains an optional extension, then I revert to a Don't Care =
position.

=20

=20

On Wed, Jul 20, 2011 at 2:28 PM, Alexander Philippou <alex@noemax.com> =
wrote:

But currently it is an extension and thus optional, isn't it?

=20

=20

From: hybi-bounces@ietf.org [mailto:hybi-bounces@ietf.org] On Behalf Of =
David Endicott
Sent: Wednesday 20 July 2011 21:10
To: Brian
Cc: Hybi; Greg Wilkins


Subject: Re: [hybi] deflate-stream and masking

=20

Unlurking to agree.    =20

=20

When it was optional it was avoidable and non-impairing.   It should not =
be mandatory.   =20

=20

+6.02 x10^23 votes for removal from specification (which when deflated =
is just +1, I believe)

=20

=20

On Wed, Jul 20, 2011 at 5:50 AM, Brian <theturtle32@gmail.com> wrote:

+500.  deflate-stream has always been utterly ridiculous in light of =
masking.  It really should get the axe, and with extreme prejudice.  Why =
is it still in the spec?  I don't recall anyone citing a reasonable =
reason for keeping it, but there have been many very good arguments =
against it.  It's not even very clearly specified in the document -- =
when I implemented it in my Flash client, I had to read the source code =
of Andy Green's implementation to figure out how it was supposed to work =
-- his implementation became my specification for the extension.

=20

Brian

=20

On Tue, Jul 19, 2011 at 7:34 PM, Greg Wilkins <gregw@intalio.com> wrote:

I've just noticed that the w3c is currently intending to make support
for deflate-stream mandatory!

 http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917

This moves this extension from being useless, but mostly harmless, to
being a major impost on servers and intermediaries.
If the browser make this mandatory, then servers will obviously have
to support it at a cost of extra CPU, extra buffers but for no
significant savings in bandwidth.
Intermediaries that wish to act on frame boundaries will also have to
implement it.

This illustrate that having silly options always puts you at risk of
people taking you up on those options.

This extension is demonstrably broken and needs to be either fixed or =
removed.

regards




On 20 June 2011 16:33, Greg Wilkins <gregw@intalio.com> wrote:
> As part of my continuing campaign against including deflate-stream in
> the specification as a standard extension, I did a quick test of how
> well it works when applied to masked frames.
>
> I took a days worth of traffic from an IRC channel and wrapped it up
> as JSON messages sent as websocket frames.
> There were 487 message that looked like:
>
>     {channel:"#webtide", username:"tbecker", text:"joakime: jenkins
> had issues pulling from github a couple of times  last week"}
>
> As an unmasked WS stream, it was 50675 bytes, and as a masked stream
> is was 52623 bytes.
> I then compressed both these streams with gzip and got 13306 bytes for
> unmasked and 51704 bytes for the masked!!!!
>
> So for this very typical example, masking was sufficiently random to
> completely negate the benefits of compression.
>
> So the deflate-stream "extension" is:
>
>  + next to useless for inbound traffic
>  + breaks all the rules of what an extension can do
>  + is potentially vulnerable to injection as attackers can send
> repeated patterns that may subvert masking
>  + can be replaced by the in-frame compression extension already =
proposed.
>  + was inserted in the draft with little or no discussion and without
> clear consensus.
>
> Can I call for a straw poll of who wants to keep this extension in the =
spec?
>
>
>
> regards
>
_______________________________________________
hybi mailing list
hybi@ietf.org
https://www.ietf.org/mailman/listinfo/hybi

=20


_______________________________________________
hybi mailing list
hybi@ietf.org
https://www.ietf.org/mailman/listinfo/hybi

=20

=20


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>I vote for keeping it in the spec as it is now =E2=80=93 an optional =
extension. Otherwise we will either remove deflate-stream and ship the =
protocol w/o compression and be subject to understandable criticism, or =
we will start discussing which is the optimal compression and further =
delay standardization until we complete deliberations. Otoh if we =
proceed with the spec as is then compression will be there, it will be =
optional, and we can return to add additional compression methods in the =
immediate future. No harm done to keep it in the spec as is, imho. =
Let=E2=80=99s ship. <o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><div =
style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'><div><div style=3D'border:none;border-top:solid #B5C4DF =
1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
David Endicott [mailto:dendicott@gmail.com] <br><b>Sent:</b> Wednesday =
20 July 2011 21:55<br><b>To:</b> Alexander Philippou<br><b>Cc:</b> =
Hybi<br><b>Subject:</b> Re: [hybi] deflate-stream and =
masking<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>I'm basing =
my position on Greg's comment above that:<o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span class=3Dapple-style-span><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&gt; I've =
just noticed that the w3c is currently intending to make =
support</span></span><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><br><span =
class=3Dapple-style-span>&gt; for deflate-stream =
mandatory!</span><br><span class=3Dapple-style-span>&gt;&nbsp;<a =
href=3D"http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917" =
target=3D"_blank"><span =
style=3D'color:#1C51A8'>http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D1=
2917</span></a></span></span><o:p></o:p></p></div><div><p =
class=3DMsoNormal>If I cannot avoid it in my implementations (because =
browsers make it mandatory), then I'm strongly =
opposed.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>If it remains an optional extension, then I revert to =
a Don't Care position.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal>On Wed, =
Jul 20, 2011 at 2:28 PM, Alexander Philippou &lt;<a =
href=3D"mailto:alex@noemax.com">alex@noemax.com</a>&gt; =
wrote:<o:p></o:p></p><div><div><p>But currently it is an extension and =
thus optional, isn't it?<o:p></o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;color:#1F497D'>&nbsp;</span><o:p></o:p></p><div=
 style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'><div><div style=3D'border:none;border-top:solid #B5C4DF =
1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span =
style=3D'font-size:10.0pt'>From:</span></b><span =
style=3D'font-size:10.0pt'> <a href=3D"mailto:hybi-bounces@ietf.org" =
target=3D"_blank">hybi-bounces@ietf.org</a> [mailto:<a =
href=3D"mailto:hybi-bounces@ietf.org" =
target=3D"_blank">hybi-bounces@ietf.org</a>] <b>On Behalf Of </b>David =
Endicott<br><b>Sent:</b> Wednesday 20 July 2011 21:10<br><b>To:</b> =
Brian<br><b>Cc:</b> Hybi; Greg Wilkins</span><o:p></o:p></p><div><p =
class=3DMsoNormal><br><b>Subject:</b> Re: [hybi] deflate-stream and =
masking<o:p></o:p></p></div></div></div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Unlurking =
to agree. &nbsp; &nbsp;&nbsp;<o:p></o:p></p><div><div><div><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p></div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>When it was =
optional it was avoidable and non-impairing. &nbsp; It should not be =
mandatory. &nbsp; &nbsp;<o:p></o:p></p><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p></div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>+6.02 =
x10^23 votes for removal from specification (which when deflated is just =
+1, I believe)<o:p></o:p></p></div><div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p></div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;margin-bottom:12.0pt'>&nbsp;<o:p></o:p><=
/p><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>On Wed, Jul =
20, 2011 at 5:50 AM, Brian &lt;<a href=3D"mailto:theturtle32@gmail.com" =
target=3D"_blank">theturtle32@gmail.com</a>&gt; wrote:<o:p></o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>+500. =
&nbsp;deflate-stream has always been utterly ridiculous in light of =
masking. &nbsp;It really should get the axe, and with extreme prejudice. =
&nbsp;Why is it still in the spec? &nbsp;I don't recall anyone citing a =
reasonable reason for keeping it, but there have been many very good =
arguments against it. &nbsp;It's not even very clearly specified in the =
document -- when I implemented it in my Flash client, I had to read the =
source code of Andy Green's implementation to figure out how it was =
supposed to work -- his implementation became my specification for the =
extension.<o:p></o:p></p><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p></div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;margin-bottom:12.0pt'><span =
style=3D'color:#888888'>Brian</span><o:p></o:p></p></div><div><div><div><=
p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>On Tue, Jul =
19, 2011 at 7:34 PM, Greg Wilkins &lt;<a =
href=3D"mailto:gregw@intalio.com" =
target=3D"_blank">gregw@intalio.com</a>&gt; wrote:<o:p></o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>I've just =
noticed that the w3c is currently intending to make support<br>for =
deflate-stream mandatory!<br><br>&nbsp;<a =
href=3D"http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917" =
target=3D"_blank">http://www.w3.org/Bugs/Public/show_bug.cgi?id=3D12917</=
a><br><br>This moves this extension from being useless, but mostly =
harmless, to<br>being a major impost on servers and =
intermediaries.<br>If the browser make this mandatory, then servers will =
obviously have<br>to support it at a cost of extra CPU, extra buffers =
but for no<br>significant savings in bandwidth.<br>Intermediaries that =
wish to act on frame boundaries will also have to<br>implement =
it.<br><br>This illustrate that having silly options always puts you at =
risk of<br>people taking you up on those options.<br><br>This extension =
is demonstrably broken and needs to be either fixed or =
removed.<br><br>regards<o:p></o:p></p><div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><br><br><br>=
On 20 June 2011 16:33, Greg Wilkins &lt;<a =
href=3D"mailto:gregw@intalio.com" =
target=3D"_blank">gregw@intalio.com</a>&gt; wrote:<br>&gt; As part of my =
continuing campaign against including deflate-stream in<br>&gt; the =
specification as a standard extension, I did a quick test of how<br>&gt; =
well it works when applied to masked frames.<br>&gt;<br>&gt; I took a =
days worth of traffic from an IRC channel and wrapped it up<br>&gt; as =
JSON messages sent as websocket frames.<br>&gt; There were 487 message =
that looked like:<br>&gt;<br>&gt; &nbsp; &nbsp; =
{channel:&quot;#webtide&quot;, username:&quot;tbecker&quot;, =
text:&quot;joakime: jenkins<br>&gt; had issues pulling from github a =
couple of times &nbsp;last week&quot;}<br>&gt;<br>&gt; As an unmasked WS =
stream, it was 50675 bytes, and as a masked stream<br>&gt; is was 52623 =
bytes.<br>&gt; I then compressed both these streams with gzip and got =
13306 bytes for<br>&gt; unmasked and 51704 bytes for the =
masked!!!!<br>&gt;<br>&gt; So for this very typical example, masking was =
sufficiently random to<br>&gt; completely negate the benefits of =
compression.<br>&gt;<br>&gt; So the deflate-stream &quot;extension&quot; =
is:<br>&gt;<br>&gt; &nbsp;+ next to useless for inbound traffic<br>&gt; =
&nbsp;+ breaks all the rules of what an extension can do<br>&gt; &nbsp;+ =
is potentially vulnerable to injection as attackers can send<br>&gt; =
repeated patterns that may subvert masking<br>&gt; &nbsp;+ can be =
replaced by the in-frame compression extension already proposed.<br>&gt; =
&nbsp;+ was inserted in the draft with little or no discussion and =
without<br>&gt; clear consensus.<br>&gt;<br>&gt; Can I call for a straw =
poll of who wants to keep this extension in the =
spec?<br>&gt;<br>&gt;<br>&gt;<br>&gt; =
regards<br>&gt;<br>_______________________________________________<br>hyb=
i mailing list<br><a href=3D"mailto:hybi@ietf.org" =
target=3D"_blank">hybi@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/hybi" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/hybi</a><o:p></o:=
p></p></div></div></div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p></div></div></div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;margin-bottom:12.0pt'><br>______________=
_________________________________<br>hybi mailing list<br><a =
href=3D"mailto:hybi@ietf.org" target=3D"_blank">hybi@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/hybi" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/hybi</a><o:p></o:=
p></p></div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p></div></div></div></div></div></div></div></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></div></body></html>
------=_NextPart_000_00E8_01CC472E.E9ECF4F0--


From stpeter@stpeter.im  Wed Jul 20 12:48:59 2011
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Cc: Hybi <hybi@ietf.org>
Subject: Re: [hybi] deflate-stream and masking
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<hat type='individual'/>

On 7/20/11 1:43 PM, Patrick McManus wrote:

> As for compression, the most productive thing to do seems to me to be to
> define in a separate draft deflate-frame and push it through the
> adoption/standardization process. No matter what is defined the
> extensions mechanism should (and does!) provide a mechanism for
> migration for any sensible protocol implementations.
>
> While I don't want to go to the mat for deflate-stream it certainly has
> a number of use cases where it does well - most downstream traffic of>
> tiny-sized messages, and large upstream messages - so I'm not in the
> camp of "just throw it out".

My reading of the discussion is that people don't necessarily want to 
throw out deflate-stream, but they don't think it belongs in the base 
spec and they would prefer to see it defined in a separate draft, just 
as you propose for deflate-frame.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From Gabriel.Montenegro@microsoft.com  Wed Jul 20 13:00:11 2011
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From: Gabriel Montenegro <Gabriel.Montenegro@microsoft.com>
To: Peter Saint-Andre <stpeter@stpeter.im>, Patrick McManus <pmcmanus@mozilla.com>
Thread-Topic: [hybi] deflate-stream and masking
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<As individual>

That is my reading as well.=20

<as chair>

Patrick, would you be able to own that separate spec?

> -----Original Message-----
> From: hybi-bounces@ietf.org [mailto:hybi-bounces@ietf.org] On Behalf Of
> Peter Saint-Andre
> Sent: Wednesday, July 20, 2011 12:49
> To: Patrick McManus
> Cc: Hybi
> Subject: Re: [hybi] deflate-stream and masking
>=20
> <hat type=3D'individual'/>
>=20
> On 7/20/11 1:43 PM, Patrick McManus wrote:
>=20
> > As for compression, the most productive thing to do seems to me to be
> > to define in a separate draft deflate-frame and push it through the
> > adoption/standardization process. No matter what is defined the
> > extensions mechanism should (and does!) provide a mechanism for
> > migration for any sensible protocol implementations.
> >
> > While I don't want to go to the mat for deflate-stream it certainly
> > has a number of use cases where it does well - most downstream traffic
> > of> tiny-sized messages, and large upstream messages - so I'm not in
> > the camp of "just throw it out".
>=20
> My reading of the discussion is that people don't necessarily want to thr=
ow out
> deflate-stream, but they don't think it belongs in the base spec and they=
 would
> prefer to see it defined in a separate draft, just as you propose for def=
late-frame.
>=20
> Peter
>=20
> --
> Peter Saint-Andre
> https://stpeter.im/
>=20
>=20
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi


From pmcmanus@mozilla.com  Wed Jul 20 13:04:40 2011
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I really believe the base spec must have a compression definition.

The point of my message was that the W3C editor's draft should not turn
into a change in direction on compression on our part because the scope
of that issue is much greater.

On Wed, 2011-07-20 at 20:00 +0000, Gabriel Montenegro wrote:
> <As individual>
> 
> That is my reading as well. 
> 
> <as chair>
> 
> Patrick, would you be able to own that separate spec?
> 
> > -----Original Message-----
> > From: hybi-bounces@ietf.org [mailto:hybi-bounces@ietf.org] On Behalf Of
> > Peter Saint-Andre
> > Sent: Wednesday, July 20, 2011 12:49
> > To: Patrick McManus
> > Cc: Hybi
> > Subject: Re: [hybi] deflate-stream and masking
> > 
> > <hat type='individual'/>
> > 
> > On 7/20/11 1:43 PM, Patrick McManus wrote:
> > 
> > > As for compression, the most productive thing to do seems to me to be
> > > to define in a separate draft deflate-frame and push it through the
> > > adoption/standardization process. No matter what is defined the
> > > extensions mechanism should (and does!) provide a mechanism for
> > > migration for any sensible protocol implementations.
> > >
> > > While I don't want to go to the mat for deflate-stream it certainly
> > > has a number of use cases where it does well - most downstream traffic
> > > of> tiny-sized messages, and large upstream messages - so I'm not in
> > > the camp of "just throw it out".
> > 
> > My reading of the discussion is that people don't necessarily want to throw out
> > deflate-stream, but they don't think it belongs in the base spec and they would
> > prefer to see it defined in a separate draft, just as you propose for deflate-frame.
> > 
> > Peter
> > 
> > --
> > Peter Saint-Andre
> > https://stpeter.im/
> > 
> > 
> > _______________________________________________
> > hybi mailing list
> > hybi@ietf.org
> > https://www.ietf.org/mailman/listinfo/hybi
> 



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From: John Tamplin <jat@google.com>
Date: Wed, 20 Jul 2011 16:29:19 -0400
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Subject: Re: [hybi] deflate-stream and masking
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On Wed, Jul 20, 2011 at 4:04 PM, Patrick McManus <pmcmanus@mozilla.com>wrote:

> I really believe the base spec must have a compression definition.
>
> The point of my message was that the W3C editor's draft should not turn
> into a change in direction on compression on our part because the scope
> of that issue is much greater.


I agree with concerns that deflate-stream is less useful than it was
originally since C->S masking has been added, but that doesn't mean it is of
no use at all.  It was originally added to provide "better than nothing"
compression and an example extension in the spec.  Those benefits still
exist, so I would prefer not to simply remove it.

I have no objection to replacing it with deflate-frame if we can get
agreement on it quickly.

I also don't understand Ian Hickson's position, but I don't think that
alters how we define the wire protocol and should be addressed there and not
here.

-- 
John A. Tamplin
Software Engineer (GWT), Google

--000e0cd32b946f755104a88617a5
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<div class=3D"gmail_quote">On Wed, Jul 20, 2011 at 4:04 PM, Patrick McManus=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:pmcmanus@mozilla.com">pmcmanus@moz=
illa.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

I really believe the base spec must have a compression definition.<br>
<br>
The point of my message was that the W3C editor&#39;s draft should not turn=
<br>
into a change in direction on compression on our part because the scope<br>
of that issue is much greater.</blockquote><div><br></div><div>I agree with=
 concerns that deflate-stream is less useful than it was originally since C=
-&gt;S masking has been added, but that doesn&#39;t mean it is of no use at=
 all. =C2=A0It was originally added to provide &quot;better than nothing&qu=
ot; compression and an example extension in the spec. =C2=A0Those benefits =
still exist, so I would prefer not to simply remove it.</div>

<div><br></div><div>I have no objection to replacing it with deflate-frame =
if we can get agreement on it quickly.</div><div><br></div><div>I also don&=
#39;t understand Ian Hickson&#39;s position, but I don&#39;t think that alt=
ers how we define the wire protocol and should be addressed there and not h=
ere.</div>

</div><br>-- <br>John A. Tamplin<br>Software Engineer (GWT), Google<br>

--000e0cd32b946f755104a88617a5--

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Date: Wed, 20 Jul 2011 16:36:04 -0400
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Whereas, I really believe it shouldn't.   (But not enough to delay
realization of WS)

TCP doesn't have compression.  HTTP does.    TCP is a transport layer.  HTTP
is an application layer.   Websockets is transport layer.  Et voila, my
reasoning.

I've always understood WS as a protocol that defines how content is
exchanged asynchronously between "web" peers.   It is a framing mechanism
for TCP.   It is not an application layer (it doesn't define "verbs", like
GET, POST, HEAD, etc.).    Transport layers should be content agnostic.
Compression is a function of that which uses the transport layer (ie.
application layer).    But, I've said this before, and there is no reason
to belabor the point.

If I have to implement the extension as a mandatory component, I'll be
pissed off, but I'll do it.   My needs for a non-XHR kludge to get async
communication out-weighs my distaste for the extension.  I understand other
peoples needs may differ from mine.




On Wed, Jul 20, 2011 at 4:04 PM, Patrick McManus <pmcmanus@mozilla.com>wrote:

> I really believe the base spec must have a compression definition.
>
> The point of my message was that the W3C editor's draft should not turn
> into a change in direction on compression on our part because the scope
> of that issue is much greater.
>
> On Wed, 2011-07-20 at 20:00 +0000, Gabriel Montenegro wrote:
> > <As individual>
> >
> > That is my reading as well.
> >
> > <as chair>
> >
> > Patrick, would you be able to own that separate spec?
> >
> > > -----Original Message-----
> > > From: hybi-bounces@ietf.org [mailto:hybi-bounces@ietf.org] On Behalf
> Of
> > > Peter Saint-Andre
> > > Sent: Wednesday, July 20, 2011 12:49
> > > To: Patrick McManus
> > > Cc: Hybi
> > > Subject: Re: [hybi] deflate-stream and masking
> > >
> > > <hat type='individual'/>
> > >
> > > On 7/20/11 1:43 PM, Patrick McManus wrote:
> > >
> > > > As for compression, the most productive thing to do seems to me to be
> > > > to define in a separate draft deflate-frame and push it through the
> > > > adoption/standardization process. No matter what is defined the
> > > > extensions mechanism should (and does!) provide a mechanism for
> > > > migration for any sensible protocol implementations.
> > > >
> > > > While I don't want to go to the mat for deflate-stream it certainly
> > > > has a number of use cases where it does well - most downstream
> traffic
> > > > of> tiny-sized messages, and large upstream messages - so I'm not in
> > > > the camp of "just throw it out".
> > >
> > > My reading of the discussion is that people don't necessarily want to
> throw out
> > > deflate-stream, but they don't think it belongs in the base spec and
> they would
> > > prefer to see it defined in a separate draft, just as you propose for
> deflate-frame.
> > >
> > > Peter
> > >
> > > --
> > > Peter Saint-Andre
> > > https://stpeter.im/
> > >
> > >
> > > _______________________________________________
> > > hybi mailing list
> > > hybi@ietf.org
> > > https://www.ietf.org/mailman/listinfo/hybi
> >
>
>
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>

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Whereas, I really believe it shouldn&#39;t. =A0 (But not enough to delay re=
alization of WS)<div><br></div><div>TCP doesn&#39;t have compression. =A0HT=
TP does. =A0 =A0TCP is a transport layer. =A0HTTP is an application layer. =
=A0=A0Websockets is transport layer. =A0Et voila, my reasoning.</div>
<div><br></div><div>I&#39;ve always understood WS as a protocol that define=
s how content is exchanged asynchronously between &quot;web&quot; peers. =
=A0 It is a framing mechanism for TCP. =A0 It is not an application layer (=
it doesn&#39;t define &quot;verbs&quot;, like GET, POST, HEAD, etc.). =A0 =
=A0Transport layers should be content agnostic. =A0 Compression is a functi=
on of that which uses the transport layer (ie. application layer). =A0 =A0B=
ut, I&#39;ve said this before, and there is no reason to=A0belabor=A0the po=
int.</div>
<div><br></div><div>If I have to implement the extension as a mandatory com=
ponent, I&#39;ll be pissed off, but I&#39;ll do it. =A0 My needs for a non-=
XHR kludge to get async communication out-weighs my distaste for the extens=
ion. =A0I understand other peoples needs may differ from mine.</div>
<div><br></div><div><br></div><div><br></div><div><br><div class=3D"gmail_q=
uote">On Wed, Jul 20, 2011 at 4:04 PM, Patrick McManus <span dir=3D"ltr">&l=
t;<a href=3D"mailto:pmcmanus@mozilla.com">pmcmanus@mozilla.com</a>&gt;</spa=
n> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">I really believe the base spec must have a =
compression definition.<br>
<br>
The point of my message was that the W3C editor&#39;s draft should not turn=
<br>
into a change in direction on compression on our part because the scope<br>
of that issue is much greater.<br>
<div><div></div><div class=3D"h5"><br>
On Wed, 2011-07-20 at 20:00 +0000, Gabriel Montenegro wrote:<br>
&gt; &lt;As individual&gt;<br>
&gt;<br>
&gt; That is my reading as well.<br>
&gt;<br>
&gt; &lt;as chair&gt;<br>
&gt;<br>
&gt; Patrick, would you be able to own that separate spec?<br>
&gt;<br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: <a href=3D"mailto:hybi-bounces@ietf.org">hybi-bounces@ietf.=
org</a> [mailto:<a href=3D"mailto:hybi-bounces@ietf.org">hybi-bounces@ietf.=
org</a>] On Behalf Of<br>
&gt; &gt; Peter Saint-Andre<br>
&gt; &gt; Sent: Wednesday, July 20, 2011 12:49<br>
&gt; &gt; To: Patrick McManus<br>
&gt; &gt; Cc: Hybi<br>
&gt; &gt; Subject: Re: [hybi] deflate-stream and masking<br>
&gt; &gt;<br>
&gt; &gt; &lt;hat type=3D&#39;individual&#39;/&gt;<br>
&gt; &gt;<br>
&gt; &gt; On 7/20/11 1:43 PM, Patrick McManus wrote:<br>
&gt; &gt;<br>
&gt; &gt; &gt; As for compression, the most productive thing to do seems to=
 me to be<br>
&gt; &gt; &gt; to define in a separate draft deflate-frame and push it thro=
ugh the<br>
&gt; &gt; &gt; adoption/standardization process. No matter what is defined =
the<br>
&gt; &gt; &gt; extensions mechanism should (and does!) provide a mechanism =
for<br>
&gt; &gt; &gt; migration for any sensible protocol implementations.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; While I don&#39;t want to go to the mat for deflate-stream i=
t certainly<br>
&gt; &gt; &gt; has a number of use cases where it does well - most downstre=
am traffic<br>
&gt; &gt; &gt; of&gt; tiny-sized messages, and large upstream messages - so=
 I&#39;m not in<br>
&gt; &gt; &gt; the camp of &quot;just throw it out&quot;.<br>
&gt; &gt;<br>
&gt; &gt; My reading of the discussion is that people don&#39;t necessarily=
 want to throw out<br>
&gt; &gt; deflate-stream, but they don&#39;t think it belongs in the base s=
pec and they would<br>
&gt; &gt; prefer to see it defined in a separate draft, just as you propose=
 for deflate-frame.<br>
&gt; &gt;<br>
&gt; &gt; Peter<br>
&gt; &gt;<br>
&gt; &gt; --<br>
&gt; &gt; Peter Saint-Andre<br>
&gt; &gt; <a href=3D"https://stpeter.im/" target=3D"_blank">https://stpeter=
.im/</a><br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; hybi mailing list<br>
&gt; &gt; <a href=3D"mailto:hybi@ietf.org">hybi@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"=
_blank">https://www.ietf.org/mailman/listinfo/hybi</a><br>
&gt;<br>
<br>
<br>
_______________________________________________<br>
hybi mailing list<br>
<a href=3D"mailto:hybi@ietf.org">hybi@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/hybi</a><br>
</div></div></blockquote></div><br></div>

--0016364d2c7973ca6804a8862e88--

From jat@google.com  Wed Jul 20 13:45:00 2011
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From: John Tamplin <jat@google.com>
Date: Wed, 20 Jul 2011 16:44:36 -0400
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To: David Endicott <dendicott@gmail.com>
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Subject: Re: [hybi] deflate-stream and masking
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On Wed, Jul 20, 2011 at 4:36 PM, David Endicott <dendicott@gmail.com> wrote:
>
> I've always understood WS as a protocol that defines how content is
> exchanged asynchronously between "web" peers.   It is a framing mechanism
> for TCP.   It is not an application layer (it doesn't define "verbs", like
> GET, POST, HEAD, etc.).    Transport layers should be content agnostic.
> Compression is a function of that which uses the transport layer (ie.
> application layer).    But, I've said this before, and there is no reason
> to belabor the point.
>

The point is that practically all content which will be going over
WebSockets will benefit from compression (see my experiments documented here
a year and a half ago, which included clients sending binary data).  Without
compression built into the spec, developers may be forced into a position of
continuing to use XHR hacks because the lack of compression makes WS slower
(for example, Google Instant search).

-- 
John A. Tamplin
Software Engineer (GWT), Google

--000e0cd32b94276b3004a8864eec
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<div class=3D"gmail_quote">On Wed, Jul 20, 2011 at 4:36 PM, David Endicott =
<span dir=3D"ltr">&lt;<a href=3D"mailto:dendicott@gmail.com">dendicott@gmai=
l.com</a>&gt;</span> wrote:<blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

<div></div><div>I&#39;ve always understood WS as a protocol that defines ho=
w content is exchanged asynchronously between &quot;web&quot; peers. =C2=A0=
 It is a framing mechanism for TCP. =C2=A0 It is not an application layer (=
it doesn&#39;t define &quot;verbs&quot;, like GET, POST, HEAD, etc.). =C2=
=A0 =C2=A0Transport layers should be content agnostic. =C2=A0 Compression i=
s a function of that which uses the transport layer (ie. application layer)=
. =C2=A0 =C2=A0But, I&#39;ve said this before, and there is no reason to=C2=
=A0belabor=C2=A0the point.</div>

</blockquote><div><br></div><div>The point is that practically all content =
which will be going over WebSockets will benefit from compression (see my e=
xperiments documented here a year and a half ago, which included clients se=
nding binary data). =C2=A0Without compression built into the spec, develope=
rs may be forced into a position of continuing to use XHR hacks because the=
 lack of compression makes WS slower (for example, Google Instant search).<=
/div>

</div><br>-- <br>John A. Tamplin<br>Software Engineer (GWT), Google<br>

--000e0cd32b94276b3004a8864eec--

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From: David Endicott <dendicott@gmail.com>
To: John Tamplin <jat@google.com>
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Subject: Re: [hybi] deflate-stream and masking
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Agreed.   The vast majority of (unencrypted, unmasked, unmangled) content
will benefit from compression.   That is a truth I cannot debate.

My use cases will not.    My "slowness" concern is latency, not throughput.
  My traffic will be many small messages and arrives at the WS layer already
encrypted, so compression doesn't work for me to any great extent.   The
turn-around latency of XHR is my biggest concern.   The HTTP
request/response cycle is my problem.   WS solves my latency concerns, and
beyond that I'd like it to just stay out of my way.

I acknowledge that other peoples needs will differ from mine.   I'm just
lazy enough to want to avoid doing work I don't need.


On Wed, Jul 20, 2011 at 4:44 PM, John Tamplin <jat@google.com> wrote:

> On Wed, Jul 20, 2011 at 4:36 PM, David Endicott <dendicott@gmail.com>wrote:
>>
>> I've always understood WS as a protocol that defines how content is
>> exchanged asynchronously between "web" peers.   It is a framing mechanism
>> for TCP.   It is not an application layer (it doesn't define "verbs", like
>> GET, POST, HEAD, etc.).    Transport layers should be content agnostic.
>> Compression is a function of that which uses the transport layer (ie.
>> application layer).    But, I've said this before, and there is no reason
>> to belabor the point.
>>
>
> The point is that practically all content which will be going over
> WebSockets will benefit from compression (see my experiments documented here
> a year and a half ago, which included clients sending binary data).  Without
> compression built into the spec, developers may be forced into a position of
> continuing to use XHR hacks because the lack of compression makes WS slower
> (for example, Google Instant search).
>
> --
> John A. Tamplin
> Software Engineer (GWT), Google
>

--000e0cd342b64cc14604a88675af
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Agreed. =A0 The vast majority of (unencrypted, unmasked, unmangled) content=
 will benefit from compression. =A0 That is a truth I cannot debate.<div><b=
r></div><div>My use cases will not. =A0 =A0My &quot;slowness&quot; concern =
is latency, not throughput. =A0 My traffic will be many small messages and =
arrives at the WS layer already encrypted, so compression doesn&#39;t work =
for me to any great extent. =A0 The turn-around latency of XHR is my bigges=
t concern. =A0 The HTTP request/response cycle is my problem. =A0 WS solves=
 my latency concerns, and beyond that I&#39;d like it to just stay out of m=
y way.</div>
<div><br></div><div><meta http-equiv=3D"content-type" content=3D"text/html;=
 charset=3Dutf-8">I acknowledge that other peoples needs will differ from m=
ine. =A0 I&#39;m just lazy enough to want to avoid doing work I don&#39;t n=
eed.</div>
<div><br></div><div><br><div class=3D"gmail_quote">On Wed, Jul 20, 2011 at =
4:44 PM, John Tamplin <span dir=3D"ltr">&lt;<a href=3D"mailto:jat@google.co=
m">jat@google.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div class=3D"gmail_quote"><div class=3D"im">On Wed, Jul 20, 2011 at 4:36 P=
M, David Endicott <span dir=3D"ltr">&lt;<a href=3D"mailto:dendicott@gmail.c=
om" target=3D"_blank">dendicott@gmail.com</a>&gt;</span> wrote:<blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex">


<div></div><div>I&#39;ve always understood WS as a protocol that defines ho=
w content is exchanged asynchronously between &quot;web&quot; peers. =A0 It=
 is a framing mechanism for TCP. =A0 It is not an application layer (it doe=
sn&#39;t define &quot;verbs&quot;, like GET, POST, HEAD, etc.). =A0 =A0Tran=
sport layers should be content agnostic. =A0 Compression is a function of t=
hat which uses the transport layer (ie. application layer). =A0 =A0But, I&#=
39;ve said this before, and there is no reason to=A0belabor=A0the point.</d=
iv>


</blockquote><div><br></div></div><div>The point is that practically all co=
ntent which will be going over WebSockets will benefit from compression (se=
e my experiments documented here a year and a half ago, which included clie=
nts sending binary data). =A0Without compression built into the spec, devel=
opers may be forced into a position of continuing to use XHR hacks because =
the lack of compression makes WS slower (for example, Google Instant search=
).</div>


</div><div><div></div><div class=3D"h5"><br>-- <br>John A. Tamplin<br>Softw=
are Engineer (GWT), Google<br>
</div></div></blockquote></div><br></div>

--000e0cd342b64cc14604a88675af--

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From: John Tamplin <jat@google.com>
Date: Wed, 20 Jul 2011 16:58:21 -0400
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To: David Endicott <dendicott@gmail.com>
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Cc: Hybi <hybi@ietf.org>, Patrick McManus <pmcmanus@mozilla.com>, Gabriel Montenegro <Gabriel.Montenegro@microsoft.com>
Subject: Re: [hybi] deflate-stream and masking
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On Wed, Jul 20, 2011 at 4:55 PM, David Endicott <dendicott@gmail.com> wrote:

> Agreed.   The vast majority of (unencrypted, unmasked, unmangled) content
> will benefit from compression.   That is a truth I cannot debate.
>
> My use cases will not.    My "slowness" concern is latency, not throughput.
>   My traffic will be many small messages and arrives at the WS layer already
> encrypted, so compression doesn't work for me to any great extent.   The
> turn-around latency of XHR is my biggest concern.   The HTTP
> request/response cycle is my problem.   WS solves my latency concerns, and
> beyond that I'd like it to just stay out of my way.
>
> I acknowledge that other peoples needs will differ from mine.   I'm just
> lazy enough to want to avoid doing work I don't need.
>

Then even if the w3c position that all clients request deflate-stream
stands, then your server is free to refuse that extension.  Likewise if
deflate-frame is requested.

-- 
John A. Tamplin
Software Engineer (GWT), Google

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<br><br><div class=3D"gmail_quote">On Wed, Jul 20, 2011 at 4:55 PM, David E=
ndicott <span dir=3D"ltr">&lt;<a href=3D"mailto:dendicott@gmail.com">dendic=
ott@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

Agreed. =C2=A0 The vast majority of (unencrypted, unmasked, unmangled) cont=
ent will benefit from compression. =C2=A0 That is a truth I cannot debate.<=
div><br></div><div>My use cases will not. =C2=A0 =C2=A0My &quot;slowness&qu=
ot; concern is latency, not throughput. =C2=A0 My traffic will be many smal=
l messages and arrives at the WS layer already encrypted, so compression do=
esn&#39;t work for me to any great extent. =C2=A0 The turn-around latency o=
f XHR is my biggest concern. =C2=A0 The HTTP request/response cycle is my p=
roblem. =C2=A0 WS solves my latency concerns, and beyond that I&#39;d like =
it to just stay out of my way.</div>


<div><br></div><div>I acknowledge that other peoples needs will differ from=
 mine. =C2=A0 I&#39;m just lazy enough to want to avoid doing work I don&#3=
9;t need.</div></blockquote><div><br></div><div>Then even if the w3c positi=
on that all clients request deflate-stream stands, then your server is free=
 to refuse that extension. =C2=A0Likewise if deflate-frame is requested.=C2=
=A0</div>

</div><br>-- <br>John A. Tamplin<br>Software Engineer (GWT), Google<br>

--000e0cd5c4be44021e04a8867fa7--

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From: David Endicott <dendicott@gmail.com>
To: John Tamplin <jat@google.com>
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Subject: Re: [hybi] deflate-stream and masking
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My reading and my concern (and the de-lurking to comment) is that it seems
its possible the client might request it, get refused and close the
connection instead of continuing.

But, of course, browsers would never do anything like that to influence
protocol implementations, would they?


On Wed, Jul 20, 2011 at 4:58 PM, John Tamplin <jat@google.com> wrote:

>
>
> On Wed, Jul 20, 2011 at 4:55 PM, David Endicott <dendicott@gmail.com>wrote:
>
>> Agreed.   The vast majority of (unencrypted, unmasked, unmangled) content
>> will benefit from compression.   That is a truth I cannot debate.
>>
>> My use cases will not.    My "slowness" concern is latency, not
>> throughput.   My traffic will be many small messages and arrives at the WS
>> layer already encrypted, so compression doesn't work for me to any great
>> extent.   The turn-around latency of XHR is my biggest concern.   The HTTP
>> request/response cycle is my problem.   WS solves my latency concerns, and
>> beyond that I'd like it to just stay out of my way.
>>
>> I acknowledge that other peoples needs will differ from mine.   I'm just
>> lazy enough to want to avoid doing work I don't need.
>>
>
> Then even if the w3c position that all clients request deflate-stream
> stands, then your server is free to refuse that extension.  Likewise if
> deflate-frame is requested.
>
> --
> John A. Tamplin
> Software Engineer (GWT), Google
>

--000e0cd3b54e467b2604a88698c2
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My reading and my concern (and the de-lurking to comment) is that it seems =
its possible the client might request it, get refused and close the connect=
ion instead of continuing. =A0=A0<div><br></div><div>But, of course, browse=
rs would never do anything like that to influence protocol implementations,=
 would they?<div>
<br><br><div class=3D"gmail_quote">On Wed, Jul 20, 2011 at 4:58 PM, John Ta=
mplin <span dir=3D"ltr">&lt;<a href=3D"mailto:jat@google.com">jat@google.co=
m</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<br><br><div class=3D"gmail_quote"><div class=3D"im">On Wed, Jul 20, 2011 a=
t 4:55 PM, David Endicott <span dir=3D"ltr">&lt;<a href=3D"mailto:dendicott=
@gmail.com" target=3D"_blank">dendicott@gmail.com</a>&gt;</span> wrote:<br>=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">


Agreed. =A0 The vast majority of (unencrypted, unmasked, unmangled) content=
 will benefit from compression. =A0 That is a truth I cannot debate.<div><b=
r></div><div>My use cases will not. =A0 =A0My &quot;slowness&quot; concern =
is latency, not throughput. =A0 My traffic will be many small messages and =
arrives at the WS layer already encrypted, so compression doesn&#39;t work =
for me to any great extent. =A0 The turn-around latency of XHR is my bigges=
t concern. =A0 The HTTP request/response cycle is my problem. =A0 WS solves=
 my latency concerns, and beyond that I&#39;d like it to just stay out of m=
y way.</div>



<div><br></div><div>I acknowledge that other peoples needs will differ from=
 mine. =A0 I&#39;m just lazy enough to want to avoid doing work I don&#39;t=
 need.</div></blockquote><div><br></div></div><div>Then even if the w3c pos=
ition that all clients request deflate-stream stands, then your server is f=
ree to refuse that extension. =A0Likewise if deflate-frame is requested.=A0=
</div>


</div><div><div></div><div class=3D"h5"><br>-- <br>John A. Tamplin<br>Softw=
are Engineer (GWT), Google<br>
</div></div></blockquote></div><br></div></div>

--000e0cd3b54e467b2604a88698c2--

From w@1wt.eu  Wed Jul 20 14:11:07 2011
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Date: Wed, 20 Jul 2011 23:11:01 +0200
From: Willy Tarreau <w@1wt.eu>
To: Alexander Philippou <alex@noemax.com>
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Cc: 'Hybi' <hybi@ietf.org>
Subject: Re: [hybi] deflate-stream and masking
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On Wed, Jul 20, 2011 at 10:46:43PM +0300, Alexander Philippou wrote:
> I vote for keeping it in the spec as it is now ??? an optional extension. Otherwise we will either remove deflate-stream and ship the protocol w/o compression and be subject to understandable criticism, or we will start discussing which is the optimal compression and further delay standardization until we complete deliberations. Otoh if we proceed with the spec as is then compression will be there, it will be optional, and we can return to add additional compression methods in the immediate future. No harm done to keep it in the spec as is, imho. Let???s ship. 

The issue is important for intermediaries. They will have to uncompress
then compress stream in order to forward it. This can be extremlely
expensive in terms of CPU usage and added latency, and I already expect
that a number of intermediaries will recompress with null-compression
in order to save CPU cycles where they can. This will result in an
inflated data stream in the end and higher latencies.

So I vote for removing it and quickly work on defining the deflate-frame
extension as well.

Regards,
Willy


From ibc@aliax.net  Wed Jul 20 14:13:04 2011
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
To: Dave Cridland <dave@cridland.net>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/19 Dave Cridland <dave@cridland.net>:
>> Hi, I assume there is no interest in making DNS SRV mechanism exposed
>> in http://tools.ietf.org/html/draft-ibc-websocket-dns-srv-02 part of
>> the WebSocket core specification, neither referencing it (in the same
>> way RFC 3261 "SIP protocol" mandates the usage of RFC 3263 "Locating
>> SIP servers").
>>
>> As said before, making such DNS SRV specification an extension (so
>> present in other document) will mean no success at all, as WebSocket
>> client implementors (i.e. webbrowser vendors) will not be mandated to
>> implement it and service providers could not rely on the support of
>> DNS SRV in web browsers. So nobody will use them (because IE10 decided
>> not to implement it, for example). IMHO this is sad due the real
>> advantages DNS SRV provides for a protocol like WebSocket.
>>
>> Yes, in HTTP there is no special DNS stuff, all the load-balancing and
>> failover mechanism are done at server side with very complex and
>> expensive solutions (www.facebook.com resolves to a single IPv4 !!!!).
>> The question is: should we also inherit every HTTP limitation in
>> WebSocket?
>
> I agree wholeheartedly with this, and strongly recommend that mandatory u=
se
> of SRV is included in the core protocol.
>
> I think with HTTP's very short lived requests, then it's possible to work
> around the lack of SRV support (at a cost), but the benefit is markedly
> higher with the long-lived, stateful sessions we're anticipating with
> WebSocket.


Unfortunaltely it seems that the debate about DNS SRV support does not
interest to the core WG authors. I would like, at least, to receive
good arguments not to include/mandate DNS SRV support in the draft. If
not, the proposal is just being ignored with no reason.

Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: Greg Wilkins <gregw@intalio.com>
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Subject: Re: [hybi] deflate-stream and masking
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On 21 July 2011 05:48, Peter Saint-Andre <stpeter@stpeter.im> wrote:
> My reading of the discussion is that people don't necessarily want to throw
> out deflate-stream, but they don't think it belongs in the base spec and
> they would prefer to see it defined in a separate draft, just as you propose
> for deflate-frame.

My reading is that there is a mix of opinions.  Some want it thrown
out, others want to keep it.

What I want to know is how (yet again) has a feature with such divided
support made it into the spec.  Why does this WG insist on putting non
consensus features into the spec and then having the argument to
remove them or not.   The status quo should not be to have a
non-consensus feature included.    It should be for the proponents of
a feature to make their case for inclusion and not for the opponents
to make the case for exclusion.

While it would be good for the base spec to have both compression and
an exemplar extension - that is no reason to just accept any old
extension that is neither good compression nor a good exemplar.

I also understand the argument that deflate-frame should just be
written up in another draft - by why does that not also apply to
deflate-stream?

If we are going to have a favoured son, can't we pick the healthy one
with some good prospects rather than the black sheep that is going to
end up dead or in jail?

regards

From jat@google.com  Wed Jul 20 19:37:40 2011
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From: John Tamplin <jat@google.com>
Date: Wed, 20 Jul 2011 22:37:15 -0400
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On Wed, Jul 20, 2011 at 10:29 PM, Greg Wilkins <gregw@intalio.com> wrote:

> What I want to know is how (yet again) has a feature with such divided
>
support made it into the spec.  Why does this WG insist on putting non
> consensus features into the spec and then having the argument to
> remove them or not.   The status quo should not be to have a
> non-consensus feature included.    It should be for the proponents of
> a feature to make their case for inclusion and not for the opponents
> to make the case for exclusion.
>

At the time, it wasn't divided.  I don't recall any serious objections at
the time it was proposed and included in the spec, including from you.
 Since then, we added masking which reduces the value in the client->server
direction, but it still has value since most of the large data will be going
from the server to the client.

-- 
John A. Tamplin
Software Engineer (GWT), Google

--000e0cd6668a45a31004a88b3b48
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<div class=3D"gmail_quote">On Wed, Jul 20, 2011 at 10:29 PM, Greg Wilkins <=
span dir=3D"ltr">&lt;<a href=3D"mailto:gregw@intalio.com">gregw@intalio.com=
</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin=
:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

<div class=3D"im">What I want to know is how (yet again) has a feature with=
 such divided</div></blockquote><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
support made it into the spec. =C2=A0Why does this WG insist on putting non=
<br>
consensus features into the spec and then having the argument to<br>
remove them or not. =C2=A0 The status quo should not be to have a<br>
non-consensus feature included. =C2=A0 =C2=A0It should be for the proponent=
s of<br>
a feature to make their case for inclusion and not for the opponents<br>
to make the case for exclusion.<br></blockquote><div><br></div><div>At the =
time, it wasn&#39;t divided. =C2=A0I don&#39;t recall any serious objection=
s at the time it was proposed and included in the spec, including from you.=
 =C2=A0Since then, we added masking which reduces the value in the client-&=
gt;server direction, but it still has value since most of the large data wi=
ll be going from the server to the client.</div>

</div><br>-- <br>John A. Tamplin<br>Software Engineer (GWT), Google<br>

--000e0cd6668a45a31004a88b3b48--

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Subject: [hybi] Fwd: Gen-ART last call review of draft-ietf-hybi-thewebsocketprotocol-10
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FYI, this was sent to ietf@ietf.org but not the WG list.

-------- Original Message --------
Subject: Gen-ART  last call review of 
draft-ietf-hybi-thewebsocketprotocol-10
Date: Tue, 19 Jul 2011 23:01:53 -0400
From: Richard L. Barnes <rbarnes@bbn.com>
To: General Area Review Team <gen-art@ietf.org>, 
draft-ietf-websec-thewebsocketprotocol@tools.ietf.org,	IETF Discussion 
<ietf@ietf.org>

I am the assigned Gen-ART reviewer for this draft. For background on
Gen-ART, please see the FAQ at
<http://wiki.tools.ietf.org/area/gen/trac/wiki/GenArtfaq>.

Please resolve these comments along with any other Last Call comments
you may receive.

Document: draft-ietf-hybi-thewebsocketprotocol-10.txt
Reviewer: Richard Barnes
Review Date: 19 July 2011
IETF LC End Date: 25 July 2011
IESG Telechat date: (if known) -

Summary:
Not ready.

Major issues:

[Is GET/Upgrade appropriate?]
It seems like the use of GET here goes beyond the normal "safe" 
semantics for that method.  To quote RFC 2616:
"
    In particular, the convention has been established that the GET and
    HEAD methods SHOULD NOT have the significance of taking an action
    other than retrieval.
"
In addition, alternative protocols specified in Upgrade headers are 
generally optional, whereas in this case, the transaction fails overall 
if the server does not support the websocket protocol.  It doesn't look 
to me like any of the current methods provide much better semantics 
(except possiblyCONNECT or the use of OPTIONS as in RFC 2817), and it 
seems like defining a new method could help get around the Upgrade issue 
as well.

[Huge buffers]
The frame length can be 7, 16, or 64 bits long.  Since the client is 
expected to buffer data until the end of a frame, this is asking clients 
to buffer 128 B, 64 KB, or 16 EB.  If it were 32 bits, the max would be 
4 MB.  Why not just make this a 32-bit fixed length field?

[Why is masking necessary?]
I seriously question the necessity of the masking of data frames.  As I 
understand it, the goal is to prevent proxies that don't understand 
Upgrade from confusing WebSocket data with HTTP data.  This risk seems a 
little dubious to me; has such a poisoning attack been demonstrated?  It 
seems like there are much simpler ways of doing this, like using a 
method other than GET (either CONNECT or something new).

[Why only client-to-server masking?]
Why isn't masking required on server-to-client frames?

[Unlimited buffering with fragmentation]
Much like with the frame length issue above, the fragmentation mechanism 
here seems like it imposes a heavy burden on the receive side.  Since 
the receiving client is supposed to buffer data until the end of a 
frame, it seems like fragmentation could be used to cause a receiving 
client to buffer a frame of indefinite size.


Minor issues:

[Setting protocol]
In the NOTE in Section 1.3, the document observes that Sec- headers 
can't be set in XHRs.  But clearly the Sec-WebSocket-Protocol header can 
be set through some API that applications use to get to this protocol. 
So it seems a little misleading to imply that scripts can't set these 
headers (besides -Version and -Origin).

[Use existing Origin header]
Why is this document creating a separate Sec-WebSocket-Origin header, 
rather than using the Origin header from draft-ietf-websec-origin?

[Need a reference to CORS]
It would be helpful to have a reference to Cross-Origin Resource Sharing 
in the NOTE in this section, since the pre-request check that is 
described is exactly a CORS priming request.

[Key is over-engineered]
What is the need for the hash and GUID in the handshake processing of 
the key?  It seems like it would be sufficient for the server to do 
anything deterministic with the key, even something as simple as 
incrementing it or flipping the bits.



Nits / Editorial comments:

[Why not plain sockets?]
The introduction makes clear why this protocol is needed instead of 
HTTP, but not why this protocol improves over providing a plain socket 
interface.  Presumably this is because the HTTP header provides a space 
where the browser can inject trusted information?

[Put the handshake section first]
It would really increase readability to put all the 
negotiation/handshake parts up front, before you get to the data format 
(Section 4).  As it is, when I got to the extension data part, I didn't 
realize that there was an extension mechanism, since it hadn't been 
mentioned yet.

[Stray sections]
It seems like Section 8 (Error Handling) should be moved up to Section 
4.6, and Section 9 (Extensions) should be part of section 4.

_______________________________________________
Ietf mailing list
Ietf@ietf.org
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You sound frustrated, Greg.    I understand that.

On the one hand, we could throw our hands up in frustration and either start
over, or fork the existing work.   Either way, the browser vendors would
need to ratify (by implementing) the protocol.

On the other hand, we could push the current design to standardization, so
that the browser vendors can do final implementation and we all can start
using the 20% of it that isn't borked.  Of course, if we remove a feature or
three beforehand, then the browser folks have to do less and can ship
sooner.

Please try to forgive the dreamers who have gilded our kitchen sink.   It is
a thrilling and intoxicating thing to do protocol design - it leads one to
do stuff because you can imagine it, not because it's actually needed.

I'd be happy to help with whichever of the above courses of action that
 gives me a lightweight asynchronous persistant connection between scripts
running in an easily available, off the shelf, bog standard browser client
and whatever server abomination I come up with, as soon as possible.



On Wed, Jul 20, 2011 at 10:29 PM, Greg Wilkins <gregw@intalio.com> wrote:

> On 21 July 2011 05:48, Peter Saint-Andre <stpeter@stpeter.im> wrote:
> > My reading of the discussion is that people don't necessarily want to
> throw
> > out deflate-stream, but they don't think it belongs in the base spec and
> > they would prefer to see it defined in a separate draft, just as you
> propose
> > for deflate-frame.
>
> My reading is that there is a mix of opinions.  Some want it thrown
> out, others want to keep it.
>
> What I want to know is how (yet again) has a feature with such divided
> support made it into the spec.  Why does this WG insist on putting non
> consensus features into the spec and then having the argument to
> remove them or not.   The status quo should not be to have a
> non-consensus feature included.    It should be for the proponents of
> a feature to make their case for inclusion and not for the opponents
> to make the case for exclusion.
>
> While it would be good for the base spec to have both compression and
> an exemplar extension - that is no reason to just accept any old
> extension that is neither good compression nor a good exemplar.
>
> I also understand the argument that deflate-frame should just be
> written up in another draft - by why does that not also apply to
> deflate-stream?
>
> If we are going to have a favoured son, can't we pick the healthy one
> with some good prospects rather than the black sheep that is going to
> end up dead or in jail?
>
> regards
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>

--000e0cd342b651830804a88b5f06
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

You sound frustrated, Greg. =A0 =A0I understand that.<div><br></div><div>On=
 the one hand, we could throw our hands up in frustration and either start =
over, or fork the existing work. =A0 Either way, the browser vendors would =
need to ratify (by implementing) the protocol.</div>
<div><br></div><div>On the other hand, we could push the current design to =
standardization, so that the browser vendors can do final implementation an=
d we all can start using the 20% of it that isn&#39;t borked. =A0Of course,=
 if we remove a feature or three beforehand, then the browser folks have to=
 do less and can ship sooner.</div>
<div><br></div><div>Please try to forgive the dreamers who have gilded our =
kitchen sink. =A0 It is a thrilling and intoxicating thing to do protocol d=
esign - it leads one to do stuff because you can imagine it, not because it=
&#39;s actually needed.</div>
<div><br></div><div>I&#39;d be happy to help with whichever of the above co=
urses of action that =A0gives me a lightweight=A0asynchronous=A0persistant =
connection between scripts running in an easily available, off the shelf, b=
og standard browser client and whatever server abomination I come up with, =
as soon as possible.</div>
<div><br></div><div><br></div><div><br><div class=3D"gmail_quote">On Wed, J=
ul 20, 2011 at 10:29 PM, Greg Wilkins <span dir=3D"ltr">&lt;<a href=3D"mail=
to:gregw@intalio.com">gregw@intalio.com</a>&gt;</span> wrote:<br><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex;">
<div class=3D"im">On 21 July 2011 05:48, Peter Saint-Andre &lt;<a href=3D"m=
ailto:stpeter@stpeter.im">stpeter@stpeter.im</a>&gt; wrote:<br>
&gt; My reading of the discussion is that people don&#39;t necessarily want=
 to throw<br>
&gt; out deflate-stream, but they don&#39;t think it belongs in the base sp=
ec and<br>
&gt; they would prefer to see it defined in a separate draft, just as you p=
ropose<br>
&gt; for deflate-frame.<br>
<br>
</div>My reading is that there is a mix of opinions. =A0Some want it thrown=
<br>
out, others want to keep it.<br>
<br>
What I want to know is how (yet again) has a feature with such divided<br>
support made it into the spec. =A0Why does this WG insist on putting non<br=
>
consensus features into the spec and then having the argument to<br>
remove them or not. =A0 The status quo should not be to have a<br>
non-consensus feature included. =A0 =A0It should be for the proponents of<b=
r>
a feature to make their case for inclusion and not for the opponents<br>
to make the case for exclusion.<br>
<br>
While it would be good for the base spec to have both compression and<br>
an exemplar extension - that is no reason to just accept any old<br>
extension that is neither good compression nor a good exemplar.<br>
<br>
I also understand the argument that deflate-frame should just be<br>
written up in another draft - by why does that not also apply to<br>
deflate-stream?<br>
<br>
If we are going to have a favoured son, can&#39;t we pick the healthy one<b=
r>
with some good prospects rather than the black sheep that is going to<br>
end up dead or in jail?<br>
<div><div></div><div class=3D"h5"><br>
regards<br>
_______________________________________________<br>
hybi mailing list<br>
<a href=3D"mailto:hybi@ietf.org">hybi@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/hybi</a><br>
</div></div></blockquote></div><br></div>

--000e0cd342b651830804a88b5f06--

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Cc: Hybi <hybi@ietf.org>, Patrick McManus <pmcmanus@mozilla.com>, Greg Wilkins <gregw@intalio.com>
Subject: Re: [hybi] deflate-stream and masking
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On 7/20/11 8:37 PM, John Tamplin wrote:

> most of the large
> data will be going from the server to the client.

Doesn't that depend on the application?

/psa


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Subject: Re: [hybi] Fwd: Gen-ART last call review of draft-ietf-hybi-thewebsocketprotocol-10
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Thanks for this Peter.   I believe the reviewer actually summarized the
consensus of the dissenters rather well.


Summary:
> Not ready.
>


> Major issues:
>
> [Is GET/Upgrade appropriate?]
> [Huge buffers]
> [Why is masking necessary?]
> [Why only client-to-server masking?]
> [Unlimited buffering with fragmentation]
>
>

> Minor issues:
>
> [Setting protocol]
> [Use existing Origin header]
> [Need a reference to CORS]
> [Key is over-engineered]
>
>

> Nits / Editorial comments:
>
> [Why not plain sockets?]
> [Put the handshake section first]
>

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<div class=3D"gmail_quote"><div>Thanks for this Peter. =A0 I believe the re=
viewer actually summarized the consensus of the=A0dissenters=A0rather well.=
 =A0</div><div>=A0</div><div><br></div><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Summary:<br>
Not ready.<br></blockquote><div>=A0</div><blockquote class=3D"gmail_quote" =
style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Major issues:<br>
<br>
[Is GET/Upgrade appropriate?]<br>[Huge buffers]<br>[Why is masking necessar=
y?]<br>[Why only client-to-server masking?]<br>[Unlimited buffering with fr=
agmentation]<br><br></blockquote><div>=A0</div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x;">

Minor issues:<br>
<br>
[Setting protocol]<br>[Use existing Origin header]<br>[Need a reference to =
CORS]<br>[Key is over-engineered]<br><br>
</blockquote><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Nits / Editorial=
 comments:<br>
<br>
[Why not plain sockets?]<br>[Put the handshake section first]<br></blockquo=
te><div><br></div><div>=A0</div></div>

--0016364d2c79d2a3d104a88bcc5e--

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From: John Tamplin <jat@google.com>
Date: Wed, 20 Jul 2011 23:23:27 -0400
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Subject: Re: [hybi] Fwd: Gen-ART last call review of draft-ietf-hybi-thewebsocketprotocol-10
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On Wed, Jul 20, 2011 at 10:46 PM, Peter Saint-Andre <stpeter@stpeter.im> wr=
ote:
> -------- Original Message --------
> Subject: Gen-ART =C2=A0last call review of draft-ietf-hybi-thewebsocketpr=
otocol-10
> Date: Tue, 19 Jul 2011 23:01:53 -0400
> From: Richard L. Barnes <rbarnes@bbn.com>
> To: General Area Review Team <gen-art@ietf.org>, draft-ietf-websec-theweb=
socketprotocol@tools.ietf.org, IETF Discussion <ietf@ietf.org>
>
> I am the assigned Gen-ART reviewer for this draft. For background on
> Gen-ART, please see the FAQ at
> <http://wiki.tools.ietf.org/area/gen/trac/wiki/GenArtfaq>.
>
> Please resolve these comments along with any other Last Call comments
> you may receive.
>
> Document: draft-ietf-hybi-thewebsocketprotocol-10.txt
> Reviewer: Richard Barnes
> Review Date: 19 July 2011
> IETF LC End Date: 25 July 2011
> IESG Telechat date: (if known) -
>
> Summary:
> Not ready.
>
> Major issues:
>
> [Huge buffers]
> The frame length can be 7, 16, or 64 bits long. =C2=A0Since the client is=
 expected to buffer data until the end of a frame,
> this is asking clients to buffer 128 B, 64 KB, or 16 EB. =C2=A0If it were=
 32 bits, the max would be 4 MB. =C2=A0Why not just
> make this a 32-bit fixed length field?

It is a compromise to get everyone to agree. =C2=A0There were several
requirements different groups were interested in:
=C2=A0- Small frames should have minimal overhead. =C2=A0A chat program sen=
ding
individual keystrokes should not have to pay
=C2=A0 =C2=A0an extra 3 bytes of length, and there are small packet thresho=
lds
to consider for mobile environments, where larger
=C2=A0 =C2=A0packets require powering up the radio to a higher power state.
=C2=A0- Some people wanted to be able to use facilities like Unix's
sendfile to transmit an entire file without having to stay in
=C2=A0 =C2=A0the loop and write framing. =C2=A0For future compatibility, th=
is
requires a larger size than 4G. =C2=A0Just like IPv6, it seemed
=C2=A0 =C2=A0better to use an "obviously large enough" value rather than ar=
gue
further about exactly how large it would need to be.
=C2=A0- Having a variable-size length field means the cost to support the
very large buffers is small (one value wasted of the
=C2=A0 =C2=A0lead length byte), while saving a significant percentage of th=
e
frame size for very small payloads.

As there were holdouts on both the side of wanting small headers for
small frames and wanting to send large messages without having to
fragment, this compromise was necessary to make progress.

> [Why is masking necessary?]
> I seriously question the necessity of the masking of data frames. =C2=A0A=
s I understand it, the goal is to prevent
> proxies that don't understand Upgrade from confusing WebSocket data with =
HTTP data. =C2=A0This risk seems a
> little dubious to me; has such a poisoning attack been demonstrated? =C2=
=A0It seems like there are much simpler
> ways of doing this, like using a method other than GET (either CONNECT or=
 something new).

There was a very long, contentious discussion about this based on
security research that found transparent proxies could be fooled into
believing the content following the WebSocket handshake was HTTP,
allowing poisoning a transparent cache -- imagine if an attacker could
replace the contents of www.google.com/ga.js on some cache serving
many users, basically it is a wildcard XSS for users of those caches.
There was no attack demonstrated using WebSocket framing, but it was
demonstrated using just the WS handshake followed by user-controlled
data.

The scenario here is that the attacker controls the server completely
and controls the JS code running in the client - what is to be
protected is transparent intermediaries which by their nature cannot
participate in the protocol so we cannot determine their compatibility
in order to proceed. =C2=A0While there was some discussion of approaches to
limit the risk to currently discovered vulnerabilities (such as
sending a bogus CONNECT message as part of the handshake), a few
people were concerned that further vulnerabilities could be discovered
and that masking the client->server traffic prevented attacker control
over those bytes at an acceptable cost.

> [Why only client-to-server masking?]
> Why isn't masking required on server-to-client frames?

In the attack scenario, the server is under complete control of the
attacker and can send any bytes it chooses anyway.

> [Unlimited buffering with fragmentation]
> Much like with the frame length issue above, the fragmentation mechanism =
here seems like it imposes a
> heavy burden on the receive side. =C2=A0Since the receiving client is sup=
posed to buffer data until the end of a
> frame, it seems like fragmentation could be used to cause a receiving cli=
ent to buffer a frame of indefinite size.

Obviously, an implementation will have to have a maximum size message
that it can support. =C2=A0In the spec as written, the only recourse when
this size is exceed is to terminate the connection (perhaps retrying
sending smaller messages). =C2=A0There have been some proposals to allow
each side to state their maximum frame and/or message sizes in the
handshake, but there hasn't been agreement to put them in the spec.

> [Why not plain sockets?]
> The introduction makes clear why this protocol is needed instead of HTTP,=
 but not why this protocol
> improves over providing a plain socket interface. =C2=A0Presumably this i=
s because the HTTP header provides
> a space where the browser can inject trusted information?

The browser is executing code on behalf of a potential attacker, and
would not give access to raw TCP sockets to such code as that would
allow circumvention of many protections, such as scanning machines
behind a corporate firewall, for example. =C2=A0If you mean just opening a
socket subject to the same origin restrictions, you would have to have
a special handshake to validate those restrictions, and you need some
framing to delineate messages since TCP is just a stream of bytes and
the API is message-oriented. =C2=A0If you do those things, then you have
essentially WebSockets (of course you could solve the same problems in
different ways, but it is the same class of solution).

--
John A. Tamplin
Software Engineer (GWT), Google

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On Wed, Jul 20, 2011 at 11:37 PM, Thomson, Martin
<Martin.Thomson@commscope.com> wrote:
> To those providing responses to review comments like this, consider for a moment that perhaps the draft
> does not - and should - provide the answer.

There was discussion about that to, and the majority opinion (not
mine) was that rationale did not belong in the spec.  If warranted, a
separate rationale document could be written.

-- 
John A. Tamplin
Software Engineer (GWT), Google

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Speaking as an unauthorized spokesperson for dissenters....


On Wed, Jul 20, 2011 at 11:23 PM, John Tamplin <jat@google.com> wrote:

> On Wed, Jul 20, 2011 at 10:46 PM, Peter Saint-Andre <stpeter@stpeter.im>
> wrote:
> > From: Richard L. Barnes <rbarnes@bbn.com>
> > Document: draft-ietf-hybi-thewebsocketprotocol-10.txt
> > Reviewer: Richard Barnes
> > Review Date: 19 July 2011
> >
> > Summary:
> > Not ready.
> >
> > Major issues:
> >
> > [Huge buffers]
> > The frame length can be 7, 16, or 64 bits long.  Since the client is
> expected to buffer data until the end of a frame,
> > this is asking clients to buffer 128 B, 64 KB, or 16 EB.  If it were 32
> bits, the max would be 4 MB.  Why not just
> > make this a 32-bit fixed length field?
>
> It is a compromise to get everyone to agree.  There were several
> requirements different groups were interested in:

As there were holdouts on both the side of wanting small headers for
> small frames and wanting to send large messages without having to
> fragment, this compromise was necessary to make progress.
>

It was my perception that in general people were agreed on the obvious
choices (minimize overhead, be flexible), nobody had the energy/interest to
challenge the ridiculous, as we'd all just ignore it or crash in our real
world implementations.


>
>> > [Why is masking necessary?]
>
> There was a very long, contentious discussion about this based on
> security research that found transparent proxies could be fooled into
>

I've not seen this research.

believing the content following the WebSocket handshake was HTTP,
> allowing poisoning a transparent cache -- imagine if an attacker could
> replace the contents of www.google.com/ga.js on some cache serving
> many users, basically it is a wildcard XSS for users of those caches.
> There was no attack demonstrated using WebSocket framing, but it was
>

Ah.


> demonstrated using just the WS handshake followed by user-controlled
> data.
>
> The scenario here is....


Giving me qualms of plausibility.  But I would challenge instead that this
is a fault of the intermediaries and not our problem.


>
> > [Why only client-to-server masking?]
> > Why isn't masking required on server-to-client frames?
>
> In the attack scenario, the server is under complete control of the
> attacker and can send any bytes it chooses anyway.
>

The dissenting consensus was that half an ass of security was not a donkey
we wanted to hitch our cart to.


>
> > [Unlimited buffering with fragmentation]
> > Much like with the frame length issue above, the fragmentation mechanism
> here seems like it imposes a
> > heavy burden on the receive side.  Since the receiving client is supposed
> to buffer data until the end of a
> > frame, it seems like fragmentation could be used to cause a receiving
> client to buffer a frame of indefinite size.
>
> Obviously, an implementation will have to have a maximum size message
> that it can support.  In the spec as written, the only recourse when
> this size is exceed is to terminate the connection (perhaps retrying
> sending smaller messages).  There have been some proposals to allow
> each side to state their maximum frame and/or message sizes in the
> handshake, but there hasn't been agreement to put them in the spec.
>

Allowing streaming is a laudable goal, but WS comes out of the gate as a
framing protocol.


>
> > [Why not plain sockets?]
> > The introduction makes clear why this protocol is needed instead of HTTP,
> but not why this protocol
> > improves over providing a plain socket interface.  Presumably this is
> because the HTTP header provides
> > a space where the browser can inject trusted information?
>
> The browser is executing code on behalf of a potential attacker, and
> would not give access to raw TCP sockets to such code as that would
> allow circumvention of many protections, such as scanning machines
> behind a corporate firewall, for example.  If you mean just opening a
> socket subject to the same origin restrictions, you would have to have
> a special handshake to validate those restrictions, and you need some
> framing to delineate messages since TCP is just a stream of bytes and
> the API is message-oriented.  If you do those things, then you have
> essentially WebSockets (of course you could solve the same problems in
> different ways, but it is the same class of solution).
>

I don't believe anyone suggested raw TCP sockets.    In John's example, he
is referring strictly to browser clients.   As a programmer guy, I can write
a websocket client in whatever language I want and pretend to be a websocket
as polite or malicious as I want.

I believe there are some who were hoping websockets would be a lightweight
transport protocol (a la TCP) that could be made available in a browser with
the understanding that the browser implementation of such a client would
apply appropriate security models.     Some of us are worried that the
design has suffered from concerns outside it's problem domain.



>
> --
> John A. Tamplin
> Software Engineer (GWT), Google
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>

--20cf301fb94d28093b04a88c2a2c
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Speaking as an unauthorized spokesperson for dissenters....<div><br><br><di=
v class=3D"gmail_quote">On Wed, Jul 20, 2011 at 11:23 PM, John Tamplin <spa=
n dir=3D"ltr">&lt;<a href=3D"mailto:jat@google.com">jat@google.com</a>&gt;<=
/span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;"><div class=3D"im">On Wed, Jul 20, 2011 at 1=
0:46 PM, Peter Saint-Andre &lt;<a href=3D"mailto:stpeter@stpeter.im">stpete=
r@stpeter.im</a>&gt; wrote:<br>
&gt; From: Richard L. Barnes &lt;<a href=3D"mailto:rbarnes@bbn.com">rbarnes=
@bbn.com</a>&gt;<br>&gt; Document: draft-ietf-hybi-thewebsocketprotocol-10.=
txt<br>
&gt; Reviewer: Richard Barnes<br>
&gt; Review Date: 19 July 2011<br>&gt;<br>
&gt; Summary:<br>
&gt; Not ready.<br>
&gt;<br>
&gt; Major issues:<br>
&gt;<br>
</div><div class=3D"im">&gt; [Huge buffers]<br>
&gt; The frame length can be 7, 16, or 64 bits long. =A0Since the client is=
 expected to buffer data until the end of a frame,<br>
&gt; this is asking clients to buffer 128 B, 64 KB, or 16 EB. =A0If it were=
 32 bits, the max would be 4 MB. =A0Why not just<br>
&gt; make this a 32-bit fixed length field?<br>
<br>
</div>It is a compromise to get everyone to agree. =A0There were several<br=
>
requirements different groups were interested in:</blockquote><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex;">As there were holdouts on both the side of wanting small=
 headers for<br>

small frames and wanting to send large messages without having to<br>
fragment, this compromise was necessary to make progress.<br></blockquote><=
div>=A0</div><div>It was my perception that in general people were agreed o=
n the obvious choices (minimize overhead, be flexible), nobody had the ener=
gy/interest to challenge the ridiculous, as we&#39;d all just ignore it or =
crash in our real world implementations.</div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex;"><div class=3D"im"><blockquote=
 class=3D"gmail_quote" style=3D"margin-top: 0px; margin-right: 0px; margin-=
bottom: 0px; margin-left: 0.8ex; border-left-width: 1px; border-left-color:=
 rgb(204, 204, 204); border-left-style: solid; padding-left: 1ex; ">
<br></blockquote>
&gt; [Why is masking necessary?]<br><br>
</div>There was a very long, contentious discussion about this based on<br>
security research that found transparent proxies could be fooled into<br></=
blockquote><div><br></div><div>I&#39;ve not seen this research.=A0</div><di=
v><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex;">

believing the content following the WebSocket handshake was HTTP,<br>
allowing poisoning a transparent cache -- imagine if an attacker could<br>
replace the contents of <a href=3D"http://www.google.com/ga.js" target=3D"_=
blank">www.google.com/ga.js</a> on some cache serving<br>
many users, basically it is a wildcard XSS for users of those caches.<br>
There was no attack demonstrated using WebSocket framing, but it was<br></b=
lockquote><div><br></div><div>Ah.</div><div>=A0</div><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex;">

demonstrated using just the WS handshake followed by user-controlled<br>
data.<br>
<br>
The scenario here is....</blockquote><div><br></div><div>Giving me qualms o=
f plausibility. =A0But I would challenge instead that this is a fault of th=
e intermediaries and not our problem.</div><div>=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex;">
<div class=3D"im"><br>
&gt; [Why only client-to-server masking?]<br>
&gt; Why isn&#39;t masking required on server-to-client frames?<br>
<br>
</div>In the attack scenario, the server is under complete control of the<b=
r>
attacker and can send any bytes it chooses anyway.<br></blockquote><div><br=
></div><div>The dissenting consensus was that half an ass of security was n=
ot a donkey we wanted to hitch our cart to.</div><div>=A0</div><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex;">

<div class=3D"im"><br>
&gt; [Unlimited buffering with fragmentation]<br>
&gt; Much like with the frame length issue above, the fragmentation mechani=
sm here seems like it imposes a<br>
&gt; heavy burden on the receive side. =A0Since the receiving client is sup=
posed to buffer data until the end of a<br>
&gt; frame, it seems like fragmentation could be used to cause a receiving =
client to buffer a frame of indefinite size.<br>
<br>
</div>Obviously, an implementation will have to have a maximum size message=
<br>
that it can support. =A0In the spec as written, the only recourse when<br>
this size is exceed is to terminate the connection (perhaps retrying<br>
sending smaller messages). =A0There have been some proposals to allow<br>
each side to state their maximum frame and/or message sizes in the<br>
handshake, but there hasn&#39;t been agreement to put them in the spec.<br>=
</blockquote><div><br></div><div>Allowing streaming is a laudable goal, but=
 WS comes out of the gate as a framing protocol. =A0=A0</div><div>=A0</div>=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">

<div class=3D"im"><br>
&gt; [Why not plain sockets?]<br>
&gt; The introduction makes clear why this protocol is needed instead of HT=
TP, but not why this protocol<br>
&gt; improves over providing a plain socket interface. =A0Presumably this i=
s because the HTTP header provides<br>
&gt; a space where the browser can inject trusted information?<br>
<br>
</div>The browser is executing code on behalf of a potential attacker, and<=
br>
would not give access to raw TCP sockets to such code as that would<br>
allow circumvention of many protections, such as scanning machines<br>
behind a corporate firewall, for example. =A0If you mean just opening a<br>
socket subject to the same origin restrictions, you would have to have<br>
a special handshake to validate those restrictions, and you need some<br>
framing to delineate messages since TCP is just a stream of bytes and<br>
the API is message-oriented. =A0If you do those things, then you have<br>
essentially WebSockets (of course you could solve the same problems in<br>
different ways, but it is the same class of solution).<br></blockquote><div=
><br></div><div>I don&#39;t believe anyone suggested raw TCP sockets. =A0 =
=A0In John&#39;s example, he is referring strictly to browser clients. =A0 =
As a programmer guy, I can write a websocket client in whatever language I =
want and pretend to be a websocket as polite or malicious as I want. =A0 =
=A0</div>
<div><br></div><div>I believe there are some who were hoping websockets wou=
ld be a lightweight transport protocol (a la TCP) that could be made availa=
ble in a browser with the understanding that the browser implementation of =
such a client would apply appropriate security models. =A0 =A0 Some of us a=
re worried that the design has suffered from concerns outside it&#39;s prob=
lem domain.</div>
<div><br></div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<font color=3D"#888888"><br>
--<br>
John A. Tamplin<br>
Software Engineer (GWT), Google<br>
</font><div><div></div><div class=3D"h5">__________________________________=
_____________<br>
hybi mailing list<br>
<a href=3D"mailto:hybi@ietf.org">hybi@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/hybi</a><br>
</div></div></blockquote></div><br></div>

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Subject: Re: [hybi] Fwd: Gen-ART last call review of draft-ietf-hybi-thewebsocketprotocol-10
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I hope you understand the anxiety around this issue.

>We're doing masking
Why?
> Because (somebody said they researched and besides we can imagine...)
Explain please.
> No.  If anyone else asks, somebody might blog some reasoning



On Wed, Jul 20, 2011 at 11:44 PM, John Tamplin <jat@google.com> wrote:

> On Wed, Jul 20, 2011 at 11:37 PM, Thomson, Martin
> <Martin.Thomson@commscope.com> wrote:
> > To those providing responses to review comments like this, consider for a
> moment that perhaps the draft
> > does not - and should - provide the answer.
>
> There was discussion about that to, and the majority opinion (not
> mine) was that rationale did not belong in the spec.  If warranted, a
> separate rationale document could be written.
>
> --
> John A. Tamplin
> Software Engineer (GWT), Google
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
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I hope you understand the anxiety around this issue.<div><br></div><div>&gt=
;We&#39;re doing masking</div><div>Why?</div><div>&gt; Because (somebody sa=
id they researched and besides we can imagine...)</div><div>Explain please.=
</div>
<div>&gt; No. =A0If anyone else asks, somebody might blog some reasoning</d=
iv><div><br></div><div>=A0</div><div><br><div class=3D"gmail_quote">On Wed,=
 Jul 20, 2011 at 11:44 PM, John Tamplin <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:jat@google.com">jat@google.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;"><div class=3D"im">On Wed, Jul 20, 2011 at 1=
1:37 PM, Thomson, Martin<br>
&lt;<a href=3D"mailto:Martin.Thomson@commscope.com">Martin.Thomson@commscop=
e.com</a>&gt; wrote:<br>
&gt; To those providing responses to review comments like this, consider fo=
r a moment that perhaps the draft<br>
&gt; does not - and should - provide the answer.<br>
<br>
</div>There was discussion about that to, and the majority opinion (not<br>
mine) was that rationale did not belong in the spec. =A0If warranted, a<br>
separate rationale document could be written.<br>
<font color=3D"#888888"><br>
--<br>
</font><div class=3D"im">John A. Tamplin<br>
Software Engineer (GWT), Google<br>
_______________________________________________<br>
</div><div><div></div><div class=3D"h5">hybi mailing list<br>
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</div></div></blockquote></div><br></div>

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On Wed, Jul 20, 2011 at 11:44 PM, David Endicott <dendicott@gmail.com> wrot=
e:
> Giving me qualms of plausibility. =C2=A0But I would challenge instead tha=
t this
> is a fault of the intermediaries and not our problem.

Sure. But the publication of the research led Safari and Firefox to
disable draft versions of WebSocket in their browsers because they
didn't want to be on the news for "Browser <xxx> led to an attack on
<yyy> thousand users ...".

By the nature of transparent intermediaries, it is hard to discern
information about them to do anything sensible with vulnerable ones,
such as disallow WebSocket operation, notify administrators, etc.

You could say users browsing to sites containing hostile code or
phishing is the fault of the user and it is their problem, but browser
vendors still want to help prevent damage to such users.

> I don't believe anyone suggested raw TCP sockets. =C2=A0 =C2=A0In John's =
example, he
> is referring strictly to browser clients. =C2=A0 As a programmer guy, I c=
an write
> a websocket client in whatever language I want and pretend to be a websoc=
ket
> as polite or malicious as I want.

Certainly you can write WebSocket clients other than browsers.
However, most of the security restrictions come about because it is to
be used by a browser executing potentially hostile code.  If you
aren't a browser and are running your code directly on the client
machine, then obviously you can already do whatever you want.

--=20
John A. Tamplin
Software Engineer (GWT), Google

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Subject: Re: [hybi] Fwd: Gen-ART last call review of draft-ietf-hybi-thewebsocketprotocol-10
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On Wed, Jul 20, 2011 at 11:57:26PM -0400, John Tamplin wrote:
> On Wed, Jul 20, 2011 at 11:44 PM, David Endicott <dendicott@gmail.com> wrote:
> > Giving me qualms of plausibility.  But I would challenge instead that this
> > is a fault of the intermediaries and not our problem.
> 
> Sure. But the publication of the research led Safari and Firefox to
> disable draft versions of WebSocket in their browsers because they
> didn't want to be on the news for "Browser <xxx> led to an attack on
> <yyy> thousand users ...".
> 
> By the nature of transparent intermediaries, it is hard to discern
> information about them to do anything sensible with vulnerable ones,
> such as disallow WebSocket operation, notify administrators, etc.
> 
> You could say users browsing to sites containing hostile code or
> phishing is the fault of the user and it is their problem, but browser
> vendors still want to help prevent damage to such users.

And with the current masking, even if we don't like it much, at least it
comes at very little cost and serves the purpose quite well.

> > I don't believe anyone suggested raw TCP sockets.    In John's example, he
> > is referring strictly to browser clients.   As a programmer guy, I can write
> > a websocket client in whatever language I want and pretend to be a websocket
> > as polite or malicious as I want.
> 
> Certainly you can write WebSocket clients other than browsers.
> However, most of the security restrictions come about because it is to
> be used by a browser executing potentially hostile code.  If you
> aren't a browser and are running your code directly on the client
> machine, then obviously you can already do whatever you want.

I would also add that in many environments, proxies are configured as
mandatory for browsers. Relying on HTTP ensures that the browser will
use the proxy and not try to scan local networks. And it's the only
way to enable WS connections to be initiated outside via proxies in
those environments.

Willy


From gregw@intalio.com  Thu Jul 21 00:02:57 2011
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Subject: Re: [hybi] deflate-stream and masking
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On 21 July 2011 12:37, John Tamplin <jat@google.com> wrote:
> At the time, it wasn't divided. =A0I don't recall any serious objections =
at
> the time it was proposed and included in the spec, including from you.

I don't recall any discussion at all about the extension.   There was
some talk about the effect of compression, to which there was a
response that spoke against stream compression
http://www.ietf.org/mail-archive/web/hybi/current/msg01811.html   The
deflate-stream extension then just appeared in the a draft...  I can't
actually find any thread that proposed it's addition.   I've looked
several times and failed to find any discussion.

Note that I'm not saying that somebody has done some evil deception to
get this into the spec.  Just that when objections have been raised,
then the default position should be that the feature is excluded
unless rough consensus can be achieved.


On 21 July 2011 12:47, David Endicott wrote:
> You sound frustrated, Greg.

Because this is an process anti-pattern that I think is mostly
responsible for much of the difficulties in this WG.   When an idea is
put into the spec, then it has unfair advantage over alternate ideas
and the entire discussion is tainted by the natural tendency to status
quo. Also I just don't understand why when there are many opponents to
this optional feature that we don't just drop it and let the
extensions be proposed in their own drafts and prosper or otherwise on
their own merits - not because they lucked out and became the favoured
son of the core specification.   I'm seen the results of the J2EE
specification process where favoured-son have been picked (Servlets,
no JSPs, no JSFs, no ....) and it is a disaster.


regards

From tomas.kukosa@siemens-enterprise.com  Thu Jul 21 01:28:53 2011
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To: "hybi@ietf.org" <hybi@ietf.org>
Date: Thu, 21 Jul 2011 10:28:51 +0200
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Subject: [hybi] Fragmented text message
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Hi,

I have a one question regarding fragmented text messages.

If the text message is fragmented must be each fragment a valid UTF-8 strin=
g or only complete defragmented message must be a valid UTF-8 string?
I.e. may I during receiving decode each fragment by UTF-8 and than join str=
ings or do I need to receive all fragments and then decode only defragmente=
d message?

Thanks,
  Tomas=

From theturtle32@gmail.com  Thu Jul 21 01:47:19 2011
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From: Brian <theturtle32@gmail.com>
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Subject: Re: [hybi] Fragmented text message
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--0015174c3ca610e1ad04a89065db
Content-Type: text/plain; charset=ISO-8859-1

Unless I'm mistaken, the fragmentation may occur in the middle of a
multi-byte character sequence.  Your code should be aware of that when
decoding.  My initial implementation buffers all fragments and then decodes
the whole message into a string at once.  I imagine you could probably
inspect the last four bytes of a fragment to determine whether there's a
partial utf-8 character.  If there is, you could buffer just those few bytes
and decode the rest of the fragment.  Then when the next fragment comes in,
prepend those bytes to the new payload and continue. Depending on your use
case and what you're optimizing for, it may be more efficient to just buffer
the whole message and then decode.

Brian


On Thu, Jul 21, 2011 at 1:28 AM, Kukosa, Tomas <
tomas.kukosa@siemens-enterprise.com> wrote:

> Hi,
>
> I have a one question regarding fragmented text messages.
>
> If the text message is fragmented must be each fragment a valid UTF-8
> string or only complete defragmented message must be a valid UTF-8 string?
> I.e. may I during receiving decode each fragment by UTF-8 and than join
> strings or do I need to receive all fragments and then decode only
> defragmented message?
>
> Thanks,
>  Tomas
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>

--0015174c3ca610e1ad04a89065db
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Unless I&#39;m mistaken, the fragmentation may occur in the middle of a mul=
ti-byte character sequence. =A0Your code should be aware of that when decod=
ing. =A0My initial implementation buffers all fragments and then decodes th=
e whole message into a string at once. =A0I imagine you could probably insp=
ect the last four bytes of a fragment to determine whether there&#39;s a pa=
rtial utf-8 character. =A0If there is, you could buffer just those few byte=
s and decode the rest of the fragment. =A0Then when the next fragment comes=
 in, prepend those bytes to the new payload and continue. Depending on your=
 use case and what you&#39;re optimizing for, it may be more efficient to j=
ust buffer the whole message and then decode.<div>
<br></div><div>Brian</div><div><br><br><div class=3D"gmail_quote">On Thu, J=
ul 21, 2011 at 1:28 AM, Kukosa, Tomas <span dir=3D"ltr">&lt;<a href=3D"mail=
to:tomas.kukosa@siemens-enterprise.com">tomas.kukosa@siemens-enterprise.com=
</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">Hi,<br>
<br>
I have a one question regarding fragmented text messages.<br>
<br>
If the text message is fragmented must be each fragment a valid UTF-8 strin=
g or only complete defragmented message must be a valid UTF-8 string?<br>
I.e. may I during receiving decode each fragment by UTF-8 and than join str=
ings or do I need to receive all fragments and then decode only defragmente=
d message?<br>
<br>
Thanks,<br>
 =A0Tomas<br>
_______________________________________________<br>
hybi mailing list<br>
<a href=3D"mailto:hybi@ietf.org">hybi@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/hybi</a><br>
</blockquote></div><br></div>

--0015174c3ca610e1ad04a89065db--

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From: John Tamplin <jat@google.com>
Date: Thu, 21 Jul 2011 04:48:21 -0400
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Subject: Re: [hybi] deflate-stream and masking
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On Thu, Jul 21, 2011 at 3:02 AM, Greg Wilkins <gregw@intalio.com> wrote:
> I don't recall any discussion at all about the extension. =C2=A0 There wa=
s
> some talk about the effect of compression, to which there was a
> response that spoke against stream compression
> http://www.ietf.org/mail-archive/web/hybi/current/msg01811.html =C2=A0 Th=
e
> deflate-stream extension then just appeared in the a draft... =C2=A0I can=
't
> actually find any thread that proposed it's addition. =C2=A0 I've looked
> several times and failed to find any discussion.

"hybi deflate-stream" found it immediately for me in my mailbox, and
then it was easy to find the other threads from there.

Patrick first discussed it in a thread starting at
http://www.ietf.org/mail-archive/web/hybi/current/msg04071.html, where
you commented only about allocating bits in a fixed manner and how
extensions were named, nothing about the basic operation of the
extension.

Patrick then made an official proposal in the thread starting at
http://www.ietf.org/mail-archive/web/hybi/current/msg04117.html.

The only dissent was me suggesting changing the name from deflate to
deflate-stream, so it went into the next revision of the spec.

> Note that I'm not saying that somebody has done some evil deception to
> get this into the spec. =C2=A0Just that when objections have been raised,
> then the default position should be that the feature is excluded
> unless rough consensus can be achieved.

I believe the first objections to it were raised months after it was
added to the spec, only after we had argued through masking.  So, I
think it is not correct to say that this was added to the spec over
objections and then it is unfairly harder to remove later.  There were
cases earlier where the previous editor did take unilateral action to
put new things into the spec, but that has not been the case here.

Once we do agree on something, I think it is entirely correct that
there is a higher bar necessary to remove it or else we will never
make progress.  Having new information to bring to the discussion,
such as after we added masking, is an adequate reason to reopen the
discussion, which is what we have been doing.

Personally, I think having a weak compression extension in the base
spec is better than no compression at all, so I would object to
removing it but would not object to replacing it with deflate-frame,
if doing so does not delay finalizing the spec.

--=20
John A. Tamplin
Software Engineer (GWT), Google

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Date: Thu, 21 Jul 2011 18:22:25 +0900
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--000e0cd306b643318704a890e424
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Agreed. Nothing in the spec disallows splitting UTF-8 byte sequence into
separate frames.

Impatient receivers must use some intelligent UTF-8 decoder like Brian
explained to get code points decoded. For example in python,
codecs.getincrementaldecoder('utf-8')() does this. Maybe most of major
platforms have library like it.

We may also add a constraint that the text message must be fragmented at
UTF-8 byte sequence boundary, but it complicates fragmentation code. I'm not
for that.

On Thu, Jul 21, 2011 at 17:47, Brian <theturtle32@gmail.com> wrote:

> Unless I'm mistaken, the fragmentation may occur in the middle of a
> multi-byte character sequence.  Your code should be aware of that when
> decoding.  My initial implementation buffers all fragments and then decodes
> the whole message into a string at once.  I imagine you could probably
> inspect the last four bytes of a fragment to determine whether there's a
> partial utf-8 character.  If there is, you could buffer just those few bytes
> and decode the rest of the fragment.  Then when the next fragment comes in,
> prepend those bytes to the new payload and continue. Depending on your use
> case and what you're optimizing for, it may be more efficient to just buffer
> the whole message and then decode.
>
> Brian
>
>
> On Thu, Jul 21, 2011 at 1:28 AM, Kukosa, Tomas <
> tomas.kukosa@siemens-enterprise.com> wrote:
>
>> If the text message is fragmented must be each fragment a valid UTF-8
>> string or only complete defragmented message must be a valid UTF-8 string?
>> I.e. may I during receiving decode each fragment by UTF-8 and than join
>> strings or do I need to receive all fragments and then decode only
>> defragmented message?
>>
>

--000e0cd306b643318704a890e424
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Agreed. Nothing in the spec disallows splitting UTF-8 byte sequence into se=
parate frames.<div><br></div><div>Impatient receivers must use=A0some intel=
ligent UTF-8 decoder like Brian explained to get code points decoded.=A0For=
 example in python, codecs.getincrementaldecoder(&#39;utf-8&#39;)() does th=
is. Maybe most of major platforms have library like it.</div>

<div><br></div><div>We may also add a constraint that the text message must=
 be fragmented at UTF-8 byte sequence boundary, but it complicates fragment=
ation code.=A0I&#39;m not for that.</div><div><br></div><div class=3D"gmail=
_quote">

On Thu, Jul 21, 2011 at 17:47, Brian <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:theturtle32@gmail.com">theturtle32@gmail.com</a>&gt;</span> wrote:<br><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #=
ccc solid;padding-left:1ex;">

Unless I&#39;m mistaken, the fragmentation may occur in the middle of a mul=
ti-byte character sequence. =A0Your code should be aware of that when decod=
ing. =A0My initial implementation buffers all fragments and then decodes th=
e whole message into a string at once. =A0I imagine you could probably insp=
ect the last four bytes of a fragment to determine whether there&#39;s a pa=
rtial utf-8 character. =A0If there is, you could buffer just those few byte=
s and decode the rest of the fragment. =A0Then when the next fragment comes=
 in, prepend those bytes to the new payload and continue. Depending on your=
 use case and what you&#39;re optimizing for, it may be more efficient to j=
ust buffer the whole message and then decode.<div>


<br></div><font color=3D"#888888"><div>Brian</div></font><div><div></div><d=
iv class=3D"h5"><div><br><br><div class=3D"gmail_quote">On Thu, Jul 21, 201=
1 at 1:28 AM, Kukosa, Tomas <span dir=3D"ltr">&lt;<a href=3D"mailto:tomas.k=
ukosa@siemens-enterprise.com" target=3D"_blank">tomas.kukosa@siemens-enterp=
rise.com</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">If the text message is fragmented must be ea=
ch fragment a valid UTF-8 string or only complete defragmented message must=
 be a valid UTF-8 string?<br>


I.e. may I during receiving decode each fragment by UTF-8 and than join str=
ings or do I need to receive all fragments and then decode only defragmente=
d message?<br></blockquote></div></div></div></div></blockquote></div>


--000e0cd306b643318704a890e424--

From gregw@intalio.com  Thu Jul 21 02:31:50 2011
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From: Greg Wilkins <gregw@intalio.com>
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Subject: Re: [hybi] deflate-stream and masking
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John,

I do recall the first thread, but it is all about framed based
compression.  This was a well discussed thread with 30 posts from 10
authors.   There was good support for frame based compression, a bit
of dissent about reserved bit and just a hint of support for stream
based compression.

The second thread, I missed at the time and missed again while
searching. It only has 5 responses from 3 authors.  Reading it now, I
see that opinion was split in these responses with yourself still
advocating frame compression.  Hardly wide consensus that could be
classified as "once we do agree on something".

But maybe I'm being a bit precious and this is probably not the best
example of the anti-pattern that I'm whining about.  So I'll stop
whipping this particular dead horse.

Regardless, there was significant opposition to stream compression
then (pre masking) and there is more now.    I think we are setting a
very low  quality/consensus standard for this extension.   But having
said all this before, I'll now go easy on this not-so-healthy horse.

cheers

From stpeter@stpeter.im  Thu Jul 21 05:20:09 2011
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Subject: [hybi] Fwd: [apps-discuss] Review of draft-ietf-hybi-thewebsocketprotocol for	apps-review
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FYI, another review.

/psa


-------- Original Message --------
Subject: 	[apps-discuss] Review of draft-ietf-hybi-thewebsocketprotocol
for apps-review
Date: 	Wed, 20 Jul 2011 22:09:23 -0700
From: 	Lisa Dusseault <lisa.dusseault@gmail.com>
To: 	apps-discuss@ietf.org




I have been selected as the Applications Area Review Team reviewer for
this draft (for background on apps-review, please see
http://www.apps.ietf.org/content/applications-area-review-team).
Please resolve these comments along with any other Last Call comments
you may receive. Please wait for direction from your document shepherd
or AD before posting a new version of the draft.

Document: draft-ietf-hybi-thewebsocketprotocol (reviewed -10)
Title: The WebSocket protocol
Reviewer: Lisa Dusseault
Review Date: July 20, 2011

Readiness Summary: This draft is almost ready for publication on the
Standards Track, but it has a few issues that should be fixed or thought
about before publication.

Document Summary: The Websocket protocol defines a HTTP handshake to
transition from a Web context to a bi-directional connection. It also
defines framing and masking for use in the bi-directional context. It
addresses a number of security concerns involving untrusted application
code running in browsers, but assumes that browsers are trusted and
servers hosting Websockets are trusted to a greater extent.

Major Issues

1. Masking

It would be good to state what the purpose of masking is. The sentence
is "Frames sent from the client to the server are masked to avoid
confusing network intermediaries", but I didn't upon reading this, or
looking at the specification text for masking, understand why network
intermediaries would be confused. A citation or further explanation
would be good. I see later there's an explanation in page 50, so a
forward reference would also work.

2. Cookie handling

Section 5.1,
The request MAY include headers associated with sending cookies,
as defined by the appropriate specifications [RFC6265]. These
headers are referred to as _Headers to Send Appropriate
Cookies_.

and

"Additionally, if any headers in the
server's handshake indicate that cookies should be set (as defined by
[RFC6265]), these cookies are referred to as _Cookies Set During the
Server's Opening Handshake_."

and

Section 5.2.1, point 8
8. Optionally, other headers, such as those used to send cookies to
a server. Unknown headers MUST be ignored.

First, a nit: these important underlined terms are referred to where? I
didn't see anywhere where these terms are used other than where they are
defined here.

The more substantive issue: I'm left unclear as to whether cookies are
really expected to be used, or how the client might know that it needs
to use cookies or else the application will not work. In many Web sites,
the site will not work if cookies are not used by the client, and this
is sufficiently rare that it's OK. Is that OK for a Websockets app? How
will the user know how to fix the problem? Since Websockets can't as
easily reply with a Web page to explain how to enable cookies, it would
be good to be more clear on this.

3. Failing a WebSocket Connection

Section 7.1.7 is supposed to describe how to _Fail the WebSocket
Connection_. It explains how the client does so. However, there's at
least one case in section 9.1 where the server receiving a malformed
Sec-WebSocket-Extensions header must _Fail the WebSocket Connection_.
How does the server do this?

4. Dropping with extreme prejudice

  From the Security Considerations:

"If at any time a server is faced with data that it does not
understand, or that violates some criteria by which the server
determines safety of input, or when the server sees an opening
handshake that does not correspond to the values the server is
expecting (e.g. incorrect path or origin), the server SHOULD just
disconnect. It is always safe to disconnect."

This seems pretty excessive. When the server just disconnects, the
client can't tell much about what went wrong, and whether an automated
retry would be any use at all. This is bad for deployed use, and even
worse for development/debugging. Yet we're recommending servers be that
unhelpful? Wouldn't it be better to recommend that the server reply with
an error response (some HTTP status codes defined here, too) that can
help the client diagnose an incorrect origin? An error like that is
often going to be a mis-configuration on one side or the other, rather
than an attack.

I'd also like to know what the websockets server implementations
currently do.


Minor Issues

The reservation of different ranges of error codes for use by
extensions, libraries and frameworks, and application code, seems to be
intended to avoid conflicts between those kinds of use. I don't think it
will work very well. I don't understand where a library ends and
application code begins, even in code I write where I write an
error_handler to be called by the library I'm using (which might be a
"library" only for convenience sake or might be a library that dozens of
other software programs use).

The border between code and "extensions" is not even very clear; if
somebody writes an extension intending to publish an internet-draft but
never gets around to it, they may have ended up using the wrong range
with the right intentions.

The border between extensions and "this protocol" is not very clear; if
I submit a internet-draft with a feature proposed to be a requirement
for websockets, I might prefer to use a 1000-1999 range error code but
if the feature is relegated to an optional draft it should change to
2000-2999 range.

This kind of ambiguity leads to arguments at best, unnecessary changes
to code (when specs are changed late after such arguments) at worst. I'm
not sure what to recommend; perhaps only one range reserved for "this
protocol and extensions published within the IETF process", and another
range for "libraries, frameworks and application code".


Thanks,
Lisa Dusseault

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From ibc@aliax.net  Thu Jul 21 06:00:02 2011
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2011/7/21 Peter Saint-Andre <stpeter@stpeter.im>:
> The more substantive issue: I'm left unclear as to whether cookies are
> really expected to be used, or how the client might know that it needs
> to use cookies or else the application will not work. In many Web sites,
> the site will not work if cookies are not used by the client, and this
> is sufficiently rare that it's OK. Is that OK for a Websockets app? How
> will the user know how to fix the problem? Since Websockets can't as
> easily reply with a Web page to explain how to enable cookies, it would
> be good to be more clear on this.

And this clearly shows the lack of specification for any
authentication mechanism in WebSocket. It seems that someone though
"perhaps Cookies are good solution" and just added that to the spec
without describing it. Too much vague for a protocol specification
IMHO.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From w@1wt.eu  Thu Jul 21 06:23:31 2011
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From: Willy Tarreau <w@1wt.eu>
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On Thu, Jul 21, 2011 at 03:00:00PM +0200, Iñaki Baz Castillo wrote:
> 2011/7/21 Peter Saint-Andre <stpeter@stpeter.im>:
> > The more substantive issue: I'm left unclear as to whether cookies are
> > really expected to be used, or how the client might know that it needs
> > to use cookies or else the application will not work. In many Web sites,
> > the site will not work if cookies are not used by the client, and this
> > is sufficiently rare that it's OK. Is that OK for a Websockets app? How
> > will the user know how to fix the problem? Since Websockets can't as
> > easily reply with a Web page to explain how to enable cookies, it would
> > be good to be more clear on this.
> 
> And this clearly shows the lack of specification for any
> authentication mechanism in WebSocket. It seems that someone though
> "perhaps Cookies are good solution" and just added that to the spec
> without describing it. Too much vague for a protocol specification
> IMHO.

Cookies are not only used for authentication. They're used for persistence
and QoS too.

Willy


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2011/7/21 Willy Tarreau <w@1wt.eu>:
> Cookies are not only used for authentication. They're used for persistenc=
e
> and QoS too.

Yes, but in WebSocket they are not specified.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From evnikita2@gmail.com  Wed Jul 20 20:28:01 2011
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Cc: Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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I support the opinion that SRV mechanism for use with WebSocket should 
be incorporated in the core WS specification.  Mykyta

19.07.2011 16:26, Dave Cridland wrote:
> On Tue Jul 19 12:51:16 2011, Iñaki Baz Castillo wrote:
>> 2011/7/11 The IESG <iesg-secretary@ietf.org>:
>> > The IESG has received a request from the BiDirectional or
>> > Server-Initiated HTTP WG (hybi) to consider the following document:
>> > - 'The WebSocket protocol'
>> > <draft-ietf-hybi-thewebsocketprotocol-10.txt> as a Proposed Standard
>>
>> Hi, I assume there is no interest in making DNS SRV mechanism exposed
>> in http://tools.ietf.org/html/draft-ibc-websocket-dns-srv-02 part of
>> the WebSocket core specification, neither referencing it (in the same
>> way RFC 3261 "SIP protocol" mandates the usage of RFC 3263 "Locating
>> SIP servers").
>>
>> As said before, making such DNS SRV specification an extension (so
>> present in other document) will mean no success at all, as WebSocket
>> client implementors (i.e. webbrowser vendors) will not be mandated to
>> implement it and service providers could not rely on the support of
>> DNS SRV in web browsers. So nobody will use them (because IE10 decided
>> not to implement it, for example). IMHO this is sad due the real
>> advantages DNS SRV provides for a protocol like WebSocket.
>>
>> Yes, in HTTP there is no special DNS stuff, all the load-balancing and
>> failover mechanism are done at server side with very complex and
>> expensive solutions (www.facebook.com resolves to a single IPv4 !!!!).
>> The question is: should we also inherit every HTTP limitation in
>> WebSocket?
>
> I agree wholeheartedly with this, and strongly recommend that 
> mandatory use of SRV is included in the core protocol.
>
> I think with HTTP's very short lived requests, then it's possible to 
> work around the lack of SRV support (at a cost), but the benefit is 
> markedly higher with the long-lived, stateful sessions we're 
> anticipating with WebSocket.
>
> Dave.


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Subject: Re: [hybi] Fragmented text message
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My understanding is that each websocket entity that uses UTF-8 must have a =
valid UTF-8 stream as a payload. This covers all UTF-8 frames including con=
tinuation frames.

I slightly more prefer approach where UTF-8 message must contain valid sequ=
ence rather than each continuation frame (simpler for intermediaries, endpo=
ints that stream data to application still must handle partial UTF-8 code p=
oint encoding case). Fine with any approach as long as it is well defined.

From: hybi-bounces@ietf.org [mailto:hybi-bounces@ietf.org] On Behalf Of Tak=
eshi Yoshino
Sent: Thursday, July 21, 2011 2:22 AM
To: Brian
Cc: hybi@ietf.org
Subject: Re: [hybi] Fragmented text message

Agreed. Nothing in the spec disallows splitting UTF-8 byte sequence into se=
parate frames.

Impatient receivers must use some intelligent UTF-8 decoder like Brian expl=
ained to get code points decoded. For example in python, codecs.getincremen=
taldecoder('utf-8')() does this. Maybe most of major platforms have library=
 like it.

We may also add a constraint that the text message must be fragmented at UT=
F-8 byte sequence boundary, but it complicates fragmentation code. I'm not =
for that.

On Thu, Jul 21, 2011 at 17:47, Brian <theturtle32@gmail.com<mailto:theturtl=
e32@gmail.com>> wrote:
Unless I'm mistaken, the fragmentation may occur in the middle of a multi-b=
yte character sequence.  Your code should be aware of that when decoding.  =
My initial implementation buffers all fragments and then decodes the whole =
message into a string at once.  I imagine you could probably inspect the la=
st four bytes of a fragment to determine whether there's a partial utf-8 ch=
aracter.  If there is, you could buffer just those few bytes and decode the=
 rest of the fragment.  Then when the next fragment comes in, prepend those=
 bytes to the new payload and continue. Depending on your use case and what=
 you're optimizing for, it may be more efficient to just buffer the whole m=
essage and then decode.

Brian

On Thu, Jul 21, 2011 at 1:28 AM, Kukosa, Tomas <tomas.kukosa@siemens-enterp=
rise.com<mailto:tomas.kukosa@siemens-enterprise.com>> wrote:
If the text message is fragmented must be each fragment a valid UTF-8 strin=
g or only complete defragmented message must be a valid UTF-8 string?
I.e. may I during receiving decode each fragment by UTF-8 and than join str=
ings or do I need to receive all fragments and then decode only defragmente=
d message?

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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">My understanding is that =
each websocket entity that uses UTF-8 must have a valid UTF-8 stream as a p=
ayload. This covers all UTF-8 frames including continuation
 frames.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I slightly more prefer ap=
proach where UTF-8 message must contain valid sequence rather than each con=
tinuation frame (simpler for intermediaries, endpoints that
 stream data to application still must handle partial UTF-8 code point enco=
ding case). Fine with any approach as long as it is well defined.<o:p></o:p=
></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> hybi-bou=
nces@ietf.org [mailto:hybi-bounces@ietf.org]
<b>On Behalf Of </b>Takeshi Yoshino<br>
<b>Sent:</b> Thursday, July 21, 2011 2:22 AM<br>
<b>To:</b> Brian<br>
<b>Cc:</b> hybi@ietf.org<br>
<b>Subject:</b> Re: [hybi] Fragmented text message<o:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Agreed. Nothing in the spec disallows splitting UTF-=
8 byte sequence into separate frames.<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Impatient receivers must use&nbsp;some intelligent U=
TF-8 decoder like Brian explained to get code points decoded.&nbsp;For exam=
ple in python, codecs.getincrementaldecoder('utf-8')() does this. Maybe mos=
t of major platforms have library like it.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">We may also add a constraint that the text message m=
ust be fragmented at UTF-8 byte sequence boundary, but it complicates fragm=
entation code.&nbsp;I'm not for that.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">On Thu, Jul 21, 2011 at 17:47, Brian &lt;<a href=3D"=
mailto:theturtle32@gmail.com">theturtle32@gmail.com</a>&gt; wrote:<o:p></o:=
p></p>
<p class=3D"MsoNormal">Unless I'm mistaken, the fragmentation may occur in =
the middle of a multi-byte character sequence. &nbsp;Your code should be aw=
are of that when decoding. &nbsp;My initial implementation buffers all frag=
ments and then decodes the whole message into
 a string at once. &nbsp;I imagine you could probably inspect the last four=
 bytes of a fragment to determine whether there's a partial utf-8 character=
. &nbsp;If there is, you could buffer just those few bytes and decode the r=
est of the fragment. &nbsp;Then when the next fragment
 comes in, prepend those bytes to the new payload and continue. Depending o=
n your use case and what you're optimizing for, it may be more efficient to=
 just buffer the whole message and then decode.<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">Brian<o:p></o:p></span=
></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Thu, Jul 21, 2011 at 1:28 AM, Kukosa, Tomas &lt;<=
a href=3D"mailto:tomas.kukosa@siemens-enterprise.com" target=3D"_blank">tom=
as.kukosa@siemens-enterprise.com</a>&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal">If the text message is fragmented must be each fragm=
ent a valid UTF-8 string or only complete defragmented message must be a va=
lid UTF-8 string?<br>
I.e. may I during receiving decode each fragment by UTF-8 and than join str=
ings or do I need to receive all fragments and then decode only defragmente=
d message?<o:p></o:p></p>
</div>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

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From: David Endicott <dendicott@gmail.com>
To: =?ISO-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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I am opposed to inclusion in core specification.  I would accept it as
optional extension.

DNS resolution is not a function of a transport protocol.  DNS SRV has no
special association with WS.    It is my opinion that this would be
additional cruft that is only marginally related to the purpose and functio=
n
of websockets.    It does not address a general use case.   DNS SRV applies
only to a (small?) subset of server-side implementations.    It is a good
and useful mechanism, but I do not believe it should be tied tightly to
websockets, nor included as part of the core spec.


On Wed, Jul 20, 2011 at 5:12 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> 2011/7/19 Dave Cridland <dave@cridland.net>:
> >> Hi, I assume there is no interest in making DNS SRV mechanism exposed
> >> in http://tools.ietf.org/html/draft-ibc-websocket-dns-srv-02 part of
> >> the WebSocket core specification, neither referencing it (in the same
> >> way RFC 3261 "SIP protocol" mandates the usage of RFC 3263 "Locating
> >> SIP servers").
> >>
> >> As said before, making such DNS SRV specification an extension (so
> >> present in other document) will mean no success at all, as WebSocket
> >> client implementors (i.e. webbrowser vendors) will not be mandated to
> >> implement it and service providers could not rely on the support of
> >> DNS SRV in web browsers. So nobody will use them (because IE10 decided
> >> not to implement it, for example). IMHO this is sad due the real
> >> advantages DNS SRV provides for a protocol like WebSocket.
> >>
> >> Yes, in HTTP there is no special DNS stuff, all the load-balancing and
> >> failover mechanism are done at server side with very complex and
> >> expensive solutions (www.facebook.com resolves to a single IPv4 !!!!).
> >> The question is: should we also inherit every HTTP limitation in
> >> WebSocket?
> >
> > I agree wholeheartedly with this, and strongly recommend that mandatory
> use
> > of SRV is included in the core protocol.
> >
> > I think with HTTP's very short lived requests, then it's possible to wo=
rk
> > around the lack of SRV support (at a cost), but the benefit is markedly
> > higher with the long-lived, stateful sessions we're anticipating with
> > WebSocket.
>
>
> Unfortunaltely it seems that the debate about DNS SRV support does not
> interest to the core WG authors. I would like, at least, to receive
> good arguments not to include/mandate DNS SRV support in the draft. If
> not, the proposal is just being ignored with no reason.
>
> Regards.
>
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>

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I am opposed to inclusion in core specification. =A0I would accept it as op=
tional=A0extension.<div><br></div><div>DNS resolution is not a function of =
a transport protocol. =A0DNS SRV has no special association with WS. =A0 =
=A0It is my opinion that this would be additional cruft that is only margin=
ally related to the purpose and function of websockets. =A0 =A0It does not =
address a general use case. =A0 DNS SRV applies only to a (small?) subset o=
f server-side implementations. =A0 =A0It is a good and useful mechanism, bu=
t I do not believe it should be tied tightly to websockets, nor included as=
 part of the core spec.</div>
<div><br><br><div class=3D"gmail_quote">On Wed, Jul 20, 2011 at 5:12 PM, I=
=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net">=
ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
2011/7/19 Dave Cridland &lt;<a href=3D"mailto:dave@cridland.net">dave@cridl=
and.net</a>&gt;:<br>
<div class=3D"im">&gt;&gt; Hi, I assume there is no interest in making DNS =
SRV mechanism exposed<br>
&gt;&gt; in <a href=3D"http://tools.ietf.org/html/draft-ibc-websocket-dns-s=
rv-02" target=3D"_blank">http://tools.ietf.org/html/draft-ibc-websocket-dns=
-srv-02</a> part of<br>
&gt;&gt; the WebSocket core specification, neither referencing it (in the s=
ame<br>
&gt;&gt; way RFC 3261 &quot;SIP protocol&quot; mandates the usage of RFC 32=
63 &quot;Locating<br>
&gt;&gt; SIP servers&quot;).<br>
&gt;&gt;<br>
&gt;&gt; As said before, making such DNS SRV specification an extension (so=
<br>
&gt;&gt; present in other document) will mean no success at all, as WebSock=
et<br>
&gt;&gt; client implementors (i.e. webbrowser vendors) will not be mandated=
 to<br>
&gt;&gt; implement it and service providers could not rely on the support o=
f<br>
&gt;&gt; DNS SRV in web browsers. So nobody will use them (because IE10 dec=
ided<br>
&gt;&gt; not to implement it, for example). IMHO this is sad due the real<b=
r>
&gt;&gt; advantages DNS SRV provides for a protocol like WebSocket.<br>
&gt;&gt;<br>
&gt;&gt; Yes, in HTTP there is no special DNS stuff, all the load-balancing=
 and<br>
&gt;&gt; failover mechanism are done at server side with very complex and<b=
r>
&gt;&gt; expensive solutions (<a href=3D"http://www.facebook.com" target=3D=
"_blank">www.facebook.com</a> resolves to a single IPv4 !!!!).<br>
&gt;&gt; The question is: should we also inherit every HTTP limitation in<b=
r>
&gt;&gt; WebSocket?<br>
&gt;<br>
&gt; I agree wholeheartedly with this, and strongly recommend that mandator=
y use<br>
&gt; of SRV is included in the core protocol.<br>
&gt;<br>
&gt; I think with HTTP&#39;s very short lived requests, then it&#39;s possi=
ble to work<br>
&gt; around the lack of SRV support (at a cost), but the benefit is markedl=
y<br>
&gt; higher with the long-lived, stateful sessions we&#39;re anticipating w=
ith<br>
&gt; WebSocket.<br>
<br>
<br>
</div>Unfortunaltely it seems that the debate about DNS SRV support does no=
t<br>
interest to the core WG authors. I would like, at least, to receive<br>
good arguments not to include/mandate DNS SRV support in the draft. If<br>
not, the proposal is just being ignored with no reason.<br>
<br>
Regards.<br>
<div class=3D"im"><br>
<br>
--<br>
I=F1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
_______________________________________________<br>
</div><div><div></div><div class=3D"h5">hybi mailing list<br>
<a href=3D"mailto:hybi@ietf.org">hybi@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/hybi</a><br>
</div></div></blockquote></div><br></div>

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Date: Thu, 21 Jul 2011 12:11:03 -0400
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Subject: Re: [hybi] Fragmented text message
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Would that not then require that the websocket peer and any websocket aware
intermediaries examine and understand the content of the frames?   (ie. to
be able to tell if a UTF sequence is being split).

That seems, to me, to be highly impractical.  Or even impossible.

Also, I consider WS to be transport protocol and as such, it must be content
agnostic.

So long as the endpoints deliver via their API's a valid UTF-8 stream, what
does it matter if it's scattered on the wire?


On Thu, Jul 21, 2011 at 12:01 PM, Piotr Kulaga <piotrku@microsoft.com>wrote:

>  My understanding is that each websocket entity that uses UTF-8 must have
> a valid UTF-8 stream as a payload. This covers all UTF-8 frames including
> continuation frames.****
>
> ** **
>
> I slightly more prefer approach where UTF-8 message must contain valid
> sequence rather than each continuation frame (simpler for intermediaries,
> endpoints that stream data to application still must handle partial UTF-8
> code point encoding case). Fine with any approach as long as it is well
> defined.****
>
> ** **
>
> *From:* hybi-bounces@ietf.org [mailto:hybi-bounces@ietf.org] *On Behalf Of
> *Takeshi Yoshino
> *Sent:* Thursday, July 21, 2011 2:22 AM
> *To:* Brian
> *Cc:* hybi@ietf.org
> *Subject:* Re: [hybi] Fragmented text message****
>
> ** **
>
> Agreed. Nothing in the spec disallows splitting UTF-8 byte sequence into
> separate frames.****
>
> ** **
>
> Impatient receivers must use some intelligent UTF-8 decoder like Brian
> explained to get code points decoded. For example in python,
> codecs.getincrementaldecoder('utf-8')() does this. Maybe most of major
> platforms have library like it.****
>
> ** **
>
> We may also add a constraint that the text message must be fragmented at
> UTF-8 byte sequence boundary, but it complicates fragmentation code. I'm not
> for that.****
>
> ** **
>
> On Thu, Jul 21, 2011 at 17:47, Brian <theturtle32@gmail.com> wrote:****
>
> Unless I'm mistaken, the fragmentation may occur in the middle of a
> multi-byte character sequence.  Your code should be aware of that when
> decoding.  My initial implementation buffers all fragments and then decodes
> the whole message into a string at once.  I imagine you could probably
> inspect the last four bytes of a fragment to determine whether there's a
> partial utf-8 character.  If there is, you could buffer just those few bytes
> and decode the rest of the fragment.  Then when the next fragment comes in,
> prepend those bytes to the new payload and continue. Depending on your use
> case and what you're optimizing for, it may be more efficient to just buffer
> the whole message and then decode.****
>
> ** **
>
> Brian****
>
> ** **
>
> On Thu, Jul 21, 2011 at 1:28 AM, Kukosa, Tomas <
> tomas.kukosa@siemens-enterprise.com> wrote:****
>
> If the text message is fragmented must be each fragment a valid UTF-8
> string or only complete defragmented message must be a valid UTF-8 string?
> I.e. may I during receiving decode each fragment by UTF-8 and than join
> strings or do I need to receive all fragments and then decode only
> defragmented message?****
>
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>
>

--000e0ce0b4f07c9a2904a89698d5
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Would that not then require that the websocket peer and any websocket aware=
 intermediaries examine and understand the content of the frames? =A0 (ie. =
to be able to tell if a UTF sequence is being split).<div><br></div><div>Th=
at seems, to me, to be highly impractical. =A0Or even impossible.</div>
<div><br></div><div>Also, I consider WS to be transport protocol and as suc=
h, it must be content agnostic. =A0 =A0</div><div><br></div><div>So long as=
 the endpoints deliver via their API&#39;s a valid UTF-8 stream, what does =
it matter if it&#39;s scattered on the wire?</div>
<div><br></div><div><br><div class=3D"gmail_quote">On Thu, Jul 21, 2011 at =
12:01 PM, Piotr Kulaga <span dir=3D"ltr">&lt;<a href=3D"mailto:piotrku@micr=
osoft.com">piotrku@microsoft.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex;">






<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D">My un=
derstanding is that each websocket entity that uses UTF-8 must have a valid=
 UTF-8 stream as a payload. This covers all UTF-8 frames including continua=
tion
 frames.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D"><u></=
u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D">I sli=
ghtly more prefer approach where UTF-8 message must contain valid sequence =
rather than each continuation frame (simpler for intermediaries, endpoints =
that
 stream data to application still must handle partial UTF-8 code point enco=
ding case). Fine with any approach as long as it is well defined.<u></u><u>=
</u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;color:#1F497D"><u></=
u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt">From:</span></b>=
<span style=3D"font-size:10.0pt"> <a href=3D"mailto:hybi-bounces@ietf.org" =
target=3D"_blank">hybi-bounces@ietf.org</a> [mailto:<a href=3D"mailto:hybi-=
bounces@ietf.org" target=3D"_blank">hybi-bounces@ietf.org</a>]
<b>On Behalf Of </b>Takeshi Yoshino<br>
<b>Sent:</b> Thursday, July 21, 2011 2:22 AM<br>
<b>To:</b> Brian<br>
<b>Cc:</b> <a href=3D"mailto:hybi@ietf.org" target=3D"_blank">hybi@ietf.org=
</a><br>
<b>Subject:</b> Re: [hybi] Fragmented text message<u></u><u></u></span></p>=
<div><div></div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">Agreed. Nothing in the spec disallows splitting UTF-=
8 byte sequence into separate frames.<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Impatient receivers must use=A0some intelligent UTF-=
8 decoder like Brian explained to get code points decoded.=A0For example in=
 python, codecs.getincrementaldecoder(&#39;utf-8&#39;)() does this. Maybe m=
ost of major platforms have library like it.<u></u><u></u></p>

</div>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">We may also add a constraint that the text message m=
ust be fragmented at UTF-8 byte sequence boundary, but it complicates fragm=
entation code.=A0I&#39;m not for that.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">On Thu, Jul 21, 2011 at 17:47, Brian &lt;<a href=3D"=
mailto:theturtle32@gmail.com" target=3D"_blank">theturtle32@gmail.com</a>&g=
t; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">Unless I&#39;m mistaken, the fragmentation may occur=
 in the middle of a multi-byte character sequence. =A0Your code should be a=
ware of that when decoding. =A0My initial implementation buffers all fragme=
nts and then decodes the whole message into
 a string at once. =A0I imagine you could probably inspect the last four by=
tes of a fragment to determine whether there&#39;s a partial utf-8 characte=
r. =A0If there is, you could buffer just those few bytes and decode the res=
t of the fragment. =A0Then when the next fragment
 comes in, prepend those bytes to the new payload and continue. Depending o=
n your use case and what you&#39;re optimizing for, it may be more efficien=
t to just buffer the whole message and then decode.<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#888888">Brian<u></u><u></u></s=
pan></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Thu, Jul 21, 2011 at 1:28 AM, Kukosa, Tomas &lt;<=
a href=3D"mailto:tomas.kukosa@siemens-enterprise.com" target=3D"_blank">tom=
as.kukosa@siemens-enterprise.com</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">If the text message is fragmented must be each fragm=
ent a valid UTF-8 string or only complete defragmented message must be a va=
lid UTF-8 string?<br>
I.e. may I during receiving decode each fragment by UTF-8 and than join str=
ings or do I need to receive all fragments and then decode only defragmente=
d message?<u></u><u></u></p>
</div>
</div>
</div>
</div>
</div>
</div></div></div>
</div>

<br>_______________________________________________<br>
hybi mailing list<br>
<a href=3D"mailto:hybi@ietf.org">hybi@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/hybi</a><br>
<br></blockquote></div><br></div>

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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/21 David Endicott <dendicott@gmail.com>:
> DNS resolution is not a function of a transport protocol. =C2=A0DNS SRV h=
as no
> special association with WS. =C2=A0 =C2=A0It is my opinion that this woul=
d be
> additional cruft that is only marginally related to the purpose and funct=
ion
> of websockets. =C2=A0 =C2=A0It does not address a general use case. =C2=
=A0 DNS SRV applies
> only to a (small?) subset of server-side implementations. =C2=A0 =C2=A0It=
 is a good
> and useful mechanism, but I do not believe it should be tied tightly to
> websockets, nor included as part of the core spec.

An WebSocket URI is given to a WebSocket client, and the client MUST
locate the corresponding WS server, right? and for locating the server
the client MUST follows a procedures which, for now, involve (if it's
not an IP) DNS A/AAAA resolution, right? So now imagine that the
location mechanism is a bit more powerful and also involves SRV
queries (not always).

If you think that a transport protocol (like WebSocket) must not
resolve a server destination then also remove the WS URI inspection
and resolution from the core spec, don't you agree? or just DNS A/AAA
is valid?

I don't agree with your opinion at all. Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Thu Jul 21 17:06:59 2011, David Endicott wrote:
> DNS resolution is not a function of a transport protocol. 

I entirely agree, but the specification already includes DNS  
resolution as part of URI resolution and URI scheme definition, and  
as such, if you want all these things - which are, I concur, not a  
function of a transport protocol - removed from this specification,  
then we'll need a distinct document which adds them back in, as it  
were.

Dave.
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Date: Thu, 21 Jul 2011 18:39:10 +0200
From: Willy Tarreau <w@1wt.eu>
To: =?iso-8859-1?Q?I=F1aki?= Baz Castillo <ibc@aliax.net>
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Cc: Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Thu, Jul 21, 2011 at 06:27:49PM +0200, Iñaki Baz Castillo wrote:
> 2011/7/21 David Endicott <dendicott@gmail.com>:
> > DNS resolution is not a function of a transport protocol.  DNS SRV has no
> > special association with WS.    It is my opinion that this would be
> > additional cruft that is only marginally related to the purpose and function
> > of websockets.    It does not address a general use case.   DNS SRV applies
> > only to a (small?) subset of server-side implementations.    It is a good
> > and useful mechanism, but I do not believe it should be tied tightly to
> > websockets, nor included as part of the core spec.
> 
> An WebSocket URI is given to a WebSocket client, and the client MUST
> locate the corresponding WS server, right? and for locating the server
> the client MUST follows a procedures which, for now, involve (if it's
> not an IP) DNS A/AAAA resolution, right? So now imagine that the
> location mechanism is a bit more powerful and also involves SRV
> queries (not always).
> 
> If you think that a transport protocol (like WebSocket) must not
> resolve a server destination then also remove the WS URI inspection
> and resolution from the core spec, don't you agree? or just DNS A/AAA
> is valid?
> 
> I don't agree with your opinion at all. Regards.

Iñaki,

I understand the point David is making. DNS is something independant of
WS and conversely. It is one way of resolving addresses, just like there
will be people using hosts files. At no place the protocol specification
dictates how a client should resolve a name to an IP address. The protocol
specifies the transport part only.

This is the same for other protocols. For instance, neither FTP nor HTTP
explain how a client is supposed to resolve a host name, still the later
explains how to parse a URI. DNS SRV is a DNS extension which only concerns
resolvers. Not all clients will be using resolvers, just a part of them.
Some others will simply forward the request to their HTTP proxy which will
apply whatever DNS resolving method they know, including possibly DNS SRV.

In practice, if there are new elements of DNS SRV that are specific to WS,
they should probably be added to the DNS SRV spec and not the WS spec.
Maybe the WS spec should mention that it addresses transport only and
not address resolving. It may recommend to follow some principles to
perform the resolving but should not specify how to do it.

Hoping it's a bit clearer,
Willy


From ibc@aliax.net  Thu Jul 21 10:15:15 2011
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Date: Thu, 21 Jul 2011 19:15:13 +0200
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/21 Willy Tarreau <w@1wt.eu>:
> I understand the point David is making. DNS is something independant of
> WS and conversely. It is one way of resolving addresses, just like there
> will be people using hosts files. At no place the protocol specification
> dictates how a client should resolve a name to an IP address. The protoco=
l
> specifies the transport part only.

Host files just replace A/AAAA resolution. Still is possible to query
SRV records and later perform hostfiles check for retrieved hostnames
(instead of performing DNS A/AAAA for them).


> This is the same for other protocols. For instance, neither FTP nor HTTP
> explain how a client is supposed to resolve a host name, still the later
> explains how to parse a URI.

Maybe because DNS SRV born later than FTP or HTTP. But now look at SIP
and XMPP protocols which make *extensive* usage of DNS SRV. And SIP is
just a "signaling protocol", you could also consider it a "transport
protocol" which transports sessions and other kind of data. It's at
the very same level than WS, IMHO.



> DNS SRV is a DNS extension which only concerns
> resolvers. Not all clients will be using resolvers, just a part of them.
> Some others will simply forward the request to their HTTP proxy which wil=
l
> apply whatever DNS resolving method they know, including possibly DNS SRV=
.

If WS spec does not mandate DNS SRV resolution in WS clients (so
webbrowsers mainly) then let's forget SRV balancing/failover
capabilities. If the WS core draft does not want to handle this topic,
then refer to another document mandating it (in the same way SIP RFC
3261 mandates the usage of DNS NAPTR/SRV in RFC 3263 for SIP clients).

So you can split WS in:
- transport specification RFC (handshake, frames and so).
- location WS servers (a MUST for WS clients when resolving WS URI's).


> In practice, if there are new elements of DNS SRV that are specific to WS=
,
> they should probably be added to the DNS SRV spec and not the WS spec.

No. DNS SRV spec (RFC 2782) just explains the DNS SRV mechanism. Any
protocol can decide wheter to include/mandate it or not. Each protocol
can register in IANA new "service" values SRV, in the same way SIP and
XMPP RFC's create "sip" and "xmpp-server" values.


> Maybe the WS spec should mention that it addresses transport only and
> not address resolving. It may recommend to follow some principles to
> perform the resolving but should not specify how to do it.

That would be great, so another document/spec would *mandate* how WS
clients MUST resolve WS destinations. But that is not true now as WS
core spec states simple A/AAAA resolution for locating a WS server.


Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From dendicott@gmail.com  Thu Jul 21 10:18:34 2011
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From: David Endicott <dendicott@gmail.com>
To: Willy Tarreau <w@1wt.eu>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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Thanks Willy, you made my point better than I did.

It is my opinion that name resolution (however done) is outside the purview
of WS.   It may be handled in any number of ways by the client, and must
happen *before* WS establishes it's TCP connection and begins handshaking.

DNS, DNS SRV, etc. are good and useful tools, but are not part of WS.

A document showing how to use DNS SRV with WS would be useful, but it's not
part of the core WS spec.


On Thu, Jul 21, 2011 at 12:39 PM, Willy Tarreau <w@1wt.eu> wrote:

> On Thu, Jul 21, 2011 at 06:27:49PM +0200, I=F1aki Baz Castillo wrote:
> > 2011/7/21 David Endicott <dendicott@gmail.com>:
> > > DNS resolution is not a function of a transport protocol.  DNS SRV ha=
s
> no
> > > special association with WS.    It is my opinion that this would be
> > > additional cruft that is only marginally related to the purpose and
> function
> > > of websockets.    It does not address a general use case.   DNS SRV
> applies
> > > only to a (small?) subset of server-side implementations.    It is a
> good
> > > and useful mechanism, but I do not believe it should be tied tightly =
to
> > > websockets, nor included as part of the core spec.
> >
> > An WebSocket URI is given to a WebSocket client, and the client MUST
> > locate the corresponding WS server, right? and for locating the server
> > the client MUST follows a procedures which, for now, involve (if it's
> > not an IP) DNS A/AAAA resolution, right? So now imagine that the
> > location mechanism is a bit more powerful and also involves SRV
> > queries (not always).
> >
> > If you think that a transport protocol (like WebSocket) must not
> > resolve a server destination then also remove the WS URI inspection
> > and resolution from the core spec, don't you agree? or just DNS A/AAA
> > is valid?
> >
> > I don't agree with your opinion at all. Regards.
>
> I=F1aki,
>
> I understand the point David is making. DNS is something independant of
> WS and conversely. It is one way of resolving addresses, just like there
> will be people using hosts files. At no place the protocol specification
> dictates how a client should resolve a name to an IP address. The protoco=
l
> specifies the transport part only.
>
> This is the same for other protocols. For instance, neither FTP nor HTTP
> explain how a client is supposed to resolve a host name, still the later
> explains how to parse a URI. DNS SRV is a DNS extension which only concer=
ns
> resolvers. Not all clients will be using resolvers, just a part of them.
> Some others will simply forward the request to their HTTP proxy which wil=
l
> apply whatever DNS resolving method they know, including possibly DNS SRV=
.
>
> In practice, if there are new elements of DNS SRV that are specific to WS=
,
> they should probably be added to the DNS SRV spec and not the WS spec.
> Maybe the WS spec should mention that it addresses transport only and
> not address resolving. It may recommend to follow some principles to
> perform the resolving but should not specify how to do it.
>
> Hoping it's a bit clearer,
> Willy
>
>

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Thanks Willy, you made my point better than I did.<div><br></div><div>It is=
 my opinion that name resolution (however done) is outside the=A0purview of=
 WS. =A0 It may be handled in any number of ways by the client, and must ha=
ppen *before* WS establishes=A0it&#39;s TCP connection and begins handshaki=
ng.</div>
<div><br></div><div>DNS, DNS SRV, etc. are good and useful tools, but are n=
ot part of WS.</div><div><br></div><div>A document showing how to use DNS S=
RV with WS would be useful, but it&#39;s not part of the core WS spec.</div=
>
<div><br><br><div class=3D"gmail_quote">On Thu, Jul 21, 2011 at 12:39 PM, W=
illy Tarreau <span dir=3D"ltr">&lt;<a href=3D"mailto:w@1wt.eu">w@1wt.eu</a>=
&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0=
 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div><div></div><div class=3D"h5">On Thu, Jul 21, 2011 at 06:27:49PM +0200,=
 I=F1aki Baz Castillo wrote:<br>
&gt; 2011/7/21 David Endicott &lt;<a href=3D"mailto:dendicott@gmail.com">de=
ndicott@gmail.com</a>&gt;:<br>
&gt; &gt; DNS resolution is not a function of a transport protocol. =A0DNS =
SRV has no<br>
&gt; &gt; special association with WS. =A0 =A0It is my opinion that this wo=
uld be<br>
&gt; &gt; additional cruft that is only marginally related to the purpose a=
nd function<br>
&gt; &gt; of websockets. =A0 =A0It does not address a general use case. =A0=
 DNS SRV applies<br>
&gt; &gt; only to a (small?) subset of server-side implementations. =A0 =A0=
It is a good<br>
&gt; &gt; and useful mechanism, but I do not believe it should be tied tigh=
tly to<br>
&gt; &gt; websockets, nor included as part of the core spec.<br>
&gt;<br>
&gt; An WebSocket URI is given to a WebSocket client, and the client MUST<b=
r>
&gt; locate the corresponding WS server, right? and for locating the server=
<br>
&gt; the client MUST follows a procedures which, for now, involve (if it&#3=
9;s<br>
&gt; not an IP) DNS A/AAAA resolution, right? So now imagine that the<br>
&gt; location mechanism is a bit more powerful and also involves SRV<br>
&gt; queries (not always).<br>
&gt;<br>
&gt; If you think that a transport protocol (like WebSocket) must not<br>
&gt; resolve a server destination then also remove the WS URI inspection<br=
>
&gt; and resolution from the core spec, don&#39;t you agree? or just DNS A/=
AAA<br>
&gt; is valid?<br>
&gt;<br>
&gt; I don&#39;t agree with your opinion at all. Regards.<br>
<br>
</div></div>I=F1aki,<br>
<br>
I understand the point David is making. DNS is something independant of<br>
WS and conversely. It is one way of resolving addresses, just like there<br=
>
will be people using hosts files. At no place the protocol specification<br=
>
dictates how a client should resolve a name to an IP address. The protocol<=
br>
specifies the transport part only.<br>
<br>
This is the same for other protocols. For instance, neither FTP nor HTTP<br=
>
explain how a client is supposed to resolve a host name, still the later<br=
>
explains how to parse a URI. DNS SRV is a DNS extension which only concerns=
<br>
resolvers. Not all clients will be using resolvers, just a part of them.<br=
>
Some others will simply forward the request to their HTTP proxy which will<=
br>
apply whatever DNS resolving method they know, including possibly DNS SRV.<=
br>
<br>
In practice, if there are new elements of DNS SRV that are specific to WS,<=
br>
they should probably be added to the DNS SRV spec and not the WS spec.<br>
Maybe the WS spec should mention that it addresses transport only and<br>
not address resolving. It may recommend to follow some principles to<br>
perform the resolving but should not specify how to do it.<br>
<br>
Hoping it&#39;s a bit clearer,<br>
<font color=3D"#888888">Willy<br>
<br>
</font></blockquote></div><br></div>

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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/21 David Endicott <dendicott@gmail.com>:
> It is my opinion that name resolution (however done) is outside the=C2=A0=
purview
> of WS. =C2=A0 It may be handled in any number of ways by the client, and =
must
> happen *before* WS establishes=C2=A0it's TCP connection and begins handsh=
aking.
> DNS, DNS SRV, etc. are good and useful tools, but are not part of WS.
> A document showing how to use DNS SRV with WS would be useful, but it's n=
ot
> part of the core WS spec.

Ok, there you have a proposal:

  http://tools.ietf.org/html/draft-ibc-websocket-dns-srv-02

:)

Of course, the core WS spec should reference and *mandate* it, or
maybe not and instead a W3C specification should mandate it for
webbrowsers.
And in this case, the WS URI resolution section in core WS spec (DNS A
stuff) should be entirely removed.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [hybi] Fragmented text message
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On Thu, Jul 21, 2011 at 12:11 PM, David Endicott <dendicott@gmail.com>wrote:

> Would that not then require that the websocket peer and any websocket aware
> intermediaries examine and understand the content of the frames?   (ie. to
> be able to tell if a UTF sequence is being split).
>
> That seems, to me, to be highly impractical.  Or even impossible.
>

Correct -- it seems much better to allow fragmentation at any byte boundary.
 Defragmentation of a text message already requires keeping state -- having
to keep state for a partial character seems hardly any extra burden.

-- 
John A. Tamplin
Software Engineer (GWT), Google

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<div class=3D"gmail_quote">On Thu, Jul 21, 2011 at 12:11 PM, David Endicott=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:dendicott@gmail.com">dendicott@gma=
il.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

Would that not then require that the websocket peer and any websocket aware=
 intermediaries examine and understand the content of the frames? =C2=A0 (i=
e. to be able to tell if a UTF sequence is being split).<div><br></div><div=
>

That seems, to me, to be highly impractical. =C2=A0Or even impossible.</div=
></blockquote><div><br></div><div>Correct -- it seems much better to allow =
fragmentation at any byte boundary. =C2=A0Defragmentation of a text message=
 already requires keeping state -- having to keep state for a partial chara=
cter seems hardly any extra burden.=C2=A0</div>

</div><br>-- <br>John A. Tamplin<br>Software Engineer (GWT), Google<br>

--001636459338ac144704a897a4df--

From w@1wt.eu  Thu Jul 21 10:33:46 2011
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On Thu, Jul 21, 2011 at 07:15:13PM +0200, Iñaki Baz Castillo wrote:
> If WS spec does not mandate DNS SRV resolution in WS clients (so
> webbrowsers mainly) then let's forget SRV balancing/failover
> capabilities. If the WS core draft does not want to handle this topic,
> then refer to another document mandating it (in the same way SIP RFC
> 3261 mandates the usage of DNS NAPTR/SRV in RFC 3263 for SIP clients).
> 
> So you can split WS in:
> - transport specification RFC (handshake, frames and so).
> - location WS servers (a MUST for WS clients when resolving WS URI's).

But you can't make usage of DNS a MUST. While it generally is acceptable
over the wide Internet, it's almost always anti-pattern on internal
networks. Application developers in particular can never rely on this
because the guy who manages the DNS is not the same who manages servers
and generally doesn't accept to add any input for an application that is
not in production. Similarly, DNS which announce "public" records for
some services will cause trouble to local clients which would always
make use of the DNS first because of the MUST, and would not be able
to connect to the local address:port which is not announced anywhere.

If someone were to develop a backup/restore solution based on WS, it would
be funny to discover that it cannot be used to restore the DNS server when
this one fails...

However in my opinion it could be good to document a recommended DNS
architecture for WS, that is judged optimal and the most interoperable.
This could clearly be a separate RFC suggesting how clients and server
can make use of DNS SRV records for HA and LB.

> > In practice, if there are new elements of DNS SRV that are specific to WS,
> > they should probably be added to the DNS SRV spec and not the WS spec.
> 
> No. DNS SRV spec (RFC 2782) just explains the DNS SRV mechanism. Any
> protocol can decide wheter to include/mandate it or not. Each protocol
> can register in IANA new "service" values SRV, in the same way SIP and
> XMPP RFC's create "sip" and "xmpp-server" values.

Then maybe the name should be reserved as soon as the protocol spec is
released, and the document should refer to it.

> > Maybe the WS spec should mention that it addresses transport only and
> > not address resolving. It may recommend to follow some principles to
> > perform the resolving but should not specify how to do it.
> 
> That would be great, so another document/spec would *mandate* how WS
> clients MUST resolve WS destinations.

MAY

> But that is not true now as WS
> core spec states simple A/AAAA resolution for locating a WS server.

It does not even say that. It only talks about DNS when suggesting how
a client may select a limit on the number of concurrent connections (#5.1).

Regards,
Willy


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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Thu Jul 21 18:18:31 2011, David Endicott wrote:
> It is my opinion that name resolution (however done) is outside the  
> purview
> of WS.   It may be handled in any number of ways by the client, and  
> must
> happen *before* WS establishes it's TCP connection and begins  
> handshaking.

The URI scheme is defined here, and absolutely must explain whether  
it's a host (for direct address lookup), or a domain (for SRV).

It's proven impossible, despite effort, to retrofit SRV onto HTTP;  
there is no way it'll be possible to retrofit onto WS.

Dave.
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2011/7/21 Dave Cridland <dave@cridland.net>:
> It's proven impossible, despite effort, to retrofit SRV onto HTTP; there =
is
> no way it'll be possible to retrofit onto WS.

Right. If WS borns with no SRV (as a MUST for WS clients) then just
forget it and let inherit all the ugly limitations from HTTP protocol.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: David Endicott <dendicott@gmail.com>
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I am strongly opposed to any MUST definition for any type of URL resolution=
.


I'm ok with inheriting / mimicking HTTP.    Since it is intended to live in
the same universe as HTTP, I'm ok with it sharing mechanisms / limitations.



On Thu, Jul 21, 2011 at 1:52 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> 2011/7/21 Dave Cridland <dave@cridland.net>:
> > It's proven impossible, despite effort, to retrofit SRV onto HTTP; ther=
e
> is
> > no way it'll be possible to retrofit onto WS.
>
> Right. If WS borns with no SRV (as a MUST for WS clients) then just
> forget it and let inherit all the ugly limitations from HTTP protocol.
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
>

--20cf301fb94dfefd3504a89821be
Content-Type: text/html; charset=ISO-8859-1
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I am strongly opposed to any MUST definition for any type of URL resolution=
. =A0=A0<div><br></div><div>I&#39;m ok with inheriting /=A0mimicking=A0HTTP=
. =A0 =A0Since it is intended to live in the same universe as HTTP, I&#39;m=
 ok with it sharing mechanisms / limitations.</div>
<div><br></div><div><br></div><div><br><div class=3D"gmail_quote">On Thu, J=
ul 21, 2011 at 1:52 PM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=
=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex;">
2011/7/21 Dave Cridland &lt;<a href=3D"mailto:dave@cridland.net">dave@cridl=
and.net</a>&gt;:<br>
<div class=3D"im">&gt; It&#39;s proven impossible, despite effort, to retro=
fit SRV onto HTTP; there is<br>
&gt; no way it&#39;ll be possible to retrofit onto WS.<br>
<br>
</div>Right. If WS borns with no SRV (as a MUST for WS clients) then just<b=
r>
forget it and let inherit all the ugly limitations from HTTP protocol.<br>
<div><div></div><div class=3D"h5"><br>
--<br>
I=F1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
</div></div></blockquote></div><br></div>

--20cf301fb94dfefd3504a89821be--

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2011/7/21 David Endicott <dendicott@gmail.com>:
> I am strongly opposed to any MUST definition for any type of URL resoluti=
on.

SIP and XMPP mandate (MUST) a resolution mechanism based on NAPTR, SRV
and A/AAAA records. Are they also wrong? do you also oppose to the DNS
MX resolution (as mandatory) for a mailto: URI? Do you imagine that a
mail server admin could not assume that SMTP clients would always use
MX resolution as the first choice? annoying that you say that, sorry.


> I'm ok with inheriting /=C2=A0mimicking=C2=A0HTTP. =C2=A0 =C2=A0Since it =
is intended to live in
> the same universe as HTTP, I'm ok with it sharing mechanisms / limitation=
s.

Yes, I assume many people in the HTTP warden is fine with this. That
is the problem: forcing a *new* protocol to inherit ugly limitations
just because "people is used to them". I don't understand how you can
prefer to ignore cool NEW solutions/mechanisms. This should not be a
valid argument in a new protocol design.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From dendicott@gmail.com  Thu Jul 21 11:43:17 2011
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I do not know SIP or XMPP well enough to comment on why they mandated the
name resolution mechanisms they did.    However, XMPP is used in a highly
dynamic host environment, so having flexible & extensible name resolution i=
s
likely an excellent choice.

I do not oppose DNS's MX records for SMTP as email addresses are name@domai=
n,
so obviously the Domain Name System is appropriate.    Also, I fail to see
why a mail admin should care how the SMTP clients arrive at the server.  DN=
S
MX is a reasonable solution, but there may be other methods, any and all of
which are irrelevant to the SMTP server.    Especially when the SMTP server
supports multiple email domains....

Since WS is intended as a browser supported protocol, WS should follow the
same URI resolution mechanisms as HTTP (or how URI resolution is done in
general)  Having URLs that could resolve differently for a HTTP request and
a WS setup is a problem.


However....I agree in the sense that I don't think it's been well thought
out what the ramifications are of introducing a new URI *scheme*.  (ie.
ws:// and wss://)    Since everything after the double-slash is "scheme
specific", defining a scheme implies many things (including name resolution
policies, well-known ports, transport protocols, etc.)   But, I don't want
to get side tracked over this.



On Thu, Jul 21, 2011 at 2:10 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> 2011/7/21 David Endicott <dendicott@gmail.com>:
> > I am strongly opposed to any MUST definition for any type of URL
> resolution.
>
> SIP and XMPP mandate (MUST) a resolution mechanism based on NAPTR, SRV
> and A/AAAA records. Are they also wrong? do you also oppose to the DNS
> MX resolution (as mandatory) for a mailto: URI? Do you imagine that a
> mail server admin could not assume that SMTP clients would always use
> MX resolution as the first choice? annoying that you say that, sorry.
>
>
> > I'm ok with inheriting / mimicking HTTP.    Since it is intended to liv=
e
> in
> > the same universe as HTTP, I'm ok with it sharing mechanisms /
> limitations.
>
> Yes, I assume many people in the HTTP warden is fine with this. That
> is the problem: forcing a *new* protocol to inherit ugly limitations
> just because "people is used to them". I don't understand how you can
> prefer to ignore cool NEW solutions/mechanisms. This should not be a
> valid argument in a new protocol design.
>
>
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
>

--0016364d2c795297c804a898b8e9
Content-Type: text/html; charset=ISO-8859-1
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I do not know SIP or XMPP well enough to comment on why they mandated the n=
ame resolution mechanisms they did. =A0 =A0However, XMPP is used in a highl=
y dynamic host=A0environment, so having flexible &amp; extensible name reso=
lution is likely an excellent choice.<div>
<br></div><div>I do not oppose DNS&#39;s MX records for SMTP as email addre=
sses are name@domain, so obviously the Domain Name System is appropriate. =
=A0 =A0Also, I fail to see why a mail admin should care how the SMTP client=
s arrive at the server. =A0DNS MX is a reasonable solution, but there may b=
e other methods, any and all of which are irrelevant to the SMTP server. =
=A0 =A0Especially when the SMTP server supports multiple email domains....<=
/div>
<div><br></div><div>Since WS is intended as a browser supported protocol, W=
S should follow the same URI resolution mechanisms as HTTP (or how URI reso=
lution is done in general) =A0Having URLs that could resolve differently fo=
r a HTTP request and a WS setup is a problem.</div>
<div><br></div><div><br></div><div>However....I agree in the sense that I d=
on&#39;t think it&#39;s been well thought out what the ramifications are of=
 introducing a new URI *scheme*. =A0(ie. ws:// and wss://) =A0 =A0Since eve=
rything after the double-slash is &quot;scheme specific&quot;, defining a s=
cheme implies many things (including name resolution policies, well-known p=
orts, transport protocols, etc.) =A0 But, I don&#39;t want to get side trac=
ked over this.</div>
<div><br></div><div><br><br><div class=3D"gmail_quote">On Thu, Jul 21, 2011=
 at 2:10 PM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:i=
bc@aliax.net">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex;">
<div class=3D"im">2011/7/21 David Endicott &lt;<a href=3D"mailto:dendicott@=
gmail.com">dendicott@gmail.com</a>&gt;:<br>
</div><div class=3D"im">&gt; I am strongly opposed to any MUST definition f=
or any type of URL resolution.<br>
<br>
</div>SIP and XMPP mandate (MUST) a resolution mechanism based on NAPTR, SR=
V<br>
and A/AAAA records. Are they also wrong? do you also oppose to the DNS<br>
MX resolution (as mandatory) for a mailto: URI? Do you imagine that a<br>
mail server admin could not assume that SMTP clients would always use<br>
MX resolution as the first choice? annoying that you say that, sorry.<br>
<div class=3D"im"><br>
<br>
&gt; I&#39;m ok with inheriting /=A0mimicking=A0HTTP. =A0 =A0Since it is in=
tended to live in<br>
&gt; the same universe as HTTP, I&#39;m ok with it sharing mechanisms / lim=
itations.<br>
<br>
</div>Yes, I assume many people in the HTTP warden is fine with this. That<=
br>
is the problem: forcing a *new* protocol to inherit ugly limitations<br>
just because &quot;people is used to them&quot;. I don&#39;t understand how=
 you can<br>
prefer to ignore cool NEW solutions/mechanisms. This should not be a<br>
valid argument in a new protocol design.<br>
<font color=3D"#888888"><br>
<br>
<br>
--<br>
</font><div><div></div><div class=3D"h5">I=F1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
</div></div></blockquote></div><br></div>

--0016364d2c795297c804a898b8e9--

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From: Dave Cridland <dave@cridland.net>
To: David Endicott <dendicott@gmail.com>, Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>, Willy Tarreau <w@1wt.eu>, =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Thu Jul 21 19:43:07 2011, David Endicott wrote:
> I do not know SIP or XMPP well enough to comment on why they  
> mandated the
> name resolution mechanisms they did.    However, XMPP is used in a  
> highly
> dynamic host environment, so having flexible & extensible name  
> resolution is
> likely an excellent choice.
> 
> 
I have no idea what you might mean by "highly dynamic host  
environment" in this context, but XMPP servers are normally found at  
the same location consistently. However, it is *not* always (or  
typically) the same location as a simple A record lookup:

;; ANSWER SECTION:
_xmpp-server._tcp.isode.com. 259200 IN	SRV	5 0 5269 xmpp.isode.com.

;; ADDITIONAL SECTION:
xmpp.isode.com.		225364	IN	A	62.3.217.250

;; ANSWER SECTION:
isode.com.		73130	IN	A	87.106.143.99

This property alone is very useful - in a websockets case this would  
mean being able to provide websockets services from a different host  
(or network) to the traditional web services in a simple manner,  
fully compatible with SOP.

The fact that this also allows cheap lightweight load balancing and  
fallback control is also useful in other cases; none of this relates  
to dynamic hosts, but simply richer service location.


> I do not oppose DNS's MX records for SMTP as email addresses are  
> name@domain,
> so obviously the Domain Name System is appropriate.    Also, I fail  
> to see
> why a mail admin should care how the SMTP clients arrive at the  
> server.  DNS
> MX is a reasonable solution, but there may be other methods, any  
> and all of
> which are irrelevant to the SMTP server.    Especially when the  
> SMTP server
> supports multiple email domains....
> 
> 
A mail admin does need to care *that* the service is located, and  
therefore will care *how* it is located.

You can be as theoretical as you like, by all means, but in practical  
terms, your email address (and my XMPP address) work because they use  
a defined, interoperable, service location mechanism, which operates  
via DNS record lookup.

(Also, I have no idea what multiple domains has to do with this.)


> Since WS is intended as a browser supported protocol, WS should  
> follow the
> same URI resolution mechanisms as HTTP (or how URI resolution is  
> done in
> general)  Having URLs that could resolve differently for a HTTP  
> request and
> a WS setup is a problem.
> 
> 
But they do resolve differently anyway. You don't get a page from a  
'ws' scheme URI, you get a transport protocol connection. This is  
good, indeed, it's kind of the point.

Your suggestion of "how URI resolution is done in general" is  
somewhat self-defeating, too, since aside from 'http' and 'https',  
there are 'mailto', which uses MX, 'sip' and 'xmpp', which both use  
SRV.

I think opponents of SRV records need to mount a stronger argument  
than the kind of luddite argument that if it's hard for one protocol  
in use by the browser, it should be hard for them all.

Dave.
-- 
Dave Cridland - mailto:dave@cridland.net - xmpp:dwd@dave.cridland.net
  - acap://acap.dave.cridland.net/byowner/user/dwd/bookmarks/
  - http://dave.cridland.net/
Infotrope Polymer - ACAP, IMAP, ESMTP, and Lemonade

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>
> I have no idea what you might mean by "highly dynamic host environment" in
> this context, but XMPP servers are normally found at the same location
> consistently. However, it is *not* always (or typically) the same location
> as a simple A record lookup:
>

That's what I meant.  XMPP systems have hosts that change around (for many
reasons) and having a name resolution that handles that is good.



> This property alone is very useful - in a websockets case this would mean
> being able to provide websockets services from a different host (or network)
> to the traditional web services in a simple manner, fully compatible with
> SOP.   The fact that this also allows cheap lightweight load balancing and
> fallback control is also useful in other cases; none of this relates to
> dynamic hosts, but simply richer service location.


Yes, those are all excellent reasons to use DNS SRV.   None of them are a
reason to mandate that WS require it.   Because something is good for some
(or many) use cases, does not mean it is appropriate for everything and
certainly is not a reason to mandate it as a requirement.
 System implementer should be free to pick and choose tools and mechanisms
appropriate for their tasks.   DNS SRV would likely be an excellent choice
for many people.   But it should not be the one and only choice.   That's
really all I'm saying - don't force people to use something without an
overwhelming reason to make it the only option.



> I do not oppose DNS's MX records for SMTP as email addresses are
>> name@domain,
>> so obviously the Domain Name System is appropriate.    Also, I fail to see
>> why a mail admin should care how the SMTP clients arrive at the server.
>>  DNS
>> MX is a reasonable solution, but there may be other methods, any and all
>> of
>> which are irrelevant to the SMTP server.    Especially when the SMTP
>> server
>> supports multiple email domains...
>>
>>  A mail admin does need to care *that* the service is located, and
> therefore will care *how* it is located.

You can be as theoretical as you like, by all means, but in practical terms,
> your email address (and my XMPP address) work because they use a defined,
> interoperable, service location mechanism, which operates via DNS record
> lookup.
>
> (Also, I have no idea what multiple domains has to do with this.)


Imagine I'm a SMTP server.   People connect to me.   They do SMTP
transactions.    I do not care how they found me.   Perhaps they used DNS to
find the MX server.  Perhaps they had it cached from before.  Perhaps they
guessed.  Perhaps it's in a hosts file.   I don't care.     I answer VRFY
and RCPT TO commands as appropriate.   If the "name" they are trying to
mailwith is one I recognize, I process it.  If I don't, it's an error.
Just because DNS-MX said that @foobar was handled at <addr>, doesn't mean
the dave@foobar is going to work.

Yes, DNS MX is a well known mechanism for determining what SMTP server to
connect with, but like I tried to say above, it's not mandated by the SMTP
protocol.   DNS MX is independent of SMTP and the two mechanisms
operate separately, but with a common goal.  I can use DNS to resolve a name
and never send email/message.  I can send a email/message via SMTP and never
use DNS to resolve a name.    Or I can use one to do the other.

When a SMTP server handles mail for multiple domains, the SMTP server has to
process the @domain part of the RCPT TO request - DNS is not involved at
that point.   This process is unrelated to any DNS MX definitions.    I used
that as an example of how some name resolutions are sometimes done outside
of any DNS framework.



> Since WS is intended as a browser supported protocol, WS should follow the
>> same URI resolution mechanisms as HTTP (or how URI resolution is done in
>> general)  Having URLs that could resolve differently for a HTTP request
>> and
>> a WS setup is a problem.
>>
>>  But they do resolve differently anyway. You don't get a page from a 'ws'
> scheme URI, you get a transport protocol connection. This is good, indeed,
> it's kind of the point.
>

Do they?   A http uri and a ws uri have the same host/path construction.
 It's really only the scheme that differs - and that identifies the
transport protocol to be used.   Resolution of host name/addresses and
mapping of paths "should" be consistent.

WS is a connection that is semantically related to the URI of the request.


e.g. I could ws://host/davesaid  and get live traffic of what Dave is
saying, and then I could ws://host/bobsaid  and get traffic of what Bob
says.  I wouldn't get Bob on /davesaid and I wouldn't get Dave on /bobsaid.
   Dynamic content identified by a URI

And if I http://host/davesaid  I could get a <li> of what Dave said.
Static content of a URI.

It could be problematic if  ws://host/davesaid resolves to a different
address than http://host/davesaid.     (Or it could be advantage - not for
us to decide, however)


> Your suggestion of "how URI resolution is done in general" is somewhat
> self-defeating, too, since aside from 'http' and 'https', there are
> 'mailto', which uses MX, 'sip' and 'xmpp', which both use SRV.
>

As you just said, the universe is bigger than just xmpp, sip, and http.

>
> I think opponents of SRV records need to mount a stronger argument than the
> kind of luddite argument that if it's hard for one protocol in use by the
> browser, it should be hard for them all.


I think you misinterpret my position.  And I resent the luddite slight.   I
think DNS SRV is an awesome tool and would greatly benefit many
implementations.

My position is that it should not be a *requirement*.     It should be an
optional mechanism that can be used if desired.   Further, since WS is a
bastard cousin to HTTP, they should share a similar name resolution
mechanism.



>
>
> Dave.
> --
> Dave Cridland - mailto:dave@cridland.net - xmpp:dwd@dave.cridland.net
>  - acap://acap.dave.cridland.net/**byowner/user/dwd/bookmarks/<http://acap.dave.cridland.net/byowner/user/dwd/bookmarks/>
>  - http://dave.cridland.net/
> Infotrope Polymer - ACAP, IMAP, ESMTP, and Lemonade
>

--001517592bd091cf6f04a89a9883
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">I have no idea w=
hat you might mean by &quot;highly dynamic host environment&quot; in this c=
ontext, but XMPP servers are normally found at the same location consistent=
ly. However, it is *not* always (or typically) the same location as a simpl=
e A record lookup:<br>
</blockquote><div><br></div><div>That&#39;s what I meant. =A0XMPP systems h=
ave hosts that change around (for many reasons) and having a name resolutio=
n that handles that is good.</div><div><br></div><div>=A0</div><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex;">

This property alone is very useful - in a websockets case this would mean b=
eing able to provide websockets services from a different host (or network)=
 to the traditional web services in a simple manner, fully compatible with =
SOP. =A0=A0The fact that this also allows cheap lightweight load balancing =
and fallback control is also useful in other cases; none of this relates to=
 dynamic hosts, but simply richer service location.</blockquote>
<div><br></div><div>Yes, those are all excellent reasons to use DNS SRV. =
=A0 None of them are a reason to mandate that WS require it. =A0 Because so=
mething is good for some (or many) use cases, does not mean it is appropria=
te for everything and certainly is not a reason to mandate it as a requirem=
ent. =A0 =A0System=A0implementer=A0should be free to pick and choose tools =
and mechanisms appropriate for their tasks. =A0 DNS SRV would likely be an =
excellent choice for many people. =A0 But it should not be the one and only=
 choice. =A0 That&#39;s really all I&#39;m saying - don&#39;t force people =
to use something without an overwhelming reason to make it the only option.=
</div>
<div><br></div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div class=3D"=
im"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-lef=
t:1px #ccc solid;padding-left:1ex">
I do not oppose DNS&#39;s MX records for SMTP as email addresses are name@d=
omain,<br>
so obviously the Domain Name System is appropriate. =A0 =A0Also, I fail to =
see<br>
why a mail admin should care how the SMTP clients arrive at the server. =A0=
DNS<br>
MX is a reasonable solution, but there may be other methods, any and all of=
<br>
which are irrelevant to the SMTP server. =A0 =A0Especially when the SMTP se=
rver<br>
supports multiple email domains...<br>
<br>
</blockquote></div>
A mail admin does need to care *that* the service is located, and therefore=
 will care *how* it is located.</blockquote><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"=
>

You can be as theoretical as you like, by all means, but in practical terms=
, your email address (and my XMPP address) work because they use a defined,=
 interoperable, service location mechanism, which operates via DNS record l=
ookup.<br>

<br>
(Also, I have no idea what multiple domains has to do with this.)</blockquo=
te><div><br></div><meta http-equiv=3D"content-type" content=3D"text/html; c=
harset=3Dutf-8"><div>Imagine I&#39;m a SMTP server. =A0 People connect to m=
e. =A0 They do SMTP transactions. =A0 =A0I do not care how they found me. =
=A0 Perhaps they used DNS to find the MX server. =A0Perhaps they had it cac=
hed from before. =A0Perhaps they guessed. =A0Perhaps it&#39;s in a hosts fi=
le. =A0 I don&#39;t care. =A0 =A0 I answer VRFY and RCPT TO commands as app=
ropriate. =A0 If the &quot;name&quot; they are trying to mailwith is one I =
recognize, I process it. =A0If I don&#39;t, it&#39;s an error. =A0 Just bec=
ause DNS-MX said that @foobar was handled at &lt;addr&gt;, doesn&#39;t mean=
 the dave@foobar is going to work.</div>
<div><br></div><div>Yes, DNS MX is a well known mechanism for determining w=
hat SMTP server to connect with, but like I tried to say above, it&#39;s no=
t mandated by the SMTP protocol. =A0 DNS MX is independent of SMTP and the =
two mechanisms operate=A0separately, but with a common goal. =A0I can use D=
NS to resolve a name and never send email/message. =A0I can send a email/me=
ssage via SMTP and never use DNS to resolve a name. =A0 =A0Or I can use one=
 to do the other.</div>
<div><br></div><div>When a SMTP server handles mail for multiple domains, t=
he SMTP server has to process the @domain part of the RCPT TO request - DNS=
 is not involved at that point. =A0 This process is unrelated to any DNS MX=
 definitions. =A0 =A0I used that as an example of how some name resolutions=
 are sometimes done outside of any DNS framework.</div>
<div><br></div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div class=3D"=
im"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-lef=
t:1px #ccc solid;padding-left:1ex">
Since WS is intended as a browser supported protocol, WS should follow the<=
br>
same URI resolution mechanisms as HTTP (or how URI resolution is done in<br=
>
general) =A0Having URLs that could resolve differently for a HTTP request a=
nd<br>
a WS setup is a problem.<br>

<br>
</blockquote></div>
But they do resolve differently anyway. You don&#39;t get a page from a &#3=
9;ws&#39; scheme URI, you get a transport protocol connection. This is good=
, indeed, it&#39;s kind of the point.<br></blockquote><div>=A0</div><div>
Do they? =A0 A http uri and a ws uri have the same host/path construction. =
=A0It&#39;s really only the scheme that differs - and that identifies the t=
ransport protocol to be used. =A0 Resolution of host name/addresses and map=
ping of paths &quot;should&quot; be consistent.</div>
<div><br></div><div>WS is a connection that is semantically related to the =
URI of the request. =A0=A0</div><div><br></div><div>e.g. I could ws://host/=
davesaid =A0and get live traffic of what Dave is saying, and then I could w=
s://host/bobsaid =A0and get traffic of what Bob says. =A0I wouldn&#39;t get=
 Bob on /davesaid and I wouldn&#39;t get Dave on /bobsaid. =A0 =A0Dynamic c=
ontent identified by a URI =A0 =A0 =A0</div>
<div><br></div><div>And if I <a href=3D"http://host/davesaid">http://host/d=
avesaid</a> =A0I could get a &lt;li&gt; of what Dave said. =A0 Static conte=
nt of a URI.</div><div><br></div><div>It could be problematic if =A0ws://ho=
st/davesaid resolves to a different address than <a href=3D"http://host/dav=
esaid">http://host/davesaid</a>. =A0 =A0 (Or it could be advantage - not fo=
r us to decide, however)</div>
<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex;">
<br>
Your suggestion of &quot;how URI resolution is done in general&quot; is som=
ewhat self-defeating, too, since aside from &#39;http&#39; and &#39;https&#=
39;, there are &#39;mailto&#39;, which uses MX, &#39;sip&#39; and &#39;xmpp=
&#39;, which both use SRV.<br>
</blockquote><div><br></div><div>As you just said, the universe is bigger t=
han just xmpp, sip, and http. =A0=A0</div><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<br>
I think opponents of SRV records need to mount a stronger argument than the=
 kind of luddite argument that if it&#39;s hard for one protocol in use by =
the browser, it should be hard for them all.</blockquote><div><br></div>
<div>I think you misinterpret my position. =A0And=A0I resent the luddite sl=
ight. =A0=A0I think DNS SRV is an awesome tool and would greatly benefit ma=
ny implementations.</div><meta http-equiv=3D"content-type" content=3D"text/=
html; charset=3Dutf-8"><div>
<br></div><div>My position is that it should not be a *requirement*. =A0 =
=A0 It should be an optional mechanism that can be used if desired. =A0 Fur=
ther, since WS is a bastard cousin to HTTP, they should share a similar nam=
e resolution mechanism. =A0 =A0=A0</div>
<div><br></div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div><div></di=
v><div class=3D"h5"><br>
<br>
Dave.<br>
-- <br>
Dave Cridland - mailto:<a href=3D"mailto:dave@cridland.net" target=3D"_blan=
k">dave@cridland.net</a> - <a href=3D"mailto:xmpp%3Adwd@dave.cridland.net" =
target=3D"_blank">xmpp:dwd@dave.cridland.net</a><br>
=A0- acap://<a href=3D"http://acap.dave.cridland.net/byowner/user/dwd/bookm=
arks/" target=3D"_blank">acap.dave.cridland.net/<u></u>byowner/user/dwd/boo=
kmarks/</a><br>
=A0- <a href=3D"http://dave.cridland.net/" target=3D"_blank">http://dave.cr=
idland.net/</a><br>
Infotrope Polymer - ACAP, IMAP, ESMTP, and Lemonade<br>
</div></div></blockquote></div><br>

--001517592bd091cf6f04a89a9883--

From bruce@callenish.com  Thu Jul 21 14:01:28 2011
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Subject: Re: [hybi] Fwd: Gen-ART last call review of draft-ietf-hybi-thewebsocketprotocol-10
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I highly suggest that anyone with concerns around masking take a look at 
the FAQ[1]. Skip down to the section labelled "FAQS", that is where the 
useful information is to be found, particularly the odyssey involved in 
coming to a decision on masking to be found in the answer to the 
question "Why does the WebSockets protocol use masking from the client 
to the server?".

There are lots of issues, lots of solutions, and lots of objections to 
every one of the solutions. If a reviewer is going to question the 
utility of masking they should make themselves aware of that FAQ so that 
they can properly understand the context.

Should some portion of that answer be put into the spec? The WG decided 
that it was too complicated and distracted from defining the protocol. 
If a reviewer thinks some portion SHOULD be included, then my personal 
opinion is that wording should be presented to the working group to be 
considered for consensus and adoption.

[1] http://wiki.tools.ietf.org/wg/hybi/trac/wiki/FAQ

On 20/07/2011 8:48 PM, David Endicott wrote:
> I hope you understand the anxiety around this issue.
>
> >We're doing masking
> Why?
> > Because (somebody said they researched and besides we can imagine...)
> Explain please.
> > No.  If anyone else asks, somebody might blog some reasoning
>
>
> On Wed, Jul 20, 2011 at 11:44 PM, John Tamplin <jat@google.com 
> <mailto:jat@google.com>> wrote:
>
>     On Wed, Jul 20, 2011 at 11:37 PM, Thomson, Martin
>     <Martin.Thomson@commscope.com
>     <mailto:Martin.Thomson@commscope.com>> wrote:
>     > To those providing responses to review comments like this,
>     consider for a moment that perhaps the draft
>     > does not - and should - provide the answer.
>
>     There was discussion about that to, and the majority opinion (not
>     mine) was that rationale did not belong in the spec.  If warranted, a
>     separate rationale document could be written.
>
>     --
>     John A. Tamplin
>     Software Engineer (GWT), Google
>     _______________________________________________
>     hybi mailing list
>     hybi@ietf.org <mailto:hybi@ietf.org>
>     https://www.ietf.org/mailman/listinfo/hybi
>
>
>
>
> _______________________________________________
> hybi mailing list
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> https://www.ietf.org/mailman/listinfo/hybi


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<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    I highly suggest that anyone with concerns around masking take a
    look at the FAQ[1]. Skip down to the section labelled "FAQS", that
    is where the useful information is to be found, particularly the
    odyssey involved in coming to a decision on masking to be found in
    the answer to the question "Why does the WebSockets protocol use
    masking from the client to the server?".<br>
    <br>
    There are lots of issues, lots of solutions, and lots of objections
    to every one of the solutions. If a reviewer is going to question
    the utility of masking they should make themselves aware of that FAQ
    so that they can properly understand the context.<br>
    <br>
    Should some portion of that answer be put into the spec? The WG
    decided that it was too complicated and distracted from defining the
    protocol. If a reviewer thinks some portion SHOULD be included, then
    my personal opinion is that wording should be presented to the
    working group to be considered for consensus and adoption. <br>
    <br>
    [1] <a class="moz-txt-link-freetext" href="http://wiki.tools.ietf.org/wg/hybi/trac/wiki/FAQ">http://wiki.tools.ietf.org/wg/hybi/trac/wiki/FAQ</a><br>
    <br>
    On 20/07/2011 8:48 PM, David Endicott wrote:
    <blockquote
cite="mid:CAP992=F9zipxN8hmh_9bjRDdyG0C0O-7UatNo5RFhSRq2kxdag@mail.gmail.com"
      type="cite">I hope you understand the anxiety around this issue.
      <div><br>
      </div>
      <div>&gt;We're doing masking</div>
      <div>Why?</div>
      <div>&gt; Because (somebody said they researched and besides we
        can imagine...)</div>
      <div>Explain please.</div>
      <div>&gt; No. &nbsp;If anyone else asks, somebody might blog some
        reasoning</div>
      <div><br>
      </div>
      <div>&nbsp;</div>
      <div><br>
        <div class="gmail_quote">On Wed, Jul 20, 2011 at 11:44 PM, John
          Tamplin <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:jat@google.com">jat@google.com</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex;">
            <div class="im">On Wed, Jul 20, 2011 at 11:37 PM, Thomson,
              Martin<br>
              &lt;<a moz-do-not-send="true"
                href="mailto:Martin.Thomson@commscope.com">Martin.Thomson@commscope.com</a>&gt;
              wrote:<br>
              &gt; To those providing responses to review comments like
              this, consider for a moment that perhaps the draft<br>
              &gt; does not - and should - provide the answer.<br>
              <br>
            </div>
            There was discussion about that to, and the majority opinion
            (not<br>
            mine) was that rationale did not belong in the spec. &nbsp;If
            warranted, a<br>
            separate rationale document could be written.<br>
            <font color="#888888"><br>
              --<br>
            </font>
            <div class="im">John A. Tamplin<br>
              Software Engineer (GWT), Google<br>
              _______________________________________________<br>
            </div>
            <div>
              <div class="h5">hybi mailing list<br>
                <a moz-do-not-send="true" href="mailto:hybi@ietf.org">hybi@ietf.org</a><br>
                <a moz-do-not-send="true"
                  href="https://www.ietf.org/mailman/listinfo/hybi"
                  target="_blank">https://www.ietf.org/mailman/listinfo/hybi</a><br>
              </div>
            </div>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
hybi mailing list
<a class="moz-txt-link-abbreviated" href="mailto:hybi@ietf.org">hybi@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/hybi">https://www.ietf.org/mailman/listinfo/hybi</a>
</pre>
    </blockquote>
    <br>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Thu Jul 21 21:57:23 2011, David Endicott wrote:
> >
> > I have no idea what you might mean by "highly dynamic host  
> environment" in
> > this context, but XMPP servers are normally found at the same  
> location
> > consistently. However, it is *not* always (or typically) the same  
> location
> > as a simple A record lookup:
> >
> 
> That's what I meant.  XMPP systems have hosts that change around  
> (for many
> reasons) and having a name resolution that handles that is good.
> 
> 
But that statement makes no sense.

Firstly, XMPP servers simply *don't* change around. Really. I don't  
think Google's servers have changed since the service launched:

;; ANSWER SECTION:
_xmpp-server._tcp.gmail.com. 26125 IN	SRV	5 0 5269  
xmpp-server.l.google.com.
_xmpp-server._tcp.gmail.com. 26125 IN	SRV	20 0 5269  
xmpp-server1.l.google.com.
_xmpp-server._tcp.gmail.com. 26125 IN	SRV	20 0 5269  
xmpp-server2.l.google.com.
_xmpp-server._tcp.gmail.com. 26125 IN	SRV	20 0 5269  
xmpp-server3.l.google.com.
_xmpp-server._tcp.gmail.com. 26125 IN	SRV	20 0 5269  
xmpp-server4.l.google.com.

The *clients* move, but they're *doing* SRV resolution, in order to  
locate the servers for their domain.

Secondly, there's nothing dynamic or magical about SRV lookup; the  
records are no more or less static than any other. SRV handles  
roaming hosts as targets no better or worse than A records,  
therefore. All SRV adds is a way of adding indirection, host/port  
discovery, and pushing fallback and load balancing toward the client.

I'm concerned that you may have critically misunderstood what SRV  
records are useful for.

> > This property alone is very useful - in a websockets case this  
> would mean
> > being able to provide websockets services from a different host  
> (or network)
> > to the traditional web services in a simple manner, fully  
> compatible with
> > SOP.   The fact that this also allows cheap lightweight load  
> balancing and
> > fallback control is also useful in other cases; none of this  
> relates to
> > dynamic hosts, but simply richer service location.
> 
> 
> Yes, those are all excellent reasons to use DNS SRV.   None of them  
> are a
> reason to mandate that WS require it.   Because something is good  
> for some
> (or many) use cases, does not mean it is appropriate for everything  
> and
> certainly is not a reason to mandate it as a requirement.
>  System implementer should be free to pick and choose tools and  
> mechanisms
> appropriate for their tasks.   DNS SRV would likely be an excellent  
> choice
> for many people.   But it should not be the one and only choice.    
> That's
> really all I'm saying - don't force people to use something without  
> an
> overwhelming reason to make it the only option.

SRV records simply can't be bolted on afterwards. That's been proven  
with HTTP itself. That's an overwhelming reason, in my opinion.

> Imagine I'm a SMTP server.   People connect to me.   They do SMTP
> transactions.    I do not care how they found me.   Perhaps they  
> used DNS to
> find the MX server.  Perhaps they had it cached from before.   
> Perhaps they
> guessed.  Perhaps it's in a hosts file.   I don't care.     I  
> answer VRFY
> and RCPT TO commands as appropriate.   If the "name" they are  
> trying to
> mailwith is one I recognize, I process it.  If I don't, it's an  
> error.
> Just because DNS-MX said that @foobar was handled at <addr>,  
> doesn't mean
> the dave@foobar is going to work.
> 
> 
Erm. OK...

In the real world, people do care that their mailserver can be found,  
and so they publish MX records, in the confidence that other mail  
servers will use them. This is similar to being a webmaster and  
wanting your website to be found, thus putting A records in. But  
cleverer.

Now, I agree that, in theory, it would be possible to not bother with  
DNS, and simply ask people to put your hostname into their hosts  
files, but you know, I don't think this would work as well, and it's  
certainly not an argument against SRV. (Iñaki's proposal, as I  
recall, even has fallbacks to allow server administrators to avoid  
SRV records in some cases).


> Yes, DNS MX is a well known mechanism for determining what SMTP  
> server to
> connect with, but like I tried to say above, it's not mandated by  
> the SMTP
> protocol.   DNS MX is independent of SMTP and the two mechanisms
> operate separately, but with a common goal.  I can use DNS to  
> resolve a name
> and never send email/message.  I can send a email/message via SMTP  
> and never
> use DNS to resolve a name.    Or I can use one to do the other.
> 
> 
In practical terms, the MX record and SMTP are interlinked. Moreover,  
RFC 5321 says:

   Only resolvable, fully-qualified domain names (FQDNs) are permitted
   when domain names are used in SMTP.  In other words, names that can
   be resolved to MX RRs or address (i.e., A or AAAA) RRs (as  
discussed
   in Section 5) are permitted, as are CNAME RRs whose targets can be
   resolved, in turn, to MX or address RRs.  Local nicknames or
   unqualified names MUST NOT be used.

So there is an argument that it does, indeed, mandate DNS.


> When a SMTP server handles mail for multiple domains, the SMTP  
> server has to
> process the @domain part of the RCPT TO request - DNS is not  
> involved at
> that point.   This process is unrelated to any DNS MX definitions.   
>   I used
> that as an example of how some name resolutions are sometimes done  
> outside
> of any DNS framework.

But the name resolutions are done in exactly the same way, whether  
the mailserver handles one or many domains.

So a mail administrator must ensure that MX records for all the  
domains point to the mailserver host.


> Do they?   A http uri and a ws uri have the same host/path  
> construction.
>  It's really only the scheme that differs - and that identifies the
> transport protocol to be used.   Resolution of host name/addresses  
> and
> mapping of paths "should" be consistent.
> 
> 
So you're expecting xmpp://dave.cridland@isode.com to do what,  
exactly? Because it doesn't do an A record lookup of isode.com, for  
starters.


> WS is a connection that is semantically related to the URI of the  
> request.
> 
> 
> e.g. I could ws://host/davesaid  and get live traffic of what Dave  
> is
> saying, and then I could ws://host/bobsaid  and get traffic of what  
> Bob
> says.  I wouldn't get Bob on /davesaid and I wouldn't get Dave on  
> /bobsaid.
>    Dynamic content identified by a URI
> 
> And if I http://host/davesaid  I could get a <li> of what Dave said.
> Static content of a URI.
> 
> It could be problematic if  ws://host/davesaid resolves to a  
> different
> address than http://host/davesaid.     (Or it could be advantage -  
> not for
> us to decide, however)

I have no idea what any of this means, but it sounds largely  
theoretical, and orthogonal to the point at hand.

Let's put it in practical terms.

'ws' and 'http' URIs have to be treated in different ways.

Therefore, we can treat them in different ways.

> > Your suggestion of "how URI resolution is done in general" is  
> somewhat
> > self-defeating, too, since aside from 'http' and 'https', there  
> are
> > 'mailto', which uses MX, 'sip' and 'xmpp', which both use SRV.
> >
> 
> As you just said, the universe is bigger than just xmpp, sip, and  
> http.
> 
> 
Erm.

Yes.

What this has to do with SRV record usage in websockets is entirely  
beyond me.


> >
> > I think opponents of SRV records need to mount a stronger  
> argument than the
> > kind of luddite argument that if it's hard for one protocol in  
> use by the
> > browser, it should be hard for them all.
> 
> 
> I think you misinterpret my position.  And I resent the luddite  
> slight.   I
> think DNS SRV is an awesome tool and would greatly benefit many
> implementations.
> 
> My position is that it should not be a *requirement*.     It should  
> be an
> optional mechanism that can be used if desired.   Further, since WS  
> is a
> bastard cousin to HTTP, they should share a similar name resolution
> mechanism.

My argument is that it cannot be made optional, so if we want to ever  
take advantage of this "awesome tool", we need to bake it in from the  
start.

Dave.
-- 
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From: Dave Cridland <dave@cridland.net>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Thu Jul 21 18:33:38 2011, Willy Tarreau wrote:
> If someone were to develop a backup/restore solution based on WS,  
> it would
> be funny to discover that it cannot be used to restore the DNS  
> server when
> this one fails...
> 
> 
If SRV (with a fallback) is defined as part of the core spec, this  
usage would use the fallback.


> However in my opinion it could be good to document a recommended DNS
> architecture for WS, that is judged optimal and the most  
> interoperable.
> This could clearly be a separate RFC suggesting how clients and  
> server
> can make use of DNS SRV records for HA and LB.

Okay, a thought experiment for you.

Let's assume we go ahead with dumb A/AAAA lookups.

How will it be possible for a server to later inform browser it  
wishes to use SRV lookups?

If you have an answer for this, we can make it optional, and moreover  
we can deploy SRV for HTTP as well.

Dave.
-- 
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On Thu, Jul 21, 2011 at 9:57 PM, David Endicott <dendicott@gmail.com> wrote=
:
>> I have no idea what you might mean by "highly dynamic host environment" =
in
>> this context, but XMPP servers are normally found at the same location
>> consistently. However, it is *not* always (or typically) the same locati=
on
>> as a simple A record lookup:
>
> That's what I meant. =A0XMPP systems have hosts that change around (for m=
any
> reasons) and having a name resolution that handles that is good.
>
>>
>> This property alone is very useful - in a websockets case this would mea=
n
>> being able to provide websockets services from a different host (or netw=
ork)
>> to the traditional web services in a simple manner, fully compatible wit=
h
>> SOP. =A0=A0The fact that this also allows cheap lightweight load balanci=
ng and
>> fallback control is also useful in other cases; none of this relates to
>> dynamic hosts, but simply richer service location.
>
> Yes, those are all excellent reasons to use DNS SRV. =A0 None of them are=
 a
> reason to mandate that WS require it. =A0 Because something is good for s=
ome
> (or many) use cases, does not mean it is appropriate for everything and
> certainly is not a reason to mandate it as a requirement.
> =A0System=A0implementer=A0should be free to pick and choose tools and mec=
hanisms
> appropriate for their tasks. =A0 DNS SRV would likely be an excellent cho=
ice
> for many people. =A0 But it should not be the one and only choice. =A0 Th=
at's
> really all I'm saying - don't force people to use something without an
> overwhelming reason to make it the only option.
>
>>>
>>> I do not oppose DNS's MX records for SMTP as email addresses are
>>> name@domain,
>>> so obviously the Domain Name System is appropriate. =A0 =A0Also, I fail=
 to
>>> see
>>> why a mail admin should care how the SMTP clients arrive at the server.
>>> =A0DNS
>>> MX is a reasonable solution, but there may be other methods, any and al=
l
>>> of
>>> which are irrelevant to the SMTP server. =A0 =A0Especially when the SMT=
P
>>> server
>>> supports multiple email domains...
>>>
>> A mail admin does need to care *that* the service is located, and
>> therefore will care *how* it is located.
>>
>> You can be as theoretical as you like, by all means, but in practical
>> terms, your email address (and my XMPP address) work because they use a
>> defined, interoperable, service location mechanism, which operates via D=
NS
>> record lookup.
>>
>> (Also, I have no idea what multiple domains has to do with this.)
>
> Imagine I'm a SMTP server. =A0 People connect to me. =A0 They do SMTP
> transactions. =A0 =A0I do not care how they found me. =A0 Perhaps they us=
ed DNS to
> find the MX server. =A0Perhaps they had it cached from before. =A0Perhaps=
 they
> guessed. =A0Perhaps it's in a hosts file. =A0 I don't care. =A0 =A0 I ans=
wer VRFY
> and RCPT TO commands as appropriate. =A0 If the "name" they are trying to
> mailwith is one I recognize, I process it. =A0If I don't, it's an error.
> Just because DNS-MX said that @foobar was handled at <addr>, doesn't mean
> the dave@foobar is going to work.
> Yes, DNS MX is a well known mechanism for determining what SMTP server to
> connect with, but like I tried to say above, it's not mandated by the SMT=
P
> protocol. =A0 DNS MX is independent of SMTP and the two mechanisms
> operate=A0separately, but with a common goal. =A0I can use DNS to resolve=
 a name
> and never send email/message. =A0I can send a email/message via SMTP and =
never
> use DNS to resolve a name. =A0 =A0Or I can use one to do the other.
> When a SMTP server handles mail for multiple domains, the SMTP server has=
 to
> process the @domain part of the RCPT TO request - DNS is not involved at
> that point. =A0 This process is unrelated to any DNS MX definitions. =A0 =
=A0I used
> that as an example of how some name resolutions are sometimes done outsid=
e
> of any DNS framework.
>
>>>
>>> Since WS is intended as a browser supported protocol, WS should follow
>>> the
>>> same URI resolution mechanisms as HTTP (or how URI resolution is done i=
n
>>> general) =A0Having URLs that could resolve differently for a HTTP reque=
st
>>> and
>>> a WS setup is a problem.
>>>
>> But they do resolve differently anyway. You don't get a page from a 'ws'
>> scheme URI, you get a transport protocol connection. This is good, indee=
d,
>> it's kind of the point.
>
>
> Do they? =A0 A http uri and a ws uri have the same host/path construction=
.
> =A0It's really only the scheme that differs - and that identifies the
> transport protocol to be used. =A0 Resolution of host name/addresses and
> mapping of paths "should" be consistent.
> WS is a connection that is semantically related to the URI of the request=
.
>
> e.g. I could ws://host/davesaid =A0and get live traffic of what Dave is
> saying, and then I could ws://host/bobsaid =A0and get traffic of what Bob
> says. =A0I wouldn't get Bob on /davesaid and I wouldn't get Dave on /bobs=
aid.
> =A0 =A0Dynamic content identified by a URI
> And if I http://host/davesaid =A0I could get a <li> of what Dave said.
> Static content of a URI.
> It could be problematic if =A0ws://host/davesaid resolves to a different
> address than http://host/davesaid. =A0 =A0 (Or it could be advantage - no=
t for
> us to decide, however)
>>
>> Your suggestion of "how URI resolution is done in general" is somewhat
>> self-defeating, too, since aside from 'http' and 'https', there are
>> 'mailto', which uses MX, 'sip' and 'xmpp', which both use SRV.
>
> As you just said, the universe is bigger than just xmpp, sip, and http.
>>
>> I think opponents of SRV records need to mount a stronger argument than
>> the kind of luddite argument that if it's hard for one protocol in use b=
y
>> the browser, it should be hard for them all.
>
> I think you misinterpret my position. =A0And=A0I resent the luddite sligh=
t. =A0=A0I
> think DNS SRV is an awesome tool and would greatly benefit many
> implementations.
> My position is that it should not be a *requirement*. =A0 =A0 It should b=
e an
> optional mechanism that can be used if desired. =A0 Further, since WS is =
a
> bastard cousin to HTTP, they should share a similar name resolution
> mechanism.

I=F1aki's proposal only imposes a requirement on clients, not on
server-side WS deployments, since clients will fall back to A/AAAA if
no SRV records are available. The mechanism (on the client-side)
cannot be optional though, it would make SRV for WS mostly useless in
practice.

-Philippe.

>
>>
>> Dave.
>> --
>> Dave Cridland - mailto:dave@cridland.net - xmpp:dwd@dave.cridland.net
>> =A0- acap://acap.dave.cridland.net/byowner/user/dwd/bookmarks/
>> =A0- http://dave.cridland.net/
>> Infotrope Polymer - ACAP, IMAP, ESMTP, and Lemonade
>
>
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>
>

From bruce@callenish.com  Thu Jul 21 15:15:51 2011
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Cc: Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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I think there is some misunderstanding as to what is being proposed, at 
least as I understand it. Iñaki, please correct me if I am wrong.

You are right that it would be impossible to require all environments 
that wanted to add Websockets support, whether client or server, to 
change their DNS to have NAPTR and SRV records. After all, Websockets is 
intended to integrate easily into the already existing HTTP infrastructure.

What is being proposed, though, is to require clients to resolve the 
hostname portion of ws: and wss: URLs by _first_ looking for NAPTR and 
SRV records (unless, of course, the hostname is already an IP address). 
If a NAPTR record is found, use it to look up an SRV record (otherwise 
use a default). If an SRV record is found, use it to look up the A or 
AAAA record. If no SRV record is found, look up the record exactly the 
same as if you were looking up an HTTP host, by using the host name from 
the URL.

So if you have no control over the DNS, it is not a problem. The host 
will be resolved exactly the same way as it is now, using a hosts file 
or A record or whatever. The only change is that the client is required 
to try to use the more advanced mechanism if it is available.

I admit that I find it a little troubling to use MUST for the client to 
follow this procedure as there is a burden on implementers to understand 
how to code this since it isn't done by default in the standard 
libraries the way that ordinary name resolution is. Making it the 
recognized best practice with a SHOULD would be preferable all else 
being equal.

It can be argued that not using a MUST may well open up interoperability 
problems because some Websockets clients contact the wrong host. 
However, keep in mind that in the older SIP RFC2543 it provided two 
resolution mechanisms. It specified that clients SHOULD look up address 
records, but MAY use the DNS SRV mechanism. SIP survived that without 
too much of a hassle. And specifying that Websockets clients SHOULD use 
DNS SRV, but MAY use address records alone looks like an improvement.

As for where the enhanced location definition should live, I think a 
separate document is the best choice. No need to hold up this one.

On 21/07/2011 11:01 AM, David Endicott wrote:
> I am strongly opposed to any MUST definition for any type of URL 
> resolution.
>
> I'm ok with inheriting / mimicking HTTP.    Since it is intended to 
> live in the same universe as HTTP, I'm ok with it sharing mechanisms / 
> limitations.
>
>
>
> On Thu, Jul 21, 2011 at 1:52 PM, Iñaki Baz Castillo <ibc@aliax.net 
> <mailto:ibc@aliax.net>> wrote:
>
>     2011/7/21 Dave Cridland <dave@cridland.net
>     <mailto:dave@cridland.net>>:
>     > It's proven impossible, despite effort, to retrofit SRV onto
>     HTTP; there is
>     > no way it'll be possible to retrofit onto WS.
>
>     Right. If WS borns with no SRV (as a MUST for WS clients) then just
>     forget it and let inherit all the ugly limitations from HTTP protocol.
>
>     --
>     Iñaki Baz Castillo
>     <ibc@aliax.net <mailto:ibc@aliax.net>>
>
>
>
>
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi


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<html>
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      http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    I think there is some misunderstanding as to what is being proposed,
    at least as I understand it. I&ntilde;aki, please correct me if I am wrong.<br>
    <br>
    You are right that it would be impossible to require all
    environments that wanted to add Websockets support, whether client
    or server, to change their DNS to have NAPTR and SRV records. After
    all, Websockets is intended to integrate easily into the already
    existing HTTP infrastructure.<br>
    <br>
    What is being proposed, though, is to require clients to resolve the
    hostname portion of ws: and wss: URLs by _first_ looking for NAPTR
    and SRV records (unless, of course, the hostname is already an IP
    address). If a NAPTR record is found, use it to look up an SRV
    record (otherwise use a default). If an SRV record is found, use it
    to look up the A or AAAA record. If no SRV record is found, look up
    the record exactly the same as if you were looking up an HTTP host,
    by using the host name from the URL.<br>
    <br>
    So if you have no control over the DNS, it is not a problem. The
    host will be resolved exactly the same way as it is now, using a
    hosts file or A record or whatever. The only change is that the
    client is required to try to use the more advanced mechanism if it
    is available.<br>
    <br>
    I admit that I find it a little troubling to use MUST for the client
    to follow this procedure as there is a burden on implementers to
    understand how to code this since it isn't done by default in the
    standard libraries the way that ordinary name resolution is. Making
    it the recognized best practice with a SHOULD would be preferable
    all else being equal.<br>
    <br>
    It can be argued that not using a MUST may well open up
    interoperability problems because some Websockets clients contact
    the wrong host. However, keep in mind that in the older SIP RFC2543
    it provided two resolution mechanisms. It specified that clients
    SHOULD look up address records, but MAY use the DNS SRV mechanism.
    SIP survived that without too much of a hassle. And specifying that
    Websockets clients SHOULD use DNS SRV, but MAY use address records
    alone looks like an improvement.<br>
    <br>
    As for where the enhanced location definition should live, I think a
    separate document is the best choice. No need to hold up this one.<br>
    <br>
    On 21/07/2011 11:01 AM, David Endicott wrote:
    <blockquote
cite="mid:CAP992=FJymFPKcPVWrF-LkcEtNUz=Kt9L_ex+kLtjiGjL1T46w@mail.gmail.com"
      type="cite">I am strongly opposed to any MUST definition for any
      type of URL resolution. &nbsp;&nbsp;
      <div><br>
      </div>
      <div>I'm ok with inheriting /&nbsp;mimicking&nbsp;HTTP. &nbsp; &nbsp;Since it is
        intended to live in the same universe as HTTP, I'm ok with it
        sharing mechanisms / limitations.</div>
      <div><br>
      </div>
      <div><br>
      </div>
      <div><br>
        <div class="gmail_quote">On Thu, Jul 21, 2011 at 1:52 PM, I&ntilde;aki
          Baz Castillo <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex;">
            2011/7/21 Dave Cridland &lt;<a moz-do-not-send="true"
              href="mailto:dave@cridland.net">dave@cridland.net</a>&gt;:<br>
            <div class="im">&gt; It's proven impossible, despite effort,
              to retrofit SRV onto HTTP; there is<br>
              &gt; no way it'll be possible to retrofit onto WS.<br>
              <br>
            </div>
            Right. If WS borns with no SRV (as a MUST for WS clients)
            then just<br>
            forget it and let inherit all the ugly limitations from HTTP
            protocol.<br>
            <div>
              <div class="h5"><br>
                --<br>
                I&ntilde;aki Baz Castillo<br>
                &lt;<a moz-do-not-send="true"
                  href="mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
              </div>
            </div>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
hybi mailing list
<a class="moz-txt-link-abbreviated" href="mailto:hybi@ietf.org">hybi@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/hybi">https://www.ietf.org/mailman/listinfo/hybi</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

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Date: Thu, 21 Jul 2011 23:21:07 +0100
From: Dave Cridland <dave@cridland.net>
To: Bruce Atherton <bruce@callenish.com>, Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>, David Endicott <dendicott@gmail.com>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Thu Jul 21 23:15:59 2011, Bruce Atherton wrote:
> So if you have no control over the DNS, it is not a problem. The  
> host will be resolved exactly the same way as it is now, using a  
> hosts file or A record or whatever. The only change is that the  
> client is required to try to use the more advanced mechanism if it  
> is available.
> 
> 
Right.


> I admit that I find it a little troubling to use MUST for the  
> client to follow this procedure as there is a burden on  
> implementers to understand how to code this since it isn't done by  
> default in the standard libraries the way that ordinary name  
> resolution is. Making it the recognized best practice with a SHOULD  
> would be preferable all else being equal.
> 
> 
SRV lookup is pretty commonplace now in libraries. XMPP and SIP  
clients have no difficulty finding this functionality in a wide  
variety of environments. For the web, where there are substantially  
fewer web browsers than there are XMPP clients, I don't think this  
would pose any kind of problem.


> It can be argued that not using a MUST may well open up  
> interoperability problems because some Websockets clients contact  
> the wrong host. However, keep in mind that in the older SIP RFC2543  
> it provided two resolution mechanisms. It specified that clients  
> SHOULD look up address records, but MAY use the DNS SRV mechanism.  
> SIP survived that without too much of a hassle. And specifying that  
> Websockets clients SHOULD use DNS SRV, but MAY use address records  
> alone looks like an improvement.
> 
> 
SIP survived because it was very small. I don't see WS making a  
transition, in the same way that repeated attempts have failed to  
move HTTP to SRV.

Dave.
-- 
Dave Cridland - mailto:dave@cridland.net - xmpp:dwd@dave.cridland.net
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From w@1wt.eu  Thu Jul 21 15:30:01 2011
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Date: Fri, 22 Jul 2011 00:29:52 +0200
From: Willy Tarreau <w@1wt.eu>
To: Dave Cridland <dave@cridland.net>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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Hi Dave,

On Thu, Jul 21, 2011 at 10:52:07PM +0100, Dave Cridland wrote:
> On Thu Jul 21 18:33:38 2011, Willy Tarreau wrote:
> >If someone were to develop a backup/restore solution based on WS,  
> >it would
> >be funny to discover that it cannot be used to restore the DNS  
> >server when
> >this one fails...
> >
> >
> If SRV (with a fallback) is defined as part of the core spec, this  
> usage would use the fallback.

Which would be indicated by which component since the DNS is down ?

> >However in my opinion it could be good to document a recommended DNS
> >architecture for WS, that is judged optimal and the most  
> >interoperable.
> >This could clearly be a separate RFC suggesting how clients and  
> >server
> >can make use of DNS SRV records for HA and LB.
> 
> Okay, a thought experiment for you.
> 
> Let's assume we go ahead with dumb A/AAAA lookups.
> 
> How will it be possible for a server to later inform browser it  
> wishes to use SRV lookups?

The same way it has been stated that browsers should use AAAA records
before A records when they exist. At one point it will be possible to
say "if an SRV record exists, then the browser should preferably use
it".

My point is not that DNS is not useful, just that in *some* cases, it
is all but desirable. Have you ever wondered why every cisco router
on the planet has "no ip-domain-lookups" in its config ? Basically
because core infrastructure component must be able to run with very
low dependencies, and DNS certainly is one dependency that is not
desirable for a number of core infrastructure components. And there
is no reason why those components could not be WS clients.

> If you have an answer for this, we can make it optional, and moreover  
> we can deploy SRV for HTTP as well.

It's not that easy either for HTTP. As I said in an earlier mail, HTTP
(as well as WS) has something special called proxies, which exempt
clients from having to deal with DNS. The simple fact that the server
does not know if the client will be able to use the DNS or rely on a
chain of intermediaries that may use it in various ways is a problem
for making use of DNS extensions mandatory.

You couldn't make DNS SRV mandatory for WS when WS starts with a protocol
upgrade from HTTP which does not mandate use of DNS SRV : the DNS resolving
had to be performed by the HTTP chain well before the connection gets
upgraded to WS. If any existing HTTP component in the chain does not make
use of DNS SRV, everything falls apart.

Experience has shown that a spec is not a way to force the world to change
to accomodate its requirements, rather it's a way to tell the world how to
adopt it at a reasonable cost and effort. The higher the expectations, the
lower the adoption. The most important thing is that the spec is designed
to support smooth upgrades so that real world usage drives it forwards.

Best regards,
Willy


From ibc@aliax.net  Thu Jul 21 16:19:43 2011
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Cc: Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/21 David Endicott <dendicott@gmail.com>:
> Yes, those are all excellent reasons to use DNS SRV. =C2=A0 None of them =
are a
> reason to mandate that WS require it. =C2=A0 Because something is good fo=
r some
> (or many) use cases, does not mean it is appropriate for everything and
> certainly is not a reason to mandate it as a requirement.

Hi David, I will be away for some days until I can read all the new
mails in this thread (I'll do it and will reply in a few days).
In the meanwhile I strongly ask you to read the draft about DNS SRV in
WebSocket:

  http://tools.ietf.org/html/draft-ibc-websocket-dns-srv-02

Sepecially some sections:


   1. Introducion

   [...]

   DNS SRV mechanism facilitates network applications scalability
   without requiring an intermediary node distributing the traffic in
   load-balancing or failover fashion.  Instead, DNS SRV mechanism just
   requires a proper DNS setup.

   By introducing DNS SRV records into WebSocket protocol
   [I-D.ietf-hybi-thewebsocketprotocol], WebSocket providers can,
   optionally, take same advantages and provide scalable services with a
   minimal infrastructure.

   This specification mandates the usage of DNS SRV resource records by
   WebSocket clients when resolving a "ws:" or "wss:" URI [RFC3986], but
   still leaves the decision of using SRV records up to the service
   administrator.


  3. Implementation

  [...]

   It is up to the system administrator whether to set, or not, DNS SRV
   resource records for the WebSocket protocol within the provided
   service.  This specification allows the system administrator to use
   the DNS SRV [RFC2782] mechanism to improve the service reliability by
   providing load-balancing and failover capabilities, but does not
   mandate it (the system administrator could choose whichever
   scalability strategy).



Section 4 "Client Usage" in which is shown how the WS client should
perform SRV resolution just in case the WS URI is a domain and no port
is present in the URI:

   To clarify it, a WebSocket URI like "ws://example.org/myservice"
   requires the client to perform SRV resolution while
   "ws://example.org:80/myservice" does not (as the port is explicitly
   present in the URI).


   step 3:  If there is no SRV result, attempt the fallback process describ=
ed
       in Section 4.2 and omit next steps.



   4.2.  Fallback Process

   The fallback process SHOULD be a normal A or AAAA address record
   resolution to determine the IPv4 or IPv6 address of the URI host
   component (or URI host value without DNS resolution if it contains an
   IP address).

   The server connection port is obtained as stated in Section 3.1 of
   [I-D.ietf-hybi-thewebsocketprotocol].

NOTE: The section 3.1 has changed in the new WS draft. It pointed to
http://tools.ietf.org/html/draft-ietf-hybi-thewebsocketprotocol-06#section-=
3.1
(3.1.  Parsing WebSocket URIs)


Also, the section 5 (Examples) should clarify it even more.


Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From ibc@aliax.net  Thu Jul 21 16:22:17 2011
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Cc: Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/22 Bruce Atherton <bruce@callenish.com>:
> You are right that it would be impossible to require all environments tha=
t
> wanted to add Websockets support, whether client or server, to change the=
ir
> DNS to have NAPTR and SRV records. After all, Websockets is intended to
> integrate easily into the already existing HTTP infrastructure.
>
> What is being proposed, though, is to require clients to resolve the
> hostname portion of ws: and wss: URLs by _first_ looking for NAPTR and SR=
V
> records (unless, of course, the hostname is already an IP address). If a
> NAPTR record is found, use it to look up an SRV record (otherwise use a
> default). If an SRV record is found, use it to look up the A or AAAA reco=
rd.
> If no SRV record is found, look up the record exactly the same as if you
> were looking up an HTTP host, by using the host name from the URL.

Well, in SIP there are NAPTR records because SIP can work over
different transports (UDP, TCP, TLS-TCP. SCTP, TLS-SCTP). In case of
WebSocket, it just defined for TCP so NAPTR records don't make sense.

So just SRV is required:

a)  _ws._tcp.DOMAIN.COM    for WS URI
b)  _wss._tcp.DOMAIN.COM   for WSS URI


Regards.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/21 David Endicott <dendicott@gmail.com>:
> Do they? =C2=A0 A http uri and a ws uri have the same host/path construct=
ion.
> =C2=A0It's really only the scheme that differs - and that identifies the
> transport protocol to be used. =C2=A0 Resolution of host name/addresses a=
nd
> mapping of paths "should" be consistent.
> WS is a connection that is semantically related to the URI of the request=
.
>
> e.g. I could ws://host/davesaid =C2=A0and get live traffic of what Dave i=
s
> saying, and then I could ws://host/bobsaid =C2=A0and get traffic of what =
Bob
> says. =C2=A0I wouldn't get Bob on /davesaid and I wouldn't get Dave on /b=
obsaid.
> =C2=A0 =C2=A0Dynamic content identified by a URI
> And if I http://host/davesaid =C2=A0I could get a <li> of what Dave said.
> Static content of a URI.
> It could be problematic if =C2=A0ws://host/davesaid resolves to a differe=
nt
> address than http://host/davesaid. =C2=A0 =C2=A0 (Or it could be advantag=
e - not for
> us to decide, however)

David, this does not make sense at all. Let see this case:

a) mailto:alice@google.com
b) xmpp:alice@google.com
c) sip:alice@google,com
d) http://google.com
e) ws://google.com

Do you really expect that all those URI's should point to the same
server??? not, right? then, why e) should behave like d)?

And of course, protocols defining a kind of URI (specific for such
protocol) CAN and probably MUST also define how to locate such URI
destination. In case of http just poor A/AAAA is done, but in other
cases we all do know that other kind of DNS queries are done.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From ibc@aliax.net  Thu Jul 21 16:31:34 2011
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/22 Willy Tarreau <w@1wt.eu>:
> You couldn't make DNS SRV mandatory for WS when WS starts with a protocol
> upgrade from HTTP which does not mandate use of DNS SRV : the DNS resolvi=
ng
> had to be performed by the HTTP chain well before the connection gets
> upgraded to WS. If any existing HTTP component in the chain does not make
> use of DNS SRV, everything falls apart.

I understand that the websocket client is *co-located* within the
webbrowser client, but it's a different entity. Nothing prevents it in
using a different DNS mechanism.

Honestly I'm not reading good arguments against DNS SRV for WebSocket
protocol. Perhaps the topic about the HTTP proxies, but there should
be some way to deal with it without forcing WebSocket to inherit all
the limitations and poor capabilities of HTTP (when referring to
scalability and failover).

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From bruce@callenish.com  Thu Jul 21 16:47:30 2011
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Cc: Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On 21/07/2011 3:21 PM, Dave Cridland wrote:
>
> SRV lookup is pretty commonplace now in libraries. XMPP and SIP 
> clients have no difficulty finding this functionality in a wide 
> variety of environments. For the web, where there are substantially 
> fewer web browsers than there are XMPP clients, I don't think this 
> would pose any kind of problem.

Oh, it isn't hard, but it violates the principle of least surprise. Most 
people think they know how to code name resolution by using 
gethostbyname() or equivalent. And to do it efficiently, especially when 
we are operating in a world that is predominantly driven by HTTP 
servers, requires complexities like dealing with asynchronous calls to 
fetch the A and SRV records at the same time under the assumption that 
the SRV record probably won't exist.

>
>> It can be argued that not using a MUST may well open up 
>> interoperability problems because some Websockets clients contact the 
>> wrong host. However, keep in mind that in the older SIP RFC2543 it 
>> provided two resolution mechanisms. It specified that clients SHOULD 
>> look up address records, but MAY use the DNS SRV mechanism. SIP 
>> survived that without too much of a hassle. And specifying that 
>> Websockets clients SHOULD use DNS SRV, but MAY use address records 
>> alone looks like an improvement.
>>
>>
> SIP survived because it was very small. I don't see WS making a 
> transition, in the same way that repeated attempts have failed to move 
> HTTP to SRV.
>
> Dave.

As I understand it, the issue that caused the various drafts for HTTP 
SRV to die without getting much traction is one of efficiency. It is 
inefficient to make all these extra DNS calls, especially when it is 
unlikely that many of the records you are blocking on will exist. Other 
than the inefficiency, HTTP SRV is just an incremental technology you 
add to your existing DNS without hurting what already exists. Since 
Websockets uses the same infrastructure the records are unlikely to 
exist for it either, so the inefficiency issues are still present.

Hmm, I seem to be arguing myself out of supporting DNS SRV as the 
preferred location mechanism. That is not the case. I just think that we 
face some of the same challenges migrating Websockets to SRV as are 
faced by HTTP because we share the same environment for the most part. 
Willy's example of a proxy that doesn't know how to resolve host names 
using the SRV method is a good example. But by advocating for the 
deployment of SRV records as a best practice for Websockets, we may 
improve things in the HTTP world as well. It is a shame there is no 
standard defined for HTTP SRV to point to.


From Gabriel.Montenegro@microsoft.com  Thu Jul 21 17:14:28 2011
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From: Gabriel Montenegro <Gabriel.Montenegro@microsoft.com>
To: Bruce Atherton <bruce@callenish.com>, Dave Cridland <dave@cridland.net>
Thread-Topic: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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[As an individual]

I think the lack of DNS SRV for HTTP could also be because of the guidance =
in http://wiki.tools.ietf.org/html/draft-iab-dns-applications-02#section-4 =
that argues that DNS might not be the best approach for *all* HTTP applicat=
ions. Perhaps also for WS applications...

It could be a good alternative to have, so I agree with pursuing it as a se=
parate option in a different document.
=09
> -----Original Message-----
> From: hybi-bounces@ietf.org [mailto:hybi-bounces@ietf.org] On Behalf Of
> Bruce Atherton
> Sent: Thursday, July 21, 2011 16:48
> To: Dave Cridland
> Cc: Server-Initiated HTTP; IETF-Discussion
> Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.t=
xt> (The
> WebSocket protocol) to Proposed Standard
>=20
> On 21/07/2011 3:21 PM, Dave Cridland wrote:
> >
> > SRV lookup is pretty commonplace now in libraries. XMPP and SIP
> > clients have no difficulty finding this functionality in a wide
> > variety of environments. For the web, where there are substantially
> > fewer web browsers than there are XMPP clients, I don't think this
> > would pose any kind of problem.
>=20
> Oh, it isn't hard, but it violates the principle of least surprise. Most =
people think
> they know how to code name resolution by using
> gethostbyname() or equivalent. And to do it efficiently, especially when =
we are
> operating in a world that is predominantly driven by HTTP servers, requir=
es
> complexities like dealing with asynchronous calls to fetch the A and SRV =
records
> at the same time under the assumption that the SRV record probably won't =
exist.
>=20
> >
> >> It can be argued that not using a MUST may well open up
> >> interoperability problems because some Websockets clients contact the
> >> wrong host. However, keep in mind that in the older SIP RFC2543 it
> >> provided two resolution mechanisms. It specified that clients SHOULD
> >> look up address records, but MAY use the DNS SRV mechanism. SIP
> >> survived that without too much of a hassle. And specifying that
> >> Websockets clients SHOULD use DNS SRV, but MAY use address records
> >> alone looks like an improvement.
> >>
> >>
> > SIP survived because it was very small. I don't see WS making a
> > transition, in the same way that repeated attempts have failed to move
> > HTTP to SRV.
> >
> > Dave.
>=20
> As I understand it, the issue that caused the various drafts for HTTP SRV=
 to die
> without getting much traction is one of efficiency. It is inefficient to =
make all
> these extra DNS calls, especially when it is unlikely that many of the re=
cords you
> are blocking on will exist. Other than the inefficiency, HTTP SRV is just=
 an
> incremental technology you add to your existing DNS without hurting what
> already exists. Since Websockets uses the same infrastructure the records=
 are
> unlikely to exist for it either, so the inefficiency issues are still pre=
sent.
>=20
> Hmm, I seem to be arguing myself out of supporting DNS SRV as the preferr=
ed
> location mechanism. That is not the case. I just think that we face some =
of the
> same challenges migrating Websockets to SRV as are faced by HTTP because =
we
> share the same environment for the most part.
> Willy's example of a proxy that doesn't know how to resolve host names us=
ing
> the SRV method is a good example. But by advocating for the deployment of
> SRV records as a best practice for Websockets, we may improve things in t=
he
> HTTP world as well. It is a shame there is no standard defined for HTTP S=
RV to
> point to.
>=20
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi


From dendicott@gmail.com  Thu Jul 21 19:24:44 2011
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--000e0ce0b4f00f3d8404a89f2b39
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On Thu, Jul 21, 2011 at 7:47 PM, Bruce Atherton <bruce@callenish.com> wrote:

>
> As I understand it, the issue that caused the various drafts for HTTP SRV
> to die without getting much traction is one of efficiency. It is inefficient
> to make all these extra DNS calls, especially when it is unlikely that many
> of the records you are blocking on will exist. Other than the inefficiency,
> HTTP SRV is just an incremental technology you add to your existing DNS
> without hurting what already exists. Since Websockets uses the same
> infrastructure the records are unlikely to exist for it either, so the
> inefficiency issues are still present.
>
> Hmm, I seem to be arguing myself out of supporting DNS SRV as the preferred
> location mechanism. That is not the case. I just think that we face some of
> the same challenges migrating Websockets to SRV as are faced by HTTP because
> we share the same environment for the most part. Willy's example of a proxy
> that doesn't know how to resolve host names using the SRV method is a good
> example. But by advocating for the deployment of SRV records as a best
> practice for Websockets, we may improve things in the HTTP world as well. It
> is a shame there is no standard defined for HTTP SRV to point to.
>

I find myself reminded of my reservations about HTTP Upgrade as the opening
handshake.  It is clever, efficient and reflects some of the shared nature
between HTTP and WS.   However, I felt it should be considered one of
several mechanisms for opening a WS connection, one especially suited for
a co-mingled environment.   But not explicitly the only such method.  (I was
unable to convince many of my position on that, so I could very well be in
the minority about this issue as well)    I think DNS SRV is in a similar
area.   It's a useful technology that if the client uses could be of
benefit.   I'm just not convinced there is overwhelming cause to make it a
mandatory requirement.    Saying that nobody will use it if it's not
legislated does not strike me as a good enough reason.   The technological
advantages are worthy, when it's used, but when it doesn't come into play,
there are added inefficiencies.   Also the name resolution of the HTTP that
serves the Javascript that opens the WS should remain constant.   If WS
resolves the host/domain to a different address than the HTTP it was spawned
from, it becomes a method to bypass same-origin / CORS restrictions.

I favor a minimalist core with extensibility.    Name resolution happens
before the WS opening handshake, so I continue to see this as outside the
domain of the WS protocol.   I would prefer that name resolution be provided
by a selectable method.  That way, in 20 years, when name resolution needs
have again changed, we'll have the ability to adapt.

--000e0ce0b4f00f3d8404a89f2b39
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br><br><div class=3D"gmail_quote">On Thu, Jul 21, 2011 at 7:47 PM, Bruce A=
therton <span dir=3D"ltr">&lt;<a href=3D"mailto:bruce@callenish.com">bruce@=
callenish.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div class=3D"im"><br></div>
As I understand it, the issue that caused the various drafts for HTTP SRV t=
o die without getting much traction is one of efficiency. It is inefficient=
 to make all these extra DNS calls, especially when it is unlikely that man=
y of the records you are blocking on will exist. Other than the inefficienc=
y, HTTP SRV is just an incremental technology you add to your existing DNS =
without hurting what already exists. Since Websockets uses the same infrast=
ructure the records are unlikely to exist for it either, so the inefficienc=
y issues are still present.<br>

<br>
Hmm, I seem to be arguing myself out of supporting DNS SRV as the preferred=
 location mechanism. That is not the case. I just think that we face some o=
f the same challenges migrating Websockets to SRV as are faced by HTTP beca=
use we share the same environment for the most part. Willy&#39;s example of=
 a proxy that doesn&#39;t know how to resolve host names using the SRV meth=
od is a good example. But by advocating for the deployment of SRV records a=
s a best practice for Websockets, we may improve things in the HTTP world a=
s well. It is a shame there is no standard defined for HTTP SRV to point to=
.<br>
</blockquote><div><br></div><div>I find myself reminded of my reservations =
about HTTP Upgrade as the opening handshake. =A0It is clever, efficient and=
 reflects some of the shared nature between HTTP and WS. =A0 However, I fel=
t it should be considered one of several mechanisms for opening a WS connec=
tion, one especially suited for a=A0co-mingled=A0environment. =A0 But=A0not=
 explicitly the only such method. =A0(I was unable to convince many of my p=
osition on that, so I could very well be in the minority about this issue a=
s well) =A0 =A0I think DNS SRV is in a similar area. =A0 It&#39;s a useful =
technology that if the client uses could be of benefit. =A0 I&#39;m just no=
t convinced there is overwhelming cause to make it a mandatory requirement.=
 =A0 =A0Saying that nobody will use it if it&#39;s not legislated does not =
strike me as a good enough reason. =A0 The technological advantages are wor=
thy, when it&#39;s used, but when it doesn&#39;t come into play, there are =
added inefficiencies. =A0=A0Also the name resolution of the HTTP that serve=
s the Javascript that opens the WS should remain constant. =A0 If WS resolv=
es the host/domain to a different address than the HTTP it was spawned from=
, it becomes a method to bypass same-origin / CORS restrictions. =A0 =A0</d=
iv>
<div><br></div><meta http-equiv=3D"content-type" content=3D"text/html; char=
set=3Dutf-8"><div>I favor a minimalist core with extensibility. =A0 =A0Name=
 resolution happens before the WS opening handshake, so I continue to see t=
his as outside the domain of the WS protocol. =A0 I would prefer that name =
resolution be provided by a selectable method. =A0That way, in 20 years, wh=
en name resolution needs have again changed, we&#39;ll have the ability to =
adapt.</div>
<div><br></div></div>

--000e0ce0b4f00f3d8404a89f2b39--

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From: John Tamplin <jat@google.com>
Date: Thu, 21 Jul 2011 22:45:17 -0400
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To: David Endicott <dendicott@gmail.com>
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Cc: Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Thu, Jul 21, 2011 at 10:24 PM, David Endicott <dendicott@gmail.com> wrot=
e:
> I find myself reminded of my reservations about HTTP Upgrade as the openi=
ng
> handshake. =C2=A0It is clever, efficient and reflects some of the shared =
nature
> between HTTP and WS. =C2=A0 However, I felt it should be considered one o=
f
> several mechanisms for opening a WS connection, one especially suited for
> a=C2=A0co-mingled=C2=A0environment. =C2=A0 But=C2=A0not explicitly the on=
ly such method. =C2=A0(I was
> unable to convince many of my position on that, so I could very well be i=
n
> the minority about this issue as well) =C2=A0 =C2=A0I think DNS SRV is in=
 a similar
> area. =C2=A0 It's a useful technology that if the client uses could be of
> benefit. =C2=A0 I'm just not convinced there is overwhelming cause to mak=
e it a
> mandatory requirement. =C2=A0 =C2=A0Saying that nobody will use it if it'=
s not
> legislated does not strike me as a good enough reason. =C2=A0 The technol=
ogical
> advantages are worthy, when it's used, but when it doesn't come into play=
,
> there are added inefficiencies. =C2=A0=C2=A0Also the name resolution of t=
he HTTP that
> serves the Javascript that opens the WS should remain constant. =C2=A0 If=
 WS
> resolves the host/domain to a different address than the HTTP it was spaw=
ned
> from, it becomes a method to bypass same-origin / CORS restrictions.
> I favor a minimalist core with extensibility. =C2=A0 =C2=A0Name resolutio=
n happens
> before the WS opening handshake, so I continue to see this as outside the
> domain of the WS protocol. =C2=A0 I would prefer that name resolution be =
provided
> by a selectable method. =C2=A0That way, in 20 years, when name resolution=
 needs
> have again changed, we'll have the ability to adapt.

How about some language like "SHOULD use SRV records to locate the
host unless specifically configured for an alternate name resolution
method"?  I think that leaves open the possibility for clients that
know better to do so, but strongly encourages most client
implementations to use SRV records.

--=20
John A. Tamplin
Software Engineer (GWT), Google

From gregw@intalio.com  Thu Jul 21 19:51:05 2011
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From: Greg Wilkins <gregw@intalio.com>
To: David Endicott <dendicott@gmail.com>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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I agree that DNS SRV is a matter outside the scope of  websockets,
which (for better or for worse) use a connection that is established
via a HTTP request.   Thus I do not think that the establishment of a
HTTP connection for websockets should differ in any way from the name
resolution handling of other HTTP connections made by a user-agent.
If the advocates of DNS SRV wish for websocket to use it, then they
need to convince HTTPbis (I think that is the right WG?) to adopt it
for all of HTTP and then websockets will inherit that feature.

If in future, there is a proposal to directly establish websocket
connections without a HTTP upgrade (as David Endicott has alluded to),
then I think DNS SRV should definitely be supported.

regards

From w@1wt.eu  Thu Jul 21 22:43:59 2011
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Date: Fri, 22 Jul 2011 07:43:45 +0200
From: Willy Tarreau <w@1wt.eu>
To: =?iso-8859-1?Q?I=F1aki?= Baz Castillo <ibc@aliax.net>
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Cc: Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Fri, Jul 22, 2011 at 01:22:15AM +0200, Iñaki Baz Castillo wrote:
> Well, in SIP there are NAPTR records because SIP can work over
> different transports (UDP, TCP, TLS-TCP. SCTP, TLS-SCTP). In case of
> WebSocket, it just defined for TCP so NAPTR records don't make sense.
> 
> So just SRV is required:
> 
> a)  _ws._tcp.DOMAIN.COM    for WS URI
> b)  _wss._tcp.DOMAIN.COM   for WSS URI

Iñaki, what we're saying is that the resolving applies first to HTTP
well before it is WS. For instance, a client could connect to an HTTP
server, fetch a few objects, then decide to upgrade the connection to
switch to WebSocket. DNS resolving for WS would not even be involved
here.

I agree with what others have been saying : if/when a different handshake
is supported, eg. on a specific port without the HTTP upgrade, then it
will make sense. But as of now we're relying on the lower layer. As Greg
said it, without a deep change in HTTP you won't be able to make the rule
a MUST for WS. However, John's suggest of using a SHOULD when the record
exists and the client can see it looks fine. What's the problem if not all
of your clients go to the same hosts ? You can even announce all of your
servers with A/AAAA and with SRV as well as long as they're running on the
same ports. Those who can use SRV just have more information than the other
ones and can be served better.

Regards,
Willy


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Date: Fri, 22 Jul 2011 10:55:19 +0100
From: Dave Cridland <dave@cridland.net>
To: David Endicott <dendicott@gmail.com>, Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>, Bruce Atherton <bruce@callenish.com>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Fri Jul 22 03:24:41 2011, David Endicott wrote:
> there are added inefficiencies.   Also the name resolution of the  
> HTTP that
> serves the Javascript that opens the WS should remain constant.    
> If WS
> resolves the host/domain to a different address than the HTTP it  
> was spawned
> from, it becomes a method to bypass same-origin / CORS restrictions.

That's an unfortunate misunderstanding.

All protocols that use SRV records maintain the target domain.

So a ws://example.com/xyz would still send a Host header of  
"example.com", whether SRV or not, so there is no impact on same  
origin policy, CORS, etc.

Dave.
-- 
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From henrikn@microsoft.com  Fri Jul 22 03:44:08 2011
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From: Henrik Frystyk Nielsen <henrikn@microsoft.com>
To: Greg Wilkins <gregw@intalio.com>, "Roy T. Fielding" <fielding@gbiv.com>
Thread-Topic: [hybi] Questions and comments on draft-hybi-thewebsocketprotocol-10
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Date: Fri, 22 Jul 2011 10:44:06 +0000
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Cc: Mark Nottingham <mnot@mnot.net>, "hybi@ietf.org HTTP" <hybi@ietf.org>
Subject: Re: [hybi] Questions and comments on draft-hybi-thewebsocketprotocol-10
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Greg,

There was a related thread on http-wg mailing list(see [1]).  Please see in=
 particular [2] and [3].

Hope this helps,

Henrik

[1] http://lists.w3.org/Archives/Public/ietf-http-wg/2011JulSep/thread.html=
#msg112
[2] http://lists.w3.org/Archives/Public/ietf-http-wg/2011JulSep/0158.html
[3] http://lists.w3.org/Archives/Public/ietf-http-wg/2011JulSep/0170.html

> Ok I think I'm getting you more now.   It is not that you want
> WebSockets to give an immediate semantic final response to the GET reques=
t,
> rather you don't want websocket to consider the 101 to be that final resp=
onse
> because it is by definition a non final response.
> So you say that this could be resolved by the spec making it clear that t=
he event
> stream accesses by the ws handshake is semantically the response to the
> handshake GET?


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From: David Endicott <dendicott@gmail.com>
To: Dave Cridland <dave@cridland.net>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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Good to know, thank you.

On Fri, Jul 22, 2011 at 5:55 AM, Dave Cridland <dave@cridland.net> wrote:

> On Fri Jul 22 03:24:41 2011, David Endicott wrote:
>
>> there are added inefficiencies.   Also the name resolution of the HTTP
>> that
>> serves the Javascript that opens the WS should remain constant.   If WS
>> resolves the host/domain to a different address than the HTTP it was
>> spawned
>> from, it becomes a method to bypass same-origin / CORS restrictions.
>>
>
> That's an unfortunate misunderstanding.
>
> All protocols that use SRV records maintain the target domain.
>
> So a ws://example.com/xyz would still send a Host header of "example.com",
> whether SRV or not, so there is no impact on same origin policy, CORS, etc.
>
>
> Dave.
> --
> Dave Cridland - mailto:dave@cridland.net - xmpp:dwd@dave.cridland.net
>  - acap://acap.dave.cridland.net/**byowner/user/dwd/bookmarks/<http://acap.dave.cridland.net/byowner/user/dwd/bookmarks/>
>  - http://dave.cridland.net/
> Infotrope Polymer - ACAP, IMAP, ESMTP, and Lemonade
>

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Good to know, thank you.<br><br><div class=3D"gmail_quote">On Fri, Jul 22, =
2011 at 5:55 AM, Dave Cridland <span dir=3D"ltr">&lt;<a href=3D"mailto:dave=
@cridland.net">dave@cridland.net</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex;">
<div class=3D"im">On Fri Jul 22 03:24:41 2011, David Endicott wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
there are added inefficiencies. =A0 Also the name resolution of the HTTP th=
at<br>
serves the Javascript that opens the WS should remain constant. =A0 If WS<b=
r>
resolves the host/domain to a different address than the HTTP it was spawne=
d<br>
from, it becomes a method to bypass same-origin / CORS restrictions.<br>
</blockquote>
<br></div>
That&#39;s an unfortunate misunderstanding.<br>
<br>
All protocols that use SRV records maintain the target domain.<br>
<br>
So a ws://<a href=3D"http://example.com/xyz" target=3D"_blank">example.com/=
xyz</a> would still send a Host header of &quot;<a href=3D"http://example.c=
om" target=3D"_blank">example.com</a>&quot;, whether SRV or not, so there i=
s no impact on same origin policy, CORS, etc.<div>
<div></div><div class=3D"h5"><br>
<br>
Dave.<br>
-- <br>
Dave Cridland - mailto:<a href=3D"mailto:dave@cridland.net" target=3D"_blan=
k">dave@cridland.net</a> - <a href=3D"mailto:xmpp%3Adwd@dave.cridland.net" =
target=3D"_blank">xmpp:dwd@dave.cridland.net</a><br>
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kmarks/</a><br>
=A0- <a href=3D"http://dave.cridland.net/" target=3D"_blank">http://dave.cr=
idland.net/</a><br>
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</div></div></blockquote></div><br>

--000e0cd342b6c245a504a8a899d0--

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From: David Endicott <dendicott@gmail.com>
To: Dave Cridland <dave@cridland.net>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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serves the Javascript that opens the WS should remain constant.   If WS

> resolves the host/domain to a different address than the HTTP it was
>>> spawned
>>> from, it becomes a method to bypass same-origin / CORS restrictions.
>>>
>>
>> That's an unfortunate misunderstanding.
>>
>> All protocols that use SRV records maintain the target domain.
>>
>> So a ws://example.com/xyz would still send a Host header of "example.com",
>> whether SRV or not, so there is no impact on same origin policy, CORS, etc.
>>
>>
> Good to know, thank you.

Actually....I wasn't talking about the Host: header - that is totally
spoofable...I was concerned about:

1. Browser client resolves example.com via old style DNS to x.x.x.x and
fetches HTTP
2. Received HTML starts JS which starts WS connection
3. WS resolves example.com via DNS SRV to y.y.y.y and opens
4. WS now has access outside origin.

Please note, I did not specify why DNS SRV resolved differently than old
style DNS - could be malicious, could be an simple mistake.     I am
assuming the DNS SRV and old DNS might be answered from different servers.

Do browsers restrict origin / cross-site access based on name or on address?

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<br>serves the Javascript that opens the WS should remain constant. =A0 If =
WS<br><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D=
"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div><div =
class=3D"h5">
<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex">

resolves the host/domain to a different address than the HTTP it was spawne=
d<br>
from, it becomes a method to bypass same-origin / CORS restrictions.<br>
</blockquote>
<br></div>
That&#39;s an unfortunate misunderstanding.<br>
<br>
All protocols that use SRV records maintain the target domain.<br>
<br>
So a ws://<a href=3D"http://example.com/xyz" target=3D"_blank">example.com/=
xyz</a> would still send a Host header of &quot;<a href=3D"http://example.c=
om" target=3D"_blank">example.com</a>&quot;, whether SRV or not, so there i=
s no impact on same origin policy, CORS, etc.<div>

<div></div><div><br></div></div></blockquote></div></div></div></blockquote=
><meta http-equiv=3D"content-type" content=3D"text/html; charset=3Dutf-8"><=
br class=3D"Apple-interchange-newline">&gt; Good to know, thank you.<div><d=
iv>
</div><div class=3D"h5"><br></div></div><div class=3D"h5">Actually....I was=
n&#39;t talking about the Host: header - that is totally spoofable...I was =
concerned about:</div><div class=3D"h5"><br></div><div class=3D"h5">1. Brow=
ser client resolves <a href=3D"http://example.com">example.com</a> via old =
style DNS to x.x.x.x and fetches HTTP</div>
<div class=3D"h5">2. Received HTML starts JS which starts WS connection</di=
v><div class=3D"h5">3. WS resolves <a href=3D"http://example.com">example.c=
om</a> via DNS SRV to y.y.y.y and opens</div><div class=3D"h5">4. WS now ha=
s access outside origin.</div>
<div class=3D"h5"><br></div><div class=3D"h5">Please note, I did not specif=
y why DNS SRV resolved differently than old style DNS - could be=A0maliciou=
s, could be an simple mistake. =A0 =A0 I am assuming the DNS SRV and old DN=
S might be answered from different servers.</div>
<div class=3D"h5"><br></div><div class=3D"h5">Do browsers restrict origin /=
 cross-site access based on name or on address? =A0=A0</div><div class=3D"h=
5"><br></div><div>=A0</div></div>

--0016364d2c791977f404a8a8b552--

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From: Takeshi Yoshino <tyoshino@google.com>
Date: Fri, 22 Jul 2011 23:38:06 +0900
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--0016e646076e12325004a8a96be5
Content-Type: text/plain; charset=ISO-8859-1

I've implemented this proposal on pywebsocket. pywebsocket's deflate-stream
implementation flushes compressed data for each frame using Z_SYNC_FLUSH.
Compression level is Z_DEFAULT_COMPRESSION. Default Huffman tree is used. No
dictionary is attached.

Here is the result of quick experiment. I exchanged some trivial text
messages with echo server and dumped occupation length on wire (strictly
speaking, occupation in TCP payload) for each frame. HyBi 07 is used.

Significant difference between deflate-stream and deflate-application-data
is
- deflate-stream cannot utilize history from previous frames' contents well
in C -> S direction due to different mask between frames

Non-significant one is
- output of deflate-stream are basically bigger than
deflate-application-data since Z_SYNC_FLUSH wastes 4 octets for each flush
operation

deflate-application-data compared to no compression
- for short messages, it just increases size because of ~2 octets for two
block headers and end of block symbol

In experiment #2, the size of deflate-stream output (C <- S) converges to 8.
This contains 5 octets and 5 bits (4 octet + 3 bit * 2 + 7 bit) for header
and Z_SYNC_FLUSH stuff. This means that the original 3 octet frame became 2
octets + 3 bits. This is the benefit of including WebSocket header to
compression target. If multiple similar frames are buffered and compressed
together at once, this benefit will be big, but I'm not sure if such case is
popular in practical use case.

----

#1: Send text frame 'a', 'b', 'c', 'd', 'e', then close

C -> S
plain:
7 7 7 7 7 8
deflate-stream:
13 13 13 13 13 15
deflate-application-data:
9 9 9 9 9 8

C <- S
plain:
3 3 3 3 3 4
deflate-stream:
9 9 9 9 9 10
deflate-application-data:
5 5 5 5 5 4

#2: Send text frame 'a', 'a', 'a', 'a', 'a', then close
C -> S
plain:
7 7 7 7 7 8
deflate-stream:
13 13 13 13 13 14
deflate-application-data:
9 9 9 9 9 8

C <- S
plain:
3 3 3 3 3 4
deflate-stream:
9 9 8 8 8 10
deflate-application-data:
5 5 5 5 5 4

#3: Send text frame 'HelloWorldWebSocketProtocol' 5 times, then close

C -> S
plain:
33 33 33 33 33 8
deflate-stream:
43 43 41 40 43 14
deflate-application-data:
35 11 10 10 10 8

C <- S
plain:
29 29 29 29 29 4
deflate-stream:
35 9 8 8 8 10
deflate-application-data:
31 7 6 6 6 4

#4: Send text frame '60b725f10c9c85c70d97880dfe8191b3'
,'3b5d5c3712955042212316173ccf37be' ,'2cd6ee2c70b0bde53fbe6cac3c8b8bb1',
'e29311f6f1bf1af907f9ef9f44b8328b', '9ffbf43126e33be52cd2bf7e01d627f9', then
close

C -> S
plain:
38 38 38 38 38 8
deflate-stream:
48 48 45 48 48 14
deflate-application-data:
40 40 39 40 38 8

C <- S
plain:
34 34 34 34 34 4
deflate-stream:
40 40 39 40 38 10
deflate-application-data:
36 36 35 36 34 4

#5: Send text frame 'aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa',
then close

C -> S
plain:
56 8
deflate-stream:
18 14
deflate-application-data:
12 8

C <- S
plain:
52 4
deflate-stream:
11 10
deflate-application-data:
8 4

--0016e646076e12325004a8a96be5
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I&#39;ve implemented this proposal on pywebsocket. pywebsocket&#39;s=A0defl=
ate-stream implementation flushes compressed data for each frame using Z_SY=
NC_FLUSH. Compression level is Z_DEFAULT_COMPRESSION. Default Huffman tree =
is used. No dictionary is attached.<div>

<br></div><div>Here is the result of quick experiment. I exchanged some tri=
vial text messages with echo server and dumped occupation length on wire (s=
trictly speaking, occupation in TCP payload) for each frame.=A0HyBi 07 is u=
sed.</div>


<div><br></div><div>Significant difference between deflate-stream and defla=
te-application-data is</div><div>- deflate-stream cannot utilize history fr=
om previous frames&#39; contents well in C -&gt; S direction due to differe=
nt mask between frames</div>

<div><br></div><div>Non-significant one is</div><div>- output of deflate-st=
ream are basically bigger than deflate-application-data since Z_SYNC_FLUSH =
wastes 4 octets for each flush operation</div><div><br></div><div>deflate-a=
pplication-data compared to no compression</div>

<div>- for short messages, it just increases size because of ~2 octets for =
two block headers and end of block symbol</div><div><br></div><div>In exper=
iment #2, the size of deflate-stream output (C &lt;- S) converges to 8. Thi=
s contains 5 octets and 5 bits (4 octet + 3 bit * 2 + 7 bit) for header and=
 Z_SYNC_FLUSH stuff. This means that the original 3 octet frame became 2 oc=
tets + 3 bits. This is the benefit of including WebSocket header to compres=
sion target.=A0If multiple similar frames are buffered and compressed toget=
her at once, this benefit will be big, but I&#39;m not sure if such case is=
 popular in practical use case.</div>

<div><br></div><div>----</div><div><br></div><div>#1: Send text frame &#39;=
a&#39;, &#39;b&#39;, &#39;c&#39;, &#39;d&#39;, &#39;e&#39;, then close</div=
>
<div><br></div><div>C -&gt; S</div><div>plain:</div><div>7 7 7 7 7 8</div><=
div>deflate-stream:</div><div>13 13 13 13 13 15</div><div>deflate-applicati=
on-data:</div><div>9 9 9 9 9 8</div><div><br></div><div>C &lt;- S</div>


<div>plain:</div><div>3 3 3 3 3 4</div><div>deflate-stream:</div><div>9 9 9=
 9 9 10</div><div>deflate-application-data:</div><div>5 5 5 5 5 4</div><div=
><br></div><div><div>#2: Send text frame &#39;a&#39;, &#39;a&#39;, &#39;a&#=
39;, &#39;a&#39;, &#39;a&#39;, then close</div>


</div><div><div>C -&gt; S</div><div>plain:</div><div>7 7 7 7 7 8</div><div>=
deflate-stream:</div><div>13 13 13 13 13 14</div><div>deflate-application-d=
ata:</div><div>9 9 9 9 9 8</div><div><br></div><div>C &lt;- S</div><div>


plain:</div><div>3 3 3 3 3 4</div><div>deflate-stream:</div><div>9 9 8 8 8 =
10</div><div>deflate-application-data:</div><div>5 5 5 5 5 4</div><div><br>=
</div></div><div>#3: Send text frame &#39;HelloWorldWebSocketProtocol&#39; =
5 times, then close</div>


<div><br></div><div>C -&gt; S</div><div>plain:</div><div>33 33 33 33 33 8</=
div><div>deflate-stream:</div><div><a href=3D"tel:43%2043%2041%2040%2043" v=
alue=3D"+14343414043" target=3D"_blank">43 43 41 40 43</a> 14</div><div>def=
late-application-data:</div>

<div>35 11 10 10 10 8</div><div><br></div><div><div>
C &lt;- S</div><div>plain:</div><div>29 29 29 29 29 4</div><div>deflate-str=
eam:</div><div>35 9 8 8 8 10</div><div>deflate-application-data:</div><div>=
31 7 6 6 6 4</div><div><br></div></div><div><div>#4: Send text frame &#39;6=
0b725f10c9c85c70d97880dfe8191b3&#39; ,&#39;3b5d5c3712955042212316173ccf37be=
&#39; ,&#39;2cd6ee2c70b0bde53fbe6cac3c8b8bb1&#39;, &#39;e29311f6f1bf1af907f=
9ef9f44b8328b&#39;, &#39;9ffbf43126e33be52cd2bf7e01d627f9&#39;, then close<=
/div>


</div><div><br></div><div><div>C -&gt; S</div><div>plain:</div><div>38 38 3=
8 38 38 8</div><div>deflate-stream:</div><div>48 <a href=3D"tel:48%2045%204=
8%2048%2014" value=3D"+14845484814" target=3D"_blank">48 45 48 48 14</a></d=
iv>

<div>deflate-application-data:</div><div>40 40 39 40 38 8</div><div><br></d=
iv>
<div><div>C &lt;- S</div><div>plain:</div><div><a href=3D"tel:34%2034%2034%=
2034%2034" value=3D"+13434343434" target=3D"_blank">34 34 34 34 34</a> 4</d=
iv><div>deflate-stream:</div><div>40 <a href=3D"tel:40%2039%2040%2038%2010"=
 value=3D"+14039403810" target=3D"_blank">40 39 40 38 10</a></div>

<div>deflate-application-data:</div><div>36 36 35 36 34 4</div></div></div>=
<div><br></div><div><div>
<div>#5: Send text frame &#39;aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa=
aaaaa&#39;, then close</div></div><div><br></div><div><div>C -&gt; S</div><=
div>plain:</div><div>56 8</div><div>deflate-stream:</div><div>18 14</div>


<div>deflate-application-data:</div><div>12 8</div><div><br></div><div><div=
>C &lt;- S</div><div>plain:</div><div>52 4</div><div>deflate-stream:</div><=
div>11 10</div><div>deflate-application-data:</div><div>8 4</div></div>

</div></div><div><br></div>

--0016e646076e12325004a8a96be5--

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From: Greg Wilkins <gregw@intalio.com>
To: Takeshi Yoshino <tyoshino@google.com>
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Cc: "hybi@ietf.org" <hybi@ietf.org>
Subject: Re: [hybi] Payload only compression extension, again
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On 23 July 2011 00:38, Takeshi Yoshino <tyoshino@google.com> wrote:
> I've implemented this proposal on pywebsocket. pywebsocket's=A0deflate-st=
ream
> implementation flushes compressed data for each frame using Z_SYNC_FLUSH.
> Compression level is Z_DEFAULT_COMPRESSION. Default Huffman tree is used.=
 No
> dictionary is attached.

Great - I'll make sure we have an implementation in  the next release of Je=
tty.

> - output of deflate-stream are basically bigger than
> deflate-application-data since Z_SYNC_FLUSH wastes 4 octets for each flus=
h
> operation

I think you mean deflate-application-data frames are bigger....

> - for short messages, it just increases size because of ~2 octets for two
> block headers and end of block symbol

Does this not suggest that we should use a reserve bit to indicate if
the frame is compressed or not?  This will allow uncompressed frames
(either because of small size or incompressible content) to be sent.


cheers

From fielding@gbiv.com  Fri Jul 22 15:44:15 2011
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Cc: Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Jul 21, 2011, at 10:52 AM, I=F1aki Baz Castillo wrote:

> 2011/7/21 Dave Cridland <dave@cridland.net>:
>> It's proven impossible, despite effort, to retrofit SRV onto HTTP; =
there is
>> no way it'll be possible to retrofit onto WS.
>=20
> Right. If WS borns with no SRV (as a MUST for WS clients) then just
> forget it and let inherit all the ugly limitations from HTTP protocol.

I am tired of this.  SRV is not used for HTTP because SRV adds latency
to the initial request for no useful purpose whatsoever.  SRV records =
for
XMPP and MX records for mail are useful because there is only one such
server expected per domain and it is *very* desirable to maintain =
central
control over that routing.  In contrast, HTTP is deployed in an anarchic
manner in which there are often several HTTP servers per machine
(e.g., tests, staging, production, CUPS, etc,).  AFAICT, WebSockets is
even more anarchic than HTTP -- it will have to be, given that the sane
network admins will block it by default.

In short, SRV is not used by the Web because it is inappropriate for =
HTTP.
I have seen no reason to believe that it would be appropriate for =
WebSockets.
If you want SRV to be part of the proposed standard, then you have to =
convince
the people implementing WS to use SRV.  None have done so, yet, so we =
can't
expect the editor to add it to the spec just because you have an =
opinion.

....Roy=

From ted.ietf@gmail.com  Sat Jul 23 11:41:14 2011
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From: Ted Hardie <ted.ietf@gmail.com>
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--20cf305b0a5e1999f804a8c0edb5
Content-Type: text/plain; charset=ISO-8859-1

On Fri, Jul 22, 2011 at 9:47 AM, David Endicott <dendicott@gmail.com> wrote:

>
> Actually....I wasn't talking about the Host: header - that is totally
> spoofable...I was concerned about:
>
> 1. Browser client resolves example.com via old style DNS to x.x.x.x and
> fetches HTTP
> 2. Received HTML starts JS which starts WS connection
> 3. WS resolves example.com via DNS SRV to y.y.y.y and opens
> 4. WS now has access outside origin.
>
> Please note, I did not specify why DNS SRV resolved differently than old
> style DNS - could be malicious, could be an simple mistake.     I am
> assuming the DNS SRV and old DNS might be answered from different servers.
>
>
You definitely could set it up such that the results from an SRV lookup
points to a different server than that resulting from a lookup of AAAA or A;
that's kind of the point.  The SRV lookup is to a service within the
original domain, but the resulting looking up could have results outside it.
 To go back to Dave Cridland's example, you can see that the result of the
SRV is another name requiring lookup.

;; ANSWER SECTION:
_xmpp-server._tcp.gmail.com. 26125 IN   SRV     5 0 5269
xmpp-server.l.google.com.
_xmpp-server._tcp.gmail.com. 26125 IN   SRV     20 0 5269
xmpp-server1.l.google.com.
_xmpp-server._tcp.gmail.com. 26125 IN   SRV     20 0 5269
xmpp-server2.l.google.com.
_xmpp-server._tcp.gmail.com. 26125 IN   SRV     20 0 5269
xmpp-server3.l.google.com.
_xmpp-server._tcp.gmail.com. 26125 IN   SRV     20 0 5269
xmpp-server4.l.google.com.

You'd have to avoid the results triggering the antibodies to a cross-site
scripting attack in order to deploy this well, in my opinion.

regards,

Ted



Do browsers restrict origin / cross-site access based on name or on address?
>
>
>
>
> _______________________________________________
> Ietf mailing list
> Ietf@ietf.org
> https://www.ietf.org/mailman/listinfo/ietf
>
>

--20cf305b0a5e1999f804a8c0edb5
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

On Fri, Jul 22, 2011 at 9:47 AM, David Endicott <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:dendicott@gmail.com">dendicott@gmail.com</a>&gt;</span> wrote:=
<br><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div class=3D"im"><br></div><div class=3D"gmail_quote"><div>Actually....I w=
asn&#39;t talking about the Host: header - that is totally spoofable...I wa=
s concerned about:</div><div><br></div><div>1. Browser client resolves <a h=
ref=3D"http://example.com" target=3D"_blank">example.com</a> via old style =
DNS to x.x.x.x and fetches HTTP</div>

<div>2. Received HTML starts JS which starts WS connection</div><div>3. WS =
resolves <a href=3D"http://example.com" target=3D"_blank">example.com</a> v=
ia DNS SRV to y.y.y.y and opens</div><div>4. WS now has access outside orig=
in.</div>

<div><br></div><div>Please note, I did not specify why DNS SRV resolved dif=
ferently than old style DNS - could be=A0malicious, could be an simple mist=
ake. =A0 =A0 I am assuming the DNS SRV and old DNS might be answered from d=
ifferent servers.</div>

<div><br></div></div></blockquote><div><br></div><div>You definitely could =
set it up such that the results from an SRV lookup points to a different se=
rver than that resulting from a lookup of AAAA or A; that&#39;s kind of the=
 point. =A0The SRV lookup is to a service within the original domain, but t=
he resulting looking up could have results outside it. =A0To go back to Dav=
e Cridland&#39;s example, you can see that the result of the SRV is another=
 name requiring lookup. =A0</div>
<div><font class=3D"Apple-style-span" color=3D"#222222" face=3D"arial, sans=
-serif" size=3D"3"><span class=3D"Apple-style-span" style=3D"border-collaps=
e: collapse; font-size: 11px;"><font class=3D"Apple-style-span" color=3D"#0=
00000" face=3D"arial"><span class=3D"Apple-style-span" style=3D"border-coll=
apse: separate; font-size: small;"><br>
</span></font></span></font></div><div><span class=3D"Apple-style-span" sty=
le=3D"font-family: arial, sans-serif; font-size: 10.8333px; border-collapse=
: collapse; color: rgb(34, 34, 34); ">;; ANSWER SECTION:<br>_xmpp-server._<=
a href=3D"http://tcp.gmail.com/" target=3D"_blank" style=3D"color: rgb(53, =
66, 88); ">tcp.gmail.com</a>. 26125 IN =A0 SRV =A0 =A0 5 0 5269=A0<a href=
=3D"http://xmpp-server.l.google.com/" target=3D"_blank" style=3D"color: rgb=
(53, 66, 88); ">xmpp-server.l.google.com</a>.<br>
_xmpp-server._<a href=3D"http://tcp.gmail.com/" target=3D"_blank" style=3D"=
color: rgb(53, 66, 88); ">tcp.gmail.com</a>. 26125 IN =A0 SRV =A0 =A0 20 0 =
5269=A0<a href=3D"http://xmpp-server1.l.google.com/" target=3D"_blank" styl=
e=3D"color: rgb(53, 66, 88); ">xmpp-server1.l.google.com</a>.<br>
_xmpp-server._<a href=3D"http://tcp.gmail.com/" target=3D"_blank" style=3D"=
color: rgb(53, 66, 88); ">tcp.gmail.com</a>. 26125 IN =A0 SRV =A0 =A0 20 0 =
5269=A0<a href=3D"http://xmpp-server2.l.google.com/" target=3D"_blank" styl=
e=3D"color: rgb(53, 66, 88); ">xmpp-server2.l.google.com</a>.<br>
_xmpp-server._<a href=3D"http://tcp.gmail.com/" target=3D"_blank" style=3D"=
color: rgb(53, 66, 88); ">tcp.gmail.com</a>. 26125 IN =A0 SRV =A0 =A0 20 0 =
5269=A0<a href=3D"http://xmpp-server3.l.google.com/" target=3D"_blank" styl=
e=3D"color: rgb(53, 66, 88); ">xmpp-server3.l.google.com</a>.<br>
_xmpp-server._<a href=3D"http://tcp.gmail.com/" target=3D"_blank" style=3D"=
color: rgb(53, 66, 88); ">tcp.gmail.com</a>. 26125 IN =A0 SRV =A0 =A0 20 0 =
5269=A0<a href=3D"http://xmpp-server4.l.google.com/" target=3D"_blank" styl=
e=3D"color: rgb(53, 66, 88); ">xmpp-server4.l.google.com</a>.<br>
</span></div><div>=A0</div><div>You&#39;d have to avoid the results trigger=
ing the antibodies to a cross-site scripting attack in order to deploy this=
 well, in my opinion.</div><div><br></div><div>regards,</div><div><br></div=
>
<div>Ted</div><div><br></div><div><br></div><div><br></div><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex;"><div class=3D"gmail_quote"><div></div><div>Do browsers rest=
rict origin / cross-site access based on name or on address? =A0=A0</div>
<div><br></div><div>=A0</div></div>
<br>_______________________________________________<br>
Ietf mailing list<br>
<a href=3D"mailto:Ietf@ietf.org">Ietf@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/ietf" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/ietf</a><br>
<br></blockquote></div><br>

--20cf305b0a5e1999f804a8c0edb5--

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In message <4E28A51F.4020704@callenish.com>, Bruce Atherton writes:
>
> I admit that I find it a little troubling to use MUST for the client to 
> follow this procedure as there is a burden on implementers to understand 
> how to code this since it isn't done by default in the standard 
> libraries the way that ordinary name resolution is. Making it the 
> recognized best practice with a SHOULD would be preferable all else 
> being equal.
 
No.  MUST is what is needed.  It's a new protocol.  Do what's best from
day one.

-- 
Mark Andrews, ISC
1 Seymour St., Dundas Valley, NSW 2117, Australia
PHONE: +61 2 9871 4742                 INTERNET: marka@isc.org

From moore@network-heretics.com  Sat Jul 23 16:25:07 2011
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Cc: Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Jul 23, 2011, at 5:23 PM, Mark Andrews wrote:

> In message <4E28A51F.4020704@callenish.com>, Bruce Atherton writes:
>>=20
>> I admit that I find it a little troubling to use MUST for the client =
to=20
>> follow this procedure as there is a burden on implementers to =
understand=20
>> how to code this since it isn't done by default in the standard=20
>> libraries the way that ordinary name resolution is. Making it the=20
>> recognized best practice with a SHOULD would be preferable all else=20=

>> being equal.
>=20
> No.  MUST is what is needed.  It's a new protocol.  Do what's best =
from
> day one.


Sort of agree.  If use of SRV for this protocol is really appropriate =
(which I doubt, but I haven't looked at it closely) then the protocol =
specification should say "MUST use SRV".
If use of SRV for this protocol is not appropriate, or if it's not clear =
that it's appropriate, then the specification should probably say "MUST =
NOT use SRV".=20

Either way, provide clear direction to implementors and don't leave the =
decision as to whether to use SRV up to the implementation.  That would =
create different behaviors in different implementations, which is =
clearly not desirable.

Keith


From dave@cridland.net  Sun Jul 24 02:35:54 2011
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From: Dave Cridland <dave@cridland.net>
To: Willy Tarreau <w@1wt.eu>, Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>, =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Fri Jul 22 06:43:45 2011, Willy Tarreau wrote:
> Iñaki, what we're saying is that the resolving applies first to HTTP
> well before it is WS. For instance, a client could connect to an  
> HTTP
> server, fetch a few objects, then decide to upgrade the connection  
> to
> switch to WebSocket. DNS resolving for WS would not even be involved
> here.
> 
> 
I get where you're coming from.

You're saying that you have a nebulous connection thing, that you  
pump HTTP requests down, and you may want to use that for WebSocket  
upgrade requests as well, by taking the ws URI and performing a  
simple transformation on it into an HTTP URI to make the upgrade  
request on.

So, given ws://example.com/foo/bar, you'll make the upgrade request  
on http://example.com/foo/bar

What Iñaki and I are arguing for is that for WS, there is simply one  
additional step prior to finding a suitable connection, which is  
performing an SRV lookup/selection.

And this new step can actually be performed before any of the HTTP  
connection steps. So, roughly, in order to "do" WebSocket for  
ws://example.com/foo/bar you would first do SRV lookup, which would  
give you an equivalent HTTP URI to perform the upgrade request. So  
for example, if you got:

_ws._tcp.example.com SRV 1 0 ws.example.com

You'd make the upgrade request on the HTTP URI of:

http://ws.example.com/foo/bar

Of course, if this failed to connect, you can find another HTTP URI  
to try - something you cannot do without SRV.

Dave.
-- 
Dave Cridland - mailto:dave@cridland.net - xmpp:dwd@dave.cridland.net
  - acap://acap.dave.cridland.net/byowner/user/dwd/bookmarks/
  - http://dave.cridland.net/
Infotrope Polymer - ACAP, IMAP, ESMTP, and Lemonade

From marka@isc.org  Sun Jul 24 00:34:27 2011
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In message <B2C17B21-EA8A-4698-8C41-F55A9AA140D4@gbiv.com>, "Roy T. Fielding" writes:
> On Jul 21, 2011, at 10:52 AM, I=F1aki Baz Castillo wrote:
> 
> > 2011/7/21 Dave Cridland <dave@cridland.net>:
> >> It's proven impossible, despite effort, to retrofit SRV onto HTTP; there=
>  is
> >> no way it'll be possible to retrofit onto WS.
> > =
> 
> > Right. If WS borns with no SRV (as a MUST for WS clients) then just
> > forget it and let inherit all the ugly limitations from HTTP protocol.
> 
> I am tired of this.  SRV is not used for HTTP because SRV adds latency
> to the initial request for no useful purpose whatsoever.

How do you solve the problem of hosting just "http://example.com/"
on "s1.joes-web-service.com" and not redirect everything else at
example.com?  People have been complaining about this for about as
long as the web has existed.

> SRV records for
> XMPP and MX records for mail are useful because there is only one such
> server expected per domain and it is *very* desirable to maintain central
> control over that routing.  In contrast, HTTP is deployed in an anarchic
> manner in which there are often several HTTP servers per machine
> (e.g., tests, staging, production, CUPS, etc,).  AFAICT, WebSockets is
> even more anarchic than HTTP -- it will have to be, given that the sane
> network admins will block it by default.
> 
> In short, SRV is not used by the Web because it is inappropriate for HTTP.
> I have seen no reason to believe that it would be appropriate for WebSocket=
> s.
> If you want SRV to be part of the proposed standard, then you have to convi=
> nce
> the people implementing WS to use SRV.  None have done so, yet, so we can't
> expect the editor to add it to the spec just because you have an opinion.
> 
> ....Roy
> _______________________________________________
> Ietf mailing list
> Ietf@ietf.org
> https://www.ietf.org/mailman/listinfo/ietf
-- 
Mark Andrews, ISC
1 Seymour St., Dundas Valley, NSW 2117, Australia
PHONE: +61 2 9871 4742                 INTERNET: marka@isc.org

From w@1wt.eu  Sun Jul 24 03:52:35 2011
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From: Willy Tarreau <w@1wt.eu>
To: Dave Cridland <dave@cridland.net>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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[ should we leave ietf@ietf.org in CC or not ? I'm suspecting that people
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On Sun, Jul 24, 2011 at 10:35:45AM +0100, Dave Cridland wrote:
> You're saying that you have a nebulous connection thing, that you  
> pump HTTP requests down, and you may want to use that for WebSocket  
> upgrade requests as well, by taking the ws URI and performing a  
> simple transformation on it into an HTTP URI to make the upgrade  
> request on.
> 
> So, given ws://example.com/foo/bar, you'll make the upgrade request  
> on http://example.com/foo/bar

No I'm not saying that because I don't understand what you mean here.
What I'm saying is that browsers try to reuse existing connections to
host:port. So if you want to connect to ws://example.com/foo/bar and
the browser already has a connection to example.com:80 because of a
previous HTTP request to that host, it's much advantageous for it to
reuse that existing connection.

Making an additional DNS request and a connection come with a cost.
In mobile environments, it's not uncommon to see 300-500 ms RTT. This
means that performing the additional DNS request could cost up to one
extra second.

> What Iñaki and I are arguing for is that for WS, there is simply one  
> additional step prior to finding a suitable connection, which is  
> performing an SRV lookup/selection.

That's clearly what I understood.

> And this new step can actually be performed before any of the HTTP  
> connection steps. So, roughly, in order to "do" WebSocket for  
> ws://example.com/foo/bar you would first do SRV lookup, which would  
> give you an equivalent HTTP URI to perform the upgrade request. So  
> for example, if you got:
> 
> _ws._tcp.example.com SRV 1 0 ws.example.com
> 
> You'd make the upgrade request on the HTTP URI of:
> 
> http://ws.example.com/foo/bar
> 
> Of course, if this failed to connect, you can find another HTTP URI  
> to try - something you cannot do without SRV.

If the connection fails here, HTTP will fail too, and right now, people
don't use SRV records with HTTP. I'm not judging whether it's good or
bad, I'm just saying what I'm seeing. Instead, people are using load
balancers, BGP and DNS mechanisms to ensure their site is available.

So whatever principle they're currently using with HTTP will also
provide the same mechanism for free to WS.

Yes it could be nice to have SRV on both HTTP and WS, but we need to
make it adopted for HTTP first. And if it's not adopted at all in the
case of HTTP, probably that is because it's not perceived as bringing
a lot of benefit *for that specific usage*. Instead it may even add
one DNS round trip which clearly is not desirable with highly interactive
protocols such as HTTP.

Also, the notion of "failure to connect" is hard to qualify in interactive
environments. If my browser tries to connect and waits more than a few
seconds before retrying on another server, I will probably already have
pressed the Esc key. We all do that when clicking links on Google search
results : if you don't get a result in one second, you click another one.
It is not reasonable to use even lower timeouts when using SRV records,
because it happens quite often to lose packets on the net, and sometimes
it's the client side which is slow. If the browser decides to fail within
one second and try another host, it will experience very frequent
difficulties.

Client-based failover (what SRV or MX are) is nice with protocols that
can wait long enough to ensure the server is dead. Some mailers try to
connect for one minute or even more. At that point, the client is
certain that the server is dead. In a browser, nobody waits that long,
so giving such a verdict is quite hard.

Still, I think that SRV might be used for site failover, it will be a
lot cleaner than using very short TTLs as some are doing. Such a usage
is compatible with deployed infrastructures, because nothing prevents
a client from prefering an SRV record over a CNAME record.

Now with that in mind, I don't see how one could suggest a MUST for this.
MUST that don't directly come from absolute requirements are never 100%
respected and cause much more trouble than SHOULD which are recommended
for optimal usage. If we say SHOULD, sites will ensure they have a working
DNS config which always announces valid addresses in A/AAAA or CNAME and
that SRV is used for failover. If we say MUST, sites won't bother emitting
correct addresses in A/AAAA or CNAME, or will even announce different ones,
and we'll get a lot of trouble at many places where the wrong record will
be picked by proxies or by clients which can't use SRV for whatever
reason.

Regards,
Willy


From ibc@aliax.net  Sun Jul 24 03:55:04 2011
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2011/7/22 Bruce Atherton <bruce@callenish.com>:

> Oh, it isn't hard, but it violates the principle of least surprise. Most
> people think they know how to code name resolution by using gethostbyname=
()
> or equivalent.

Should a new protocol loose cool features just because "most people"
expect a domain to be mapped to a single IP always?


> And to do it efficiently, especially when we are operating in
> a world that is predominantly driven by HTTP servers, requires complexiti=
es
> like dealing with asynchronous calls to fetch the A and SRV records at th=
e
> same time under the assumption that the SRV record probably won't exist.

You can do it serially (first SRV then A). We are speaking about a few
miliseconds just during the WS connection, no more. In SIP and XMPP
(real *realtime* protocols with IM and voice) this is not a problem.




> As I understand it, the issue that caused the various drafts for HTTP SRV=
 to
> die without getting much traction is one of efficiency.

Could you point to any reference for such a conclusion?

I really think the problem of SRV in HTTP is the fact the HTTP was
already extended when SRV appeared and migration was unfeasible.


> It is inefficient to
> make all these extra DNS calls, especially when it is unlikely that many =
of
> the records you are blocking on will exist. Other than the inefficiency,
> HTTP SRV is just an incremental technology you add to your existing DNS
> without hurting what already exists. Since Websockets uses the same
> infrastructure the records are unlikely to exist for it either, so the
> inefficiency issues are still present.

How is possible that using DNS SRV (and also DNS NAPTR in SIP !!) is
not "inefficient" for SIP and XMPP ***realtime*** protocols?



Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From w@1wt.eu  Sun Jul 24 04:02:55 2011
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Date: Sun, 24 Jul 2011 13:02:48 +0200
From: Willy Tarreau <w@1wt.eu>
To: Mark Andrews <marka@isc.org>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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Hi Mark,

On Sun, Jul 24, 2011 at 05:33:23PM +1000, Mark Andrews wrote:
> 
> In message <B2C17B21-EA8A-4698-8C41-F55A9AA140D4@gbiv.com>, "Roy T. Fielding" writes:
> > On Jul 21, 2011, at 10:52 AM, I=F1aki Baz Castillo wrote:
> > 
> > > 2011/7/21 Dave Cridland <dave@cridland.net>:
> > >> It's proven impossible, despite effort, to retrofit SRV onto HTTP; there=
> >  is
> > >> no way it'll be possible to retrofit onto WS.
> > > =
> > 
> > > Right. If WS borns with no SRV (as a MUST for WS clients) then just
> > > forget it and let inherit all the ugly limitations from HTTP protocol.
> > 
> > I am tired of this.  SRV is not used for HTTP because SRV adds latency
> > to the initial request for no useful purpose whatsoever.
> 
> How do you solve the problem of hosting just "http://example.com/"
> on "s1.joes-web-service.com" and not redirect everything else at
> example.com?  People have been complaining about this for about as
> long as the web has existed.

I would say that people complaining about this are doing the wrong thing
first. We've been using www.domain.tld, ftp.domain.tld, mail.domain.tld,
ns.domain.tld for ages, each indicating the service in their host name,
and suddenly just because www.domain.tld appears in the browser's address
bar, it should be shortened to show only "domain.tld". If people are doing
this, they must assume their choices. Sure, SRV would make that easier for
them, but it's not as if no solution had existed before.

Regards,
Willy


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2011/7/22 David Endicott <dendicott@gmail.com>:
> The technological advantages are worthy, when it's used, but when it does=
n't
> come into play, there are added inefficiencies.

~100 ms (if the DNS server is not local and there is no DNS cache for
the given domain). And just during the WS connection, no more. Taking
into account that a WS will be *usually* connected after loading a web
page, such ~100ms in a non-full-realtime protocol is insignificant.


> Also the name resolution
> of the HTTP that serves the Javascript that opens the WS should remain
> constant.

Why?


> If WS resolves the host/domain to a different address than the
> HTTP it was spawned from, it becomes a method to bypass same-origin / COR=
S
> restrictions.

No, you are wrong. You are doing invalid assumptions (in all the
thread) about how SRV works. Perhaps you should learn about it before
making such assertions. Maybe if you read
http://tools.ietf.org/html/draft-ibc-websocket-dns-srv-02#section-4.1:


   When the client constructs the WebSocket handshake HTTP request, the
   URI MUST be set as described in Section 3.2 of
   [I-D.ietf-hybi-thewebsocketprotocol] regardless of the usage of SRV
   mechanism.  This is, DNS SRV resolution for a "ws:" or "wss:" URI
   does not alter the usual construction of the WebSocket handshake
   request.



> I favor a minimalist core with extensibility. =C2=A0 =C2=A0Name resolutio=
n happens
> before the WS opening handshake, so I continue to see this as outside the
> domain of the WS protocol. =C2=A0 I would prefer that name resolution be =
provided
> by a selectable method. =C2=A0That way, in 20 years, when name resolution=
 needs
> have again changed, we'll have the ability to adapt.

This does not make sense, even if you have been saying the same during
all the thread. So, do you think that a mail client can *select* how
to resolve mailto:alice@gmail.com? why??? obviously if the mail client
does not perform DNS MX resolution for gmail.com things will not work,
why? because mail servers destinations are specified via DNS MX
resource records and because the SMTP protocol mandates it.


David, don't take me wrong but:

- You are speaking about SRV with no knowlodegde enough about it (so
making really wrong assumptions).
- You have a vision about "resolving URI's" that does not fit well in
the real world. The protocol/specification decides how resolution
takes place, rather than the client.


Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/22 Willy Tarreau <w@1wt.eu>:
> I=C3=B1aki, what we're saying is that the resolving applies first to HTTP
> well before it is WS. For instance, a client could connect to an HTTP
> server, fetch a few objects, then decide to upgrade the connection to
> switch to WebSocket. DNS resolving for WS would not even be involved
> here.

Hi. Maybe I'm the only who assumes that, usually, the WS server is not
colocated within the initial web server.

This is, your text below is valid just in the case I browse a web page
in http://some-domain.org (so port 80) and retrieve a WS URI to
connect to ws://some-domain.org (so also port 80). In that case you
say that the http/ws client would decide to upgrade the connection so
"it must assume that the WS handshake must be sent to the same IP:port
resolved for the HTTP communication".

I don't agree with yout point. Doing the "upgrade" does not mean
reusing an existing TCP connection (in which HTTP took place) for
other purpose. Instead, doing the WS upgrade means opening (or
reusing) a TCP connection, sending a HTTP GET with special semantics,
expect 101 and then start a bidirectional frame-based communication.
So sending the GET with "upgrade" has nothing to do with any previous
HTTP communication with the HTTP server.



> I agree with what others have been saying : if/when a different handshake
> is supported, eg. on a specific port without the HTTP upgrade, then it
> will make sense.

Do you mean WS as a complete separate protocol running on a specific
WS port and so? I'd really would like it (rather than the exotic
pseudo-HTTP mechanism used right now), but I expect it will never
happen.



> But as of now we're relying on the lower layer. As Greg
> said it, without a deep change in HTTP you won't be able to make the rule
> a MUST for WS. However, John's suggest of using a SHOULD when the record
> exists and the client can see it looks fine. What's the problem if not al=
l
> of your clients go to the same hosts ? You can even announce all of your
> servers with A/AAAA and with SRV as well as long as they're running on th=
e
> same ports. Those who can use SRV just have more information than the oth=
er
> ones and can be served better.

Having multiple A/AAAA records for a single domain does not provide
failover (as clients usually take just the first IP). I see your
point, but I expect no success at all.


Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/22 David Endicott <dendicott@gmail.com>:
> Actually....I wasn't talking about the Host: header - that is totally
> spoofable...I was concerned about:
> 1. Browser client resolves example.com via old style DNS to x.x.x.x and
> fetches HTTP
> 2. Received HTML starts JS which starts WS connection
> 3. WS resolves example.com via DNS SRV to y.y.y.y and opens
> 4. WS now has access outside origin.
> Please note, I did not specify why DNS SRV resolved differently than old
> style DNS - could be=C2=A0malicious, could be an simple mistake. =C2=A0 =
=C2=A0 I am
> assuming the DNS SRV and old DNS might be answered from different servers=
.
> Do browsers restrict origin / cross-site access based on name or on addre=
ss?


Now I assume that there is no SRV stuff at all:

1. Browser client resolves example.com via old style DNS to x.x.x.x and
fetches HTTP.

2. Received HTML contains a JS with a WS URI "ws://other-domain.net".

3. WS resolves other-domain.ne via old style DNS to z.z.z.z and opens.

4. WS now has access outside origin.


Is there any spec in which it's said that the WS URI must point to the
same domain as the initial web page?

NOTE: Anyhow, in the case of DNS SRV such domain can be the same. My
example above is just another.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/23 Roy T. Fielding <fielding@gbiv.com>:
>> Right. If WS borns with no SRV (as a MUST for WS clients) then just
>> forget it and let inherit all the ugly limitations from HTTP protocol.
>
> I am tired of this. =C2=A0SRV is not used for HTTP because SRV adds laten=
cy
> to the initial request for no useful purpose whatsoever.

And I'm really tired of hearing the argument of the "latency" which
nobody demostrates (but just talks about it without replying me how
the same is not a problem in realtime protocols like SIP and XMPP).



> SRV records for
> XMPP and MX records for mail are useful because there is only one such
> server expected per domain

$ host -t srv _xmpp-server._tcp.gmail.com
_xmpp-server._tcp.gmail.com has SRV record 5 0 5269 xmpp-server.l.google.co=
m.
_xmpp-server._tcp.gmail.com has SRV record 20 0 5269 xmpp-server3.l.google.=
com.
_xmpp-server._tcp.gmail.com has SRV record 20 0 5269 xmpp-server2.l.google.=
com.
_xmpp-server._tcp.gmail.com has SRV record 20 0 5269 xmpp-server4.l.google.=
com.
_xmpp-server._tcp.gmail.com has SRV record 20 0 5269 xmpp-server1.l.google.=
com.

No comments.


> and it is *very* desirable to maintain central
> control over that routing.

I don't know what this means.


>=C2=A0In contrast, HTTP is deployed in an anarchic
> manner in which there are often several HTTP servers per machine
> (e.g., tests, staging, production, CUPS, etc,).

Could you explain me why DNS A is good but DNS SRV is bad in such
"anarchic" deployments?


> =C2=A0AFAICT, WebSockets is
> even more anarchic than HTTP -- it will have to be, given that the sane
> network admins will block it by default.

------------------------
   It is up to the system administrator whether to set, or not, DNS SRV
   resource records for the WebSocket protocol within the provided
   service.  This specification allows the system administrator to use
   the DNS SRV [RFC2782] mechanism to improve the service reliability by
   providing load-balancing and failover capabilities, but does not
   mandate it (the system administrator could choose whichever
   scalability strategy).
------------------------



> In short, SRV is not used by the Web because it is inappropriate for HTTP=
.
> I have seen no reason to believe that it would be appropriate for WebSock=
ets.
> If you want SRV to be part of the proposed standard, then you have to con=
vince
> the people implementing WS to use SRV. =C2=A0None have done so, yet, so w=
e can't
> expect the editor to add it to the spec just because you have an opinion.

Ok, so the argument "I don't know SRV but I feel fine with HTTP
limitations for years" will win. Great design decissions.


Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/24 Willy Tarreau <w@1wt.eu>:
> No I'm not saying that because I don't understand what you mean here.
> What I'm saying is that browsers try to reuse existing connections to
> host:port. So if you want to connect to ws://example.com/foo/bar and
> the browser already has a connection to example.com:80 because of a
> previous HTTP request to that host, it's much advantageous for it to
> reuse that existing connection.
>
> Making an additional DNS request and a connection come with a cost.

http://tools.ietf.org/html/draft-ibc-websocket-dns-srv-02#section-5.2

--=20
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<ibc@aliax.net>

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On Sun, Jul 24, 2011 at 01:26:53PM +0200, Iñaki Baz Castillo wrote:
> 2011/7/22 Willy Tarreau <w@1wt.eu>:
> > Iñaki, what we're saying is that the resolving applies first to HTTP
> > well before it is WS. For instance, a client could connect to an HTTP
> > server, fetch a few objects, then decide to upgrade the connection to
> > switch to WebSocket. DNS resolving for WS would not even be involved
> > here.
> 
> Hi. Maybe I'm the only who assumes that, usually, the WS server is not
> colocated within the initial web server.

Web-based infrastructures make that almost mandatory at the frontend,
especially in massive hosting where you don't want to multiply IP addresses.
In general you have one single point which handles the IP:port and which
dispatches that to many servers based on the Host header, URI, file names,
cookies, etc...

> This is, your text below is valid just in the case I browse a web page
> in http://some-domain.org (so port 80) and retrieve a WS URI to
> connect to ws://some-domain.org (so also port 80). In that case you
> say that the http/ws client would decide to upgrade the connection so
> "it must assume that the WS handshake must be sent to the same IP:port
> resolved for the HTTP communication".
> 
> I don't agree with yout point. Doing the "upgrade" does not mean
> reusing an existing TCP connection (in which HTTP took place) for
> other purpose.

Huh ?

> Instead, doing the WS upgrade means opening (or
> reusing) a TCP connection, sending a HTTP GET with special semantics,
> expect 101 and then start a bidirectional frame-based communication.

I still remember how the handshake works, thank you.

> So sending the GET with "upgrade" has nothing to do with any previous
> HTTP communication with the HTTP server.

Yes it has. Either you open a fresh new connection, or you reuse an
idle existing one. But to know that the connection is idle, you must
understand the protocol that was spoken on it and this protocol must
have clearly delimited messages. HTTP supports reuse of connections
(also called "keep-alive") and since the WS handshake is HTTP, it is
possible and I'd add even recommended to reuse an existing connection
to send a WS handshake, if one such connection exists.

> > I agree with what others have been saying : if/when a different handshake
> > is supported, eg. on a specific port without the HTTP upgrade, then it
> > will make sense.
> 
> Do you mean WS as a complete separate protocol running on a specific
> WS port and so? I'd really would like it (rather than the exotic
> pseudo-HTTP mechanism used right now), but I expect it will never
> happen.

I'm sure it will happen. We need applications to be developped using
WS first. But there are places where :
  - HTTP compatibility won't be needed
  - masking will be annoying
  - HTTP overhead will be too much
  - HTTP round trip will be too much

I think that this will happen as soon as a working proposal for TLS NPN
appears, because the same requirements will exist (eg: how to specify
the resource name in a simpler way, etc...). Right now we need WS to be
able to replace long polling mechanisms which already work over HTTP, so
if we want it to be adopted, we need to deploy where previous methods
used to work. You just need to be patient :-)

> > But as of now we're relying on the lower layer. As Greg
> > said it, without a deep change in HTTP you won't be able to make the rule
> > a MUST for WS. However, John's suggest of using a SHOULD when the record
> > exists and the client can see it looks fine. What's the problem if not all
> > of your clients go to the same hosts ? You can even announce all of your
> > servers with A/AAAA and with SRV as well as long as they're running on the
> > same ports. Those who can use SRV just have more information than the other
> > ones and can be served better.
> 
> Having multiple A/AAAA records for a single domain does not provide
> failover (as clients usually take just the first IP). I see your
> point, but I expect no success at all.

That's not what I'm saying. Right now, people are using A/AAAA with short
TTLs and are updating the zones when a site fails (when I mean a site, I
mean a datacenter). This is something which happens rarely enough to be
acceptable. Using fast DNS updates for server failover does not work
because caches are everywhere and experience shows that even after one
month you still receive traffic on a server you've stopped announcing.

However, please read what I've explained in another mail about the
limitations of client-based failover in web environments.

Regards,
Willy


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On Sun, Jul 24, 2011 at 01:47:36PM +0200, Iñaki Baz Castillo wrote:
> 2011/7/24 Willy Tarreau <w@1wt.eu>:
> > No I'm not saying that because I don't understand what you mean here.
> > What I'm saying is that browsers try to reuse existing connections to
> > host:port. So if you want to connect to ws://example.com/foo/bar and
> > the browser already has a connection to example.com:80 because of a
> > previous HTTP request to that host, it's much advantageous for it to
> > reuse that existing connection.
> >
> > Making an additional DNS request and a connection come with a cost.
> 
> http://tools.ietf.org/html/draft-ibc-websocket-dns-srv-02#section-5.2

You still need the DNS request : the client does an A/AAAA request for
the HTTP host, then if you ask it to use an SRV record for the WS connection,
it must perform that request too, even if it's to conclude that it can reuse
the idle connection.

Regards,
Willy


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From: Willy Tarreau <w@1wt.eu>
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On Sun, Jul 24, 2011 at 01:42:26PM +0200, Iñaki Baz Castillo wrote:
> 2011/7/23 Roy T. Fielding <fielding@gbiv.com>:
> >> Right. If WS borns with no SRV (as a MUST for WS clients) then just
> >> forget it and let inherit all the ugly limitations from HTTP protocol.
> >
> > I am tired of this.  SRV is not used for HTTP because SRV adds latency
> > to the initial request for no useful purpose whatsoever.
> 
> And I'm really tired of hearing the argument of the "latency" which
> nobody demostrates (but just talks about it without replying me how
> the same is not a problem in realtime protocols like SIP and XMPP).

Because you have never worked in a mobile phone environment. You'd be
amazed to see what end users are paying for ! Count 300-500 ms on average
for a DNS request.

> > In contrast, HTTP is deployed in an anarchic
> > manner in which there are often several HTTP servers per machine
> > (e.g., tests, staging, production, CUPS, etc,).
> 
> Could you explain me why DNS A is good but DNS SRV is bad in such
> "anarchic" deployments?

DNS is not mandatory for HTTP. It's not "DNS A" which makes it good, but
"no mandatory DNS". This is a huge difference.

Regards,
Willy


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--000e0cd5c4be0d216204a8d150a3
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On Sun, Jul 24, 2011 at 7:11 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wr=
ote:

> 2011/7/22 David Endicott <dendicott@gmail.com>:
> > The technological advantages are worthy, when it's used, but when it
> doesn't
> > come into play, there are added inefficiencies.
>
> ~100 ms (if the DNS server is not local and there is no DNS cache for
> the given domain). And just during the WS connection, no more. Taking
> into account that a WS will be *usually* connected after loading a web
> page, such ~100ms in a non-full-realtime protocol is insignificant.


Wait, are you really saying that 100ms connect latency is unimportant?

--=20
John A. Tamplin
Software Engineer (GWT), Google

--000e0cd5c4be0d216204a8d150a3
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<div class=3D"gmail_quote">On Sun, Jul 24, 2011 at 7:11 AM, I=C3=B1aki Baz =
Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.n=
et</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"marg=
in:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

2011/7/22 David Endicott &lt;<a href=3D"mailto:dendicott@gmail.com">dendico=
tt@gmail.com</a>&gt;:<br>
<div class=3D"im">&gt; The technological advantages are worthy, when it&#39=
;s used, but when it doesn&#39;t<br>
&gt; come into play, there are added inefficiencies.<br>
<br>
</div>~100 ms (if the DNS server is not local and there is no DNS cache for=
<br>
the given domain). And just during the WS connection, no more. Taking<br>
into account that a WS will be *usually* connected after loading a web<br>
page, such ~100ms in a non-full-realtime protocol is insignificant.</blockq=
uote><div><br></div><div>Wait, are you really saying that 100ms connect lat=
ency is unimportant?</div></div><br>-- <br>John A. Tamplin<br>Software Engi=
neer (GWT), Google<br>



--000e0cd5c4be0d216204a8d150a3--

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On Sun, 2011-07-24 at 13:11 +0200, IÃ±aki Baz Castillo wrote:
> 2011/7/22 David Endicott <dendicott@gmail.com>:
> > The technological advantages are worthy, when it's used, but when it doesn't
> > come into play, there are added inefficiencies.
> 
> ~100 ms (if the DNS server is not local and there is no DNS cache for
> the given domain). And just during the WS connection, no more. Taking
> into account that a WS will be *usually* connected after loading a web
> page, such ~100ms in a non-full-realtime protocol is insignificant.
> 

reducing startup latencies is a priority for me. I would not implement
such a scheme. 

Lots of RTTs are bigger than 100ms, fwiw.





From alexey.melnikov@isode.com  Sun Jul 24 08:19:25 2011
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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Dave Cridland wrote:
> On Thu Jul 21 18:18:31 2011, David Endicott wrote:
>> It is my opinion that name resolution (however done) is outside the 
>> purview
>> of WS.   It may be handled in any number of ways by the client, and must
>> happen *before* WS establishes it's TCP connection and begins 
>> handshaking.
>
> The URI scheme is defined here, and absolutely must explain whether 
> it's a host (for direct address lookup), or a domain (for SRV).
>
> It's proven impossible, despite effort, to retrofit SRV onto HTTP; 
> there is no way it'll be possible to retrofit onto WS.
In my capacity of a WG participant:
+1



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On Jul 24, 2011, at 4:42 AM, I=F1aki Baz Castillo wrote:

> 2011/7/23 Roy T. Fielding <fielding@gbiv.com>:
>>> Right. If WS borns with no SRV (as a MUST for WS clients) then just
>>> forget it and let inherit all the ugly limitations from HTTP =
protocol.
>>=20
>> I am tired of this.  SRV is not used for HTTP because SRV adds =
latency
>> to the initial request for no useful purpose whatsoever.
>=20
> And I'm really tired of hearing the argument of the "latency" which
> nobody demostrates (but just talks about it without replying me how
> the same is not a problem in realtime protocols like SIP and XMPP).

Try using a protocol analyzer.  It is a problem with SIP and XMPP.
Chat and telephony have different user expectations about the initial
connect time, so people just don't notice it as much as they do when
a web page rendering process stalls because embedded resources require
two additional DNS requests.

>> SRV records for
>> XMPP and MX records for mail are useful because there is only one =
such
>> server expected per domain
>=20
> $ host -t srv _xmpp-server._tcp.gmail.com
> _xmpp-server._tcp.gmail.com has SRV record 5 0 5269 =
xmpp-server.l.google.com.
> _xmpp-server._tcp.gmail.com has SRV record 20 0 5269 =
xmpp-server3.l.google.com.
> _xmpp-server._tcp.gmail.com has SRV record 20 0 5269 =
xmpp-server2.l.google.com.
> _xmpp-server._tcp.gmail.com has SRV record 20 0 5269 =
xmpp-server4.l.google.com.
> _xmpp-server._tcp.gmail.com has SRV record 20 0 5269 =
xmpp-server1.l.google.com.
>=20
> No comments.

I meant server as in authority to answer for the domain, of course.

>> and it is *very* desirable to maintain central
>> control over that routing.
>=20
> I don't know what this means.

It means that admins care a great deal about what machines in the =
network
are allowed to receive mail for that network, and likewise for XMPP, =
since
both are store and forward protocols that contain messages with personal
information for individual addresses which are routed on a hop-by-hop =
basis.
HTTP has none of those characteristics.  Hence, Mail and XMPP are =
centralized
services for an entire domain, whereas HTTP services are specific to the
services provided by each individual host.

>>  In contrast, HTTP is deployed in an anarchic
>> manner in which there are often several HTTP servers per machine
>> (e.g., tests, staging, production, CUPS, etc,).
>=20
> Could you explain me why DNS A is good but DNS SRV is bad in such
> "anarchic" deployments?

Because DNS A for a server is deployed as soon as the server is made
available on the network (Intra or Inter).  It is one of the first =
things
done when installing an OS.  SRV, in contrast, requires that the zone
manager reconfigure DNS.

>>  AFAICT, WebSockets is
>> even more anarchic than HTTP -- it will have to be, given that the =
sane
>> network admins will block it by default.
>=20
> ------------------------
>   It is up to the system administrator whether to set, or not, DNS SRV
>   resource records for the WebSocket protocol within the provided
>   service.  This specification allows the system administrator to use
>   the DNS SRV [RFC2782] mechanism to improve the service reliability =
by
>   providing load-balancing and failover capabilities, but does not
>   mandate it (the system administrator could choose whichever
>   scalability strategy).
> ------------------------

*shrug*

>> In short, SRV is not used by the Web because it is inappropriate for =
HTTP.
>> I have seen no reason to believe that it would be appropriate for =
WebSockets.
>> If you want SRV to be part of the proposed standard, then you have to =
convince
>> the people implementing WS to use SRV.  None have done so, yet, so we =
can't
>> expect the editor to add it to the spec just because you have an =
opinion.
>=20
> Ok, so the argument "I don't know SRV but I feel fine with HTTP
> limitations for years" will win. Great design decissions.

No, the argument is that I know SRV, I know a lot more about HTTP, and I =
know
for a fact that SRV is not desirable for HTTP.  So, whenever you suggest =
that
lack of SRV is somehow a failing of HTTP, you should understand that it =
makes
your arguments look clueless.  Please, find some other example.

....Roy=

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On Jul 24, 2011, at 12:33 AM, Mark Andrews wrote:
> In message <B2C17B21-EA8A-4698-8C41-F55A9AA140D4@gbiv.com>, "Roy T. =
Fielding" writes:
>> On Jul 21, 2011, at 10:52 AM, I=3DF1aki Baz Castillo wrote:
>>=20
>>> 2011/7/21 Dave Cridland <dave@cridland.net>:
>>>> It's proven impossible, despite effort, to retrofit SRV onto HTTP; =
there=3D
>> is
>>>> no way it'll be possible to retrofit onto WS.
>>> =3D
>>=20
>>> Right. If WS borns with no SRV (as a MUST for WS clients) then just
>>> forget it and let inherit all the ugly limitations from HTTP =
protocol.
>>=20
>> I am tired of this.  SRV is not used for HTTP because SRV adds =
latency
>> to the initial request for no useful purpose whatsoever.
>=20
> How do you solve the problem of hosting just "http://example.com/"
> on "s1.joes-web-service.com" and not redirect everything else at
> example.com?  People have been complaining about this for about as
> long as the web has existed.

The Web has existed in usable form since 1991.  Name-based virtual
hosting wasn't even possible until we added Host in 1995.  In any case,
nobody has ever asked me to make the above a priority -- it simply isn't
a relevant problem compared to load balancing in general, and the =
general
problem does not assume that the client is friendly.

I would never try to "solve" load balancing by requiring every browser
to make an additional (failed) DNS SRV request each time it encounters
one of the 357,292,065 individual hostnames that are known to use HTTP,
not to mention the many millions more that are not exposed on the
Internet and don't even use DNS for resolution.  HTTP would not, cannot,
and never will benefit from SRV even if we had a magic wand that could
deploy it on all browsers.  SRV simply doesn't fit the Web architecture.

....Roy=

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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/24 Willy Tarreau <w@1wt.eu>:
>> Hi. Maybe I'm the only who assumes that, usually, the WS server is not
>> colocated within the initial web server.
>
> Web-based infrastructures make that almost mandatory at the frontend,
> especially in massive hosting where you don't want to multiply IP address=
es.
> In general you have one single point which handles the IP:port and which
> dispatches that to many servers based on the Host header, URI, file names=
,
> cookies, etc...

Ok, right, but that shouldn't be enough to make a specification
assuming it (IMHO).




>> I don't agree with yout point. Doing the "upgrade" does not mean
>> reusing an existing TCP connection (in which HTTP took place) for
>> other purpose.
>
> Huh ?

I just meant that there is no need of first retrieving a page from a
IP:PORT and then making the WS upgrade using the same IP:PORT.


>> Instead, doing the WS upgrade means opening (or
>> reusing) a TCP connection, sending a HTTP GET with special semantics,
>> expect 101 and then start a bidirectional frame-based communication.
>
> I still remember how the handshake works, thank you.

Sure, I didn't mean that you don't know it :)


>> So sending the GET with "upgrade" has nothing to do with any previous
>> HTTP communication with the HTTP server.
>
> Yes it has. Either you open a fresh new connection, or you reuse an
> idle existing one.

If the WS URI points to a different server, it's perfectly possible
that the WS connection has nothing to do (neither cookies usage) with
the previous HTTP/HTML traffic. I just meant it.


> But to know that the connection is idle, you must
> understand the protocol that was spoken on it and this protocol must
> have clearly delimited messages. HTTP supports reuse of connections
> (also called "keep-alive") and since the WS handshake is HTTP, it is
> possible and I'd add even recommended to reuse an existing connection
> to send a WS handshake, if one such connection exists.

Don't take me wrong, but "keep-alive" mechanism in HTTP is just a hack
since paralelism is not possible (well, it's but with too many
constrains, for example the server must send the replies in order). In
other protocols working on top of TCP (as SIP) each request contains
an id (in the case of SIP the Via branch) which allows identifying
which request a response belongs to, so I can send N requests via the
same TCP connection and receive the responses in any order. HTTP is
poor on this topic, and I wouldn't like WS to inherit it.



>> Do you mean WS as a complete separate protocol running on a specific
>> WS port and so? I'd really would like it (rather than the exotic
>> pseudo-HTTP mechanism used right now), but I expect it will never
>> happen.
>
> I'm sure it will happen. We need applications to be developped using
> WS first. But there are places where :
> =C2=A0- HTTP compatibility won't be needed
> =C2=A0- masking will be annoying
> =C2=A0- HTTP overhead will be too much
> =C2=A0- HTTP round trip will be too much
>
> I think that this will happen as soon as a working proposal for TLS NPN
> appears, because the same requirements will exist (eg: how to specify
> the resource name in a simpler way, etc...). Right now we need WS to be
> able to replace long polling mechanisms which already work over HTTP, so
> if we want it to be adopted, we need to deploy where previous methods
> used to work. You just need to be patient :-)

ok, I like what you say :)
However, imagine that in next 2 years there are a lot of WS servers
speaking HTTP (for the WS handshake, of course). If you suggest that a
different WS protocol could appear, it would require a new URI schema
so the client knows wheter to perform a HTTP WS handshake or use the
new WS protocol.



>> Having multiple A/AAAA records for a single domain does not provide
>> failover (as clients usually take just the first IP). I see your
>> point, but I expect no success at all.
>
> That's not what I'm saying. Right now, people are using A/AAAA with short
> TTLs and are updating the zones when a site fails (when I mean a site, I
> mean a datacenter). This is something which happens rarely enough to be
> acceptable. Using fast DNS updates for server failover does not work
> because caches are everywhere and experience shows that even after one
> month you still receive traffic on a server you've stopped announcing.

And there is where DNS SRV could help a lot :)



> However, please read what I've explained in another mail about the
> limitations of client-based failover in web environments.

Yes, I've read it. But I expect it's even more critical in a phone
call (my SIP phone performs SRV/A, gets the first IP:port and sends
the call request there using UDP, but the server is down, so after
some seconds *without hearing ringing* the phone sends the call to
another server). If it's not critical in SIP, why should it be so
critical in WS?

Also, if the user realizes that the connection takes too much time and
presses F5 to reload the page, why couldn't the webbrowser cache the
SRV results and mark the previous attemp as failed so next server:port
woul be tryed when the user presses F5?


Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/24 Willy Tarreau <w@1wt.eu>:
>> > Making an additional DNS request and a connection come with a cost.
>>
>> http://tools.ietf.org/html/draft-ibc-websocket-dns-srv-02#section-5.2
>
> You still need the DNS request : the client does an A/AAAA request for
> the HTTP host, then if you ask it to use an SRV record for the WS connect=
ion,
> it must perform that request too, even if it's to conclude that it can re=
use
> the idle connection.

Ok, but I don't consider a xtra DNS query to be so hard.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/24 Willy Tarreau <w@1wt.eu>:
>> And I'm really tired of hearing the argument of the "latency" which
>> nobody demostrates (but just talks about it without replying me how
>> the same is not a problem in realtime protocols like SIP and XMPP).
>
> Because you have never worked in a mobile phone environment. You'd be
> amazed to see what end users are paying for ! Count 300-500 ms on average
> for a DNS request.

Well, mobile phone world is a pain due to GPRS/3G internet
connections. But those networks should be improved rather than
assuming that all the Internet must change to work on those infernal
environments (which IMHO are not yet ready for modern internet). All I
see in mobile networks are workarounds.




>> Could you explain me why DNS A is good but DNS SRV is bad in such
>> "anarchic" deployments?
>
> DNS is not mandatory for HTTP. It's not "DNS A" which makes it good, but
> "no mandatory DNS". This is a huge difference.

So, do you mean using URI's with IP rather than domain? (take into
account that TLS connection require the certificate to match the URI
domain, but anyhow it's also possible to use IP's within the
certificate).


Cheers.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/24 John Tamplin <jat@google.com>:
>> ~100 ms (if the DNS server is not local and there is no DNS cache for
>> the given domain). And just during the WS connection, no more. Taking
>> into account that a WS will be *usually* connected after loading a web
>> page, such ~100ms in a non-full-realtime protocol is insignificant.
>
> Wait, are you really saying that 100ms connect latency is unimportant?

Open the fastest web page and tell me how long it takes. Probably you
have performed a DNS A query. I don't think that a xtra DNS query
would be the bottleneck, never.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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2011/7/24 Patrick McManus <pmcmanus@mozilla.com>:
>> ~100 ms (if the DNS server is not local and there is no DNS cache for
>> the given domain). And just during the WS connection, no more. Taking
>> into account that a WS will be *usually* connected after loading a web
>> page, such ~100ms in a non-full-realtime protocol is insignificant.
>>
>
> reducing startup latencies is a priority for me. I would not implement
> such a scheme.

Well, will you implement WebSocket (I assume in Mozilla web browsers)?
or just in case it fulfills all your own requeriments?

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: Willy Tarreau <w@1wt.eu>
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On Sun, Jul 24, 2011 at 08:25:05PM +0200, Iñaki Baz Castillo wrote:
> 2011/7/24 Willy Tarreau <w@1wt.eu>:
> >> > Making an additional DNS request and a connection come with a cost.
> >>
> >> http://tools.ietf.org/html/draft-ibc-websocket-dns-srv-02#section-5.2
> >
> > You still need the DNS request : the client does an A/AAAA request for
> > the HTTP host, then if you ask it to use an SRV record for the WS connection,
> > it must perform that request too, even if it's to conclude that it can reuse
> > the idle connection.
> 
> Ok, but I don't consider a xtra DNS query to be so hard.

I had to perform sites analysis for a customer a few months ago. Many
web pages nowadays have between 100 and 200 objects to fetch, spread
over up to 25-30 host names. If you take even only 100ms for each of
them, you're at 3 additional seconds to display the page. Granted those
requests are not WS and only HTTP, but as I said, SRV for WS won't work
before it works with HTTP, at least due to proxies.

But those 3 extra seconds will always be considered a good reason not
to make SRV mandatory for HTTP. The web is degrading very quickly due
to poor practices, and we should be careful not to suggest to make it
even worse.

Regards,
Willy


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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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* Iñaki Baz Castillo wrote:
>Open the fastest web page and tell me how long it takes.

Too long.
-- 
Björn Höhrmann · mailto:bjoern@hoehrmann.de · http://bjoern.hoehrmann.de
Am Badedeich 7 · Telefon: +49(0)160/4415681 · http://www.bjoernsworld.de
25899 Dagebüll · PGP Pub. KeyID: 0xA4357E78 · http://www.websitedev.de/ 

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From: Willy Tarreau <w@1wt.eu>
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On Sun, Jul 24, 2011 at 08:24:25PM +0200, Iñaki Baz Castillo wrote:
> > Yes it has. Either you open a fresh new connection, or you reuse an
> > idle existing one.
> 
> If the WS URI points to a different server, it's perfectly possible
> that the WS connection has nothing to do (neither cookies usage) with
> the previous HTTP/HTML traffic. I just meant it.

I agree, and there will certainly be a large number of cases where it
will be different. But that's not enough (in my opinion) to make something
mandatory considering the negative side effects.

> > But to know that the connection is idle, you must
> > understand the protocol that was spoken on it and this protocol must
> > have clearly delimited messages. HTTP supports reuse of connections
> > (also called "keep-alive") and since the WS handshake is HTTP, it is
> > possible and I'd add even recommended to reuse an existing connection
> > to send a WS handshake, if one such connection exists.
> 
> Don't take me wrong, but "keep-alive" mechanism in HTTP is just a hack
> since paralelism is not possible (well, it's but with too many
> constrains, for example the server must send the replies in order).

Pipelining improves things a lot, especially for static objects which
immediately fill the pipe.

> In
> other protocols working on top of TCP (as SIP) each request contains
> an id (in the case of SIP the Via branch) which allows identifying
> which request a response belongs to, so I can send N requests via the
> same TCP connection and receive the responses in any order.

>From what I've read, SPDY typically addresses this point.

> HTTP is poor on this topic, and I wouldn't like WS to inherit it.

But it will since there is no MUX support yet, WS frames will be delivered
in order. The application on top of WS might decide to number each message
though and use its own out-of-order mechanism though.

> >> Do you mean WS as a complete separate protocol running on a specific
> >> WS port and so? I'd really would like it (rather than the exotic
> >> pseudo-HTTP mechanism used right now), but I expect it will never
> >> happen.
> >
> > I'm sure it will happen. We need applications to be developped using
> > WS first. But there are places where :
> >  - HTTP compatibility won't be needed
> >  - masking will be annoying
> >  - HTTP overhead will be too much
> >  - HTTP round trip will be too much
> >
> > I think that this will happen as soon as a working proposal for TLS NPN
> > appears, because the same requirements will exist (eg: how to specify
> > the resource name in a simpler way, etc...). Right now we need WS to be
> > able to replace long polling mechanisms which already work over HTTP, so
> > if we want it to be adopted, we need to deploy where previous methods
> > used to work. You just need to be patient :-)
> 
> ok, I like what you say :)
> However, imagine that in next 2 years there are a lot of WS servers
> speaking HTTP (for the WS handshake, of course). If you suggest that a
> different WS protocol could appear, it would require a new URI schema
> so the client knows wheter to perform a HTTP WS handshake or use the
> new WS protocol.

That's just a minor detail. Likewise it will probably have to support a
fallback to the HTTP handshake because there will be a lot of places where
it will not pass through.

> >> Having multiple A/AAAA records for a single domain does not provide
> >> failover (as clients usually take just the first IP). I see your
> >> point, but I expect no success at all.
> >
> > That's not what I'm saying. Right now, people are using A/AAAA with short
> > TTLs and are updating the zones when a site fails (when I mean a site, I
> > mean a datacenter). This is something which happens rarely enough to be
> > acceptable. Using fast DNS updates for server failover does not work
> > because caches are everywhere and experience shows that even after one
> > month you still receive traffic on a server you've stopped announcing.
> 
> And there is where DNS SRV could help a lot :)

Not that much, because as soon as you fail your DC, you'll stop announcing
it so that visitors don't try to connect there for seconds before they
detect it's down.

> > However, please read what I've explained in another mail about the
> > limitations of client-based failover in web environments.
> 
> Yes, I've read it. But I expect it's even more critical in a phone
> call (my SIP phone performs SRV/A, gets the first IP:port and sends
> the call request there using UDP, but the server is down, so after
> some seconds *without hearing ringing* the phone sends the call to
> another server). If it's not critical in SIP, why should it be so
> critical in WS?

As I said, having it in WS means having it in HTTP too. And having a
browser rely on TCP connection timeouts to decide to use a server vs
another one would make a lot of sites unusable. I've already been used
to mark some advertiser's sites as 255.255.255.255 in my /etc/hosts to
stop waiting for their crap to respond on pages that I'm used to visit.
If a page loads objects from s1,s2,s3,s4.domain.tld and all are hosted
both on a valid and a dead DC, it can take a lot of time before all these
hosts are marked down and ignored. Nobody I know would wait for these to
fail.

> Also, if the user realizes that the connection takes too much time and
> presses F5 to reload the page, why couldn't the webbrowser cache the
> SRV results and mark the previous attemp as failed so next server:port
> woul be tryed when the user presses F5?

Yes but once again, if you have to wait one minute on each new site so
that all dead servers can be ignored, the web will look like a terrible
experience to you.

Regards,
Willy


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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Sun, 2011-07-24 at 20:33 +0200, IÃ±aki Baz Castillo wrote:
> 2011/7/24 Patrick McManus <pmcmanus@mozilla.com>:
> >> ~100 ms (if the DNS server is not local and there is no DNS cache for
> >> the given domain). And just during the WS connection, no more. Taking
> >> into account that a WS will be *usually* connected after loading a web
> >> page, such ~100ms in a non-full-realtime protocol is insignificant.
> >>
> >
> > reducing startup latencies is a priority for me. I would not implement
> > such a scheme.
> 
> Well, will you implement WebSocket (I assume in Mozilla web browsers)?
> or just in case it fulfills all your own requeriments?
> 

we don't implement protocols just because someone wrote them down.




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2011/7/24 Roy T. Fielding <fielding@gbiv.com>:

>>> SRV records for
>>> XMPP and MX records for mail are useful because there is only one such
>>> server expected per domain
>>
>> $ host -t srv _xmpp-server._tcp.gmail.com
>> _xmpp-server._tcp.gmail.com has SRV record 5 0 5269 xmpp-server.l.google=
.com.
>> _xmpp-server._tcp.gmail.com has SRV record 20 0 5269 xmpp-server3.l.goog=
le.com.
>> _xmpp-server._tcp.gmail.com has SRV record 20 0 5269 xmpp-server2.l.goog=
le.com.
>> _xmpp-server._tcp.gmail.com has SRV record 20 0 5269 xmpp-server4.l.goog=
le.com.
>> _xmpp-server._tcp.gmail.com has SRV record 20 0 5269 xmpp-server1.l.goog=
le.com.
>>
>> No comments.
>
> I meant server as in authority to answer for the domain, of course.

Ok. But I don't see the problem. What about Google Apps? My own domain
uses Gtalk and Gmail by setting Google XMPP SRV and MX records. Now
imagine that I would host my personal webpage (domain "www.aliax.net")
in Google, wouldn't be great a SRV entry for HTTP
(_http._tcp.aliax.net has SRV record 10 0 80
web-server-01.google.com)? or do we prefer the ugly HTTP 30X
redirection or ugly CNAME?



>>> and it is *very* desirable to maintain central
>>> control over that routing.
>>
>> I don't know what this means.
>
> It means that admins care a great deal about what machines in the network
> are allowed to receive mail for that network, and likewise for XMPP, sinc=
e
> both are store and forward protocols that contain messages with personal
> information for individual addresses which are routed on a hop-by-hop bas=
is.
> HTTP has none of those characteristics. =C2=A0Hence, Mail and XMPP are ce=
ntralized
> services for an entire domain, whereas HTTP services are specific to the
> services provided by each individual host.

I've never thought about it when referring to SRV for WebSocket.
Honestly I just see the SRV advantages for load-balancing and
failover.


>>> =C2=A0In contrast, HTTP is deployed in an anarchic
>>> manner in which there are often several HTTP servers per machine
>>> (e.g., tests, staging, production, CUPS, etc,).
>>
>> Could you explain me why DNS A is good but DNS SRV is bad in such
>> "anarchic" deployments?
>
> Because DNS A for a server is deployed as soon as the server is made
> available on the network (Intra or Inter). =C2=A0It is one of the first t=
hings
> done when installing an OS. =C2=A0SRV, in contrast, requires that the zon=
e
> manager reconfigure DNS.

SRV would never be mandatory for the service provider, it's just
optional. You could still play with A entries in your test
environments (or use just A entries in production). Still don't get
the problem.





>>> In short, SRV is not used by the Web because it is inappropriate for HT=
TP.
>>> I have seen no reason to believe that it would be appropriate for WebSo=
ckets.
>>> If you want SRV to be part of the proposed standard, then you have to c=
onvince
>>> the people implementing WS to use SRV. =C2=A0None have done so, yet, so=
 we can't
>>> expect the editor to add it to the spec just because you have an opinio=
n.
>>
>> Ok, so the argument "I don't know SRV but I feel fine with HTTP
>> limitations for years" will win. Great design decissions.
>
> No, the argument is that I know SRV, I know a lot more about HTTP, and I =
know
> for a fact that SRV is not desirable for HTTP. =C2=A0So, whenever you sug=
gest that
> lack of SRV is somehow a failing of HTTP, you should understand that it m=
akes
> your arguments look clueless. =C2=A0Please, find some other example.

In my opinion SRV is not possible for HTTP since SRV appeared too
late. That's the main and enough reason IMHO.
The fact that people is "confortable" enough with HTTP style-of-life
(a specification of 1999) and does not want to deal with "new
technologies" other than stuff and more stuff on top of HTTP, is not a
good argument for me. HTTP is not the layer number 5 in OSI model.


Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From w@1wt.eu  Sun Jul 24 11:57:17 2011
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On Sun, Jul 24, 2011 at 08:28:49PM +0200, Iñaki Baz Castillo wrote:
> 2011/7/24 Willy Tarreau <w@1wt.eu>:
> >> And I'm really tired of hearing the argument of the "latency" which
> >> nobody demostrates (but just talks about it without replying me how
> >> the same is not a problem in realtime protocols like SIP and XMPP).
> >
> > Because you have never worked in a mobile phone environment. You'd be
> > amazed to see what end users are paying for ! Count 300-500 ms on average
> > for a DNS request.
> 
> Well, mobile phone world is a pain due to GPRS/3G internet
> connections. But those networks should be improved rather than
> assuming that all the Internet must change to work on those infernal
> environments (which IMHO are not yet ready for modern internet). All I
> see in mobile networks are workarounds.

Those are infernal but part of the time cannot be compressed much more
due to the fact that you have to share the medium with many other people
and you have to wait for your slot to send packets. And I'm not even
counting the time it can take to forward your data across the country
between the antenna and the datacenter. Sure things will improve, but
I don't expect seeing anything below 40-50ms.

> >> Could you explain me why DNS A is good but DNS SRV is bad in such
> >> "anarchic" deployments?
> >
> > DNS is not mandatory for HTTP. It's not "DNS A" which makes it good, but
> > "no mandatory DNS". This is a huge difference.
> 
> So, do you mean using URI's with IP rather than domain? (take into
> account that TLS connection require the certificate to match the URI
> domain, but anyhow it's also possible to use IP's within the
> certificate).

On internal networks, using IP instead of URIs is not uncommon at all,
especially on developer networks where you need many instances of the
same server in different versions or for different people. Some static
servers also make use of this because it saves one roundtrip. And of
course you have it on your ADSL router's web-based configuration interface
otherwise you wouldn't be able to contact the DNS to reach the router :-)

But that's not what I meant, I meant that DNS is not the only solution
to resolve host names. WINS, NIS and /etc/hosts are usable too. When I
was a student in 94, we had all our passwords and hostnames in NIS and
no DNS was configured. It worked like a charm. DNS is not something
mandatory at all for many protocols. It just happens to be the standard
over the public Internet.

Regards,
Willy


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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/24 Willy Tarreau <w@1wt.eu>:
>> Also, if the user realizes that the connection takes too much time and
>> presses F5 to reload the page, why couldn't the webbrowser cache the
>> SRV results and mark the previous attemp as failed so next server:port
>> woul be tryed when the user presses F5?
>
> Yes but once again, if you have to wait one minute on each new site so
> that all dead servers can be ignored, the web will look like a terrible
> experience to you.

Ok, but what would be the failover solution then? any failover
solution can take some time until redirects the client's request to a
working server, it's not just a client problem.

Anyhow, don't forget that SRV is not just for failover, but also for
load-balancing (you can state that a more powerful server-01 receives
80% of the traffic and server-02 the remaining 20%).

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: Willy Tarreau <w@1wt.eu>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Sun, Jul 24, 2011 at 08:30:11PM +0200, Iñaki Baz Castillo wrote:
> 2011/7/24 John Tamplin <jat@google.com>:
> >> ~100 ms (if the DNS server is not local and there is no DNS cache for
> >> the given domain). And just during the WS connection, no more. Taking
> >> into account that a WS will be *usually* connected after loading a web
> >> page, such ~100ms in a non-full-realtime protocol is insignificant.
> >
> > Wait, are you really saying that 100ms connect latency is unimportant?
> 
> Open the fastest web page and tell me how long it takes. Probably you
> have performed a DNS A query. I don't think that a xtra DNS query
> would be the bottleneck, never.

On lossy networks such as 3G, they definitely are. A lost UDP packet is
not retransmitted nor signaled as lost, so the browser has to retry. However,
once the connection is established to the server, most losses are more or
less smoothed by TCP extensions such as SACK. So yes, it can take several
seconds to just resolve a host and then only a few hundreds of ms to retrieve
the objects. I've observed it.

Regards,
Willy


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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/24 Willy Tarreau <w@1wt.eu>:
> But that's not what I meant, I meant that DNS is not the only solution
> to resolve host names. WINS, NIS and /etc/hosts are usable too. When I
> was a student in 94, we had all our passwords and hostnames in NIS and
> no DNS was configured. It worked like a charm. DNS is not something
> mandatory at all for many protocols. It just happens to be the standard
> over the public Internet.

Ok, I get your point now :)
But, do current webbrosers resolve names using anything but DNS A?
(well, I know that they use /etc/hosts). Anyhow, WINS / NIS /etc/hosts
and such stuff just maps a hostname into a single IP. It's the
equivalent of a DNS A resource record. Think about locating a mail
server (MX is required), you need a DNS server.


Regards.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From w@1wt.eu  Sun Jul 24 12:16:51 2011
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Sun, Jul 24, 2011 at 08:52:32PM +0200, Iñaki Baz Castillo wrote:
> Ok. But I don't see the problem. What about Google Apps? My own domain
> uses Gtalk and Gmail by setting Google XMPP SRV and MX records. Now
> imagine that I would host my personal webpage (domain "www.aliax.net")
> in Google, wouldn't be great a SRV entry for HTTP
> (_http._tcp.aliax.net has SRV record 10 0 80
> web-server-01.google.com)? or do we prefer the ugly HTTP 30X
> redirection or ugly CNAME?

Well, you gave the perfect example : here, CNAME is the exact equivalent
of SRV. None is cheaper nor better than the other. You just find it ugly
because it's called CNAME.

> In my opinion SRV is not possible for HTTP since SRV appeared too
> late. That's the main and enough reason IMHO.

I'd see it differently : the balance between its extra costs vs the
expected benefits does not push in the direction to massively deploy
it. You know, it only takes 4-5 browsers to enable it and make the
whole web use it. Pat already explained why he doesn't want it.

> The fact that people is "confortable" enough with HTTP style-of-life
> (a specification of 1999) and does not want to deal with "new
> technologies" other than stuff and more stuff on top of HTTP, is not a
> good argument for me.

It's not a matter of new vs old technologies. We're not trying to
paint HTTP in a shiny color to impress friends. We're dealing with
deployed setups and users with high expectations, which are currently
more or less met and for which your proposal doesn't sensibly improve
experience but can sensibly degrade it for many people.

> HTTP is not the layer number 5 in OSI model.

Like it or not, it has inherited this role a long time ago because it
provides many advantages over plain TCP (such as easy reconnection,
authentication, etc...).

Willy


From ibc@aliax.net  Sun Jul 24 12:17:00 2011
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Hi, it's clear enough for me that DNS SRV will be not implemented in
web browsers for WebSocket URIs resolution. I understand some of the
reasons (not all however).

So I'm thinking about a future (extension) for WebSocket protocol and
API (sorry, I don't know yet the propose WS API so I use pseudo-code):


a) Let's suppose a JavaScript code contains a WS connection, something
like this:

    var mysocket =3D new WebSocket("ws://www.mydomain.org", null);

Note that there is a second parameter that is null. It would mean (as
the current spec states) that a DNS A query should be performed for
the given URI (or not if there is already a TCP connection).


b) I propose an optional second argument:

    var mysocket =3D new WebSocket("ws://www.mydomain.org", "ws://mydomain.=
org");

If the browser supports the new WS extension I'm suggesting, the
browser could *decide* whenever perform DNS A for the first URI or a
DNS SRV for the second URI. For example, WS clients within mobile
devices could decide to always use the first URI so just DNS A would
be done (or not if it was already resolved and connected). In the
other side, desktop web browsers implementing this new extension would
use, by default, the second URI with all the SRV stuff.



I don't like it too much as the service provider cannot anticipate how
clients would resolve the WS server, but for me is better than just
playing with A records and server side failover / load-balancing.


Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/24 Willy Tarreau <w@1wt.eu>:
> On Sun, Jul 24, 2011 at 08:52:32PM +0200, I=C3=B1aki Baz Castillo wrote:
>> Ok. But I don't see the problem. What about Google Apps? My own domain
>> uses Gtalk and Gmail by setting Google XMPP SRV and MX records. Now
>> imagine that I would host my personal webpage (domain "www.aliax.net")
>> in Google, wouldn't be great a SRV entry for HTTP
>> (_http._tcp.aliax.net has SRV record 10 0 80
>> web-server-01.google.com)? or do we prefer the ugly HTTP 30X
>> redirection or ugly CNAME?
>
> Well, you gave the perfect example : here, CNAME is the exact equivalent
> of SRV. None is cheaper nor better than the other. You just find it ugly
> because it's called CNAME.

Well... paying the same price (an extra DNS query) SRV offers you
balancing and failover capabilities, while CNAME is just an alias for
a DNS A resolution (like a soft link).





>> The fact that people is "confortable" enough with HTTP style-of-life
>> (a specification of 1999) and does not want to deal with "new
>> technologies" other than stuff and more stuff on top of HTTP, is not a
>> good argument for me.
>
> It's not a matter of new vs old technologies. We're not trying to
> paint HTTP in a shiny color to impress friends. We're dealing with
> deployed setups and users with high expectations, which are currently
> more or less met and for which your proposal doesn't sensibly improve
> experience but can sensibly degrade it for many people.

Anyhow I've never proposed SRV for HTTP, but just for WebSocket :)



>> HTTP is not the layer number 5 in OSI model.
>
> Like it or not, it has inherited this role a long time ago because it
> provides many advantages over plain TCP (such as easy reconnection,
> authentication, etc...).

You are right, I don't like it :)


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From w@1wt.eu  Sun Jul 24 12:29:54 2011
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Date: Sun, 24 Jul 2011 21:29:48 +0200
From: Willy Tarreau <w@1wt.eu>
To: =?iso-8859-1?Q?I=F1aki?= Baz Castillo <ibc@aliax.net>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Sun, Jul 24, 2011 at 08:57:45PM +0200, Iñaki Baz Castillo wrote:
> 2011/7/24 Willy Tarreau <w@1wt.eu>:
> >> Also, if the user realizes that the connection takes too much time and
> >> presses F5 to reload the page, why couldn't the webbrowser cache the
> >> SRV results and mark the previous attemp as failed so next server:port
> >> woul be tryed when the user presses F5?
> >
> > Yes but once again, if you have to wait one minute on each new site so
> > that all dead servers can be ignored, the web will look like a terrible
> > experience to you.
> 
> Ok, but what would be the failover solution then? any failover
> solution can take some time until redirects the client's request to a
> working server, it's not just a client problem.

This is already addressed by existing web architectures. DNS provides
geolocation and proposes you the closest working datacenter. BGP ensures
that this datacenter is always reachable via multiple links and multiple
providers. Load balancers on the site ensure that only valid servers are
used. I know some people who configure their load balancers so that a
server is removed from the pool less than 50ms after a failure or slowdown
has been noticed. You will never be able to offer that quality of service
using DNS only, because those 50ms are less than the time it takes for
most visitors to ping the site.

> Anyhow, don't forget that SRV is not just for failover, but also for
> load-balancing (you can state that a more powerful server-01 receives
> 80% of the traffic and server-02 the remaining 20%).

This is already addressed by DNS round robin and used by a lot of sites.
But keep in mind that DNS is used between *sites* and not *servers*. If
you use it for servers, you'll fail because you won't be able to silently
put them in maintenance without disturbing visitors. Load balancers are
used for servers. And between sites, people are not seeking imbalanced
distribution. If they really do, then it's easy using multiple addresses.

Don't take this as an attack (it is not), but you accused Roy of not
knowing SRV, but I think that you're not much experienced with web
infrastructures and that may be why you think that SRV is the *only*
solution to many problems that have been dealt with for ages.

I think your proposal could make sense with SHOULDs instead of MUSTs,
eventhough it's unlikely that it will be used for the latency reasons
that were suggested.

Regards,
Willy


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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Sun, Jul 24, 2011 at 09:02:59PM +0200, Iñaki Baz Castillo wrote:
> 2011/7/24 Willy Tarreau <w@1wt.eu>:
> > But that's not what I meant, I meant that DNS is not the only solution
> > to resolve host names. WINS, NIS and /etc/hosts are usable too. When I
> > was a student in 94, we had all our passwords and hostnames in NIS and
> > no DNS was configured. It worked like a charm. DNS is not something
> > mandatory at all for many protocols. It just happens to be the standard
> > over the public Internet.
> 
> Ok, I get your point now :)
> But, do current webbrosers resolve names using anything but DNS A?
> (well, I know that they use /etc/hosts). Anyhow, WINS / NIS /etc/hosts
> and such stuff just maps a hostname into a single IP. It's the
> equivalent of a DNS A resource record. Think about locating a mail
> server (MX is required), you need a DNS server.

... or a static entry (the "Smart Relay host" entry you see in on the DS
line in sendmail.cf). That's the case of any mail client in a campus, they
don't use DNS to send outgoing mail. And even within the mail relays and
servers, it's quite common to have a relaydomains file to map domains to
servers.

Willy


From w@1wt.eu  Sun Jul 24 12:56:05 2011
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From: Willy Tarreau <w@1wt.eu>
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Subject: Re: [hybi] Alternative for SRV proposal
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On Sun, Jul 24, 2011 at 09:16:58PM +0200, Iñaki Baz Castillo wrote:
> Hi, it's clear enough for me that DNS SRV will be not implemented in
> web browsers for WebSocket URIs resolution. I understand some of the
> reasons (not all however).
> 
> So I'm thinking about a future (extension) for WebSocket protocol and
> API (sorry, I don't know yet the propose WS API so I use pseudo-code):
> 
> 
> a) Let's suppose a JavaScript code contains a WS connection, something
> like this:
> 
>     var mysocket = new WebSocket("ws://www.mydomain.org", null);
> 
> Note that there is a second parameter that is null. It would mean (as
> the current spec states) that a DNS A query should be performed for
> the given URI (or not if there is already a TCP connection).
> 
> 
> b) I propose an optional second argument:
> 
>     var mysocket = new WebSocket("ws://www.mydomain.org", "ws://mydomain.org");
> 
> If the browser supports the new WS extension I'm suggesting, the
> browser could *decide* whenever perform DNS A for the first URI or a
> DNS SRV for the second URI. For example, WS clients within mobile
> devices could decide to always use the first URI so just DNS A would
> be done (or not if it was already resolved and connected). In the
> other side, desktop web browsers implementing this new extension would
> use, by default, the second URI with all the SRV stuff.

It is even worse, because it's error-prone and will make it harder for
site authors to maintain. And you'd have to convince the WHATWG too for
an API change.

Instead, why not keep the single parameter ?

Eg:
 $ host -t srv _ws-server._tcp.www.mydomain.org
 _ws-server._tcp.www.mydomain.org has SRV record 5 0 5269 ws1.mydomain.org
 _ws-server._tcp.www.mydomain.org has SRV record 5 0 5269 ws2.mydomain.org
 _ws-server._tcp.www.mydomain.org has SRV record 5 0 5269 ws3.mydomain.org

 $ host -t aaaa www.mydomain.org
 www.mydomain.org has no AAAA record

 $ host -t a www.mydomain.org
 www.mydomain.org is an alias for ws1.mydomain.org
 www.mydomain.org is an alias for ws2.mydomain.org
 www.mydomain.org is an alias for ws3.mydomain.org

The browsers configured for SRV would first run the SRV request, and fall
back to AAAA/A while the other ones would skip SRV.

We could even refine the process : you could make it mandatory that for an
SRV record to be presented, the main host would necessarily have a CNAME
record. The idea is that most of the time, CNAME records will point to the
same entries as SRV records. Thus you can avoid the slowdown for most
situations that way :

   1) resolve www.mydomain.org
   2) if it has at least a CNAME and if SRV is enabled, then retry with SRV
   3) if not, or if SRV fails, use AAAA/A as returned

That way you can make use of SRV by default without impacting too
many sites (only those with a CNAME).

I'm not saying it's beautiful, but it can be a tradeoff between costs and
chances of adoption.

Regards,
Willy


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Date: Sun, 24 Jul 2011 21:18:40 +0100
From: Dave Cridland <dave@cridland.net>
To: Willy Tarreau <w@1wt.eu>, Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>, =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Sun Jul 24 19:59:49 2011, Willy Tarreau wrote:
> On Sun, Jul 24, 2011 at 08:30:11PM +0200, Iñaki Baz Castillo wrote:
> > 2011/7/24 John Tamplin <jat@google.com>:
> > >> ~100 ms (if the DNS server is not local and there is no DNS  
> cache for
> > >> the given domain). And just during the WS connection, no more.  
> Taking
> > >> into account that a WS will be *usually* connected after  
> loading a web
> > >> page, such ~100ms in a non-full-realtime protocol is  
> insignificant.
> > >
> > > Wait, are you really saying that 100ms connect latency is  
> unimportant?
> >
> > Open the fastest web page and tell me how long it takes. Probably  
> you
> > have performed a DNS A query. I don't think that a xtra DNS query
> > would be the bottleneck, never.
> 
> On lossy networks such as 3G, they definitely are. A lost UDP  
> packet is
> not retransmitted nor signaled as lost, so the browser has to  
> retry. However,
> once the connection is established to the server, most losses are  
> more or
> less smoothed by TCP extensions such as SACK. So yes, it can take  
> several
> seconds to just resolve a host and then only a few hundreds of ms  
> to retrieve
> the objects. I've observed it.

I think what might be colouring your opinion regarding DNS resolution  
times on mobile is the difference between the first and subsequent  
RTTs.

3G sessions, in a reasonable area, drop to around 100-150ms, although  
they can go up to 300ms or higher if the network condition  
deteriorates. However, the setup of DCH, the radio state normally  
used for internet traffic (and needed for DNS requests and  
responses), takes a healthy number of round-trips, so that the first  
RTT is about three times longer.

Moreover, it's not clear to me that SRV lookup always (or even  
commonly) adds an additional round-trip. Take an XMPP client SRV  
lookup to my own server:

$ dig _xmpp-client._tcp.dave.cridland.net SRV
;; QUESTION SECTION:
;_xmpp-client._tcp.dave.cridland.net. IN	SRV

;; ANSWER SECTION:
_xmpp-client._tcp.dave.cridland.net. 86400 IN SRV 5 2 5222  
peirce.dave.cridland.net.

;; AUTHORITY SECTION:
dave.cridland.net.	86400	IN	NS	br.cridland.net.

;; ADDITIONAL SECTION:
peirce.dave.cridland.net. 86400	IN	A	217.155.137.61
peirce.dave.cridland.net.  
86400	IN	AAAA	2001:470:1f09:882:2e0:81ff:fe29:d16a

Note that the addresses of the actual server are returned in the  
additional section. My understanding is that in practical terms  
this'd always happen for in-zone cases. (If there's a large number,  
you may not get them all, since they can be discarded without error,  
but it practise you're likely to).

Finally, as I've said before, I think that any overhead involved is  
going to be swallowed up in the noise of general session startup in  
the WebSocket case. I do appreciate things are at the very least  
perceived as different in the HTTP case, although I think SRV would  
help solve issues (like off-site failover) there, too, but I think  
the moment you have long-running stateful sessions, you'll end up  
with the same impact to user experience of a few extra RTTs at  
startup as is seen in XMPP, SIP, and so on - that is, none.

100ms extra on a 100ms request/response would be bad, I agree, but  
that's not what we're talking about.

Dave.
-- 
Dave Cridland - mailto:dave@cridland.net - xmpp:dwd@dave.cridland.net
  - acap://acap.dave.cridland.net/byowner/user/dwd/bookmarks/
  - http://dave.cridland.net/
Infotrope Polymer - ACAP, IMAP, ESMTP, and Lemonade

From ibc@aliax.net  Sun Jul 24 13:21:22 2011
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Subject: Re: [hybi] Alternative for SRV proposal
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2011/7/24 Willy Tarreau <w@1wt.eu>:
> We could even refine the process : you could make it mandatory that for a=
n
> SRV record to be presented, the main host would necessarily have a CNAME
> record. The idea is that most of the time, CNAME records will point to th=
e
> same entries as SRV records. Thus you can avoid the slowdown for most
> situations that way :
>
> =C2=A0 1) resolve www.mydomain.org
> =C2=A0 2) if it has at least a CNAME and if SRV is enabled, then retry wi=
th SRV
> =C2=A0 3) if not, or if SRV fails, use AAAA/A as returned
>
> That way you can make use of SRV by default without impacting too
> many sites (only those with a CNAME).

But that would require two DNS queries (in case of CNAME), am I wrong?
and that seems to be a problem (due to explanations given in the other
thread).

Why not just making SRV optional but ensuring that the domain in the
URI also has a A/AAAA resolution? This is:

ws://mydomain.org

1) ~# host -t A mydomain.org
    1.2.3.4

2) ~# host -t srv _ws._tcp.mydomain.org
    1 50 server01.mydomain.org
    1 50 server02.mydomain.org
    2  0 server-backup.mydomain.org

2a) ~# host -t A server01.mydomain.org
       1.2.3.4

2b) ~# host -t A server02.mydomain.org
       1.2.3.5

2b) ~# host -t A server-backup.mydomain.org
       9.9.9.9


Clients with no SRV support (or disabled as mobile browsers) should
just perform step 1 while other WS clients would/could try step 2 (and
2a, 2b, 2c...).


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From derhoermi@gmx.net  Sun Jul 24 13:22:38 2011
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To: Greg Wilkins <gregw@intalio.com>
Date: Sun, 24 Jul 2011 22:22:30 +0200
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Subject: Re: [hybi] deflate-stream and masking
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* Greg Wilkins wrote:
>I took a days worth of traffic from an IRC channel and wrapped it up
>as JSON messages sent as websocket frames.
>There were 487 message that looked like:
>
>     {channel:"#webtide", username:"tbecker", text:"joakime: jenkins
>had issues pulling from github a couple of times  last week"}
>
>As an unmasked WS stream, it was 50675 bytes, and as a masked stream
>is was 52623 bytes.
>I then compressed both these streams with gzip and got 13306 bytes for
>unmasked and 51704 bytes for the masked!!!!

Deflate streams consist of blocks and the blocks consist of tables and
symbols and the symbols represent either bits from what is compressed,
most of those bits are affected by the mask, and back-references that
instruct a decoder to copy bytes -- which are not affected by the mask.
If you create two files, one has "1234" repeating, the other "abcd" re-
peating, and `gzip` them, only a few bytes will be different. You can
trivially create a Websocket stream with some byte sequence repeating.

If you want to have "GET ..." on the wire, well, that can be just some

  <append x bytes from position y to the output>
  <append x bytes from position y to the output>
  <append x bytes from position y to the output>
  ...

bits in the deflate stream. That's not easy to force because encoders
have many options in how they create the stream regardless of masking,
and masking does make it more difficult, but you are greatly aided by,
for instance, having all bytes in the mask be the same, which is not
generally the case if you try to subvert masking without compression.

Without some accepted proof to that effect, the Working Group cannot
claim that masking notably changes the security properties of the pro-
tocol under deflate-stream, which means it would be safe to use only
where it would be safe to use the same mask for all frames, in which
case you no longer have the "compresses poorly" problem, and it would
clearly rule out implementing it in web browsers that "need" masking.

If the base protocol specification were to have a feature that cannot
be implemented by a very large segment of implementations for security
reasons, it would have to have extremely important benefits; deflate-
stream however doesn't have them. Moreover, there does not seem to be
any kind of consensus that extensions should be able to arbitrarily
modify the on-the-wire format. Either way it should not be in the base
protocol specification.
-- 
Björn Höhrmann · mailto:bjoern@hoehrmann.de · http://bjoern.hoehrmann.de
Am Badedeich 7 · Telefon: +49(0)160/4415681 · http://www.bjoernsworld.de
25899 Dagebüll · PGP Pub. KeyID: 0xA4357E78 · http://www.websitedev.de/ 

From w@1wt.eu  Sun Jul 24 13:42:41 2011
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Date: Sun, 24 Jul 2011 22:42:36 +0200
From: Willy Tarreau <w@1wt.eu>
To: Dave Cridland <dave@cridland.net>
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References: <CAP992=FJymFPKcPVWrF-LkcEtNUz=Kt9L_ex+kLtjiGjL1T46w@mail.gmail.com> <4E28A51F.4020704@callenish.com> <9031.1311286867.939466@puncture> <4E28BA9D.6010501@callenish.com> <CAP992=GedTEfimykCWwdwm=BsZdwFRJO36EO0a_o7iejURJ+tQ@mail.gmail.com> <CALiegfnwNyYDy=SyrqHJXV0ZreADb7F8QySvgdHofW9Hm9miZQ@mail.gmail.com> <CABLsOLCgxDNDs54GUybBzmG8a6rNPzXzFcwf_OU25j5L+AobaQ@mail.gmail.com> <CALiegfnp+kNqnU-_x2ZBZnN8fzP=4+6WO=QPjw1rJzvZmSXEAg@mail.gmail.com> <20110724185949.GB22405@1wt.eu> <9031.1311538720.416128@puncture>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Sun, Jul 24, 2011 at 09:18:40PM +0100, Dave Cridland wrote:
> >> Open the fastest web page and tell me how long it takes. Probably  
> >you
> >> have performed a DNS A query. I don't think that a xtra DNS query
> >> would be the bottleneck, never.
> >
> >On lossy networks such as 3G, they definitely are. A lost UDP  
> >packet is
> >not retransmitted nor signaled as lost, so the browser has to  
> >retry. However,
> >once the connection is established to the server, most losses are  
> >more or
> >less smoothed by TCP extensions such as SACK. So yes, it can take  
> >several
> >seconds to just resolve a host and then only a few hundreds of ms  
> >to retrieve
> >the objects. I've observed it.
> 
> I think what might be colouring your opinion regarding DNS resolution  
> times on mobile is the difference between the first and subsequent  
> RTTs.

Note that in the point above I was not even talking about RTT but
explaining to Iñaki that sometimes DNS can be slower than the rest
of the transfer due to losses and slow retransmits.

> 3G sessions, in a reasonable area, drop to around 100-150ms, although  
> they can go up to 300ms or higher if the network condition  
> deteriorates.

I agree. However it can get a lot worse as soon as you have even just a
little bit of traffic on your link (eg: objects fetched in parallel).

> However, the setup of DCH, the radio state normally  
> used for internet traffic (and needed for DNS requests and  
> responses), takes a healthy number of round-trips, so that the first  
> RTT is about three times longer.

Yes and depending on the equipments vendor, you may even experience
losses during this phase for several seconds. But I was really not
talking about such issues that add to the bad experience and should
not be considered as the norm.

> Moreover, it's not clear to me that SRV lookup always (or even  
> commonly) adds an additional round-trip. Take an XMPP client SRV  
> lookup to my own server:
> 
> $ dig _xmpp-client._tcp.dave.cridland.net SRV
> ;; QUESTION SECTION:
> ;_xmpp-client._tcp.dave.cridland.net. IN	SRV
> 
> ;; ANSWER SECTION:
> _xmpp-client._tcp.dave.cridland.net. 86400 IN SRV 5 2 5222  
> peirce.dave.cridland.net.
> 
> ;; AUTHORITY SECTION:
> dave.cridland.net.	86400	IN	NS	br.cridland.net.
> 
> ;; ADDITIONAL SECTION:
> peirce.dave.cridland.net. 86400	IN	A	217.155.137.61
> peirce.dave.cridland.net.  
> 86400	IN	AAAA	2001:470:1f09:882:2e0:81ff:fe29:d16a
> 
> Note that the addresses of the actual server are returned in the  
> additional section. My understanding is that in practical terms  
> this'd always happen for in-zone cases. (If there's a large number,  
> you may not get them all, since they can be discarded without error,  
> but it practise you're likely to).

That's very interesting. For an unknown reason, doing the same request
from here using either dig or host only retrieves the answer and not
the authority nor the additional sections.

> Finally, as I've said before, I think that any overhead involved is  
> going to be swallowed up in the noise of general session startup in  
> the WebSocket case. I do appreciate things are at the very least  
> perceived as different in the HTTP case, although I think SRV would  
> help solve issues (like off-site failover) there, too, but I think  
> the moment you have long-running stateful sessions, you'll end up  
> with the same impact to user experience of a few extra RTTs at  
> startup as is seen in XMPP, SIP, and so on - that is, none.
>
> 100ms extra on a 100ms request/response would be bad, I agree, but  
> that's not what we're talking about.

To ensure nobody gets me wrong, I'm certain this can help solve issues
*if this is optional*. If it becomes a MUST, then the negative effects
will override the positive ones. In my opinion, the client should decide
whether to enable it or not. If architectures are built with that in
mind, then it will not be an absolute requirement, and migration to SRV
could happen smoothly just like migration to make the Host header mandatory
happened. What is needed is a way to avoid the extra DNS lookup in most
cases where the SRV record is not there. Coupling it with CNAME as I
proposed could be a reasonable solution. Surely other solutions are better.

But we have to keep in mind that for SRV to work, it cannot be made
mandatory because existing infrastructure simply does not support it.
This point alone is enough to kill the mandatory requirement. And once
optional, we have to promote its adoption with a perceived benefit for
the end user with almost zero cost. Mixing all those points with the
use cases of WS leaves little place for SRV, but maybe someone wants
to work on it and identify the *real* benefits then emit a draft.

Regards,
Willy


From w@1wt.eu  Sun Jul 24 13:51:24 2011
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From: Willy Tarreau <w@1wt.eu>
To: =?iso-8859-1?Q?I=F1aki?= Baz Castillo <ibc@aliax.net>
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Subject: Re: [hybi] Alternative for SRV proposal
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On Sun, Jul 24, 2011 at 10:21:20PM +0200, Iñaki Baz Castillo wrote:
> 2011/7/24 Willy Tarreau <w@1wt.eu>:
> > We could even refine the process : you could make it mandatory that for an
> > SRV record to be presented, the main host would necessarily have a CNAME
> > record. The idea is that most of the time, CNAME records will point to the
> > same entries as SRV records. Thus you can avoid the slowdown for most
> > situations that way :
> >
> >   1) resolve www.mydomain.org
> >   2) if it has at least a CNAME and if SRV is enabled, then retry with SRV
> >   3) if not, or if SRV fails, use AAAA/A as returned
> >
> > That way you can make use of SRV by default without impacting too
> > many sites (only those with a CNAME).
> 
> But that would require two DNS queries (in case of CNAME), am I wrong?
> and that seems to be a problem (due to explanations given in the other
> thread).

Exact, but that would only be the case where the entry is declared as a
CNAME, not always. CNAME will be needed as a fallback for SRV if SRV is
made optional anyway.

> Why not just making SRV optional but ensuring that the domain in the
> URI also has a A/AAAA resolution? This is:
> 
> ws://mydomain.org
> 
> 1) ~# host -t A mydomain.org
>     1.2.3.4
> 
> 2) ~# host -t srv _ws._tcp.mydomain.org
>     1 50 server01.mydomain.org
>     1 50 server02.mydomain.org
>     2  0 server-backup.mydomain.org
> 
> 2a) ~# host -t A server01.mydomain.org
>        1.2.3.4
> 
> 2b) ~# host -t A server02.mydomain.org
>        1.2.3.5
> 
> 2b) ~# host -t A server-backup.mydomain.org
>        9.9.9.9
> 
> 
> Clients with no SRV support (or disabled as mobile browsers) should
> just perform step 1 while other WS clients would/could try step 2 (and
> 2a, 2b, 2c...).

But that would cause the second request (SRV) to be performed all the
time when it is enabled. You can be sure that many users will disable
it when they notice that their web is faster without.

With the CNAME trick, the second request only happens if the first one
returns a CNAME.

In fact, I think we're missing a "SERVICES" record which would return
for a given host the list of services that are known to run on it. That
way you could run "host -a mydomain.org" and see :
;; ANSWER SECTION:
mydomain.org.       159     IN      A 1.2.3.4
mydomain.org.       159     IN      SERVICES _ws._tcp.
mydomain.org.       159     IN      SERVICES _http._tcp.

That way with a single request you could get the address *and* be informed
of the running services, making it worth trying again for those.

Regards,
Willy


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In message <20110724183343.GY22405@1wt.eu>, Willy Tarreau writes:
> On Sun, Jul 24, 2011 at 08:25:05PM +0200, I=F1aki Baz Castillo wrote:
> > 2011/7/24 Willy Tarreau <w@1wt.eu>:
> > >> > Making an additional DNS request and a connection come with a cost.
> > >>
> > >> http://tools.ietf.org/html/draft-ibc-websocket-dns-srv-02#section-5.2
> > >
> > > You still need the DNS request : the client does an A/AAAA request for
> > > the HTTP host, then if you ask it to use an SRV record for the WS conne=
> ction,
> > > it must perform that request too, even if it's to conclude that it can =
> reuse
> > > the idle connection.
> > =
> 
> > Ok, but I don't consider a xtra DNS query to be so hard.
> 
> I had to perform sites analysis for a customer a few months ago. Many
> web pages nowadays have between 100 and 200 objects to fetch, spread
> over up to 25-30 host names. If you take even only 100ms for each of
> them, you're at 3 additional seconds to display the page. Granted those
> requests are not WS and only HTTP, but as I said, SRV for WS won't work
> before it works with HTTP, at least due to proxies.

No. It takes 100ms extra.  Only if you serialise all the lookups
does it take 3 extra seconds.  If you are not doing the DNS lookups
in parallel today you are already doing a dis-service to your
customers.

Adding a SRV lookup should add 0ms if it isn't there as you should be
making A, AAAA and SRV lookups in parallel.  Non-existance is as
cachable as existance is.

> But those 3 extra seconds will always be considered a good reason not
> to make SRV mandatory for HTTP. The web is degrading very quickly due
> to poor practices, and we should be careful not to suggest to make it
> even worse.
> 
> Regards,
> Willy
> 
> _______________________________________________
> Ietf mailing list
> Ietf@ietf.org
> https://www.ietf.org/mailman/listinfo/ietf
-- 
Mark Andrews, ISC
1 Seymour St., Dundas Valley, NSW 2117, Australia
PHONE: +61 2 9871 4742                 INTERNET: marka@isc.org

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--000e0cd32b9480f27904a8da6ec3
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On Sun, Jul 24, 2011 at 8:46 PM, Mark Andrews <marka@isc.org> wrote:

> Adding a SRV lookup should add 0ms if it isn't there as you should be
> making A, AAAA and SRV lookups in parallel.  Non-existance is as
> cachable as existance is.
>

But you have to wait until the SRV returns even if the A/AAAA response comes
back first, so there is a non-zero cost even with an optimal implementation.
 I am not arguing whether the cost is worth the benefits, only that there is
a cost in any case.

-- 
John A. Tamplin
Software Engineer (GWT), Google

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<div class=3D"gmail_quote">On Sun, Jul 24, 2011 at 8:46 PM, Mark Andrews <s=
pan dir=3D"ltr">&lt;<a href=3D"mailto:marka@isc.org">marka@isc.org</a>&gt;<=
/span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex;">

Adding a SRV lookup should add 0ms if it isn&#39;t there as you should be<b=
r>
making A, AAAA and SRV lookups in parallel. =C2=A0Non-existance is as<br>
cachable as existance is.<br></blockquote><div><br></div><div>But you have =
to wait until the SRV returns even if the A/AAAA response comes back first,=
 so there is a non-zero cost even with an optimal implementation. =C2=A0I a=
m not arguing whether the cost is worth the benefits, only that there is a =
cost in any case.=C2=A0</div>

</div><br>-- <br>John A. Tamplin<br>Software Engineer (GWT), Google<br>

--000e0cd32b9480f27904a8da6ec3--

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2011/7/24 Willy Tarreau <w@1wt.eu>:
>> But that would require two DNS queries (in case of CNAME), am I wrong?
>> and that seems to be a problem (due to explanations given in the other
>> thread).
>
> Exact, but that would only be the case where the entry is declared as a
> CNAME, not always. CNAME will be needed as a fallback for SRV if SRV is
> made optional anyway.

Hummm, AFAIK some local DNS resolvers automatically resolve the CNAME
domain to an IP before passing the DNS result to the application, so
the application would not be aware of the existence of a CNAME record.




> In fact, I think we're missing a "SERVICES" record which would return
> for a given host the list of services that are known to run on it. That
> way you could run "host -a mydomain.org" and see :
> ;; ANSWER SECTION:
> mydomain.org. =C2=A0 =C2=A0 =C2=A0 159 =C2=A0 =C2=A0 IN =C2=A0 =C2=A0 =C2=
=A0A 1.2.3.4
> mydomain.org. =C2=A0 =C2=A0 =C2=A0 159 =C2=A0 =C2=A0 IN =C2=A0 =C2=A0 =C2=
=A0SERVICES _ws._tcp.
> mydomain.org. =C2=A0 =C2=A0 =C2=A0 159 =C2=A0 =C2=A0 IN =C2=A0 =C2=A0 =C2=
=A0SERVICES _http._tcp.
>
> That way with a single request you could get the address *and* be informe=
d
> of the running services, making it worth trying again for those.

I think you mean NAPTR records ;)

Those NAPTR records are used in SIP protocol to discover which
transport (UDP, TCP, TLS-TCP, SCTP...) to use (by preference of the
destination domain administrator).

In WS it could be something like:

  example.org  IN   NAPTR 1 0 "S" "WS+D2T" "" _ws._tcp.example.org.


But this would be an extra DNS query so there would be:

a) NAPTR + SRV +`A/AAAA

or

b) NAPTR + A/AAAA

It does not fit well with constrains exposed in the other mail thread.


Regards.
--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From marka@isc.org  Sun Jul 24 18:13:38 2011
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In message <20110724193230.GE22405@1wt.eu>, Willy Tarreau writes:
> On Sun, Jul 24, 2011 at 09:02:59PM +0200, I=F1aki Baz Castillo wrote:
> > 2011/7/24 Willy Tarreau <w@1wt.eu>:
> > > But that's not what I meant, I meant that DNS is not the only solution
> > > to resolve host names. WINS, NIS and /etc/hosts are usable too. When I
> > > was a student in 94, we had all our passwords and hostnames in NIS and
> > > no DNS was configured. It worked like a charm. DNS is not something
> > > mandatory at all for many protocols. It just happens to be the standard
> > > over the public Internet.
> > =
> 
> > Ok, I get your point now :)
> > But, do current webbrosers resolve names using anything but DNS A?
> > (well, I know that they use /etc/hosts). Anyhow, WINS / NIS /etc/hosts
> > and such stuff just maps a hostname into a single IP. It's the
> > equivalent of a DNS A resource record. Think about locating a mail
> > server (MX is required), you need a DNS server.
> 
> ... or a static entry (the "Smart Relay host" entry you see in on the DS
> line in sendmail.cf). That's the case of any mail client in a campus, they
> don't use DNS to send outgoing mail. And even within the mail relays and
> servers, it's quite common to have a relaydomains file to map domains to
> servers.
> 
> Willy

But the Smart Relay host uses MX records.  The Smart Relay host should
be seen as the same as a HTTP proxy.
 
> _______________________________________________
> Ietf mailing list
> Ietf@ietf.org
> https://www.ietf.org/mailman/listinfo/ietf
-- 
Mark Andrews, ISC
1 Seymour St., Dundas Valley, NSW 2117, Australia
PHONE: +61 2 9871 4742                 INTERNET: marka@isc.org

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--Apple-Mail-7-394893780
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On Jul 24, 2011, at 3:33 AM, Mark Andrews wrote:

> How do you solve the problem of hosting just "http://example.com/"
> on "s1.joes-web-service.com" and not redirect everything else at
> example.com?  People have been complaining about this for about as
> long as the web has existed.

Well, in a way, that's what NAPTR was for.  All of the UR
i resolution mechanisms (equally applicable to DNS-based URIs) that were =
developed and never really used grew out of the original realization in =
the early 1990s that CERN could stop hosting the original web pages if =
it wanted to, and there was no way to keep those links from going stale.

The problem never went away, but the DNS-based solutions were defined a =
long time ago and never used.   By now, I think the market has long =
since decided.  For better or worse, the mechanism the market chose to =
use with the web was HTTP redirects.

Keith


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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><div><div>On Jul 24, 2011, at 3:33 AM, Mark Andrews wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
font-family: Helvetica; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; ">How do you =
solve the problem of hosting just "<a =
href=3D"http://example.com/">http://example.com/</a>"<br>on "<a =
href=3D"http://s1.joes-web-service.com/">s1.joes-web-service.com</a>" =
and not redirect everything else at<br><a =
href=3D"http://example.com/">example.com</a>? &nbsp;People have been =
complaining about this for about as<br>long as the web has =
existed.</span></blockquote></div><br><div>Well, in a way, that's what =
NAPTR was for. &nbsp;All of the UR</div><div>i resolution mechanisms =
(equally applicable to DNS-based URIs) that were developed and never =
really used grew out of the original realization in the early 1990s that =
CERN could stop hosting the original web pages if it wanted to, and =
there was no way to keep those links from going =
stale.</div><div><br></div><div>The problem never went away, but the =
DNS-based solutions were defined a long time ago and never used. &nbsp; =
By now, I think the market has long since decided. &nbsp;For better or =
worse, the mechanism the market chose to use with the web was HTTP =
redirects.</div><div><br></div><div>Keith</div><div><br></div></body></htm=
l>=

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In message <CABLsOLD-KM6DnR8HvfGH8N1M=1bZ4z8zus0YDczaXSfocBQ+Bw@mail.gmail.com>
, John Tamplin writes:
> 
> On Sun, Jul 24, 2011 at 8:46 PM, Mark Andrews <marka@isc.org> wrote:
> 
> > Adding a SRV lookup should add 0ms if it isn't there as you should be
> > making A, AAAA and SRV lookups in parallel.  Non-existance is as
> > cachable as existance is.
> 
> But you have to wait until the SRV returns even if the A/AAAA response comes
> back first, so there is a non-zero cost even with an optimal implementation.
> I am not arguing whether the cost is worth the benefits, only that there is
> a cost in any case.

In 99.9999% of cases the three answers will come in within 1ms of
each other and you may be waiting on the A or AAAA record.  It
really doesn't take any longer to answer a SRV query compared to a
A or AAAA query.

Mark
-- 
Mark Andrews, ISC
1 Seymour St., Dundas Valley, NSW 2117, Australia
PHONE: +61 2 9871 4742                 INTERNET: marka@isc.org

From w@1wt.eu  Sun Jul 24 21:29:30 2011
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From: Willy Tarreau <w@1wt.eu>
To: Mark Andrews <marka@isc.org>
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On Mon, Jul 25, 2011 at 10:46:58AM +1000, Mark Andrews wrote:
> Adding a SRV lookup should add 0ms if it isn't there as you should be
> making A, AAAA and SRV lookups in parallel.

This does not work for a simple reason : you have to perform the lookups
on the SRV response just as you would do with a CNAME :

$ host -t srv _xmpp-server._tcp.gmail.com
_xmpp-server._tcp.gmail.com has SRV record 20 0 5269 xmpp-server1.l.google.com.
_xmpp-server._tcp.gmail.com has SRV record 5 0 5269 xmpp-server.l.google.com.
_xmpp-server._tcp.gmail.com has SRV record 20 0 5269 xmpp-server2.l.google.com.
_xmpp-server._tcp.gmail.com has SRV record 20 0 5269 xmpp-server3.l.google.com.
_xmpp-server._tcp.gmail.com has SRV record 20 0 5269 xmpp-server4.l.google.com.

And only now I can resolve xmpp-server.l.google.com and get its real address.
That's why I was saying that SRV and CNAME really are equivalent in terms of
latency, because both of them require an additional round trip.

And that is assuming that all the hosts are referenced on the main page, which
in practice is not true because a number of them are also in additional CSS
and JS.

Now if I want to be really picky, I'd add that at 100 bytes on average per
request, sending 30 additional requests means a total of 3kB of data on the
uplink, which require :
  - 1.5 second of uplink traffic on a 3G HDSPA link with 16kbps uplink
  - 375 ms of uplink traffic on a 3G HDSPA link with 64kbps uplink
  - 187 ms of uplink traffic on an ADSL 512/128

I'm not trying to dismiss the usefulness of the mechanism, I just don't
like to hear that doing something additional comes at absolutely zero cost.

Regards,
Willy


From w@1wt.eu  Sun Jul 24 21:32:09 2011
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From: Willy Tarreau <w@1wt.eu>
To: Mark Andrews <marka@isc.org>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Mon, Jul 25, 2011 at 11:12:46AM +1000, Mark Andrews wrote:
> > > But, do current webbrosers resolve names using anything but DNS A?
> > > (well, I know that they use /etc/hosts). Anyhow, WINS / NIS /etc/hosts
> > > and such stuff just maps a hostname into a single IP. It's the
> > > equivalent of a DNS A resource record. Think about locating a mail
> > > server (MX is required), you need a DNS server.
> > 
> > ... or a static entry (the "Smart Relay host" entry you see in on the DS
> > line in sendmail.cf). That's the case of any mail client in a campus, they
> > don't use DNS to send outgoing mail. And even within the mail relays and
> > servers, it's quite common to have a relaydomains file to map domains to
> > servers.
> > 
> > Willy
> 
> But the Smart Relay host uses MX records.  The Smart Relay host should
> be seen as the same as a HTTP proxy.

Exactly, which means that the browser can't be required to rely on any
information from the DNS since it does not necessarily have access to it
and other components which are not under our control might behave in a
different manner than what we'd like.

Regards,
Willy


From w@1wt.eu  Sun Jul 24 21:37:56 2011
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From: Willy Tarreau <w@1wt.eu>
To: =?iso-8859-1?Q?I=F1aki?= Baz Castillo <ibc@aliax.net>
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Subject: Re: [hybi] Alternative for SRV proposal
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On Mon, Jul 25, 2011 at 03:16:58AM +0200, Iñaki Baz Castillo wrote:
> 2011/7/24 Willy Tarreau <w@1wt.eu>:
> >> But that would require two DNS queries (in case of CNAME), am I wrong?
> >> and that seems to be a problem (due to explanations given in the other
> >> thread).
> >
> > Exact, but that would only be the case where the entry is declared as a
> > CNAME, not always. CNAME will be needed as a fallback for SRV if SRV is
> > made optional anyway.
> 
> Hummm, AFAIK some local DNS resolvers automatically resolve the CNAME
> domain to an IP before passing the DNS result to the application, so
> the application would not be aware of the existence of a CNAME record.

I'm not aware of that and it sounds strange to me.

> > In fact, I think we're missing a "SERVICES" record which would return
> > for a given host the list of services that are known to run on it. That
> > way you could run "host -a mydomain.org" and see :
> > ;; ANSWER SECTION:
> > mydomain.org.       159     IN      A 1.2.3.4
> > mydomain.org.       159     IN      SERVICES _ws._tcp.
> > mydomain.org.       159     IN      SERVICES _http._tcp.
> >
> > That way with a single request you could get the address *and* be informed
> > of the running services, making it worth trying again for those.
> 
> I think you mean NAPTR records ;)
> 
> Those NAPTR records are used in SIP protocol to discover which
> transport (UDP, TCP, TLS-TCP, SCTP...) to use (by preference of the
> destination domain administrator).
> 
> In WS it could be something like:
> 
>   example.org  IN   NAPTR 1 0 "S" "WS+D2T" "" _ws._tcp.example.org.

OK, I didn't know them, so that's fine this way.

> But this would be an extra DNS query so there would be:
> 
> a) NAPTR + SRV +`A/AAAA
> 
> or
> 
> b) NAPTR + A/AAAA
> 
> It does not fit well with constrains exposed in the other mail thread.

It does fit well as something at the option of the end user, because as
Mark says, one can send the NAPTR + A + AAAA "at zero cost" (assuming
the uplink is infinite). It does not require extra round trips, just
extra bandwidth and some extra synchronization from the client.

Willy


From w@1wt.eu  Sun Jul 24 22:01:08 2011
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From: Willy Tarreau <w@1wt.eu>
To: Mark Andrews <marka@isc.org>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Mon, Jul 25, 2011 at 02:50:37PM +1000, Mark Andrews wrote:
> 
> In message <20110725042921.GJ22405@1wt.eu>, Willy Tarreau writes:
> > On Mon, Jul 25, 2011 at 10:46:58AM +1000, Mark Andrews wrote:
> > > Adding a SRV lookup should add 0ms if it isn't there as you should be
> > > making A, AAAA and SRV lookups in parallel.
> > 
> > This does not work for a simple reason : you have to perform the lookups
> > on the SRV response just as you would do with a CNAME :
> 
> Please re-read my claim as your comment is not applicable to it.

Ah OK : "if it isn't there". Sorry for missing it.

Willy


From marka@isc.org  Sun Jul 24 20:21:56 2011
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To: Keith Moore <moore@network-heretics.com>
From: Mark Andrews <marka@isc.org>
References: <20110711140229.17432.23519.idtracker@ietfa.amsl.com> <CALiegfk0zVVRBbOP4ugsVXKmcLnryujP6DZqF6Bu_dC2C3PpeQ@mail.gmail.com> <9031.1311082001.631622@puncture> <CALiegfk_GLAhAf=yEe6hYw2bwtxEwg9aJN+f0Bm9he5QgsRavA@mail.gmail.com> <CAP992=Ft6NwG+rbcuWUP0npwVNHY_znHmXmznBQO_krMo3RT6g@mail.gmail.com> <CALiegfmTWMP3GhS1-k2aoHHXkUkB+eWqV=2+BufuWVR1s2Z-EA@mail.gmail.com> <20110721163910.GA16854@1wt.eu> <CAP992=FrX5VxP2o0JLNoJs8nXXba7wbZ6RN9wBUYC0ZSN_wbAg@mail.gmail.com> <9031.1311270000.588511@puncture> <CALiegf=pYzybvc7WB2QfPg6FKrhLxgzHuP-DpuuMfZYJV6Z7FQ@mail.gmail.com> <B2C17B21-EA8A-4698-8C41-F55A9AA140D4@gbiv.com> <20110724073323.EEAAF121E985@drugs.dv.isc.org> <4B3C19FD-B736-4DA7-9DB5-3D433320DCBC@network-heretics.com>
In-reply-to: Your message of "Sun, 24 Jul 2011 22:47:39 -0400." <4B3C19FD-B736-4DA7-9DB5-3D433320DCBC@network-heretics.com>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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In message <4B3C19FD-B736-4DA7-9DB5-3D433320DCBC@network-heretics.com>, Keith M
oore writes:
> On Jul 24, 2011, at 3:33 AM, Mark Andrews wrote:
> 
> > How do you solve the problem of hosting just "http://example.com/"
> > on "s1.joes-web-service.com" and not redirect everything else at
> > example.com?  People have been complaining about this for about as
> > long as the web has existed.
> 
> Well, in a way, that's what NAPTR was for.  All of the UR
> i resolution mechanisms (equally applicable to DNS-based URIs) that were =
> developed and never really used grew out of the original realization in =
> the early 1990s that CERN could stop hosting the original web pages if =
> it wanted to, and there was no way to keep those links from going stale.

NAPTR is not defined for HTTP.
SRV is not defined for HTTP.
 
> The problem never went away, but the DNS-based solutions were defined a =
> long time ago and never used.

No.  It was explitly NOT defined.

RFC 2782
Applicability Statement

   In general, it is expected that SRV records will be used by clients
   for applications where the relevant protocol specification indicates
   that clients should use the SRV record. Such specification MUST
   define the symbolic name to be used in the Service field of the SRV
   record as described below. It also MUST include security
   considerations. Service SRV records SHOULD NOT be used in the absence
   of such specification.

> By now, I think the market has long =
> since decided.  For better or worse, the mechanism the market chose to =
> use with the web was HTTP redirects.

If the market hasn't has a really opportunity do decide as only one
mechanism has been specified and coded.  Most people don't know of
the existence of SRV or that it would be useful for HTTP.

If the only tool you have is a hammer then you use a hammer even
if a screwdriver would be better.

Mark
-- 
Mark Andrews, ISC
1 Seymour St., Dundas Valley, NSW 2117, Australia
PHONE: +61 2 9871 4742                 INTERNET: marka@isc.org

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To: Keith Moore <moore@network-heretics.com>
From: Mark Andrews <marka@isc.org>
References: <20110711140229.17432.23519.idtracker@ietfa.amsl.com> <CALiegfk0zVVRBbOP4ugsVXKmcLnryujP6DZqF6Bu_dC2C3PpeQ@mail.gmail.com> <9031.1311082001.631622@puncture> <CALiegfk_GLAhAf=yEe6hYw2bwtxEwg9aJN+f0Bm9he5QgsRavA@mail.gmail.com> <CAP992=Ft6NwG+rbcuWUP0npwVNHY_znHmXmznBQO_krMo3RT6g@mail.gmail.com> <CALiegfmTWMP3GhS1-k2aoHHXkUkB+eWqV=2+BufuWVR1s2Z-EA@mail.gmail.com> <20110721163910.GA16854@1wt.eu> <CAP992=FrX5VxP2o0JLNoJs8nXXba7wbZ6RN9wBUYC0ZSN_wbAg@mail.gmail.com> <9031.1311270000.588511@puncture> <CALiegf=pYzybvc7WB2QfPg6FKrhLxgzHuP-DpuuMfZYJV6Z7FQ@mail.gmail.com> <B2C17B21-EA8A-4698-8C41-F55A9AA140D4@gbiv.com> <20110724073323.EEAAF121E985@drugs.dv.isc.org> <4B3C19FD-B736-4DA7-9DB5-3D433320DCBC@network-heretics.com> <20110725032136.A36701222CE2@drugs.dv.isc.org> <3BC48562-6459-4FB9-9806-731AF87FE027@network-heretics.com>
In-reply-to: Your message of "Sun, 24 Jul 2011 23:33:04 -0400." <3BC48562-6459-4FB9-9806-731AF87FE027@network-heretics.com>
Date: Mon, 25 Jul 2011 14:35:42 +1000
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Cc: "Roy T. Fielding" <fielding@gbiv.com>, Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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In message <3BC48562-6459-4FB9-9806-731AF87FE027@network-heretics.com>, Keith M
oore writes:
> On Jul 24, 2011, at 11:21 PM, Mark Andrews wrote:
> 
> >>> How do you solve the problem of hosting just "http://example.com/"
> >>> on "s1.joes-web-service.com" and not redirect everything else at
> >>> example.com?  People have been complaining about this for about as
> >>> long as the web has existed.
> >>=20
> >> Well, in a way, that's what NAPTR was for.  All of the UR
> >> i resolution mechanisms (equally applicable to DNS-based URIs) that =
> were =3D
> >> developed and never really used grew out of the original realization =
> in =3D
> >> the early 1990s that CERN could stop hosting the original web pages =
> if =3D
> >> it wanted to, and there was no way to keep those links from going =
> stale.
> >=20
> > NAPTR is not defined for HTTP.
> > SRV is not defined for HTTP.
> >=20
> >> The problem never went away, but the DNS-based solutions were defined =
> a =3D
> >> long time ago and never used.
> >=20
> > No.  It was explitly NOT defined.
> 
> Ok, fair enough.   Those of us who were working on the DNS-based URI =
> resolution mechanisms realized that they could be applied to http URIs =
> in addition to almost anything else (NAPTR is incredibly flexible if you =
> don't mind doing lots of DNS lookups).  But they were never formally =
> adopted.
> 
> But if you really want to use DNS to do redirects for http: URIs (or for =
> that matter ws: URIs or almost any other kind of URI), NAPTR was =
> tailor-made to do that.  SRV was not.

"_http._tcp.example.com SRV 100 0 80 <server>" is not a redirect.
The http client still issues "Host: example.com" not "Host: <server>".
If you want to do DNS level redirect of "www.example.com" to
"example.com" then NAPTR would be the way to do that and the http
client issues "Host: example.com" instead of "Host: www.example.com".

If web browers were using CNAME records correctly, i.e. as aliases,
then they would be treated as a DNS level redirect not as "return
the address of CNAME target but otherwise ignore that this is a
alias".  Doing this has all sorts if implications.  A lot of the
IDN issues are a direct result of HTTP clients/adminstrators abusing
CNAME.

Mark
-- 
Mark Andrews, ISC
1 Seymour St., Dundas Valley, NSW 2117, Australia
PHONE: +61 2 9871 4742                 INTERNET: marka@isc.org

From marka@isc.org  Sun Jul 24 21:51:26 2011
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To: Willy Tarreau <w@1wt.eu>
From: Mark Andrews <marka@isc.org>
References: <4E28A51F.4020704@callenish.com> <CALiegf=4K2oWfmZjGMD7J_jyaDtS3i+Mu7R0Wh75Rr+MrQCjtw@mail.gmail.com> <20110722054345.GE18126@1wt.eu> <9031.1311500145.687172@puncture> <20110724105223.GL22405@1wt.eu> <CALiegfkTVg2=k4d8rxmpqXmaRUihRmhtgfF4QRUTAKic7gBk5w@mail.gmail.com> <20110724121147.GR22405@1wt.eu> <CALiegf=GDDKdXOXgz3oognh6=qRDKFUSrRfLOtOoUucAxr4p3w@mail.gmail.com> <20110724183343.GY22405@1wt.eu> <20110725004658.A3AE71221C03@drugs.dv.isc.org> <20110725042921.GJ22405@1wt.eu>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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In message <20110725042921.GJ22405@1wt.eu>, Willy Tarreau writes:
> On Mon, Jul 25, 2011 at 10:46:58AM +1000, Mark Andrews wrote:
> > Adding a SRV lookup should add 0ms if it isn't there as you should be
> > making A, AAAA and SRV lookups in parallel.
> 
> This does not work for a simple reason : you have to perform the lookups
> on the SRV response just as you would do with a CNAME :

Please re-read my claim as your comment is not applicable to it.

Mark
-- 
Mark Andrews, ISC
1 Seymour St., Dundas Valley, NSW 2117, Australia
PHONE: +61 2 9871 4742                 INTERNET: marka@isc.org

From moore@network-heretics.com  Sun Jul 24 21:12:45 2011
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To: Mark Andrews <marka@isc.org>
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Cc: "Roy T. Fielding" <fielding@gbiv.com>, Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Jul 24, 2011, at 11:21 PM, Mark Andrews wrote:

>>> How do you solve the problem of hosting just "http://example.com/"
>>> on "s1.joes-web-service.com" and not redirect everything else at
>>> example.com?  People have been complaining about this for about as
>>> long as the web has existed.
>>=20
>> Well, in a way, that's what NAPTR was for.  All of the UR
>> i resolution mechanisms (equally applicable to DNS-based URIs) that =
were =3D
>> developed and never really used grew out of the original realization =
in =3D
>> the early 1990s that CERN could stop hosting the original web pages =
if =3D
>> it wanted to, and there was no way to keep those links from going =
stale.
>=20
> NAPTR is not defined for HTTP.
> SRV is not defined for HTTP.
>=20
>> The problem never went away, but the DNS-based solutions were defined =
a =3D
>> long time ago and never used.
>=20
> No.  It was explitly NOT defined.

Ok, fair enough.   Those of us who were working on the DNS-based URI =
resolution mechanisms realized that they could be applied to http URIs =
in addition to almost anything else (NAPTR is incredibly flexible if you =
don't mind doing lots of DNS lookups).  But they were never formally =
adopted.

But if you really want to use DNS to do redirects for http: URIs (or for =
that matter ws: URIs or almost any other kind of URI), NAPTR was =
tailor-made to do that.  SRV was not.

Keith


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From: Takeshi Yoshino <tyoshino@google.com>
Date: Mon, 25 Jul 2011 15:03:42 +0900
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To: Greg Wilkins <gregw@intalio.com>
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Subject: Re: [hybi] Payload only compression extension, again
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Hi,

On Sat, Jul 23, 2011 at 07:43, Greg Wilkins <gregw@intalio.com> wrote:

> On 23 July 2011 00:38, Takeshi Yoshino <tyoshino@google.com> wrote:
> > I've implemented this proposal on pywebsocket.
> pywebsocket's deflate-stream
> > implementation flushes compressed data for each frame using Z_SYNC_FLUSH.
> > Compression level is Z_DEFAULT_COMPRESSION. Default Huffman tree is used.
> No
> > dictionary is attached.
>
> Great - I'll make sure we have an implementation in  the next release of
> Jetty.
>
>
> - output of deflate-stream are basically bigger than
> > deflate-application-data since Z_SYNC_FLUSH wastes 4 octets for each
> flush
> > operation
>
> I think you mean deflate-application-data frames are bigger....


yes... bad typo


>  > - for short messages, it just increases size because of ~2 octets for
> two
> > block headers and end of block symbol
>
> Does this not suggest that we should use a reserve bit to indicate if
> the frame is compressed or not?  This will allow uncompressed frames
> (either because of small size or incompressible content) to be sent.
>

Yeah. Taking 1 RSV bit, we can avoid inflation of the size for this kind of
content.

The problem is, as you and John were discussing once, how to give a good
criteria to determine if a certain content of a frame is worth compress or
not. Even if frame N is incompressible, it may be referred to deflate frame
N+1. If we don't get the frame N go through compressor, N+1 cannot refer
it. So, criteria like |compressed| < |original| don't work well. Criteria
like |original| < (some pre-specified size) are also not easy to
define. It's almost clear that for communicating 1 byte key type event,
deflate shouldn't be applied. Thinking of two byte, three byte, ... it's not
trivial. I think we cannot gather real data to define such criteria
sufficient to be put in the spec soon.

Servers can get some advice about this from application level, but user
agents don't have any way. May we ask API side to add some interface to
allow JavaScript to give some hint? Something like

attribute boolean compressOutgoing;

or, more generally

attribute DOMString extensionConfig;

to dynamically instruct the user agent.

It's also an option that we introduce COMP bit in wire spec but initially
have no API exposed for JavaScript to control that.

----

Based on granularity choices are grouped like:

- Connection-wise control: deflate-application-data or no compression
extension
- Direction-wise control: need to have two separate extension
deflate-application-data-server-to-client
deflate-application-data-client-to-server
- Message-wise control: need to have API to on/off compression

Thanks,
Takeshi

--000e0cd306b6f4b9ee04a8de94c9
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div class=3D"gmail_quote">Hi,</div><div class=3D"gmail_quote"><br></div><d=
iv class=3D"gmail_quote">On Sat, Jul 23, 2011 at 07:43, Greg Wilkins <span =
dir=3D"ltr">&lt;<a href=3D"mailto:gregw@intalio.com" target=3D"_blank">greg=
w@intalio.com</a>&gt;</span> wrote:<br>



<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div>On 23 July 2011 00:38, Takeshi Yoshino =
&lt;<a href=3D"mailto:tyoshino@google.com" target=3D"_blank">tyoshino@googl=
e.com</a>&gt; wrote:<br>




&gt; I&#39;ve implemented this proposal on pywebsocket. pywebsocket&#39;s=
=A0deflate-stream<br>
&gt; implementation flushes compressed data for each frame using Z_SYNC_FLU=
SH.<br>
&gt; Compression level is Z_DEFAULT_COMPRESSION. Default Huffman tree is us=
ed. No<br>
&gt; dictionary is attached.<br>
<br>
</div>Great - I&#39;ll make sure we have an implementation in =A0the next r=
elease of Jetty.<br>
<div>=A0</div></blockquote><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>
&gt; - output of deflate-stream are basically bigger than<br>
&gt; deflate-application-data since Z_SYNC_FLUSH wastes 4 octets for each f=
lush<br>
&gt; operation<br>
<br>
</div>I think you mean deflate-application-data frames are bigger....</bloc=
kquote><div><br></div><div>yes... bad typo</div><div>=A0</div><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">


<div>
&gt; - for short messages, it just increases size because of ~2 octets for =
two<br>
&gt; block headers and end of block symbol<br>
<br>
</div>Does this not suggest that we should use a reserve bit to indicate if=
<br>
the frame is compressed or not? =A0This will allow uncompressed frames<br>
(either because of small size or incompressible content) to be sent.<br></b=
lockquote><div><br></div><div>Yeah. Taking 1 RSV bit, we can avoid inflatio=
n of the size for this kind of content.</div><div><br></div><div>
The problem is, as you and John were discussing once, how to give a good cr=
iteria to determine if=A0a certain content of a frame is=A0worth compress o=
r not. Even if frame N is incompressible, it may be referred to deflate fra=
me N+1. If we don&#39;t get the=A0frame N=A0go through compressor, N+1 cann=
ot refer it.=A0So, criteria like |compressed| &lt; |original| don&#39;t wor=
k well. Criteria like |original| &lt; (some pre-specified size) are also no=
t easy to define.=A0It&#39;s almost clear that for communicating 1 byte key=
 type event, deflate shouldn&#39;t be applied. Thinking of two byte, three =
byte, ... it&#39;s not trivial. I think we cannot gather real data to defin=
e such criteria sufficient to be put in the spec soon.</div>

<div><br></div><div>Servers can get some advice about this from application=
 level, but user agents don&#39;t have any way.=A0May we ask API side to ad=
d some interface to allow JavaScript to give some hint? Something like</div=
>

<div><br></div><div><div>attribute boolean compressOutgoing;</div></div><di=
v><div><br></div><div>or, more generally</div><div><br></div><div>attribute=
 DOMString extensionConfig;</div></div><div><br></div><div>to dynamically i=
nstruct the user agent.</div>

<div><br></div><div>It&#39;s also an option that we introduce COMP bit in w=
ire spec but initially have no API exposed for JavaScript to control that.<=
/div><div><br></div><div>----</div><div><br></div><div>Based on granularity=
 choices are grouped like:</div>

<div><br></div><div>- Connection-wise control: deflate-application-data or =
no compression extension</div><div>- Direction-wise control: need to have t=
wo separate extension deflate-application-data-server-to-client deflate-app=
lication-data-client-to-server</div>

<div>- Message-wise control: need to have API to on/off compression</div><d=
iv><br></div><div>Thanks,</div><div>Takeshi</div><div><br></div></div>

--000e0cd306b6f4b9ee04a8de94c9--

From ibc@aliax.net  Mon Jul 25 02:51:27 2011
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
To: Willy Tarreau <w@1wt.eu>
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Subject: Re: [hybi] Alternative for SRV proposal
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2011/7/25 Willy Tarreau <w@1wt.eu>:
>> Hummm, AFAIK some local DNS resolvers automatically resolve the CNAME
>> domain to an IP before passing the DNS result to the application, so
>> the application would not be aware of the existence of a CNAME record.
>
> I'm not aware of that and it sounds strange to me.

I was not exactly right. I use a local DNS cache/resolver (called
Unbound for Linux).

- I query it a DNS A for ghs.google.com.
- Unbound queries it to the DNS server.
- It receives a reply telling that ghs.google.com has just a CNAME
record (ghs.l.google.com).
- Unbound then queries DNS A for ghs.l.google.com.
- It receives "209.85.147.121".
- Then Unbound replies to my client with two answers:
    1) ghs.google.com: CNAME, cname ghs.l.google.com
    2) ghs.l.google.com: A, addr 209.85.147.121

So yes, by inspecting the DNS response the client can realize of the
presence of a CNAME for the queried domain.




>> But this would be an extra DNS query so there would be:
>>
>> a) NAPTR + SRV +`A/AAAA
>>
>> or
>>
>> b) NAPTR + A/AAAA
>>
>> It does not fit well with constrains exposed in the other mail thread.
>
> It does fit well as something at the option of the end user, because as
> Mark says, one can send the NAPTR + A + AAAA "at zero cost" (assuming
> the uplink is infinite). It does not require extra round trips, just
> extra bandwidth and some extra synchronization from the client.

It sounds good but, wouldn't it be too much not required information?
this is, a DNS ANY query retrievesa all the resource records for the
given domain.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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On Mon, Jul 25, 2011 at 2:03 AM, Takeshi Yoshino <tyoshino@google.com>wrote:

> Servers can get some advice about this from application level, but user
> agents don't have any way. May we ask API side to add some interface to
> allow JavaScript to give some hint? Something like
>

It seems unlikely to get this into the JS API anytime soon -- I think the
initial version will have to be entirely without hints from JS.

Longer term, I think you need to allow the JS code more control over
compression to get the best results, as frequently you can do much better
with domain-specific compression algorithms.  For example, chess endgame
databases get 2-3 times better compression with algorithms that know how the
index is structured.


> It's also an option that we introduce COMP bit in wire spec but initially
> have no API exposed for JavaScript to control that.
>

That would also be useful if there were heuristics to choose when to
compress the data.

However, as long as the compression state is maintained across frames, I
don't think it is a real problem to simply compress everything.  Sure, if
someone sends random data it will get bigger, but not by much.  Even on
binary data like Quake update frames, there is substantial redundancy to be
removed via compression, so with real-world data the payload is almost never
going to get bigger by compressing.

-- 
John A. Tamplin
Software Engineer (GWT), Google

--000e0cd306da0056e904a8e62746
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div class=3D"gmail_quote">On Mon, Jul 25, 2011 at 2:03 AM, Takeshi Yoshino=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:tyoshino@google.com">tyoshino@goog=
le.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

<div class=3D"gmail_quote">Servers can get some advice about this from appl=
ication level, but user agents don&#39;t have any way.=C2=A0May we ask API =
side to add some interface to allow JavaScript to give some hint? Something=
 like</div>

</blockquote><div><br></div><div>It seems unlikely to get this into the JS =
API anytime soon -- I think the initial version will have to be entirely wi=
thout hints from JS.</div><div><br></div><div>Longer term, I think you need=
 to allow the JS code more control over compression to get the best results=
, as frequently you can do much better with domain-specific compression alg=
orithms. =C2=A0For example, chess endgame databases get 2-3 times better co=
mpression with algorithms that know how the index is structured.</div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex;"><div class=3D"gmail_quote"=
><div></div><div><div>It&#39;s also an option that we introduce COMP bit in=
 wire spec but initially have no API exposed for JavaScript to control that=
.</div>

</div></div></blockquote></div><div><br></div>That would also be useful if =
there were heuristics to choose when to compress the data.<div><br></div><d=
iv>However, as long as the compression state is maintained across frames, I=
 don&#39;t think it is a real problem to simply compress everything. =C2=A0=
Sure, if someone sends random data it will get bigger, but not by much. =C2=
=A0Even on binary data like Quake update frames, there is substantial redun=
dancy to be removed via compression, so with real-world data the payload is=
 almost never going to get bigger by compressing.<br clear=3D"all">

<br>-- <br>John A. Tamplin<br>Software Engineer (GWT), Google<br>
</div>

--000e0cd306da0056e904a8e62746--

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To: John Tamplin <jat@google.com>
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Cc: "hybi@ietf.org" <hybi@ietf.org>, Greg Wilkins <gregw@intalio.com>
Subject: Re: [hybi] Payload only compression extension, again
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On Tue, Jul 26, 2011 at 12:05 AM, John Tamplin <jat@google.com> wrote:
> On Mon, Jul 25, 2011 at 2:03 AM, Takeshi Yoshino <tyoshino@google.com>
> wrote:
>> It's also an option that we introduce COMP bit in wire spec but initiall=
y
>> have no API exposed for JavaScript to control that.
>
> That would also be useful if there were heuristics to choose when to
> compress the data.
> However, as long as the compression state is maintained across frames, I
> don't think it is a real problem to simply compress everything. =A0Sure, =
if
> someone sends random data it will get bigger, but not by much. =A0Even on
> binary data like Quake update frames, there is substantial redundancy to =
be
> removed via compression, so with real-world data the payload is almost ne=
ver
> going to get bigger by compressing.

Certainly with this sort of payload that is initially being considered
normal for websocket I agree. However I expect that WS will also be
used to stream pre-compressed data to a client as well (e.g. images,
video, audio, resources in a zip at the server) for things like games.

It would also be useful if there were not heuristics to decide if the
data should be compressed or not. Often the client or server will have
external knowledge that the data will not benefit from compression,
and so having the ability to mark frames/messages as do-not-compress
is useful to allow savings of both network bytes and cpu time.

Regards,
Brodie

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Subject: Re: [hybi] Payload only compression extension, again
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On 26 July 2011 01:05, John Tamplin <jat@google.com> wrote:
> That would also be useful if there were heuristics to choose when to
> compress the data.

Once there is a RSV bit used, then the sender can use a variety of
different strategies to determine if to compress or not.  Heuristics
is a good one and I think most server gzip filters apply simple
heuristics (eg if uncompressed is > size and if compression does not
overflow buffer of same size).   Although servers also have the
benefit of mime types (eg don't try to compress image/jpg).

> However, as long as the compression state is maintained across frames, I
> don't think it is a real problem to simply compress everything.

I think that is the simplest and best heuristic to start with, and
will probably be beneficial in the vast majority of cases.

>From the browser side of things, it may be the only practical
heuristic while the API is just simple text messages.   Perhaps when
binary messages are supported something else might do better, but
that's the benefit of using a RSV bit, as the strategy can evolve over
time without the need to update the protocol specification.

cheers

From theturtle32@gmail.com  Mon Jul 25 17:56:37 2011
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--0015174c3ca64c5acf04a8ee67b2
Content-Type: text/plain; charset=ISO-8859-1

FYI, the WebSocket API draft has already been updated to support binary
messages.

Brian

On Mon, Jul 25, 2011 at 4:56 PM, Greg Wilkins <gregw@intalio.com> wrote:

> On 26 July 2011 01:05, John Tamplin <jat@google.com> wrote:
> > That would also be useful if there were heuristics to choose when to
> > compress the data.
>
> Once there is a RSV bit used, then the sender can use a variety of
> different strategies to determine if to compress or not.  Heuristics
> is a good one and I think most server gzip filters apply simple
> heuristics (eg if uncompressed is > size and if compression does not
> overflow buffer of same size).   Although servers also have the
> benefit of mime types (eg don't try to compress image/jpg).
>
> > However, as long as the compression state is maintained across frames, I
> > don't think it is a real problem to simply compress everything.
>
> I think that is the simplest and best heuristic to start with, and
> will probably be beneficial in the vast majority of cases.
>
> From the browser side of things, it may be the only practical
> heuristic while the API is just simple text messages.   Perhaps when
> binary messages are supported something else might do better, but
> that's the benefit of using a RSV bit, as the strategy can evolve over
> time without the need to update the protocol specification.
>
> cheers
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>

--0015174c3ca64c5acf04a8ee67b2
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

FYI, the WebSocket API draft has already been updated to support binary mes=
sages.<div><br></div><div>Brian<br><br><div class=3D"gmail_quote">On Mon, J=
ul 25, 2011 at 4:56 PM, Greg Wilkins <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:gregw@intalio.com">gregw@intalio.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;"><div class=3D"im">On 26 July 2011 01:05, Jo=
hn Tamplin &lt;<a href=3D"mailto:jat@google.com">jat@google.com</a>&gt; wro=
te:<br>

&gt; That would also be useful if there were heuristics to choose when to<b=
r>
&gt; compress the data.<br>
<br>
</div>Once there is a RSV bit used, then the sender can use a variety of<br=
>
different strategies to determine if to compress or not. =A0Heuristics<br>
is a good one and I think most server gzip filters apply simple<br>
heuristics (eg if uncompressed is &gt; size and if compression does not<br>
overflow buffer of same size). =A0 Although servers also have the<br>
benefit of mime types (eg don&#39;t try to compress image/jpg).<br>
<div class=3D"im"><br>
&gt; However, as long as the compression state is maintained across frames,=
 I<br>
&gt; don&#39;t think it is a real problem to simply compress everything.<br=
>
<br>
</div>I think that is the simplest and best heuristic to start with, and<br=
>
will probably be beneficial in the vast majority of cases.<br>
<br>
>From the browser side of things, it may be the only practical<br>
heuristic while the API is just simple text messages. =A0 Perhaps when<br>
binary messages are supported something else might do better, but<br>
that&#39;s the benefit of using a RSV bit, as the strategy can evolve over<=
br>
time without the need to update the protocol specification.<br>
<div><div></div><div class=3D"h5"><br>
cheers<br>
_______________________________________________<br>
hybi mailing list<br>
<a href=3D"mailto:hybi@ietf.org">hybi@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/hybi</a><br>
</div></div></blockquote></div><br></div>

--0015174c3ca64c5acf04a8ee67b2--

From gregw@intalio.com  Mon Jul 25 19:39:26 2011
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On 25 July 2011 06:22, Bjoern Hoehrmann <derhoermi@gmx.net> wrote:
...
> Either way it should not be in the base protocol specification.


Chairs / Editors,

can you make a call on this please?    Or at least a plan to resolve the issue.

There have been several calls to remove it with substantial reasons
given.   I think the silence on this matter from chairs/editor is just
encouraging repetition of the arguments.

regards

From stpeter@stpeter.im  Mon Jul 25 20:12:14 2011
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<hat type='AD'/>

On 7/25/11 10:39 PM, Greg Wilkins wrote:
> On 25 July 2011 06:22, Bjoern Hoehrmann <derhoermi@gmx.net> wrote:
> ...
>> Either way it should not be in the base protocol specification.
> 
> 
> Chairs / Editors,
> 
> can you make a call on this please?    Or at least a plan to resolve the issue.
> 
> There have been several calls to remove it with substantial reasons
> given.   I think the silence on this matter from chairs/editor is just
> encouraging repetition of the arguments.

I'll leave discussion of specific issues to the chairs. However, Alexey,
Gabriel, Sal, and I had a meeting tonight about the open issues (Ian has
not yet arrived in Quebec City). These issues will be discussed during
the WG session on Thursday. After that I would expect that the chairs or
editors will quickly close a number of issues, publish consensus calls,
post proposed resolutions, etc.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From ibc@aliax.net  Tue Jul 26 02:47:50 2011
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2011/7/24 Willy Tarreau <w@1wt.eu>:
>> Ok, but what would be the failover solution then? any failover
>> solution can take some time until redirects the client's request to a
>> working server, it's not just a client problem.
>
> This is already addressed by existing web architectures. DNS provides
> geolocation and proposes you the closest working datacenter. BGP ensures
> that this datacenter is always reachable via multiple links and multiple
> providers. Load balancers on the site ensure that only valid servers are
> used. I know some people who configure their load balancers so that a
> server is removed from the pool less than 50ms after a failure or slowdow=
n
> has been noticed. You will never be able to offer that quality of service
> using DNS only, because those 50ms are less than the time it takes for
> most visitors to ping the site.

Yes, but those are expensive server side solutions. I wouldn't like
that a scalable architecture requires so complex solutions (which are
not very suitable in case of usual web hostings). DNS SRV provides a
cheaper way of doing something similar (both ways can live together
thought).



>> Anyhow, don't forget that SRV is not just for failover, but also for
>> load-balancing (you can state that a more powerful server-01 receives
>> 80% of the traffic and server-02 the remaining 20%).
>
> This is already addressed by DNS round robin and used by a lot of sites.

I consider that a hack.


> But keep in mind that DNS is used between *sites* and not *servers*. If
> you use it for servers, you'll fail because you won't be able to silently
> put them in maintenance without disturbing visitors.

SRV provides load-balancing and failover. I never said that SRV is a
solution for temporaly put in maintenance a server.


> Load balancers are
> used for servers. And between sites, people are not seeking imbalanced
> distribution. If they really do, then it's easy using multiple addresses.
>
> Don't take this as an attack (it is not), but you accused Roy of not
> knowing SRV, but I think that you're not much experienced with web
> infrastructures and that may be why you think that SRV is the *only*
> solution to many problems that have been dealt with for ages.

Right, but I'm used to scalable and big SIP infrastructures and DNS
SRV is used for that (along with server side solutions as SIP load
balancers). What I say is that web infraestructures cannot use SRV
mechanism so it's not valid for me that just current HTTP solutions
are valid for any other new protocol.

In contrast it seems that for many people the hacks existing in HTTP
world (www.facebook.com resolves to a single IP) is the only way to go
for a scalable infraestructure "so DNS SRV is not needed". Why so many
efforts in disallowing an *optional* mechanism like SRV? If somebody
does not like it, just don't use it.


Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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2011/7/24 Willy Tarreau <w@1wt.eu>:
> To ensure nobody gets me wrong, I'm certain this can help solve issues
> *if this is optional*. If it becomes a MUST, then the negative effects
> will override the positive ones. In my opinion, the client should decide
> whether to enable it or not.

But I don't understand how a client is supposed to decide by himself
how to resolve a URI destination. If I give you my vcard containing my
SIP/XMPP/MAILTO URIs, I must expect that you would use *standarized*
mechanisms to locate the server for each service. In fact, in all
these cases (SIP, XMPP; MAILTO) the URI domain can point to several
IP:port (due to NAPTR / SRV / MX DNS records).

BTW: I know that any web browser would first lookup at the /etc/hosts
file when an URI is introduced. This would "replace" a DNS A/AAAA
query. Maybe NIS or whatever could also be used for this. It does not
break SIP/XMPP/MAILTO URI's resolutions: Initially NAPTR / SRV / MX
query would be performed and, if there is no such record (or there is
so we get hostnames for which DNS A must be performed) then the client
can check /etc/hosts for the "A" resolution (domain -> IP).

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Cc: "Roy T. Fielding" <fielding@gbiv.com>, Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>, Mark Andrews <marka@isc.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/25 Keith Moore <moore@network-heretics.com>:
> By now, I think the market has long since decided. =C2=A0For better or wo=
rse, the
> mechanism the market chose to use with the web was HTTP redirects.

That's common in HTTP world: too much "cool" stuff on top of it, and
too much noise and hacks below it.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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2011/7/25 Mark Andrews <marka@isc.org>:
>> But if you really want to use DNS to do redirects for http: URIs (or for=
 =3D
>> that matter ws: URIs or almost any other kind of URI), NAPTR was =3D
>> tailor-made to do that. =C2=A0SRV was not.
>
> "_http._tcp.example.com SRV 100 0 80 <server>" is not a redirect.
> The http client still issues "Host: example.com" not "Host: <server>".
> If you want to do DNS level redirect of "www.example.com" to
> "example.com" then NAPTR would be the way to do that and the http
> client issues "Host: example.com" instead of "Host: www.example.com".

Great explanation.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Date: Tue, 26 Jul 2011 23:23:06 +0900
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Subject: Re: [hybi] Payload only compression extension, again
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--001517570906cb948f04a8f9aca6
Content-Type: text/plain; charset=ISO-8859-1

Thanks all. It seems that we'd
- add COMP bit
but for now
- leave advanced API, heuristics unspecified

And, as discussed in this thread a year ago
http://www.ietf.org/mail-archive/web/hybi/current/msg01810.html
I'm planning to add some parameters to this extension.

- Max size of LZ77 sliding window (and any other DEFLATE parameter if
useful)
-- requiring 32kB window for each connection might cause inacceptable memory
pressure for large scale system with lots of connection or embed system with
small memory
--- see the result of experiment done by spdy folk
https://spdy-dev.googlegroups.com/attach/4051c1467338d389/zlib_20100202.html?view=1&part=4
-- So, I'd add parameter "window_bits" to allow such endpoint to limit
sliding window size used by the other peer
-- if it's hard to get this negotiation into spec soon, the spec may
initially suggest some default values and limit based on these
investigation. Say, "endpoints MUST use sliding window of 2^12 or smaller".

Additionally, I'd like to propose one more parameter.

- No compression context sharing option
-- though I'm claiming that compression context sharing across frames is
good, it's sometimes harmful for intermediaries
-- for load balancers that dispatches WebSocket frames to different
backends, shared compression context means that the load balancer have to
perform decompression. it's burden in terms of both memory pressure and CPU
power.
-- So, I'd add parameter "finish_at_eof" to allow such intermediaries to ask
clients not to keep using the same compression context across frames

--001517570906cb948f04a8f9aca6
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div>Thanks all. It seems that we&#39;d</div><div>- add COMP bit</div><div>=
but for now</div><div>- leave advanced API, heuristics unspecified</div><di=
v><br></div><div>And, as discussed in this thread a year ago</div><div><a h=
ref=3D"http://www.ietf.org/mail-archive/web/hybi/current/msg01810.html" tar=
get=3D"_blank">http://www.ietf.org/mail-archive/web/hybi/current/msg01810.h=
tml</a></div>


<div>I&#39;m planning to add some parameters to this extension.</div><div><=
div><br></div><div>- Max size of LZ77 sliding window (and any other DEFLATE=
 parameter if useful)</div><div>-- requiring 32kB window for each connectio=
n might cause inacceptable memory pressure for large scale system with lots=
 of connection or embed system with small memory</div>


<div>--- see the result of experiment done by spdy folk=A0<a href=3D"https:=
//spdy-dev.googlegroups.com/attach/4051c1467338d389/zlib_20100202.html?view=
=3D1&amp;part=3D4" target=3D"_blank">https://spdy-dev.googlegroups.com/atta=
ch/4051c1467338d389/zlib_20100202.html?view=3D1&amp;part=3D4</a></div>


<div>-- So, I&#39;d add parameter &quot;window_bits&quot; to allow such end=
point to limit sliding window size used by=A0the other peer</div><div>-- if=
 it&#39;s hard to get this negotiation into spec soon, the spec may initial=
ly suggest some default values and limit based on these investigation. Say,=
 &quot;endpoints MUST use sliding window of 2^12 or smaller&quot;.</div>


<div><br></div><div>Additionally, I&#39;d like to propose one more paramete=
r.</div><div><br></div><div>- No compression context sharing option</div></=
div><div>-- though I&#39;m claiming that compression context sharing across=
 frames is good, it&#39;s sometimes harmful for intermediaries</div>


<div>-- for load balancers that dispatches WebSocket frames to different ba=
ckends, shared compression context means that the load balancer have to per=
form decompression. it&#39;s burden in terms of both memory pressure and CP=
U power.</div>

<div>-- So, I&#39;d add parameter &quot;finish_at_eof&quot; to allow such i=
ntermediaries to ask clients not to keep using the same compression context=
 across frames</div>
<div><br></div>

--001517570906cb948f04a8f9aca6--

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From: John Tamplin <jat@google.com>
Date: Tue, 26 Jul 2011 11:11:11 -0400
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To: Takeshi Yoshino <tyoshino@google.com>
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Subject: Re: [hybi] Payload only compression extension, again
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On Tue, Jul 26, 2011 at 10:23 AM, Takeshi Yoshino <tyoshino@google.com>wrote:

> Additionally, I'd like to propose one more parameter.
>
> - No compression context sharing option
> -- though I'm claiming that compression context sharing across frames is
> good, it's sometimes harmful for intermediaries
> -- for load balancers that dispatches WebSocket frames to different
> backends, shared compression context means that the load balancer have to
> perform decompression. it's burden in terms of both memory pressure and CPU
> power.
> -- So, I'd add parameter "finish_at_eof" to allow such intermediaries to
> ask clients not to keep using the same compression context across frames
>

The problem is that if you don't share compression state across frames, then
it becomes critical to not try and compress incompressible content.  Look at
the experiment data I presented in that same thread -- it is very easy for
frame sizes to go up if you aren't sharing compression state, so I think if
this option is included we have to decide now how to turn off compression on
a frame-by-frame basis rather than delaying it for a future extension.  In
contrast, if compression state is maintained across frames, it is still
usually a win even on incompressible content because different frames have
redundancy, and the loss is never bad.

I would prefer, for the initial version, to simply have load balancers that
can't maintain compression state to simply refuse compression.

-- 
John A. Tamplin
Software Engineer (GWT), Google

--000e0cd6ae5abfc52804a8fa5888
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<div class=3D"gmail_quote">On Tue, Jul 26, 2011 at 10:23 AM, Takeshi Yoshin=
o <span dir=3D"ltr">&lt;<a href=3D"mailto:tyoshino@google.com">tyoshino@goo=
gle.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D=
"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

<div>Additionally, I&#39;d like to propose one more parameter.</div><div><d=
iv><br></div><div>- No compression context sharing option</div></div><div>-=
- though I&#39;m claiming that compression context sharing across frames is=
 good, it&#39;s sometimes harmful for intermediaries</div>




<div>-- for load balancers that dispatches WebSocket frames to different ba=
ckends, shared compression context means that the load balancer have to per=
form decompression. it&#39;s burden in terms of both memory pressure and CP=
U power.</div>



<div>-- So, I&#39;d add parameter &quot;finish_at_eof&quot; to allow such i=
ntermediaries to ask clients not to keep using the same compression context=
 across frames</div></blockquote></div><div><br></div>The problem is that i=
f you don&#39;t share compression state across frames, then it becomes crit=
ical to not try and compress incompressible content. =C2=A0Look at the expe=
riment data I presented in that same thread -- it is very easy for frame si=
zes to go up if you aren&#39;t sharing compression state, so I think if thi=
s option is included we have to decide now how to turn off compression on a=
 frame-by-frame basis rather than delaying it for a future extension. =C2=
=A0In contrast, if compression state is maintained across frames, it is sti=
ll usually a win even on incompressible content because different frames ha=
ve redundancy, and the loss is never bad.<br clear=3D"all">

<br><div>I would prefer, for the initial version, to simply have load balan=
cers that can&#39;t maintain compression state to simply refuse compression=
.</div><div><br>-- <br>John A. Tamplin<br>Software Engineer (GWT), Google<b=
r>


</div>

--000e0cd6ae5abfc52804a8fa5888--

From derhoermi@gmx.net  Tue Jul 26 08:25:57 2011
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* John Tamplin wrote:
>The problem is that if you don't share compression state across frames, then
>it becomes critical to not try and compress incompressible content.  Look at
>the experiment data I presented in that same thread -- it is very easy for
>frame sizes to go up if you aren't sharing compression state, so I think if
>this option is included we have to decide now how to turn off compression on
>a frame-by-frame basis rather than delaying it for a future extension.

Deflate stores uncompressed data in up to 2^15 byte long blocks like

  1 byte indicating no compression and whether it's a final block
  2 byte indicating the length
  2 byte one's complement of the length
  n byte uncompressed data as per the length

Are you saying the 5 additional byte per block are a problem?
-- 
Björn Höhrmann · mailto:bjoern@hoehrmann.de · http://bjoern.hoehrmann.de
Am Badedeich 7 · Telefon: +49(0)160/4415681 · http://www.bjoernsworld.de
25899 Dagebüll · PGP Pub. KeyID: 0xA4357E78 · http://www.websitedev.de/ 

From w@1wt.eu  Tue Jul 26 12:29:04 2011
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Date: Tue, 26 Jul 2011 21:28:50 +0200
From: Willy Tarreau <w@1wt.eu>
To: =?iso-8859-1?Q?I=F1aki?= Baz Castillo <ibc@aliax.net>
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Hi Iñaki,

On Tue, Jul 26, 2011 at 11:58:41AM +0200, Iñaki Baz Castillo wrote:
> 2011/7/24 Willy Tarreau <w@1wt.eu>:
> > To ensure nobody gets me wrong, I'm certain this can help solve issues
> > *if this is optional*. If it becomes a MUST, then the negative effects
> > will override the positive ones. In my opinion, the client should decide
> > whether to enable it or not.
> 
> But I don't understand how a client is supposed to decide by himself
> how to resolve a URI destination.

This is well explained in RFC3986, #3.2.2. It enumerates a number of
exiting name registration methods and makes it clear that DNS usage is
not mandatory at all, only suggested. Also, it only exposes the fact that
names are resolved to addresses. It obviously does not make any asumption
on the DNS records to use for this either.

> If I give you my vcard containing my
> SIP/XMPP/MAILTO URIs, I must expect that you would use *standarized*
> mechanisms to locate the server for each service. In fact, in all
> these cases (SIP, XMPP; MAILTO) the URI domain can point to several
> IP:port (due to NAPTR / SRV / MX DNS records).

I'd say that the common practice with DNS has always been to use A records
for IPv4, AAAA for IPv6 (or CNAME for any), unless explicitly stated
otherwise for the protocol's usage. Now you could have resolver libraries
which would try SRV before trying A/AAAA and this would be transparent to
the upper application layers.

> BTW: I know that any web browser would first lookup at the /etc/hosts
> file when an URI is introduced.

I'm not a browser developer, but I think they might even simply rely on
the system's resolver. For instance, you might have an /etc/nsswitch.conf
or /etc/hosts.conf which indicates the priorities for the resolvers.

> This would "replace" a DNS A/AAAA
> query. Maybe NIS or whatever could also be used for this. It does not
> break SIP/XMPP/MAILTO URI's resolutions: Initially NAPTR / SRV / MX
> query would be performed and, if there is no such record (or there is
> so we get hostnames for which DNS A must be performed) then the client
> can check /etc/hosts for the "A" resolution (domain -> IP).

I don't know where you want to go with that example, but as I explained
it, if you want to have any chance of making SRV *usable* with WS (or
HTTP), you have to motivate both sides by showing them that :
  - it's better for them to use it than not to use it (both servers and
    browsers)
  - the additional cost of using it is negligible
  - there are no issues with not using it
  - leaving the choices to the intermediaries will not cause disruptions

I'm pretty sure that can be done, but clearly not the way it's been
presented till now.

Willy


From marka@isc.org  Tue Jul 26 18:29:03 2011
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In message <CALiegf=e48kkF+Gky1mY7LippUB-0kZDgSGZrJxk1aZupAGkYw@mail.gmail.com>
, =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= writes:
> 2011/7/24 Willy Tarreau <w@1wt.eu>:
> >> Ok, but what would be the failover solution then? any failover
> >> solution can take some time until redirects the client's request to a
> >> working server, it's not just a client problem.
> >
> > This is already addressed by existing web architectures. DNS provides
> > geolocation and proposes you the closest working datacenter. BGP ensures
> > that this datacenter is always reachable via multiple links and multiple
> > providers. Load balancers on the site ensure that only valid servers are
> > used. I know some people who configure their load balancers so that a
> > server is removed from the pool less than 50ms after a failure or 
> slowdown
> > has been noticed. You will never be able to offer that quality of 
> service
> > using DNS only, because those 50ms are less than the time it takes for
> > most visitors to ping the site.
> 
> Yes, but those are expensive server side solutions. I wouldn't like
> that a scalable architecture requires so complex solutions (which are
> not very suitable in case of usual web hostings). DNS SRV provides a
> cheaper way of doing something similar (both ways can live together
> thought).
> 
> 
> 
> >> Anyhow, don't forget that SRV is not just for failover, but also for
> >> load-balancing (you can state that a more powerful server-01 receives
> >> 80% of the traffic and server-02 the remaining 20%).
> >
> > This is already addressed by DNS round robin and used by a lot of sites.
> 
> I consider that a hack.
> 
> 
> > But keep in mind that DNS is used between *sites* and not *servers*. If
> > you use it for servers, you'll fail because you won't be able to 
> silently
> > put them in maintenance without disturbing visitors.
> 
> SRV provides load-balancing and failover. I never said that SRV is a
> solution for temporaly put in maintenance a server.

Happy eyeballs however does allow you to take a server out of production
and not really notice it.  Note webbrowsers have had the ability to do
this for as long as webbrowsers have existed.

Billions of dollars have been wasted globally for the sake of a few hours
work by webbrowser vendors.
 
> > Load balancers are
> > used for servers. And between sites, people are not seeking imbalanced
> > distribution. If they really do, then it's easy using multiple 
> addresses.
> >
> > Don't take this as an attack (it is not), but you accused Roy of not
> > knowing SRV, but I think that you're not much experienced with web
> > infrastructures and that may be why you think that SRV is the *only*
> > solution to many problems that have been dealt with for ages.
> 
> Right, but I'm used to scalable and big SIP infrastructures and DNS
> SRV is used for that (along with server side solutions as SIP load
> balancers). What I say is that web infraestructures cannot use SRV
> mechanism so it's not valid for me that just current HTTP solutions
> are valid for any other new protocol.
> 
> In contrast it seems that for many people the hacks existing in HTTP
> world (www.facebook.com resolves to a single IP) is the only way to go
> for a scalable infraestructure "so DNS SRV is not needed". Why so many
> efforts in disallowing an *optional* mechanism like SRV? If somebody
> does not like it, just don't use it.
> 
> 
> Regards.
> 
> 
> -- 
> IÃ±aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> Ietf mailing list
> Ietf@ietf.org
> https://www.ietf.org/mailman/listinfo/ietf

-- 
Mark Andrews, ISC
1 Seymour St., Dundas Valley, NSW 2117, Australia
PHONE: +61 2 9871 4742                 INTERNET: marka@isc.org

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From: Takeshi Yoshino <tyoshino@google.com>
Date: Wed, 27 Jul 2011 12:39:01 +0900
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To: John Tamplin <jat@google.com>
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Subject: Re: [hybi] Payload only compression extension, again
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On Wed, Jul 27, 2011 at 00:11, John Tamplin <jat@google.com> wrote:

> On Tue, Jul 26, 2011 at 10:23 AM, Takeshi Yoshino <tyoshino@google.com>wrote:
>
>> Additionally, I'd like to propose one more parameter.
>>
>> - No compression context sharing option
>> -- though I'm claiming that compression context sharing across frames is
>> good, it's sometimes harmful for intermediaries
>> -- for load balancers that dispatches WebSocket frames to different
>> backends, shared compression context means that the load balancer have to
>> perform decompression. it's burden in terms of both memory pressure and CPU
>> power.
>> -- So, I'd add parameter "finish_at_eof" to allow such intermediaries to
>> ask clients not to keep using the same compression context across frames
>>
>
> The problem is that if you don't share compression state across frames,
> then it becomes critical to not try and compress incompressible content.
>  Look at the experiment data I presented in that same thread -- it is very
> easy for frame sizes to go up if you aren't sharing compression state, so I
> think if this option is included we have to decide now how to turn off
> compression on a frame-by-frame basis rather than delaying it for a future
> extension.  In contrast, if compression state is maintained across frames,
> it is still usually a win even on incompressible content because different
> frames have redundancy, and the loss is never bad.
>
>
First, I'd like to make it clear that I intended to make this option one-way
property (i.e. if S included this, no shared-context in C->S direction but
allowed for C<-S direction).

I think it's still ok that clients choose to just compress everything.

With this option, service providers have two choices
1) put this option in extension response to reject shared context while
taking in the possible low efficiency you explained
2) reject compression completely

Inclusion of this parameter implies that the server-side thinks it's more
important to reject shared context than losing compression efficiency.
Looking at your experiment, for some app, there's still some gain from
frame-by-frame compression.

Without this option, they can take only 2).

That's all, I think.


> I would prefer, for the initial version, to simply have load balancers that
> can't maintain compression state to simply refuse compression.
>
> --
> John A. Tamplin
> Software Engineer (GWT), Google
>

--000e0cd6a910368e1904a904cbc9
Content-Type: text/html; charset=ISO-8859-1
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<div class=3D"gmail_quote">On Wed, Jul 27, 2011 at 00:11, John Tamplin <spa=
n dir=3D"ltr">&lt;<a href=3D"mailto:jat@google.com">jat@google.com</a>&gt;<=
/span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex;">

<div class=3D"im"><div class=3D"gmail_quote">On Tue, Jul 26, 2011 at 10:23 =
AM, Takeshi Yoshino <span dir=3D"ltr">&lt;<a href=3D"mailto:tyoshino@google=
.com" target=3D"_blank">tyoshino@google.com</a>&gt;</span> wrote:<br><block=
quote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc=
 solid;padding-left:1ex">



<div>Additionally, I&#39;d like to propose one more parameter.</div><div><d=
iv><br></div><div>- No compression context sharing option</div></div><div>-=
- though I&#39;m claiming that compression context sharing across frames is=
 good, it&#39;s sometimes harmful for intermediaries</div>






<div>-- for load balancers that dispatches WebSocket frames to different ba=
ckends, shared compression context means that the load balancer have to per=
form decompression. it&#39;s burden in terms of both memory pressure and CP=
U power.</div>





<div>-- So, I&#39;d add parameter &quot;finish_at_eof&quot; to allow such i=
ntermediaries to ask clients not to keep using the same compression context=
 across frames</div></blockquote></div><div><br></div></div>The problem is =
that if you don&#39;t share compression state across frames, then it become=
s critical to not try and compress incompressible content. =A0Look at the e=
xperiment data I presented in that same thread -- it is very easy for frame=
 sizes to go up if you aren&#39;t sharing compression state, so I think if =
this option is included we have to decide now how to turn off compression o=
n a frame-by-frame basis rather than delaying it for a future extension. =
=A0In contrast, if compression state is maintained across frames, it is sti=
ll usually a win even on incompressible content because different frames ha=
ve redundancy, and the loss is never bad.<br clear=3D"all">



<br></blockquote><div><br></div><div>First, I&#39;d like to make it clear t=
hat I intended to make this option one-way property (i.e. if S included thi=
s, no shared-context in C-&gt;S direction but allowed for C&lt;-S direction=
).</div>

<div><br></div><div>I think it&#39;s still ok that clients choose to just c=
ompress everything.</div><div><br></div><div>With this option, service prov=
iders have two choices</div><div>1) put this option in extension response t=
o reject shared context while taking in the possible low efficiency you exp=
lained</div>

<div>2) reject compression completely</div><div><br></div><div>Inclusion of=
 this parameter implies that the server-side thinks it&#39;s more important=
 to reject shared context than losing compression efficiency. Looking at yo=
ur experiment, for some app, there&#39;s still some gain from frame-by-fram=
e compression.</div>

<div><br></div><div>Without this option, they can take only 2).</div><div><=
br></div><div>That&#39;s all, I think.</div><div>=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex;">

<div>I would prefer, for the initial version, to simply have load balancers=
 that can&#39;t maintain compression state to simply refuse compression.</d=
iv><div><div></div><div class=3D"h5"><div><br>-- <br>John A. Tamplin<br>
Software Engineer (GWT), Google<br>



</div>
</div></div></blockquote></div><br>

--000e0cd6a910368e1904a904cbc9--

From w@1wt.eu  Tue Jul 26 22:25:58 2011
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Date: Wed, 27 Jul 2011 07:25:49 +0200
From: Willy Tarreau <w@1wt.eu>
To: Mark Andrews <marka@isc.org>
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References: <CALiegf=4K2oWfmZjGMD7J_jyaDtS3i+Mu7R0Wh75Rr+MrQCjtw@mail.gmail.com> <20110722054345.GE18126@1wt.eu> <CALiegfnYm6g63JDHLiSH__r-or3kzK0XCVa3cC7RMP14KWBOSg@mail.gmail.com> <20110724120751.GQ22405@1wt.eu> <CALiegfncavmoMp4YDCeeJ3rsOfHAYQ99itKX2Q2eHB351T3X5A@mail.gmail.com> <20110724184537.GZ22405@1wt.eu> <CALiegfne620wuDMAp235n3mVcXTAnbhhNm8vpiNCy5F7+VD92A@mail.gmail.com> <20110724192948.GD22405@1wt.eu> <CALiegf=e48kkF+Gky1mY7LippUB-0kZDgSGZrJxk1aZupAGkYw@mail.gmail.com> <20110727012806.EBB811231907@drugs.dv.isc.org>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Wed, Jul 27, 2011 at 11:28:06AM +1000, Mark Andrews wrote:
> > SRV provides load-balancing and failover. I never said that SRV is a
> > solution for temporaly put in maintenance a server.
> 
> Happy eyeballs however does allow you to take a server out of production
> and not really notice it.  Note webbrowsers have had the ability to do
> this for as long as webbrowsers have existed.

Mark, could you elaborate on this point ? Surely you're not talking about
the fact that a browser retries a failed connection, but I'm not not sure
what mechanism you're talking about.

> Billions of dollars have been wasted globally for the sake of a few hours
> work by webbrowser vendors.

This analysis seems a bit simplified in my opinion. Nothing is completely
white or black.

Regards,
Willy


From dave@cridland.net  Wed Jul 27 00:38:03 2011
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To: Mark Andrews <marka@isc.org>, Server-Initiated HTTP <hybi@ietf.org>, Willy Tarreau <w@1wt.eu>, IETF-Discussion <ietf@ietf.org>, =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Wed Jul 27 02:28:06 2011, Mark Andrews wrote:
> Billions of dollars have been wasted globally for the sake of a few  
> hours
> work by webbrowser vendors.

Seems to be a recurring theme - browsers could have easily performed  
probe tests to check for vulnerable proxies and disabled WebSockets  
(or negotiated masking), but that was ruled out long ago.

Dave.
-- 
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From ibc@aliax.net  Wed Jul 27 02:44:00 2011
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Cc: Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/26 Willy Tarreau <w@1wt.eu>:
> if you want to have any chance of making SRV *usable* with WS (or
> HTTP), you have to motivate both sides by showing them that :
> =C2=A0- it's better for them to use it than not to use it (both servers a=
nd
> =C2=A0 =C2=A0browsers)
> =C2=A0- the additional cost of using it is negligible
> =C2=A0- there are no issues with not using it

These are godd points, but I never wanted to propose SRV for HTTP as I
consider it's just unfeasible at this time (take into account the
ammount of HTTP clients in the world, as browsers, libraries in any
language and so on).


> =C2=A0- leaving the choices to the intermediaries will not cause disrupti=
ons

This last point is hard to accomplish (I'm just talking about SRV for
WS, not for HTTP) because HTTP proxies should be capable of
determining that a GET request is in fact a WS handshake, and *just*
in that case perform SRV procedures over the domain (assuming that
there won't be SRV in the old, anti-fashion and technologically
limited HTTP world).


> I'm pretty sure that can be done, but clearly not the way it's been
> presented till now.

If the requeriment for including SRV in WS is also including it in
HTTP then I surrender. I don't think it will never happen, neither I'm
an expert in HTTP for such kind of proposal.


Thanks a lot.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From w@1wt.eu  Wed Jul 27 03:18:06 2011
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Date: Wed, 27 Jul 2011 12:17:57 +0200
From: Willy Tarreau <w@1wt.eu>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Wed, Jul 27, 2011 at 11:43:58AM +0200, Iñaki Baz Castillo wrote:
> 2011/7/26 Willy Tarreau <w@1wt.eu>:
> > if you want to have any chance of making SRV *usable* with WS (or
> > HTTP), you have to motivate both sides by showing them that :
> >  - it's better for them to use it than not to use it (both servers and
> >    browsers)
> >  - the additional cost of using it is negligible
> >  - there are no issues with not using it
> 
> These are godd points, but I never wanted to propose SRV for HTTP as I
> consider it's just unfeasible at this time (take into account the
> ammount of HTTP clients in the world, as browsers, libraries in any
> language and so on).

That's where I think you're mistaken. As long as you think of it as mandatory
this will not be possible. When you think of it as optional and with an added
value, then you will progressively see clients adopt it and make use of it.
Since the beginning, you're proposing the option as mandatory "just because"
and not for a perceivable added value by both sides. The downsides then
outweigh the benefits. But maybe the benefits do not really exist after all,
otherwise a valid attractive proposal would already have been made by the
time we spent discussing it.

> >  - leaving the choices to the intermediaries will not cause disruptions
> 
> This last point is hard to accomplish (I'm just talking about SRV for
> WS, not for HTTP) because HTTP proxies should be capable of
> determining that a GET request is in fact a WS handshake, and *just*
> in that case perform SRV procedures over the domain (assuming that
> there won't be SRV in the old, anti-fashion and technologically
> limited HTTP world).

Once again, if you expect *all* HTTP proxies to be able to do that, your
proposal fails. If you expect that at least *some* HTTP proxies will be
able to do that with a benefit, then something is possible. Let me repeat
it, technologies are adopted, not forced on users.

> > I'm pretty sure that can be done, but clearly not the way it's been
> > presented till now.
> 
> If the requeriment for including SRV in WS is also including it in
> HTTP then I surrender. I don't think it will never happen, neither I'm
> an expert in HTTP for such kind of proposal.

Probably that starting by enumerating in a draft the benefits it could bring
would be a good start. You seem to be very well convinced that there are
benefits, so you're probably one of the few persons able to start such a
draft.

Regards,
Willy


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2011/7/27 Willy Tarreau <w@1wt.eu>:
> That's where I think you're mistaken. As long as you think of it as manda=
tory
> this will not be possible.

Hi Willy, as I've explained several times in these threads, if a WS
client is not mandated to perform SRV given a WS URI domain, then the
service provider cannot rely on SRV records. This is, let's assume
that a WS service provider offers a WS URI like "ws://mydomain.org",
with these records (simplifying):

- SRV:   1.1.1.1:80, 1.1.1.2:80, 1.1.1.3:90
- A:        1.1.1.1

So assuming that SRV is optional for clients, the provider does not
know how many clients would use SRV or just A. Imagine the service
provider expects millons of concurrent users. He must be ready to
receive all those users in 1.1.1.1 (if they don't use SRV). So the
service provider must build a complex/expensive balancer system in
server side (BGP and all that stuf) just for the case in which most of
the clients would not perform SRV procedures. So 1.1.1.1 would be in
fact a balancer with N hosts providing the WS service behind it.

At the same time, if clients use SRV then load-balancing and failover
would be automatically provided by the SRV capabilities. No need for
like-BGS solutions in server side.

So the question is: which service provider would spend resources, time
and efforts building two different load-balancing/failover systems?.
If a provider can build the "BGP" solution then it will not need SRV.
And another provider which cannot build a "BGP" solution could not
entirely rely on SRV capabilities as maybe most of the clients would
not use it.

In the other side, if service providers must be ready to accept all
the traffic in the IP resolved via single DNS A query, then nobody
would need SRV and webbrowsers would just remove it (the Mozilla
developer has already said in these threads that he won't implement it
"due to latency reasons").

Also the fact that DNS is not mandated by HTTP neither WS makes things
even worse. In contrast, in XMPP and SIP protocols DNS is mandatory
and also SRV procedures. If you are a SIP provider hosting the service
with domain "mydomain.org" you don't ever need to have a working A
record for mydomain.org, but just SRV (and optionally NAPTR). Or you
could just have A record and not use SRV at all. SIP client procedures
for locating servers (RFC 3263) mandate first trying SRV and then
A/AAAA, so things would work in any case.

So correct me if I'm wrong, but making SRV optional for clients (in
*any* protocol) is not something I consider feasible or useful.

Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From w@1wt.eu  Wed Jul 27 05:45:53 2011
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From: Willy Tarreau <w@1wt.eu>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Wed, Jul 27, 2011 at 12:46:28PM +0200, Iñaki Baz Castillo wrote:
> Hi Willy, as I've explained several times in these threads, if a WS
> client is not mandated to perform SRV given a WS URI domain, then the
> service provider cannot rely on SRV records. This is, let's assume
> that a WS service provider offers a WS URI like "ws://mydomain.org",
> with these records (simplifying):
> 
> - SRV:   1.1.1.1:80, 1.1.1.2:80, 1.1.1.3:90
> - A:        1.1.1.1

No, he would have :

 - SRV:   1.1.1.1:80, 1.1.1.2:80, 1.1.1.3:90
 - A:     1.1.1.1, 1.1.1.2, 1.1.1.3

> So assuming that SRV is optional for clients, the provider does not
> know how many clients would use SRV or just A.

That's mainly the point : the provider would offer a service which is
designed to work both ways and which a part of the users would experience
better due to their consideration of the SRV record.

> Imagine the service
> provider expects millons of concurrent users. He must be ready to
> receive all those users in 1.1.1.1 (if they don't use SRV). So the
> service provider must build a complex/expensive balancer system in
> server side (BGP and all that stuf) just for the case in which most of
> the clients would not perform SRV procedures. So 1.1.1.1 would be in
> fact a balancer with N hosts providing the WS service behind it.
> 
> At the same time, if clients use SRV then load-balancing and failover
> would be automatically provided by the SRV capabilities. No need for
> like-BGS solutions in server side.

Once again, the goal to make SRV adopted BY USERS is not to ensure that
it tries to cover all the server-side needs, but that it offers better
quality of service to USERS. That way USERS will massively adopt it and
server will one day be able to safely rely on it. Just like neither
Javascript nor cookies nor flash are mandatory, still all of them are
very common in practice and many service providers happily rely on them.

> So the question is: which service provider would spend resources, time
> and efforts building two different load-balancing/failover systems?.
> If a provider can build the "BGP" solution then it will not need SRV.
> And another provider which cannot build a "BGP" solution could not
> entirely rely on SRV capabilities as maybe most of the clients would
> not use it.

You can't transform the Internet in one day after publishing one draft.
Look at the Set-Cookie2 header. It was a massive failure. It was thought
that the broken Set-Cookie header could be fixed by one new RFC, and
nobody adopted it. The reason is that it was proposed as a replacement.
I think that Opera might be the only browser to parse it. In contrast,
if your proposal is backwards compatible and incites users to enable it
on their side, then in a few years it can become a de-facto standard.

> In the other side, if service providers must be ready to accept all
> the traffic in the IP resolved via single DNS A query, then nobody
> would need SRV and webbrowsers would just remove it (the Mozilla
> developer has already said in these threads that he won't implement it
> "due to latency reasons").

That's why I explained that you have to find what other benefits *for
the end user* can be brought by this. End users are deciding, not site
owners.

> Also the fact that DNS is not mandated by HTTP neither WS makes things
> even worse.

That's why SRV cannot be made mandatory.

> In contrast, in XMPP and SIP protocols DNS is mandatory
> and also SRV procedures. If you are a SIP provider hosting the service
> with domain "mydomain.org" you don't ever need to have a working A
> record for mydomain.org, but just SRV (and optionally NAPTR). Or you
> could just have A record and not use SRV at all. SIP client procedures
> for locating servers (RFC 3263) mandate first trying SRV and then
> A/AAAA, so things would work in any case.
> 
> So correct me if I'm wrong, but making SRV optional for clients (in
> *any* protocol) is not something I consider feasible or useful.

Feasible yes, useful I don't know, and I'm not the one trying to argue
that it can add value. If it can bring value to the user, some users
will enable it despite the possible added latency. If it does not
bring them anything, they won't use it.

Regards,
Willy


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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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2011/7/27 Willy Tarreau <w@1wt.eu>:
> Once again, the goal to make SRV adopted BY USERS is not to ensure that
> it tries to cover all the server-side needs, but that it offers better
> quality of service to USERS. That way USERS will massively adopt it and
> server will one day be able to safely rely on it. Just like neither
> Javascript nor cookies nor flash are mandatory, still all of them are
> very common in practice and many service providers happily rely on them.

Thanks for your useful comments (as always in these threads). Let me
just a question:

What do you mean with "USERS"? Do you mean home users in front of
their web browsers? or webbrowser vendors?
I don't think home users (neither professional users) has nothing to
decide here, they will not "resolve" the WS URI retrieved from a
webpage.
So we are talking about webbrowser vendors, right? and typically there
are no more than.... 10?

Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From w@1wt.eu  Wed Jul 27 07:03:08 2011
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Wed, Jul 27, 2011 at 03:45:38PM +0200, Iñaki Baz Castillo wrote:
> 2011/7/27 Willy Tarreau <w@1wt.eu>:
> > Once again, the goal to make SRV adopted BY USERS is not to ensure that
> > it tries to cover all the server-side needs, but that it offers better
> > quality of service to USERS. That way USERS will massively adopt it and
> > server will one day be able to safely rely on it. Just like neither
> > Javascript nor cookies nor flash are mandatory, still all of them are
> > very common in practice and many service providers happily rely on them.
> 
> Thanks for your useful comments (as always in these threads). Let me
> just a question:
> 
> What do you mean with "USERS"? Do you mean home users in front of
> their web browsers? or webbrowser vendors?

Users in front of web browsers.

> I don't think home users (neither professional users) has nothing to
> decide here, they will not "resolve" the WS URI retrieved from a
> webpage.

I think you're wrong. Those are these users which ask for feature XXX or
YYY that they like because it brings them a better experience. If you can
find a real benefit for the end user, there will be an option in the browser
and some of them will enable it. It's just important to find how an end user
may benefit from making use of SRV tags when connecting to his favorite site
instead of using just CNAME or A/AAAA. Maybe being able to always connect to
less loaded servers would be appreciated, because some site maintainers will
start announcing new servers. Maybe there are solutions to provide better
geolocation using SRV than with A (ie: let the web browser decide which field
to use instead of relying on its resolver's IP address). Maybe it will be
possible for mobile users to automatically select a different port which is
not subject to annoying transparent proxies at their provider. I don't know.
You must think in terms of better experience which might be brought via
better quality of service. Surely a DNS record might provide information to
improve QoS based on the browser's decision.

> So we are talking about webbrowser vendors, right? and typically there
> are no more than.... 10?

Browsers implement what their users ask for. They don't want to add features
that are not desired and make experience worse or reduce reliability. But if
users ask for something, they'll certainly implement it.

Regards,
Willy


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2011/7/27 Willy Tarreau <w@1wt.eu>:
>> I don't think home users (neither professional users) has nothing to
>> decide here, they will not "resolve" the WS URI retrieved from a
>> webpage.
>
> I think you're wrong. Those are these users which ask for feature XXX or
> YYY that they like because it brings them a better experience. If you can
> find a real benefit for the end user, there will be an option in the brow=
ser
> and some of them will enable it. It's just important to find how an end u=
ser
> may benefit from making use of SRV tags when connecting to his favorite s=
ite
> instead of using just CNAME or A/AAAA. Maybe being able to always connect=
 to
> less loaded servers would be appreciated, because some site maintainers w=
ill
> start announcing new servers. Maybe there are solutions to provide better
> geolocation using SRV than with A (ie: let the web browser decide which f=
ield
> to use instead of relying on its resolver's IP address). Maybe it will be
> possible for mobile users to automatically select a different port which =
is
> not subject to annoying transparent proxies at their provider. I don't kn=
ow.
> You must think in terms of better experience which might be brought via
> better quality of service. Surely a DNS record might provide information =
to
> improve QoS based on the browser's decision.
>
>> So we are talking about webbrowser vendors, right? and typically there
>> are no more than.... 10?
>
> Browsers implement what their users ask for. They don't want to add featu=
res
> that are not desired and make experience worse or reduce reliability. But=
 if
> users ask for something, they'll certainly implement it.


Well, I understand (and agree) most of your text, but I still think
that the URI resolution mechanism is something transparent for an
end-user. This is not like having FlashPlayer for showing annoying and
dancing menus in a web page XD. End-users ask for FlashPlayer (and
Android 2.3 has included it for example) but end-users won't ask for
"SRV procedures".

I would translate your arguments to "web developers", those who want
to provide scalable systems and for which having some kind of QoS
mechanisms (specially for mobile devices) is a great advantage. Of
course, if this happens then end-users would be happier :)

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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On Wed, Jul 27, 2011 at 05:19:30PM +0200, Iñaki Baz Castillo wrote:
> Well, I understand (and agree) most of your text, but I still think
> that the URI resolution mechanism is something transparent for an
> end-user. This is not like having FlashPlayer for showing annoying and
> dancing menus in a web page XD. End-users ask for FlashPlayer (and
> Android 2.3 has included it for example) but end-users won't ask for
> "SRV procedures".

They won't "ask for", but when some sites pretend "we work better with",
then they'll ask for the techno to use the site better. It's similar to
the HTTP/1.0 or 1.1, keep-alive and pipelining that you can enable in
your browser. Most users won't play with it, still that's pretty
configurable. Similarly end users can decide to enable/disable IPv6 or
IDN.

> I would translate your arguments to "web developers", those who want
> to provide scalable systems and for which having some kind of QoS
> mechanisms (specially for mobile devices) is a great advantage. Of
> course, if this happens then end-users would be happier :)

I think you got it :-)

Willy


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Date: Wed, 27 Jul 2011 18:07:12 +0100
From: Dave Cridland <dave@cridland.net>
To: Willy Tarreau <w@1wt.eu>, =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>, Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>, Mark Andrews <marka@isc.org>
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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Wed Jul 27 06:25:49 2011, Willy Tarreau wrote:
> On Wed, Jul 27, 2011 at 11:28:06AM +1000, Mark Andrews wrote:
> > > SRV provides load-balancing and failover. I never said that SRV  
> is a
> > > solution for temporaly put in maintenance a server.
> >
> > Happy eyeballs however does allow you to take a server out of  
> production
> > and not really notice it.  Note webbrowsers have had the ability  
> to do
> > this for as long as webbrowsers have existed.
> 
> Mark, could you elaborate on this point ? Surely you're not talking  
> about
> the fact that a browser retries a failed connection, but I'm not  
> not sure
> what mechanism you're talking about.

Happy eyeballs - try everything as soon as you can, in parallel. Drop  
everything else when one does.

Dave.
-- 
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From w@1wt.eu  Wed Jul 27 10:13:43 2011
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From: Willy Tarreau <w@1wt.eu>
To: Dave Cridland <dave@cridland.net>
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References: <CALiegfnYm6g63JDHLiSH__r-or3kzK0XCVa3cC7RMP14KWBOSg@mail.gmail.com> <20110724120751.GQ22405@1wt.eu> <CALiegfncavmoMp4YDCeeJ3rsOfHAYQ99itKX2Q2eHB351T3X5A@mail.gmail.com> <20110724184537.GZ22405@1wt.eu> <CALiegfne620wuDMAp235n3mVcXTAnbhhNm8vpiNCy5F7+VD92A@mail.gmail.com> <20110724192948.GD22405@1wt.eu> <CALiegf=e48kkF+Gky1mY7LippUB-0kZDgSGZrJxk1aZupAGkYw@mail.gmail.com> <20110727012806.EBB811231907@drugs.dv.isc.org> <20110727052549.GB4086@1wt.eu> <9031.1311786432.357811@puncture>
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Cc: Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>, Mark Andrews <marka@isc.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Wed, Jul 27, 2011 at 06:07:12PM +0100, Dave Cridland wrote:
> On Wed Jul 27 06:25:49 2011, Willy Tarreau wrote:
> >On Wed, Jul 27, 2011 at 11:28:06AM +1000, Mark Andrews wrote:
> >> > SRV provides load-balancing and failover. I never said that SRV  
> >is a
> >> > solution for temporaly put in maintenance a server.
> >>
> >> Happy eyeballs however does allow you to take a server out of  
> >production
> >> and not really notice it.  Note webbrowsers have had the ability  
> >to do
> >> this for as long as webbrowsers have existed.
> >
> >Mark, could you elaborate on this point ? Surely you're not talking  
> >about
> >the fact that a browser retries a failed connection, but I'm not  
> >not sure
> >what mechanism you're talking about.
> 
> Happy eyeballs - try everything as soon as you can, in parallel. Drop  
> everything else when one does.

This sensibly increases load on servers with useless connections, and
is counter-efficient in mobile networks where sending uplink packets
is the limiting factor !

Willy


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Cc: Server-Initiated HTTP <hybi@ietf.org>, Mark Andrews <marka@isc.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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[dropping ietf@ietf.org]

On 7/27/11 1:13 PM, Willy Tarreau wrote:
> On Wed, Jul 27, 2011 at 06:07:12PM +0100, Dave Cridland wrote:
>> On Wed Jul 27 06:25:49 2011, Willy Tarreau wrote:
>>> On Wed, Jul 27, 2011 at 11:28:06AM +1000, Mark Andrews wrote:
>>>>> SRV provides load-balancing and failover. I never said that SRV  
>>> is a
>>>>> solution for temporaly put in maintenance a server.
>>>>
>>>> Happy eyeballs however does allow you to take a server out of  
>>> production
>>>> and not really notice it.  Note webbrowsers have had the ability  
>>> to do
>>>> this for as long as webbrowsers have existed.
>>>
>>> Mark, could you elaborate on this point ? Surely you're not talking  
>>> about
>>> the fact that a browser retries a failed connection, but I'm not  
>>> not sure
>>> what mechanism you're talking about.
>>
>> Happy eyeballs - try everything as soon as you can, in parallel. Drop  
>> everything else when one does.
> 
> This sensibly increases load on servers with useless connections, and
> is counter-efficient in mobile networks where sending uplink packets
> is the limiting factor !

Guys, it feels like you're playing ping-pong here. It's fun for us to
watch the ball go back and forth, but we're not moving forward. Can you
see a way to make progress toward consensus?

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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On Wed Jul 27 19:09:34 2011, Peter Saint-Andre wrote:
> [dropping ietf@ietf.org]
> 
> On 7/27/11 1:13 PM, Willy Tarreau wrote:
> > On Wed, Jul 27, 2011 at 06:07:12PM +0100, Dave Cridland wrote:
> >> On Wed Jul 27 06:25:49 2011, Willy Tarreau wrote:
> >>> On Wed, Jul 27, 2011 at 11:28:06AM +1000, Mark Andrews wrote:
> >>>>> SRV provides load-balancing and failover. I never said that  
> SRV
> >>> is a
> >>>>> solution for temporaly put in maintenance a server.
> >>>>
> >>>> Happy eyeballs however does allow you to take a server out of
> >>> production
> >>>> and not really notice it.  Note webbrowsers have had the  
> ability
> >>> to do
> >>>> this for as long as webbrowsers have existed.
> >>>
> >>> Mark, could you elaborate on this point ? Surely you're not  
> talking
> >>> about
> >>> the fact that a browser retries a failed connection, but I'm not
> >>> not sure
> >>> what mechanism you're talking about.
> >>
> >> Happy eyeballs - try everything as soon as you can, in parallel.  
> Drop
> >> everything else when one does.
> >
> > This sensibly increases load on servers with useless connections,  
> and
> > is counter-efficient in mobile networks where sending uplink  
> packets
> > is the limiting factor !
> 
> Guys, it feels like you're playing ping-pong here. It's fun for us  
> to
> watch the ball go back and forth, but we're not moving forward. Can  
> you
> see a way to make progress toward consensus?

No.

The opinion seems to be that WebSockets must resolve in precisely the  
same way of http URIs, and as such it's not clear to me what benefit  
a new scheme has. This means in order to sway opinion on WebSockets,  
which operationally are likely to behave far more like XMPP and SIP  
services, with long-lived connections and state, we have to also sway  
opinion on using SRV for HTTP, where the benefits are more marginal  
(due to the status quo, having solved the problems in hardware,  
mostly) and the costs much higher (due to the higher number of  
connections in HTTP).

Furthermore, at least one browser vendor has clearly stated they  
would not implement SRV lookup even if the specification mandates it.

As such, it seems pointless wasting my time and effort on this.

Dave.
-- 
Dave Cridland - mailto:dave@cridland.net - xmpp:dwd@dave.cridland.net
  - acap://acap.dave.cridland.net/byowner/user/dwd/bookmarks/
  - http://dave.cridland.net/
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From marka@isc.org  Wed Jul 27 16:10:36 2011
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References: <CALiegf=4K2oWfmZjGMD7J_jyaDtS3i+Mu7R0Wh75Rr+MrQCjtw@mail.gmail.com> <20110722054345.GE18126@1wt.eu> <CALiegfnYm6g63JDHLiSH__r-or3kzK0XCVa3cC7RMP14KWBOSg@mail.gmail.com> <20110724120751.GQ22405@1wt.eu> <CALiegfncavmoMp4YDCeeJ3rsOfHAYQ99itKX2Q2eHB351T3X5A@mail.gmail.com> <20110724184537.GZ22405@1wt.eu> <CALiegfne620wuDMAp235n3mVcXTAnbhhNm8vpiNCy5F7+VD92A@mail.gmail.com> <20110724192948.GD22405@1wt.eu> <CALiegf=e48kkF+Gky1mY7LippUB-0kZDgSGZrJxk1aZupAGkYw@mail.gmail.com> <20110727012806.EBB811231907@drugs.dv.isc.org> <20110727052549.GB4086@1wt.eu> <9031.1311786432.357811@puncture>
In-reply-to: Your message of "Wed, 27 Jul 2011 18:07:12 +0100." <9031.1311786432.357811@puncture>
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Cc: Server-Initiated HTTP <hybi@ietf.org>, IETF-Discussion <ietf@ietf.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt> (The WebSocket protocol) to Proposed Standard
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In message <9031.1311786432.357811@puncture>, Dave Cridland writes:
> On Wed Jul 27 06:25:49 2011, Willy Tarreau wrote:
> > On Wed, Jul 27, 2011 at 11:28:06AM +1000, Mark Andrews wrote:
> > > > SRV provides load-balancing and failover. I never said that SRV  
> > is a
> > > > solution for temporaly put in maintenance a server.
> > >
> > > Happy eyeballs however does allow you to take a server out of  
> > production
> > > and not really notice it.  Note webbrowsers have had the ability  
> > to do
> > > this for as long as webbrowsers have existed.
> > 
> > Mark, could you elaborate on this point ? Surely you're not talking  
> > about
> > the fact that a browser retries a failed connection, but I'm not  
> > not sure
> > what mechanism you're talking about.
> 
> Happy eyeballs - try everything as soon as you can, in parallel. Drop  
> everything else when one does.

More correctly it is try the first address and if that doesn't
connect in a short period (150...250ms) start a second connection
to the next address while continuing with the first.  If you have
more that 2 address you do something similar for the next one (I
use 1/2 the original timeout, but that is a implementation detail).
You continue to use the address that works for that session.  You
drop any other connections to other addresses that complete.

You get fast failover without lots of extra connection attempts
with the advantage that you detect and recover from network failures
that only affect the client.

Mark
-- 
Mark Andrews, ISC
1 Seymour St., Dundas Valley, NSW 2117, Australia
PHONE: +61 2 9871 4742                 INTERNET: marka@isc.org

From bruce@callenish.com  Wed Jul 27 16:56:56 2011
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Subject: [hybi] Moving forward: SHOULD use SRV?
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> Guys, it feels like you're playing ping-pong here. It's fun for us to
> watch the ball go back and forth, but we're not moving forward. Can you
> see a way to make progress toward consensus?

My read of the situation is that we have established that requiring SRV 
lookups from Websockets clients with the use of MUST is out of the 
question. At least one browser vendor has said they will not implement 
it, and in addition there are the HTTP proxy and connection reuse 
issues. It looks to me like most of the advocates of that position are 
(reluctantly) accepting that it will not be adopted.

OTOH, a number of voices have been raised in support or acceptance of 
recommending the use of SRV records to resolve hostnames in Websocket 
URLs as a best practice through the use of the SHOULD keyword. Others 
(particularly on the ietf mailing list) have said that unless it is a 
MUST you shouldn't bother doing it at all.

Perhaps consensus (or lack of it) on this idea could be looked for at 
the meeting tomorrow.


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Subject: Re: [hybi] IETF81 draft agenda
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Sal,

Are there some remote access details available? Jabber? Dialin?

thanks


On 18 July 2011 07:28, Salvatore Loreto <salvatore.loreto@ericsson.com> wro=
te:
> Hi there,
>
> here just a preliminary draft of the agenda
> ( http://www.ietf.org/proceedings/81/agenda/hybi.txt )
>
> =A0Thursday, July 28th, 2011
> =A017:40 - 19:40 Afternoon Session II
> =A0room 204 B
>
> =A0- =A0Administrative/Agenda bash =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0(Chairs - 10m)
>
> =A0- =A0status and other business about the finalization
> =A0 =A0 of the main spec =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0(Ian Fette? -
> 40m)
> =A0 =A0 (raft-ietf-hybi-thewebsocketprotocol-10)
>
> =A0- =A0extensions and other options : =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0(? - 60m)
> =A0 =A0 (e.g. among the others: Multiplex, Frame Compression,
> =A0 =A0 =A0Timeout etc.)
>
> =A0- =A0Open Discussion =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0(All - 10m)
>
>
>
> cheers
> /Sal
>
> --
> Salvatore Loreto
> www.sloreto.com
>
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>

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Greg, all,=0A=0Athere will be a virtual room reserved on the Meetecho sys=
tem.=0A=0AAccess to the on-line session (including audio and video stream=
s) will be available from 17:30 at:=0Ahttp://www.meetecho.com/ietf81/hybi=
=0A=0AThe Meetecho session automatically logs you into the standard IETF =
jabber room. So, from there, you can have an integrated experience involv=
ing all media and allowing you to interact with the room.=0A=0AA tutorial=
 of interactivity features of the tool can be found at:=0Ahttp://www.meet=
echo.com/ietf81/tutorials=0A=0AFor further information you can contact us=
 at ietf-support@meetecho.com.=0A=0ACheers,=0Athe Meetecho Team=0A=0ADa: =
hybi-bounces@ietf.org=0AA: "Salvatore Loreto" salvatore.loreto@ericsson.c=
om=0ACc: "hybi@ietf.org" hybi@ietf.org=0AData: Thu, 28 Jul 2011 10:12:32 =
+1000=0AOggetto: Re: [hybi] IETF81 draft agenda=0A=0A> Sal,=0A> =0A> Are =
there some remote access details available? Jabber? Dialin?=0A> =0A> than=
ks=0A> =0A> =0A> On 18 July 2011 07:28, Salvatore Loreto  wrote:=0A> > Hi=
 there,=0A> >=0A> > here just a preliminary draft of the agenda=0A> > ( h=
ttp://www.ietf.org/proceedings/81/agenda/hybi.txt )=0A> >=0A> > =C2=A0Thu=
rsday, July 28th, 2011=0A> > =C2=A017:40 - 19:40 Afternoon Session II=0A>=
 > =C2=A0room 204 B=0A> >=0A> > =C2=A0- =C2=A0Administrative/Agenda bash =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0(Chairs - 10m)=0A> >=0A> > =C2=A0- =
=C2=A0status and other business about the finalization=0A> > =C2=A0 =C2=A0=
 of the main spec =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0=
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0(Ian Fette? -=0A> > 40m)=0A> > =C2=A0 =C2=A0 (raft-ietf-=
hybi-thewebsocketprotocol-10)=0A> >=0A> > =C2=A0- =C2=A0extensions and ot=
her options : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0(? - 60m)=0A> > =C2=A0 =C2=A0 (e.g.=
 among the others: Multiplex, Frame Compression,=0A> > =C2=A0 =C2=A0 =C2=A0=
Timeout etc.)=0A> >=0A> > =C2=A0- =C2=A0Open Discussion =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0=
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0(All - 10m=
)=0A> >=0A> >=0A> >=0A> > cheers=0A> > /Sal=0A> >=0A> > --=0A> > Salvator=
e Loreto=0A> > www.sloreto.com=0A> >=0A> > ______________________________=
_________________=0A> > hybi mailing list=0A> > hybi@ietf.org=0A> > https=
://www.ietf.org/mailman/listinfo/hybi=0A> >=0A> _________________________=
______________________=0A> hybi mailing list=0A> hybi@ietf.org=0A> https:=
//www.ietf.org/mailman/listinfo/hybi


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In message <201107272350.p6RNodKa019978@fs4113.wdf.sap.corp>, Martin Rex writes
:
> Mark Andrews wrote:
> > 
> > Dave Cridland writes:
> > > 
> > > Happy eyeballs - try everything as soon as you can, in parallel. Drop  
> > > everything else when one does.
> > 
> > More correctly it is try the first address and if that doesn't
> > connect in a short period (150...250ms) start a second connection
> > to the next address while continuing with the first.  If you have
> > more that 2 address you do something similar for the next one (I
> > use 1/2 the original timeout, but that is a implementation detail).
> > You continue to use the address that works for that session.  You
> > drop any other connections to other addresses that complete.
> 
> Happy eyeballs means that a clients reaction to congestion is
> to perform an DoS attack, flood the network with additional
> connection requests and hammer the server with many additional
> half-open connections that will never actually get used.

It is not a DoS attack.  The client is almost certainly going to
make those connection attempts anyway if the path is congested
enough to cause the first connection attempt to fail.  The only
difference is the application gives up in 30 seconds rather than
60 or 90 seconds by doing the attempts serially.

> While this might currently "improve" the end user experience
> of clients, it simultaneously adds a deterrant to server operators
> to announce IPv6 addresses (even multiple IP addresses -- they're
> better of with IPv4 NAT if they have multiple servers at a single
> location).
-- 
Mark Andrews, ISC
1 Seymour St., Dundas Valley, NSW 2117, Australia
PHONE: +61 2 9871 4742                 INTERNET: marka@isc.org

From mrex@sap.com  Wed Jul 27 16:51:18 2011
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Mark Andrews wrote:
> 
> Dave Cridland writes:
> > 
> > Happy eyeballs - try everything as soon as you can, in parallel. Drop  
> > everything else when one does.
> 
> More correctly it is try the first address and if that doesn't
> connect in a short period (150...250ms) start a second connection
> to the next address while continuing with the first.  If you have
> more that 2 address you do something similar for the next one (I
> use 1/2 the original timeout, but that is a implementation detail).
> You continue to use the address that works for that session.  You
> drop any other connections to other addresses that complete.

Happy eyeballs means that a clients reaction to congestion is
to perform an DoS attack, flood the network with additional
connection requests and hammer the server with many additional
half-open connections that will never actually get used.

While this might currently "improve" the end user experience
of clients, it simultaneously adds a deterrant to server operators
to announce IPv6 addresses (even multiple IP addresses -- they're
better of with IPv4 NAT if they have multiple servers at a single
location).

-Martin

From sm@elandsys.com  Wed Jul 27 23:26:26 2011
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The IETF 81 HYBI WG session will held in room 202 at 5:40 p.m. (EDT) 
on Thursday, July 28.  The agenda has been posted to 
http://tools.ietf.org/wg/hybi/agenda

The audio stream is at http://ietf81streaming.dnsalias.net/ietf/ietf801.m3u

The Jabber room is xmpp:hybi@jabber.ietf.org

Regards,
S. Moonesamy
YAM WG Secretary


From salvatore.loreto@ericsson.com  Thu Jul 28 06:23:50 2011
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Hi there,

I have just update the agenda and also the slides we are going to 
present and discuss
during the today meeting.

you can find all the material here (looking for HyBi):
https://datatracker.ietf.org/meeting/81/materials.html


(the last presentation is still missing, I apologize for the 
inconvenient and I hope to be able
to upload it soon)

cheers
/Sal

-- 
Salvatore Loreto
www.sloreto.com


From stpeter@stpeter.im  Thu Jul 28 06:34:40 2011
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On 7/28/11 9:23 AM, Salvatore Loreto wrote:
> Hi there,
> 
> I have just update the agenda and also the slides we are going to
> present and discuss
> during the today meeting.
> 
> you can find all the material here (looking for HyBi):
> https://datatracker.ietf.org/meeting/81/materials.html

Specifically:

https://datatracker.ietf.org/meeting/81/materials.html#wg-hybi

For remote participants, the audio stream will be here:

http://ietf81streaming.dnsalias.net/ietf/ietf801.m3u

And the chatroom is here:

xmpp:hybi@jabber.ietf.org?join

You can listen to the audio and channel questions back to the meeting
via the jabber room.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From salvatore.loreto@ericsson.com  Thu Jul 28 11:10:48 2011
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now all the presentations have been uploaded.

cheers
/Sal

On 7/28/11 9:34 AM, Peter Saint-Andre wrote:
> On 7/28/11 9:23 AM, Salvatore Loreto wrote:
>> Hi there,
>>
>> I have just update the agenda and also the slides we are going to
>> present and discuss
>> during the today meeting.
>>
>> you can find all the material here (looking for HyBi):
>> https://datatracker.ietf.org/meeting/81/materials.html
> Specifically:
>
> https://datatracker.ietf.org/meeting/81/materials.html#wg-hybi
>
> For remote participants, the audio stream will be here:
>
> http://ietf81streaming.dnsalias.net/ietf/ietf801.m3u
>
> And the chatroom is here:
>
> xmpp:hybi@jabber.ietf.org?join
>
> You can listen to the audio and channel questions back to the meeting
> via the jabber room.
>
> Peter

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#31: Concept of "users" needs clarification

Changes (by salvatore.loreto@â€¦):

  * status:  new => closed
  * resolution:  => fixed


Comment:

 fixed in version -10

-- 
--------------------------------------+-------------------------------------
 Reporter:  g_e_montenegro@â€¦          |        Owner:        
     Type:  defect                    |       Status:  closed
 Priority:  major                     |    Milestone:        
Component:  thewebsocketprotocol      |      Version:        
 Severity:  -                         |   Resolution:  fixed 
 Keywords:                            |  
--------------------------------------+-------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/hybi/trac/ticket/31#comment:2>
hybi <http://tools.ietf.org/hybi/>
The Hypertext-Bidirectional (HyBi) working group will seek
standardization of one approach to maintain bidirectional
communications between the HTTP client, server and intermediate
entities, which will provide more efficiency compared to the current
use of hanging requests.

From trac@tools.ietf.org  Thu Jul 28 14:22:56 2011
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Subject: Re: [hybi] #14: TLS/NPN for a next version of the specification
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#14: TLS/NPN for a next version of the specification

Changes (by salvatore.loreto@â€¦):

  * component:  thewebsocketprotocol => websocket-requirements


-- 
------------------------------------+---------------------------------------
 Reporter:  sm+ietf@â€¦               |       Owner:     
     Type:  enhancement             |      Status:  new
 Priority:  major                   |   Milestone:     
Component:  websocket-requirements  |     Version:     
 Severity:  Active WG Document      |    Keywords:     
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/hybi/trac/ticket/14#comment:1>
hybi <http://tools.ietf.org/hybi/>
The Hypertext-Bidirectional (HyBi) working group will seek
standardization of one approach to maintain bidirectional
communications between the HTTP client, server and intermediate
entities, which will provide more efficiency compared to the current
use of hanging requests.

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#38: Error handling

Changes (by salvatore.loreto@â€¦):

  * status:  new => closed
  * resolution:  => fixed


Comment:

 it has been addressed in the last versions of the draft.

-- 
-------------------------------------------+--------------------------------
 Reporter:  salvatore.loreto@â€¦             |        Owner:  ifette@â€¦         
     Type:  defect                         |       Status:  closed           
 Priority:  major                          |    Milestone:                   
Component:  thewebsocketprotocol           |      Version:                   
 Severity:  -                              |   Resolution:  fixed            
 Keywords:                                 |  
-------------------------------------------+--------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/hybi/trac/ticket/38#comment:2>
hybi <http://tools.ietf.org/hybi/>
The Hypertext-Bidirectional (HyBi) working group will seek
standardization of one approach to maintain bidirectional
communications between the HTTP client, server and intermediate
entities, which will provide more efficiency compared to the current
use of hanging requests.

From stpeter@stpeter.im  Thu Jul 28 15:13:41 2011
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Date: Thu, 28 Jul 2011 18:13:36 -0400
From: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: [hybi] x- for private-use-token
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For input regarding the "x-" convention for the private-use-token
construction, please see this draft:

https://datatracker.ietf.org/doc/draft-saintandre-xdash/

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From gregw@intalio.com  Thu Jul 28 17:02:27 2011
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Subject: [hybi] websocket test suite
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All,

I'd like to restart the conversation around developing a standard set
of tests for websocket implementations.  The draft I had previously
published has expired, and reading it again now, I'm not sure it is a
good starting point as the approach adopted was a bit laissez faire
and was just a few test that was felt to check the bulk of the
specified behaviour.   I now think that what is needed is a more
rigorous approach with specific tests designed to verify each section
of the specification.  ie there should be traceability from a clause
in the spec to the test(s) that verify it.

So firstly, does anybody have suggestions of good tests suites
specified for other IETF efforts that we could use as a model to base
our tests on?

Anybody else interested in collaborating on this?

cheers

From gregw@intalio.com  Thu Jul 28 17:44:46 2011
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Subject: [hybi] Supporting OAUTH-like authentication?
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All,

Currently the protocol allows arbitrary headers, so in theory an
authentication mechanism like OAUTH could be implemented by exchanging
tokens in the headers of the handshake.    But the js API does not
allow an application to set or see HTTP header from the handshake (nor
should it), so that is not available as a general mechanism for token
passing auth like OAUTH.

So what other options do we have to pass tokens in the handshake?

Cookies - this is possible, but kind of defeats the purpose of  token
passing auth if tokens are to be sent/set in a way that facilitates
wider access to them.

Extensions - we could say that an OAUTH extensions is required to
manipulate the headers of the handshake with OAUTH tokens, but I
expect that there will be a very high barrier for an extension to be
included in browsers, so I think that is probably a non starter (at
least until a defacto-standard auth mechanism emerges).

Subprotocol/CloseEvents - Subprotocol is a user controlled field sent
by the handshake and closeEvents contain a reason string that could
originate from the server application/framework.     Thus I would not
be surprised to see these fields used (misused?) to send challenges
and pass tokens eg.

 1. Client connects to server with oauth subprotocol.
 2. Server accepts connection but immediately closes with a reason
string containing a token and/or authority URL. ( might be able to be
done with a 403 HTTP response if the HTTP reason string is passed to
the onerror call back ?? ]
 3. Client gets the token signed by an/the authority
 4. Client connections to server with oauth;token=value subprotocol
 5. Server validates authorisation of token and accepts connection.

[ forgive me if that is not exactly how oauth works or correct
terminology ... but it is something like that ]

Is this a good usage of the subprotocol field/reason fields?

Is there a better way to integrate such authentication into the handshake?


cheers

From phil127@gmail.com  Thu Jul 28 18:01:03 2011
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Date: Fri, 29 Jul 2011 03:00:48 +0200
From: Philipp Serafin <phil127@gmail.com>
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Subject: Re: [hybi] Supporting OAUTH-like authentication?
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Wouldn't it be better to move the token exchange into the actual WS
connection where we have efficient two-way communication available?

1.: Client connects to server using subprotocol "foo", which is known to
use OAuth as authentication (or e.g. "foo+oauth", if this should be
decided on per-connection basis)
2.: Client and server perform successful WS handshake
3.: Server sends token and authority URL in a WS message.
4.: Client gets token authorized
5.: Client sends authorized token back in a WS message.
6.: Server confirms that authorisation was successful.
7.: Client and server exchange messages as specified by "foo".

Am 29.07.2011 02:44, schrieb Greg Wilkins:
> All,
>
> Currently the protocol allows arbitrary headers, so in theory an
> authentication mechanism like OAUTH could be implemented by exchanging
> tokens in the headers of the handshake.    But the js API does not
> allow an application to set or see HTTP header from the handshake (nor
> should it), so that is not available as a general mechanism for token
> passing auth like OAUTH.
>
> So what other options do we have to pass tokens in the handshake?
>
> Cookies - this is possible, but kind of defeats the purpose of  token
> passing auth if tokens are to be sent/set in a way that facilitates
> wider access to them.
>
> Extensions - we could say that an OAUTH extensions is required to
> manipulate the headers of the handshake with OAUTH tokens, but I
> expect that there will be a very high barrier for an extension to be
> included in browsers, so I think that is probably a non starter (at
> least until a defacto-standard auth mechanism emerges).
>
> Subprotocol/CloseEvents - Subprotocol is a user controlled field sent
> by the handshake and closeEvents contain a reason string that could
> originate from the server application/framework.     Thus I would not
> be surprised to see these fields used (misused?) to send challenges
> and pass tokens eg.
>
>  1. Client connects to server with oauth subprotocol.
>  2. Server accepts connection but immediately closes with a reason
> string containing a token and/or authority URL. ( might be able to be
> done with a 403 HTTP response if the HTTP reason string is passed to
> the onerror call back ?? ]
>  3. Client gets the token signed by an/the authority
>  4. Client connections to server with oauth;token=value subprotocol
>  5. Server validates authorisation of token and accepts connection.
>
> [ forgive me if that is not exactly how oauth works or correct
> terminology ... but it is something like that ]
>
> Is this a good usage of the subprotocol field/reason fields?
>
> Is there a better way to integrate such authentication into the handshake?
>
>
> cheers
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi


From jlee@antwerkz.com  Thu Jul 28 18:26:39 2011
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Subject: Re: [hybi] websocket test suite
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--000e0ce0db5e2fe5fa04a92b2ce6
Content-Type: text/plain; charset=ISO-8859-1

I'm actually building up a set of unit tests in grizzly that i'm going to
use for testing 1.9 clients against 2.0 servers, and vice versa.  I was
hoping to test grizzly client code against jetty servers as well
libwebsocket servers, too.  However, having a standard set of tests to
validate against would increase my confidence that I'm implementing
everything correctly on my end.

On Thu, Jul 28, 2011 at 6:02 PM, Greg Wilkins <gregw@intalio.com> wrote:

> All,
>
> I'd like to restart the conversation around developing a standard set
> of tests for websocket implementations.  The draft I had previously
> published has expired, and reading it again now, I'm not sure it is a
> good starting point as the approach adopted was a bit laissez faire
> and was just a few test that was felt to check the bulk of the
> specified behaviour.   I now think that what is needed is a more
> rigorous approach with specific tests designed to verify each section
> of the specification.  ie there should be traceability from a clause
> in the spec to the test(s) that verify it.
>
> So firstly, does anybody have suggestions of good tests suites
> specified for other IETF efforts that we could use as a model to base
> our tests on?
>
> Anybody else interested in collaborating on this?
>
> cheers
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>



-- 
You can find me on the net at:
http://antwerkz.com
http://antwerkz.com/+
http://antwerkz.com/linkedin
http://antwerkz.com/twitter

--000e0ce0db5e2fe5fa04a92b2ce6
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I&#39;m actually building up a set of unit tests in grizzly that i&#39;m go=
ing to use for testing 1.9 clients against 2.0 servers, and vice versa.=A0 =
I was hoping to test grizzly client code against jetty servers as well libw=
ebsocket servers, too.=A0 However, having a standard set of tests to valida=
te against would increase my confidence that I&#39;m implementing everythin=
g correctly on my end.<br>
<br><div class=3D"gmail_quote">On Thu, Jul 28, 2011 at 6:02 PM, Greg Wilkin=
s <span dir=3D"ltr">&lt;<a href=3D"mailto:gregw@intalio.com">gregw@intalio.=
com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
All,<br>
<br>
I&#39;d like to restart the conversation around developing a standard set<b=
r>
of tests for websocket implementations. =A0The draft I had previously<br>
published has expired, and reading it again now, I&#39;m not sure it is a<b=
r>
good starting point as the approach adopted was a bit laissez faire<br>
and was just a few test that was felt to check the bulk of the<br>
specified behaviour. =A0 I now think that what is needed is a more<br>
rigorous approach with specific tests designed to verify each section<br>
of the specification. =A0ie there should be traceability from a clause<br>
in the spec to the test(s) that verify it.<br>
<br>
So firstly, does anybody have suggestions of good tests suites<br>
specified for other IETF efforts that we could use as a model to base<br>
our tests on?<br>
<br>
Anybody else interested in collaborating on this?<br>
<br>
cheers<br>
_______________________________________________<br>
hybi mailing list<br>
<a href=3D"mailto:hybi@ietf.org">hybi@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/hybi</a><br>
</blockquote></div><br><br clear=3D"all"><br>-- <br>You can find me on the =
net at:<div><a href=3D"http://antwerkz.com" target=3D"_blank">http://antwer=
kz.com</a><br><a href=3D"http://antwerkz.com/+" target=3D"_blank">http://an=
twerkz.com/+</a></div>
<div><a href=3D"http://antwerkz.com/linkedin" target=3D"_blank">http://antw=
erkz.com/linkedin</a><br><a href=3D"http://antwerkz.com/twitter" target=3D"=
_blank">http://antwerkz.com/twitter</a><br><br></div><br>

--000e0ce0db5e2fe5fa04a92b2ce6--

From gregw@intalio.com  Thu Jul 28 18:34:12 2011
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Subject: Re: [hybi] Supporting OAUTH-like authentication?
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Philipp,

Doing the auth within the connection is certainly possible, maybe even
desirable.  However, I think that there could be a concern about not
wanting to accept connections from non-authenticated sources, as to do
so already represents an allocation of resources.    There are also
extra RTT's involved in token exchange after the handshake.

So failing giving access to HTTP headers, doing oauth the "normal" way
in headers in the handshake will not be possible.    Those wanting
authentication in the handshake can probably kludge it now using the
protocol string.      So do we want to better facilitate auth in the
handshake.

Note also, that with your counter example, there are probably things
we could do to make it more elegant.  For instance, perhaps we want to
allow multiple sub-protocols so we don't have to come up with
"foo+oauth", "bar+oauth" etc and instead just negotiate "oauth,foo"

Eitherway, I think we need to step through (probably implement), some
common authentication mechanisms to see how they work with the current
protocol and to see how we can improve support for them.

cheers




On 29 July 2011 11:00, Philipp Serafin <phil127@gmail.com> wrote:
> Wouldn't it be better to move the token exchange into the actual WS
> connection where we have efficient two-way communication available?
>
> 1.: Client connects to server using subprotocol "foo", which is known to
> use OAuth as authentication (or e.g. "foo+oauth", if this should be
> decided on per-connection basis)
> 2.: Client and server perform successful WS handshake
> 3.: Server sends token and authority URL in a WS message.
> 4.: Client gets token authorized
> 5.: Client sends authorized token back in a WS message.
> 6.: Server confirms that authorisation was successful.
> 7.: Client and server exchange messages as specified by "foo".
>
> Am 29.07.2011 02:44, schrieb Greg Wilkins:
>> All,
>>
>> Currently the protocol allows arbitrary headers, so in theory an
>> authentication mechanism like OAUTH could be implemented by exchanging
>> tokens in the headers of the handshake. =A0 =A0But the js API does not
>> allow an application to set or see HTTP header from the handshake (nor
>> should it), so that is not available as a general mechanism for token
>> passing auth like OAUTH.
>>
>> So what other options do we have to pass tokens in the handshake?
>>
>> Cookies - this is possible, but kind of defeats the purpose of =A0token
>> passing auth if tokens are to be sent/set in a way that facilitates
>> wider access to them.
>>
>> Extensions - we could say that an OAUTH extensions is required to
>> manipulate the headers of the handshake with OAUTH tokens, but I
>> expect that there will be a very high barrier for an extension to be
>> included in browsers, so I think that is probably a non starter (at
>> least until a defacto-standard auth mechanism emerges).
>>
>> Subprotocol/CloseEvents - Subprotocol is a user controlled field sent
>> by the handshake and closeEvents contain a reason string that could
>> originate from the server application/framework. =A0 =A0 Thus I would no=
t
>> be surprised to see these fields used (misused?) to send challenges
>> and pass tokens eg.
>>
>> =A01. Client connects to server with oauth subprotocol.
>> =A02. Server accepts connection but immediately closes with a reason
>> string containing a token and/or authority URL. ( might be able to be
>> done with a 403 HTTP response if the HTTP reason string is passed to
>> the onerror call back ?? ]
>> =A03. Client gets the token signed by an/the authority
>> =A04. Client connections to server with oauth;token=3Dvalue subprotocol
>> =A05. Server validates authorisation of token and accepts connection.
>>
>> [ forgive me if that is not exactly how oauth works or correct
>> terminology ... but it is something like that ]
>>
>> Is this a good usage of the subprotocol field/reason fields?
>>
>> Is there a better way to integrate such authentication into the handshak=
e?
>>
>>
>> cheers
>> _______________________________________________
>> hybi mailing list
>> hybi@ietf.org
>> https://www.ietf.org/mailman/listinfo/hybi
>
>

From marka@isc.org  Thu Jul 28 20:05:49 2011
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In message <201107290238.p6T2cCLu021118@fs4113.wdf.sap.corp>, Martin Rex writes
:
> Mark Andrews wrote:
> > 
> > Martin Rex writes:
> > >
> > > Mark Andrews wrote:
> > > > 
> > > > More correctly it is try the first address and if that doesn't
> > > > connect in a short period (150...250ms) start a second connection
> > > > to the next address while continuing with the first.  If you have
> > > > more that 2 address you do something similar for the next one
> > > 
> > > Happy eyeballs means that a clients reaction to congestion is
> > > to perform an DoS attack, flood the network with additional
> > > connection requests and hammer the server with many additional
> > > half-open connections that will never actually get used.
> > 
> > It is not a DoS attack.  The client is almost certainly going to
> > make those connection attempts anyway if the path is congested
> > enough to cause the first connection attempt to fail.  The only
> > difference is the application gives up in 30 seconds rather than
> > 60 or 90 seconds by doing the attempts serially.
> 
> 150...250ms  ?!

Yes, that small.  For most people, most connections are "in country"
or "in continent" however there are always exceptions to this.
The times are driven by human delay tolerances.

> For a satellite link you already have started 3 parallel connects
> in non-congested(!) situations. 

Indeed.  However only one will have data sent over it.  The three
way handshake won't even complete for two of them in many cases.
For those that do complete the server won't be woken up in many
cases as no data gets sent.

> just some random IPv4 pings from my office (in germany)
> _without_congestion_:
> 
>    ping  www.asus.com.tw            300-380ms
>    ping  south-america.pool.ntp.org 280-370ms
>    ping  oceania.pool.ntp.org       340-420ms
>    ping  www.eff.org                160-170ms
>    ping  www.ietf79.cn              330-450ms
>    ping  www.ietf76.jp              270-370ms
> 
> So your approach is already hurting the network without congestion!

Only if you think a could of extra TCP connection attemps to servers
on the other side of the world is hurting the network. 
 
B.T.W. I'm well aware of speed of light issues.  Look at my signature.

Mark
-- 
Mark Andrews, ISC
1 Seymour St., Dundas Valley, NSW 2117, Australia
PHONE: +61 2 9871 4742                 INTERNET: marka@isc.org

From gregw@intalio.com  Thu Jul 28 20:15:20 2011
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Subject: Re: [hybi] websocket test suite
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Here is a first cut at a list of tests for section 4 of the spec.  Are
these the kinds of tests people were thinking about?
If so, then I'm happy to go the next step of detail and start
providing more detail for each test.

Obviously some implementation are not going to be able to send some of
the invalid frames or force fragmentation, so some suites will have to
record some tests as skipped, or only testing in one direction etc.




4.2 Base Framing Tests
----------------------

* 7 bit length: send/receive frames of length 0-125
* 16 bit length: send/receive frames of length 126-128, 65536
* 63 bit length: send/receive frames of length 65537

* Non zero RSV: send frames with non zero reserved bits and no
extension. Verify connection is failed.

* Known Opcodes: send/receive frames with text,binary,ping,pong
opcodes (continuation & close tested separately).
* Unknown Opcodes: send frames with unknown opcodes 3-7, B-F. Verify
connection is failed.


4.3 Client to Server Masking
----------------------------

???


4.4 Fragmentation
-----------------

* 1 fragment message: send/receive message in 1 fragment
* 2 fragment message: send/receive message in 2 fragments
* 3 fragment message: send/receive message in 3 fragments
* Injected control: send/receive 2 fragment message with unsolicited
pong message between frames
* Fragmented control: send fragmented ping. Verify that connection is failed.
* Interleaved fragments: send a fragment with FIN set and Continuation
opcode. Verify that connection is failed.
* Handled control: send/receive 2 fragment message with ping message
between frames and last frame delayed. Verify pong is received before
last frame is sent.
* 7 bit fragment: send/receive message with initial/middle/final
fragments of sizes 0-125
* 16 bit fragment: send/receive message with initial/middle/final
fragments of sizes 126-128,65536
* 64 bit fragment: send/receive message with
initial/middle/finalfragments of size 65537


4.5.1 Close Frames
------------------
* Orderly Close: On an idle connection, send a close and receive a close.
* Busy Close: send a close to an endpoint that has sent the first
fragment of a message.
* Simultaneous Close: send a close on both ends of a connection.
* Message after close: attempt to send a message after a close. Verify
message is not received.
* Idle close: ??
* Error Close: ??


4.5.2 Ping
----------
* Empty ping: Send an empty ping, verify pong is received.
* Non empty ping: Send a non empty ping, verify pong is received.


4.5.3 Pong
----------
* Empty pong: Send an empty ping, verify pong is received and is empty.
* Non empty pong: Send a non empty ping, verify pong is received and
has matching content.
* Outstanding ping: Send two pings with different content, reply with
content for only the second. Verify connection stays open.
* Unsolicited pong: Send/recv unsolicited pong.


4.5.4 Data Frames
-----------------
* UTF-8 Text 0000-007f: Send/receive text message with characters in
code point range  U+0000 to U+007F
* UTF-8 Text 0080-07ff: Send/receive text message with characters in
code point range  U+0080 to U+07FF
* UTF-8 Text 0800-ffff: Send/receive text message with characters in
code point range  U+0800 to U+FFFF
* UTF-8 Text 010000-10ffff: Send/receive text message with characters
in code point range  U+010000 to U+10FFFF
* Illegal bytes UTF-8: Send message with invalid UTF-8 byte sequence.
Verify message is not received.
* Illegal codes UTF-8: Send message with invalid UTF-8 code points.
Verify message is not received.
* Binary frames: Send/receive message containing illegal UTF-8 bytes
sequences and code points as binary message.

From mrex@sap.com  Thu Jul 28 19:38:52 2011
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Mark Andrews wrote:
> 
> Martin Rex writes:
> >
> > Mark Andrews wrote:
> > > 
> > > More correctly it is try the first address and if that doesn't
> > > connect in a short period (150...250ms) start a second connection
> > > to the next address while continuing with the first.  If you have
> > > more that 2 address you do something similar for the next one
> > 
> > Happy eyeballs means that a clients reaction to congestion is
> > to perform an DoS attack, flood the network with additional
> > connection requests and hammer the server with many additional
> > half-open connections that will never actually get used.
> 
> It is not a DoS attack.  The client is almost certainly going to
> make those connection attempts anyway if the path is congested
> enough to cause the first connection attempt to fail.  The only
> difference is the application gives up in 30 seconds rather than
> 60 or 90 seconds by doing the attempts serially.

150...250ms  ?!

For a satellite link you already have started 3 parallel connects
in non-congested(!) situations. 

just some random IPv4 pings from my office (in germany)
_without_congestion_:

   ping  www.asus.com.tw            300-380ms
   ping  south-america.pool.ntp.org 280-370ms
   ping  oceania.pool.ntp.org       340-420ms
   ping  www.eff.org                160-170ms
   ping  www.ietf79.cn              330-450ms
   ping  www.ietf76.jp              270-370ms

So your approach is already hurting the network without congestion!

-Martin

From ietf@meetecho.com  Fri Jul 29 00:44:37 2011
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Dear all,=0A=0Athe full recording (synchronized video, audio, slides and =
jabber room) =0Aof yesterday's HYBI session is available.=0A=0AYou can wa=
tch it by either clicking the proper link on the remote participation pag=
e (http://www.ietf.org/meeting/81/remote-participation.html#Meetecho), or=
 by directly accessing the following URL:=0Ahttp://www.meetecho.com/ietf8=
1/recordings=0A=0AFor the chair(s): please feel free to put the link to t=
he recording in the minutes, if you think this might be useful.=0A=0AIn c=
ase of problems with the playout, just drop an e-mail to =0Aietf-support@=
meetecho.com.=0A=0ACheers,=0Athe Meetecho Team


From pch-b2B3A6689@u-1.phicoh.com  Fri Jul 29 00:56:46 2011
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Cc: hybi@ietf.org, ietf@ietf.org, Mark Andrews <marka@isc.org>
Subject: Re: [hybi] Last Call: <draft-ietf-hybi-thewebsocketprotocol-10.txt>
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In your letter dated Fri, 29 Jul 2011 04:38:12 +0200 (MEST) you wrote:
>Mark Andrews wrote:
>> Martin Rex writes:
>> > Mark Andrews wrote:
>> > > 
>> > > More correctly it is try the first address and if that doesn't
>> > > connect in a short period (150...250ms) start a second connection
>> > > to the next address while continuing with the first.  If you have
>> > > more that 2 address you do something similar for the next one
>> > 
>> > Happy eyeballs means that a clients reaction to congestion is
>> > to perform an DoS attack, flood the network with additional
>> > connection requests and hammer the server with many additional
>> > half-open connections that will never actually get used.
>> 
>> It is not a DoS attack.  The client is almost certainly going to
>> make those connection attempts anyway if the path is congested
>> enough to cause the first connection attempt to fail.  The only
>> difference is the application gives up in 30 seconds rather than
>> 60 or 90 seconds by doing the attempts serially.
>
>150...250ms  ?!
>
>For a satellite link you already have started 3 parallel connects
>in non-congested(!) situations. 
>
>just some random IPv4 pings from my office (in germany)
>_without_congestion_:
>
>   ping  www.asus.com.tw            300-380ms
>   ping  south-america.pool.ntp.org 280-370ms
>   ping  oceania.pool.ntp.org       340-420ms
>   ping  www.eff.org                160-170ms
>   ping  www.ietf79.cn              330-450ms
>   ping  www.ietf76.jp              270-370ms
>
>So your approach is already hurting the network without congestion!

There are two ways to do Happy Eyeballs. A simple immediate solution that
works in most cases to use a fixed timeout value. In your case, you would
have to increase that value to a bit higher than 400ms. If HE was invented
a long time ago, then by now there could have been a parameter in DHCP
to let the network control this parameter.

The other way of doing HE is make it react to observed connect time. In that
case if you are regularly connecting to sites that are more than 400ms away
then the parameter will automatically increase to that value.

The proposal is currently discussed in v6ops and everybody seems to be happy 
with it. So a critical voice may help shape the proposal a bit.



From cgurkanerdogdu@gmail.com  Fri Jul 29 09:05:52 2011
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From: Gurkan Erdogdu <cgurkanerdogdu@gmail.com>
To: hybi@ietf.org
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Subject: [hybi] [INFO] Siwpas, Java Web Server Supporting Web Socket Protocol
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Hi all,

Siwpas, Simple Web Profile Application Server today announced the version
2.0.0 with support of Web Socket Protocol.

Siwpas main page : http://siwpas.mechsoft.com.tr

Regards;

-- 
Gurkan Erdogdu
http://gurkanerdogdu.blogspot.com

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Hi all,<br><br>Siwpas, Simple Web Profile Application Server today announced the version 2.0.0 with support of Web Socket Protocol.<br><br>Siwpas main page : <a href="http://siwpas.mechsoft.com.tr">http://siwpas.mechsoft.com.tr</a><br>
<br>Regards;<br clear="all"><br>-- <br>Gurkan Erdogdu<br><a href="http://gurkanerdogdu.blogspot.com">http://gurkanerdogdu.blogspot.com</a><br>

--00248c0d77f88f173704a937749b--

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From: Takeshi Yoshino <tyoshino@google.com>
Date: Sat, 30 Jul 2011 01:49:53 +0900
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Subject: Re: [hybi] Payload only compression extension, again
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I've conducted quick survey on zlib's memory consumption for decompress.

I injected my own alloc_func and free_func to zalloc and zfree of z_stream
to count max memory allocation on heap.

Input data is 1000000 of 10-byte long random sequences generated by rand of
stdlib. I called deflate for every 10-byte using Z_SYNC_FLUSH.

  res = deflateInit2(
      &compress_stream,
      Z_DEFAULT_COMPRESSION,
      Z_DEFLATED,
      -window_bits,
      mem_level,
      Z_DEFAULT_STRATEGY);

  res = inflateInit2(&decompress_stream, -window_bits);

deflateInit2() allocates memory as shown in this thread. deflate() never
allocates memory.
https://spdy-dev.googlegroups.com/attach/4051c1467338d389/zlib_20100202.html?view=1&part=4

window_bits=8, mem_level=1
deflateInit2: 9000, inflateInit2: 9560, inflate: 256

window_bits=8, mem_level=9
deflateInit2: 270120, inflateInit2: 9560, inflate: 256

window_bits=15, mem_level=1
deflateInit2: 138024, inflateInit2: 9560, inflate: 32768

window_bits=15, mem_level=9
deflateInit2: 399144, inflateInit2: 9560, inflate: 32768

inflateInit2()'s 9560 bytes are for inflate_state. Mainly spent for decoding
table. inflate()'s allocation is for LZ77 sliding window.

Heap necessary on compressor side ranges ~10k - ~40k.

According to the spdy survey, John suggests window bit of 10 (or 11) and
memory level of 1 shows good cost performance. For this configuration,
- allocation for compressor: 11048
- allocation for decompressor: 10584

Sounds reasonable to recommend.

mem_level seems not affecting compressor side's memory allocation (it does
affect the size of compressed data).

Thanks,
Takeshi

--000e0cd47baa4092cd04a93813a1
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<div class=3D"gmail_quote">I&#39;ve conducted quick survey on zlib&#39;s me=
mory consumption for decompress.</div><div class=3D"gmail_quote"><br></div>=
<div class=3D"gmail_quote">I injected my own alloc_func and free_func to za=
lloc and zfree of z_stream to count max memory allocation on heap.</div>


<div class=3D"gmail_quote"><br></div><div class=3D"gmail_quote">Input data =
is 1000000 of 10-byte long random sequences generated by rand of stdlib. I =
called deflate for every 10-byte using Z_SYNC_FLUSH.</div><div class=3D"gma=
il_quote">


<br></div><div class=3D"gmail_quote"><div class=3D"gmail_quote">=A0 res =3D=
 deflateInit2(</div><div class=3D"gmail_quote">=A0 =A0 =A0 &amp;compress_st=
ream,</div><div class=3D"gmail_quote">=A0 =A0 =A0 Z_DEFAULT_COMPRESSION,</d=
iv><div class=3D"gmail_quote">


=A0 =A0 =A0 Z_DEFLATED,</div><div class=3D"gmail_quote">=A0 =A0 =A0 -window=
_bits,</div><div class=3D"gmail_quote">=A0 =A0 =A0 mem_level,</div><div cla=
ss=3D"gmail_quote">=A0 =A0 =A0 Z_DEFAULT_STRATEGY);</div><div><br></div><di=
v><div>=A0 res =3D inflateInit2(&amp;decompress_stream, -window_bits);</div=
>


</div><div><br></div><div>deflateInit2() allocates memory as shown in this =
thread. deflate() never allocates memory.</div></div><div class=3D"gmail_qu=
ote"><a href=3D"https://spdy-dev.googlegroups.com/attach/4051c1467338d389/z=
lib_20100202.html?view=3D1&amp;part=3D4" target=3D"_blank">https://spdy-dev=
.googlegroups.com/attach/4051c1467338d389/zlib_20100202.html?view=3D1&amp;p=
art=3D4</a></div>


<div class=3D"gmail_quote"><br></div><div class=3D"gmail_quote">window_bits=
=3D8,=A0mem_level=3D1</div><div class=3D"gmail_quote">deflateInit2: 9000, i=
nflateInit2:=A09560, inflate: 256</div><div><br></div><div></div><div class=
=3D"gmail_quote">


window_bits=3D8,=A0mem_level=3D9</div><div class=3D"gmail_quote">deflateIni=
t2:=A0270120, inflateInit2:=A09560, inflate: 256</div><div class=3D"gmail_q=
uote"><div class=3D"gmail_quote"><br></div><div class=3D"gmail_quote">windo=
w_bits=3D15,=A0mem_level=3D1</div>


<div class=3D"gmail_quote">deflateInit2: 138024, inflateInit2:=A09560, infl=
ate: 32768</div><div class=3D"gmail_quote"></div><div class=3D"gmail_quote"=
><br></div><div class=3D"gmail_quote">window_bits=3D15,=A0mem_level=3D9</di=
v><div><div class=3D"gmail_quote">


deflateInit2: 399144, inflateInit2:=A09560, inflate: 32768</div><div class=
=3D"gmail_quote"></div></div></div><div class=3D"gmail_quote"><div class=3D=
"gmail_quote"><br></div><div class=3D"gmail_quote">inflateInit2()&#39;s 956=
0 bytes are for inflate_state. Mainly spent for decoding table. inflate()&#=
39;s allocation is for LZ77 sliding window.</div>

<div class=3D"gmail_quote"><br></div><div class=3D"gmail_quote"><div class=
=3D"gmail_quote">Heap necessary on compressor side ranges ~10k - ~40k.</div=
><div><br></div><div>According to the spdy survey, John suggests window bit=
 of 10 (or 11) and memory level of 1 shows good cost performance. For this =
configuration,</div>

<div><div>- allocation for compressor: 11048</div><div>- allocation for dec=
ompressor: 10584</div></div><div><br></div><div>Sounds reasonable to recomm=
end.</div><div><br></div></div><div class=3D"gmail_quote">mem_level seems n=
ot affecting compressor side&#39;s memory allocation (it does affect the si=
ze of compressed data).</div>


<div class=3D"gmail_quote"><br></div><div class=3D"gmail_quote">Thanks,</di=
v><div class=3D"gmail_quote">Takeshi</div></div>

--000e0cd47baa4092cd04a93813a1--

From Gabriel.Montenegro@microsoft.com  Fri Jul 29 20:33:00 2011
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From: Gabriel Montenegro <Gabriel.Montenegro@microsoft.com>
To: Server-Initiated HTTP <hybi@ietf.org>
Thread-Topic: consensus call on ability to announce max frame size
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Date: Sat, 30 Jul 2011 03:32:58 +0000
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At the meeting and on the mailing list leading up to it, the chairs saw con=
sensus for the following action:

                Add ability to announce max frame size by  both client and =
server

We're confirming on the mailing list, and will declare this consensus call =
final by next Friday Aug 5.

Thanks,

Gabriel and Salvatore
HyBi co-chairs




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 it, the chairs saw consensus for the following action:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Add ability to announce max frame si=
ze by&nbsp; both client and server<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">We&#8217;re confirming on the mailing list, and will=
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<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Gabriel and Salvatore<o:p></o:p></p>
<p class=3D"MsoNormal">HyBi co-chairs<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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--_000_CA566BAEAD6B3F4E8B5C5C4F61710C11403B6643TK5EX14MBXW604w_--

From Gabriel.Montenegro@microsoft.com  Fri Jul 29 20:40:28 2011
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From: Gabriel Montenegro <Gabriel.Montenegro@microsoft.com>
To: Server-Initiated HTTP <hybi@ietf.org>
Thread-Topic: consensus call on removing deflate-stream 
Thread-Index: AcxOaKuv70ih/eLjTUK9sH+c6yH32g==
Date: Sat, 30 Jul 2011 03:40:20 +0000
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At the meeting and on the mailing list leading up to it, the chairs saw con=
sensus for the following action:

                Remove deflate-stream from the websocket protocol specifica=
tion.

We're confirming on the mailing list, and will declare this consensus call =
final by next Friday Aug 5.

Thanks,

Gabriel and Salvatore
HyBi co-chairs


--_000_CA566BAEAD6B3F4E8B5C5C4F61710C11403B6659TK5EX14MBXW604w_
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<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Remove deflate-stream from the webso=
cket protocol specification.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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<p class=3D"MsoNormal">Gabriel and Salvatore<o:p></o:p></p>
<p class=3D"MsoNormal">HyBi co-chairs<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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--_000_CA566BAEAD6B3F4E8B5C5C4F61710C11403B6659TK5EX14MBXW604w_--

From Gabriel.Montenegro@microsoft.com  Fri Jul 29 20:40:34 2011
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From: Gabriel Montenegro <Gabriel.Montenegro@microsoft.com>
To: Server-Initiated HTTP <hybi@ietf.org>
Thread-Topic: consensus call on not specifying DNS SRV as mandatory to implement
Thread-Index: AcxOaQ6e5IrwdEjuTs6E1aKsqnlfUA==
Date: Sat, 30 Jul 2011 03:40:22 +0000
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--_000_CA566BAEAD6B3F4E8B5C5C4F61710C11403B665FTK5EX14MBXW604w_
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At the meeting and on the mailing list leading up to it, the chairs did not=
 see consensus for the following action:

                Specify DNS SRV as mandatory to implement in the websocket =
protocol specification.

We're confirming on the mailing list, and will declare this consensus call =
final by next Friday Aug 5.

Thanks,

Gabriel and Salvatore
HyBi co-chairs



--_000_CA566BAEAD6B3F4E8B5C5C4F61710C11403B665FTK5EX14MBXW604w_
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<body lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal">At the meeting and on the mailing list leading up to=
 it, the chairs did not see consensus for the following action:<o:p></o:p><=
/p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Specify DNS SRV as mandatory to impl=
ement in the websocket protocol specification.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">We&#8217;re confirming on the mailing list, and will=
 declare this consensus call final by next Friday Aug 5.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Gabriel and Salvatore<o:p></o:p></p>
<p class=3D"MsoNormal">HyBi co-chairs<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

--_000_CA566BAEAD6B3F4E8B5C5C4F61710C11403B665FTK5EX14MBXW604w_--

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From: John Tamplin <jat@google.com>
Date: Fri, 29 Jul 2011 23:40:50 -0400
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To: Gabriel Montenegro <Gabriel.Montenegro@microsoft.com>
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Subject: Re: [hybi] consensus call on ability to announce max frame size
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On Fri, Jul 29, 2011 at 11:32 PM, Gabriel Montenegro <
Gabriel.Montenegro@microsoft.com> wrote:

>  At the meeting and on the mailing list leading up to it, the chairs saw
> consensus for the following action:****
>
> ** **
>
>                 Add ability to announce max frame size by  both client an=
d
> server****
>
> ** **
>
> We=E2=80=99re confirming on the mailing list, and will declare this conse=
nsus call
> final by next Friday Aug 5.
>
> **
>

SGTM.  Will we have a concrete proposal, or does that come after declaring
consensus?

--=20
John A. Tamplin
Software Engineer (GWT), Google

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<div class=3D"gmail_quote">On Fri, Jul 29, 2011 at 11:32 PM, Gabriel Monten=
egro <span dir=3D"ltr">&lt;<a href=3D"mailto:Gabriel.Montenegro@microsoft.c=
om">Gabriel.Montenegro@microsoft.com</a>&gt;</span> wrote:<br><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex;">







<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal">At the meeting and on the mailing list leading up to=
 it, the chairs saw consensus for the following action:<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 Add ability to announce max frame s=
ize by=C2=A0 both client and server<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">We=E2=80=99re confirming on the mailing list, and wi=
ll declare this consensus call final by next Friday Aug 5.</p><p class=3D"M=
soNormal"><u></u></p></div></div></blockquote></div><div><br></div>SGTM. =
=C2=A0Will we have a concrete proposal, or does that come after declaring c=
onsensus?<br clear=3D"all">

<br>-- <br>John A. Tamplin<br>Software Engineer (GWT), Google<br>

--000e0cd32b943ba50304a9412b4a--

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From: Gabriel Montenegro <Gabriel.Montenegro@microsoft.com>
To: John Tamplin <jat@google.com>
Thread-Topic: [hybi] consensus call on ability to announce max frame size
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From duerst@it.aoyama.ac.jp  Fri Jul 29 21:23:40 2011
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Date: Sat, 30 Jul 2011 13:23:31 +0900
From: =?ISO-8859-1?Q?=22Martin_J=2E_D=FCrst=22?= <duerst@it.aoyama.ac.jp>
Organization: Aoyama Gakuin University
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Cc: Server-Initiated HTTP <hybi@ietf.org>
Subject: Re: [hybi] consensus call on removing deflate-stream
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I concur.     Martin.

On 2011/07/30 12:40, Gabriel Montenegro wrote:
> At the meeting and on the mailing list leading up to it, the chairs saw consensus for the following action:
>
>                  Remove deflate-stream from the websocket protocol specification.
>
> We're confirming on the mailing list, and will declare this consensus call final by next Friday Aug 5.
>
> Thanks,
>
> Gabriel and Salvatore
> HyBi co-chairs
>
>
>
>
>
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi

From gregw@intalio.com  Sat Jul 30 00:09:57 2011
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Date: Sat, 30 Jul 2011 17:09:54 +1000
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From: Greg Wilkins <gregw@intalio.com>
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Cc: Server-Initiated HTTP <hybi@ietf.org>, Gabriel Montenegro <Gabriel.Montenegro@microsoft.com>
Subject: Re: [hybi] consensus call on removing deflate-stream
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+1

On 30 July 2011 14:23, "Martin J. D=FCrst" <duerst@it.aoyama.ac.jp> wrote:
> I concur. =A0 =A0 Martin.
>
> On 2011/07/30 12:40, Gabriel Montenegro wrote:
>>
>> At the meeting and on the mailing list leading up to it, the chairs saw
>> consensus for the following action:
>>
>> =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Remove deflate-stream from the websocket=
 protocol
>> specification.
>>
>> We're confirming on the mailing list, and will declare this consensus ca=
ll
>> final by next Friday Aug 5.
>>
>> Thanks,
>>
>> Gabriel and Salvatore
>> HyBi co-chairs
>>
>>
>>
>>
>>
>> _______________________________________________
>> hybi mailing list
>> hybi@ietf.org
>> https://www.ietf.org/mailman/listinfo/hybi
>
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>

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Date: Sat, 30 Jul 2011 17:12:15 +1000
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Subject: Re: [hybi] consensus call on ability to announce max frame size
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+1 on the ability to declare a maximum acceptable frame sizes (this
can be implemented transparently).
-1 on the ability to declare a maximum acceptable message size (as
you'd have to tell the application what that is).



On 30 July 2011 14:07, Gabriel Montenegro
<Gabriel.Montenegro@microsoft.com> wrote:
> Concrete proposal is welcome. Of course, if something happens and consens=
us
> is not ultimately declared, it will not be taken up.
>
>
>
> But if you have something, please do share it.
>
>
>
> Thanks,
>
>
>
> Gabriel
>
>
>
> From: John Tamplin [mailto:jat@google.com]
> Sent: Friday, July 29, 2011 23:41
> To: Gabriel Montenegro
> Cc: Server-Initiated HTTP
> Subject: Re: [hybi] consensus call on ability to announce max frame size
>
>
>
> On Fri, Jul 29, 2011 at 11:32 PM, Gabriel Montenegro
> <Gabriel.Montenegro@microsoft.com> wrote:
>
> At the meeting and on the mailing list leading up to it, the chairs saw
> consensus for the following action:
>
>
>
> =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Add ability to announce max=
 frame size by=A0 both client and
> server
>
>
>
> We=92re confirming on the mailing list, and will declare this consensus c=
all
> final by next Friday Aug 5.
>
>
>
> SGTM. =A0Will we have a concrete proposal, or does that come after declar=
ing
> consensus?
>
> --
> John A. Tamplin
> Software Engineer (GWT), Google
>
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>
>

From gregw@intalio.com  Sat Jul 30 00:13:38 2011
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Subject: Re: [hybi] consensus call on not specifying DNS SRV as mandatory to implement
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-1.  DNS SRV should not be mandatory.    I think it should be
considered again if we ever have a non HTTP handshake.


On 30 July 2011 13:40, Gabriel Montenegro
<Gabriel.Montenegro@microsoft.com> wrote:
> At the meeting and on the mailing list leading up to it, the chairs did n=
ot
> see consensus for the following action:
>
>
>
> =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Specify DNS SRV as mandator=
y to implement in the websocket
> protocol specification.
>
>
>
> We=92re confirming on the mailing list, and will declare this consensus c=
all
> final by next Friday Aug 5.
>
>
>
> Thanks,
>
>
>
> Gabriel and Salvatore
>
> HyBi co-chairs
>
>
>
>
>
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>
>

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Subject: Re: [hybi] consensus call on not specifying DNS SRV as mandatory to implement
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On Sat, Jul 30, 2011 at 3:13 AM, Greg Wilkins <gregw@intalio.com> wrote:

> -1.  DNS SRV should not be mandatory.    I think it should be
> considered again if we ever have a non HTTP handshake.
>

Note "did not see consensus".

-- 
John A. Tamplin
Software Engineer (GWT), Google

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<div class=3D"gmail_quote">On Sat, Jul 30, 2011 at 3:13 AM, Greg Wilkins <s=
pan dir=3D"ltr">&lt;<a href=3D"mailto:gregw@intalio.com">gregw@intalio.com<=
/a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

-1. =C2=A0DNS SRV should not be mandatory. =C2=A0 =C2=A0I think it should b=
e<br>
considered again if we ever have a non HTTP handshake.<br></blockquote><div=
><br></div><div>Note &quot;did not see consensus&quot;.=C2=A0</div></div><b=
r>-- <br>John A. Tamplin<br>Software Engineer (GWT), Google<br>

--0016364ee904d385fe04a9442f0d--

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Subject: Re: [hybi] consensus call on ability to announce max frame size
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On Sat, Jul 30, 2011 at 05:12:15PM +1000, Greg Wilkins wrote:
> +1 on the ability to declare a maximum acceptable frame sizes (this
> can be implemented transparently).
> -1 on the ability to declare a maximum acceptable message size (as
> you'd have to tell the application what that is).

I'd also argue that HTTP has been living for a decade with support for
chunked encoding which does not set a limit on chunk size nor on message
size and has not had any trouble with that. I'm still in favor of being
able to declare a max frame size to save fairness for later mux support,
but limited message size does not seem useful to me either.

Willy


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From: John Tamplin <jat@google.com>
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Subject: Re: [hybi] consensus call on ability to announce max frame size
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On Sat, Jul 30, 2011 at 3:31 AM, Willy Tarreau <w@1wt.eu> wrote:

> I'd also argue that HTTP has been living for a decade with support for
> chunked encoding which does not set a limit on chunk size nor on message
> size and has not had any trouble with that. I'm still in favor of being
> able to declare a max frame size to save fairness for later mux support,
> but limited message size does not seem useful to me either.
>

I see no mention of maximum message size in this consensus call, only
maximum frame size.

-- 
John A. Tamplin
Software Engineer (GWT), Google

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<div class="gmail_quote">On Sat, Jul 30, 2011 at 3:31 AM, Willy Tarreau <span dir="ltr">&lt;<a href="mailto:w@1wt.eu">w@1wt.eu</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

<div class="im">I&#39;d also argue that HTTP has been living for a decade with support for</div>
chunked encoding which does not set a limit on chunk size nor on message<br>
size and has not had any trouble with that. I&#39;m still in favor of being<br>
able to declare a max frame size to save fairness for later mux support,<br>
but limited message size does not seem useful to me either.<font class="Apple-style-span" color="#888888"><br></font></blockquote></div><div><br></div>I see no mention of maximum message size in this consensus call, only maximum frame size.<br clear="all">

<br>-- <br>John A. Tamplin<br>Software Engineer (GWT), Google<br>

--0016364274c0411b1304a94498e2--

From w@1wt.eu  Sat Jul 30 01:37:21 2011
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From: Willy Tarreau <w@1wt.eu>
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Cc: Server-Initiated HTTP <hybi@ietf.org>, Greg Wilkins <gregw@intalio.com>, Gabriel Montenegro <Gabriel.Montenegro@microsoft.com>
Subject: Re: [hybi] consensus call on ability to announce max frame size
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On Sat, Jul 30, 2011 at 03:46:04AM -0400, John Tamplin wrote:
> On Sat, Jul 30, 2011 at 3:31 AM, Willy Tarreau <w@1wt.eu> wrote:
> 
> > I'd also argue that HTTP has been living for a decade with support for
> > chunked encoding which does not set a limit on chunk size nor on message
> > size and has not had any trouble with that. I'm still in favor of being
> > able to declare a max frame size to save fairness for later mux support,
> > but limited message size does not seem useful to me either.
> >
> 
> I see no mention of maximum message size in this consensus call, only
> maximum frame size.

Agreed, but as Greg mentionned it, I expressed my opinion on this
point too. I know too well how dicussion subjects diverge and how
some solutions can be replaced with other ones in this WG ;-)

Cheers,
Willy


From Martin.Thomson@commscope.com  Sat Jul 30 06:34:11 2011
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On 2011-07-29 at 23:32:58, Gabriel Montenegro wrote:
>                 Add ability to announce max frame size by  both client=20
> and server

Given that there is no constraint on message size, this capability is point=
less.  Any peer that receives this indication does not need to change their=
 behaviour in any meaningful way.  Messages of any size can be sent regardl=
ess of what frame size is supported.

That said, such a negotiation might be crucial to an effective multiplexing=
 scheme.  I suggest that this be deferred and included as part of the multi=
plexing work.

--Martin

From tobias.oberstein@tavendo.de  Sat Jul 30 17:04:21 2011
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> Here is a first cut at a list of tests for section 4 of the spec.  Are
> these the kinds of tests people were thinking about?
> If so, then I'm happy to go the next step of detail and start
> providing more detail for each test.

I am interested in this, and I think a list of tightly specified,
executable test cases which read on and exhaust the spec is definitely
desirable.

Started a little project just days ago which includes a fuzzing server:
https://github.com/oberstet/Autobahn
Using this, I found some behavior in FF>=3D7 (which is on the new protocol =
8)
in particular with respect to fragmentation and TCP cleanness (see below)
which I believe are issues.

>=20
> Obviously some implementation are not going to be able to send some of
> the invalid frames or force fragmentation, so some suites will have to
> record some tests as skipped, or only testing in one direction etc.

Besides fragmentation, I would like to suggest "TCP clean" in addition
and in combination. By that I mean: depending on implementation of an
endpoint, there might be issues related in particular to how the octet
stream is chopped up on the wire / received by the implementation.
Independent of or only in combination with WebSockets message fragmenta-
tion. For an example, pls see
https://bugzilla.mozilla.org/show_bug.cgi?id=3D664344

Obviously, an implementation must be agnostic to the chops in which it
receives octets from an octet stream (TCP).

Currently, I am testing: a) force octets to wire after each frame and
b) force out each octet onto wire separately. I might add: c) force out
"random" numbers of octets. b) is also a good test to catch "naive"
implementations doing inefficient concatenations involving reallocations.

In general, I believe it might be worth looking not only into tests
catching blunt protocol violations, but memory/performance related issues
as well. I think of at least having test drives which torture the
implementation with regard to memory leaks.

> 4.2 Base Framing Tests
> ----------------------
>=20
> * 7 bit length: send/receive frames of length 0-125

Suggestion: Control frames must have payload max. 125, but can have 0.

I wanted to ask this anyway: is it valid for data frames to have 0
payload? If yes, is a data message of payload 0 allowed?

I am asking, because current FF will happily consume messages
of payload 0 (not only individual fragments), and then generate
a JS onEvent() with message =3D empty string. I did not find anything
in the spec regarding this .. but I am not sure if its helpful for
a JS onEvent() to be fired on an empty string.

> * 16 bit length: send/receive frames of length 126-128, 65536
> * 63 bit length: send/receive frames of length 65537

Good idea to test for the integers around the cross-over points.
We could add testing for (MSB=3D=3D1 with 8 octets len) =3D> fail, to
be sure host languages which don't have unsigned longs don't
naively produce negative length when converting octets to their
signed long.

On the other hand: what do you do in an implementation upon
receiving the first frame of a fragmented message, and the frame
says it has length 2GB? I would just bail out with "frame too large"
anyway. I would find it unreasonable to allocated that amount.

Maybe we should test if, and when at which frame size the
implementation bails out with "frame too large".

>=20
> * Non zero RSV: send frames with non zero reserved bits and no
> extension. Verify connection is failed.

Yep. The tests should verify each RSV bit combination .. 0-7.

>=20
> * Known Opcodes: send/receive frames with text,binary,ping,pong
> opcodes (continuation & close tested separately).
> * Unknown Opcodes: send frames with unknown opcodes 3-7, B-F. Verify
> connection is failed.

Yep. And yes: the tests should verify each of the reserved opcodes.
For example, current FF correctly fails on reserved control frame
opcodes, but silently ignores reserved data frame opcodes.

> 4.3 Client to Server Masking
> ----------------------------
>=20
> ???

Couple of ideas:
Obviously, all c2s frames must be masked.
Mask must be !=3D 0.
Mask of frame N must be !=3D mask for frame N-1 (,N-2, N-k).
Those 2 are just to catch lazy implementations, of course no
rigorous check of "the mask must use good entropy source".

What about testing server-to-client masking? The spec says both endpoints
must be capable of processing masked frames ..

> 4.4 Fragmentation
> -----------------
>=20
> * 1 fragment message: send/receive message in 1 fragment
> * 2 fragment message: send/receive message in 2 fragments
> * 3 fragment message: send/receive message in 3 fragments
> * Injected control: send/receive 2 fragment message with unsolicited
> pong message between frames
> * Fragmented control: send fragmented ping. Verify that connection is fai=
led.
> * Interleaved fragments: send a fragment with FIN set and Continuation
> opcode. Verify that connection is failed.

Not sure if I get you right here: I see 2 classes of violations:

a) Within fragmented message, a data frame with opcode =3D=3D 1/2 and FIN =
=3D=3D 1/0
=3D> continuation not closed (with opcode =3D=3D 0 and FIN =3D=3D 1)

b) Outside fragmented message, a data frame  with opcode =3D=3D 0 and FIN =
=3D=3D 1/0
=3D> nothing to continue (no previous opcode =3D=3D 1/2 and FIN =3D=3D 0)

6 cases in total. The valid OpCode:FIN sequences

unfragmented: 1/2:1
fragmented: 1/2:0, (0:0)*, 0:1

My implicit expectation was that an endpoint must fail immediately on any o=
f
above 6 bogus cases, but I just can't find that explicit in the spec .. wha=
t
is intended?

> * Handled control: send/receive 2 fragment message with ping message
> between frames and last frame delayed. Verify pong is received before
> last frame is sent.
> * 7 bit fragment: send/receive message with initial/middle/final
> fragments of sizes 0-125
> * 16 bit fragment: send/receive message with initial/middle/final
> fragments of sizes 126-128,65536
> * 64 bit fragment: send/receive message with
> initial/middle/finalfragments of size 65537


> 4.5.1 Close Frames
> ------------------
> * Orderly Close: On an idle connection, send a close and receive a close.
> * Busy Close: send a close to an endpoint that has sent the first
> fragment of a message.
> * Simultaneous Close: send a close on both ends of a connection.
> * Message after close: attempt to send a message after a close. Verify
> message is not received.
> * Idle close: ??
> * Error Close: ??

A minor addition: the spec says, a close frame with payload > 0 must have
the first 2 octets being an status code, and may have additional content,
which then is UTF-8. So payload length =3D 1 is illegal, and if payload > 2=
,
then the rest must be UTF-8.

> 4.5.2 Ping
> ----------
> * Empty ping: Send an empty ping, verify pong is received.
> * Non empty ping: Send a non empty ping, verify pong is received.

Also something I wanted to ask anyway: the spec says

   Upon receipt of a Ping frame, an endpoint MUST send a Pong frame in
   response.  It SHOULD do so as soon as is practical.
...
   If an endpoint receives a Ping frame and has not yet sent Pong
   frame(s) in response to previous Ping frame(s), the endpoint MAY
   elect to send a Pong frame for only the most recently processed Ping
   frame.

Lets say: client has 1 large data message to send, is sending fragments
all the time. Receives Ping, but decides that it is "not practical" to
send Pong until _all_ fragments of big data message have been sent.
Is that valid?
What is the binding meaning of the words "SHOULD" and "practical"?
If there is none, how do you "verify" negatively? I mean: if there is
a Pong, fine. Conforming (at least under that load conditions). But
no Pong does not imply non-conforming, isn't it?

> 4.5.3 Pong
> ----------
> * Empty pong: Send an empty ping, verify pong is received and is empty.
> * Non empty pong: Send a non empty ping, verify pong is received and
> has matching content.
> * Outstanding ping: Send two pings with different content, reply with
> content for only the second. Verify connection stays open.
> * Unsolicited pong: Send/recv unsolicited pong.

Another point: if I've not overlooked something, the spec does not
constraint the payload (besides length). Maybe test for binary cleanness,
at least non-UTF-8 payload.

> 4.5.4 Data Frames
> -----------------
> * UTF-8 Text 0000-007f: Send/receive text message with characters in
> code point range  U+0000 to U+007F
> * UTF-8 Text 0080-07ff: Send/receive text message with characters in
> code point range  U+0080 to U+07FF
> * UTF-8 Text 0800-ffff: Send/receive text message with characters in
> code point range  U+0800 to U+FFFF
> * UTF-8 Text 010000-10ffff: Send/receive text message with characters
> in code point range  U+010000 to U+10FFFF
> * Illegal bytes UTF-8: Send message with invalid UTF-8 byte sequence.
> Verify message is not received.
> * Illegal codes UTF-8: Send message with invalid UTF-8 code points.
> Verify message is not received.
> * Binary frames: Send/receive message containing illegal UTF-8 bytes
> sequences and code points as binary message.

Maybe add tests:
- fragmented text message, fragmented within octet sequence for single code
point
- fail fast: fragmented text message with illegal UTF-8 fails already
upon receiving the frame containing the last octet of illegal sequence,
not only after receiving all message frames.




From gregw@intalio.com  Sat Jul 30 17:39:29 2011
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Subject: [hybi] max frame size. was: consensus call on ability to announce max frame size
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Changing topic to avoid debate in consensus call thread.




On 30 July 2011 17:46, John Tamplin <jat@google.com> wrote:
> I see no mention of maximum message size in this consensus call, only
> maximum frame size.

Sorry if that muddied the water. But I just wanted to be crystal clear
we are talking about frames not messages.


On 30 July 2011 17:31, Willy Tarreau <w@1wt.eu> wrote:
> I'd also argue that HTTP has been living for a decade with support for
> chunked encoding which does not set a limit on chunk size nor on message
> size and has not had any trouble with that.

But HTTP also does not have a chunk-too-large error code.
Implementations are forced to deal with large chunks.  This is often
no too difficult because stream APIs are the norm for HTTP.   Large
sizes are harder to handle in the datagram API of websocket.


On 30 July 2011 23:34, Thomson, Martin <Martin.Thomson@commscope.com> wrote=
:
> Given that there is no constraint on message size, this capability is poi=
ntless. =A0Any peer that receives this indication does not need to change t=
heir behaviour in any meaningful way. =A0Messages of any size can be sent r=
egardless of what frame size is supported.

Note sure if you are talking about max message or max frame size?

But a peer that does not respect a max frame indication and does not
change it's behaviour will then just receive frame-too-large errors in
response.   If we allow frame-too-large errors, then surely we need a
way to communicate what is not too large.

Also, while I think a max frame size should be optional to declare, I
think that once it is declared it should be mandatory for senders to
respect that limit.



> That said, such a negotiation might be crucial to an effective multiplexi=
ng scheme. =A0I suggest that this be deferred and included as part of the m=
ultiplexing work.

I think it is more than just MUX that needs this.  Because of the
datagram style API of websockets, applications will not be stream
based (blocked reading content), but will be called back when frames
and/or messages are received.  Thus there will be implementations that
have fixed buffer sizes and will not work with large frames and/or
messages.      For large messages, there is not much the
implementations can do to influence the applications, so I don't see a
limit being worthwhile.  But for a max frame size, it is pretty simple
for implementations to fragment messages to respect a limit.


regards

From andy@warmcat.com  Sat Jul 30 22:58:37 2011
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Subject: Re: [hybi] consensus call on ability to announce max frame size
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On 07/30/2011 04:32 AM, Somebody in the thread at some point said:

> At the meeting and on the mailing list leading up to it, the chairs saw
> consensus for the following action:
>
> Add ability to announce max frame size by both client and server

I think this is bogus, but it's needn't create problems.  Better to have 
removed the really really bogus 63-bit frame length and replace with 
something smaller that doesn't need to be "patched" with an "MTU".

Anyway the only thing is it would be very good to write it up so that 
implementors don't get the idea they are being told to buffer atomic 
frames.  Otherwise this kind of talk would naturally put you in mind of 
that kind of code with its attendant limitations.

-Andy

From paul@colomiets.name  Sun Jul 31 05:25:35 2011
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On Sat, Jul 30, 2011 at 6:32 AM, Gabriel Montenegro
<Gabriel.Montenegro@microsoft.com> wrote:
>
> =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Add ability to announce max=
 frame size by=A0 both client and server
>

+1 on announcing both max frame size and max message size. Unlike some
folks here
think, there *is* limit on message size in HTTP. That's reason why
Expect header exists.

--=20
Paul

From julian.reschke@gmx.de  Sun Jul 31 06:07:40 2011
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Cc: Server-Initiated HTTP <hybi@ietf.org>, Gabriel Montenegro <Gabriel.Montenegro@microsoft.com>
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On 2011-07-31 14:25, Paul Colomiets wrote:
> +1 on announcing both max frame size and max message size. Unlike some
> folks here
> think, there *is* limit on message size in HTTP. That's reason why
> Expect header exists.

Hmm, no.

*Implementations* can have limits, and Expect can help dealing with 
that. But it's not a protocol limitation.

Best regards, Julian


From tobias.oberstein@tavendo.de  Sun Jul 31 06:13:25 2011
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To: "hybi@ietf.org" <hybi@ietf.org>
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I'm implementing WebSockets and have some questions where I can't figure ou=
t the answer from the draft spec.
That could be my fault, but hints would be welcome anyway ..

1.
Are data frames of payload length 0 allowed?
If so, are data messages of length 0 (where all fragments have length 0) al=
lowed?
If so, what is a browser supposed to do? Fire an JS onMessage() event with =
0 length string?

2.
In the context of a persistent http connection, what is allowed _before_ th=
e WebSocket handshake
(that is before client sends GET with upgrade header)?
For example, are "normal" GETs allowed? In other words: is it permissable f=
or a client to do "normal"
http on a persistent connection, and then  - at any point - decide to reque=
st to upgrade to WebSocket?

3.
When the client has sent an upgrade request, the spec says on server respon=
se

   The handshake from the server is much simpler than the client
   handshake.  The first line is an HTTP Status-Line, with the status
   code 101:

        HTTP/1.1 101 Switching Protocols

   Any status code other than 101 indicates that the WebSocket handshake
   has not completed, and that the semantics of HTTP still apply.

Does that mean, the server may send not only http error codes, but also for=
 example http redirects?

4.
The spec defines 2 new URI schemes for WebSockets: ws and wss.

On the other hand,=20
http://tools.ietf.org/html/draft-ietf-websec-origin-02

"If the two origins are scheme/host/port triples, the two origins
are the same if, and only if, they have identical schemes, hosts,
and ports."

Does that mean that with browsers, a JS script coming from (http, somehost,=
 80) is not allowed
to contact (ws, somehost, 80), because http !=3D ws?

I guess that is expected to succeed, but then, what is the purpose/semantic=
s of the new schemes?






From paul@colomiets.name  Sun Jul 31 06:16:43 2011
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On Sun, Jul 31, 2011 at 4:07 PM, Julian Reschke <julian.reschke@gmx.de> wrote:
> On 2011-07-31 14:25, Paul Colomiets wrote:
>>
>> +1 on announcing both max frame size and max message size. Unlike some
>> folks here
>> think, there *is* limit on message size in HTTP. That's reason why
>> Expect header exists.
>
> Hmm, no.
>
> *Implementations* can have limits, and Expect can help dealing with that.
> But it's not a protocol limitation.

Sure, that's what we talk about here. Expect header is a way to announce
limitation of particular server setup in HTTP.

-- 
Paul

From julian.reschke@gmx.de  Sun Jul 31 06:21:18 2011
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Cc: Server-Initiated HTTP <hybi@ietf.org>, Gabriel Montenegro <Gabriel.Montenegro@microsoft.com>
Subject: Re: [hybi] consensus call on ability to announce max frame size
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On 2011-07-31 15:16, Paul Colomiets wrote:
> On Sun, Jul 31, 2011 at 4:07 PM, Julian Reschke<julian.reschke@gmx.de>  wrote:
>> On 2011-07-31 14:25, Paul Colomiets wrote:
>>>
>>> +1 on announcing both max frame size and max message size. Unlike some
>>> folks here
>>> think, there *is* limit on message size in HTTP. That's reason why
>>> Expect header exists.
>>
>> Hmm, no.
>>
>> *Implementations* can have limits, and Expect can help dealing with that.
>> But it's not a protocol limitation.
>
> Sure, that's what we talk about here. Expect header is a way to announce
> limitation of particular server setup in HTTP.

Hmm, how exactly does Expect help here?

You can do Expect: 100-continue in the client, and then the server has a 
chance to reject the request (such as a huge PUT), but that's it.

Best regards, Julian


From paul@colomiets.name  Sun Jul 31 08:37:48 2011
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From: Paul Colomiets <paul@colomiets.name>
Date: Sun, 31 Jul 2011 18:37:30 +0300
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Subject: Re: [hybi] consensus call on ability to announce max frame size
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On Sun, Jul 31, 2011 at 4:21 PM, Julian Reschke <julian.reschke@gmx.de> wrote:
>> Sure, that's what we talk about here. Expect header is a way to announce
>> limitation of particular server setup in HTTP.
>
> Hmm, how exactly does Expect help here?
>
> You can do Expect: 100-continue in the client, and then the server has a
> chance to reject the request (such as a huge PUT), but that's it.
>

I don't see you point Julian. That's exactly what max message size
announcement can do. It says that messages larger than specified will be
rejected. Then how API reacts on that is another story.

Of course maximum frame size is something different. It asks sender
to split longer messages to chunks of specified size. Thinking more
about this, I think I'm opposed to the idea of specifing frame size. The
problem with frame size limit, is that it gives lazy developers a false
impression that they can set buffer size to the maximum frame size.
In majority cases it's simply not true. And it's not very hard to split
message into frames in the intermediary either.

+1 for announcing message size
-0 for announcing frame size

-- 
Paul

From pmcmanus@mozilla.com  Sun Jul 31 09:02:48 2011
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On Sun, 2011-07-31 at 06:13 -0700, Tobias Oberstein wrote:
> I'm implementing WebSockets and have some questions where I can't figure out the answer from the draft spec.
> That could be my fault, but hints would be welcome anyway ..
> 
> 1.
> Are data frames of payload length 0 allowed?
> If so, are data messages of length 0 (where all fragments have length 0) allowed?
> If so, what is a browser supposed to do? Fire an JS onMessage() event with 0 length string?
> 

0 length strings are valid and useful strings. (e.g. the name of the
band on stage that is displayed on an electronic marquee, when nobody is
playing.) Fragmenting a 0 length string is silly, but harmless.

> 2.
> In the context of a persistent http connection, what is allowed _before_ the WebSocket handshake
> (that is before client sends GET with upgrade header)?
> For example, are "normal" GETs allowed? In other words: is it permissable for a client to do "normal"
> http on a persistent connection, and then  - at any point - decide to request to upgrade to WebSocket?
> 

prior to the 101 the connection is http. So yes, you can send the
bootstrap http request on a reused http connection. You can do anything
HTTP allows you to do, because the handshake is carried out over HTTP.
That's the point of reusing the infrastructure.

> 3.
> When the client has sent an upgrade request, the spec says on server response
> 
>    The handshake from the server is much simpler than the client
>    handshake.  The first line is an HTTP Status-Line, with the status
>    code 101:
> 
>         HTTP/1.1 101 Switching Protocols
> 
>    Any status code other than 101 indicates that the WebSocket handshake
>    has not completed, and that the semantics of HTTP still apply.
> 
> Does that mean, the server may send not only http error codes, but also for example http redirects?
> 

The server may do so as they are legitimate HTTP. Note that HTTP doesn't
require the client to follow the returned redirects, it just defines
what they mean. In the context of the current Websocket API browsers
implementing that would not follow the redirect.

> 4.
> The spec defines 2 new URI schemes for WebSockets: ws and wss.
> 
> On the other hand, 
> http://tools.ietf.org/html/draft-ietf-websec-origin-02
> 
> "If the two origins are scheme/host/port triples, the two origins
> are the same if, and only if, they have identical schemes, hosts,
> and ports."
> 
> Does that mean that with browsers, a JS script coming from (http, somehost, 80) is not allowed
> to contact (ws, somehost, 80), because http != ws?
> 

loosely - the js can open ws to any host (not just somehost) because ws
builds in an origin enforcement opportunity for the server and this
mechanism dates back to day 1 of websockets. (as opposed to http which
needs to worry about hosts that don't participate in cross origin
enforcement algorithms).




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Subject: Re: [hybi] draft-10 questions
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On Sun, Jul 31, 2011 at 12:02 PM, Patrick McManus <pmcmanus@mozilla.com>wrote:

> 0 length strings are valid and useful strings. (e.g. the name of the
> band on stage that is displayed on an electronic marquee, when nobody is
> playing.) Fragmenting a 0 length string is silly, but harmless.


You could imagine sending a 0-length fragment would be useful as sort of a
keep-alive in the middle of a streaming message when the next chunk of the
message hasn't been generated yet.  Weak, admittedly.

-- 
John A. Tamplin
Software Engineer (GWT), Google

--000e0cd51b40bb3b1804a960944e
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<div class=3D"gmail_quote">On Sun, Jul 31, 2011 at 12:02 PM, Patrick McManu=
s <span dir=3D"ltr">&lt;<a href=3D"mailto:pmcmanus@mozilla.com">pmcmanus@mo=
zilla.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

<div class=3D"im">0 length strings are valid and useful strings. (e.g. the =
name of the</div>
band on stage that is displayed on an electronic marquee, when nobody is<br=
>
playing.) Fragmenting a 0 length string is silly, but harmless.</blockquote=
><div>=C2=A0</div><div>You could imagine sending a 0-length fragment would =
be useful as sort of a keep-alive in the middle of a streaming message when=
 the next chunk of the message hasn&#39;t been generated yet. =C2=A0Weak, a=
dmittedly.</div>

</div><br>-- <br>John A. Tamplin<br>Software Engineer (GWT), Google<br>

--000e0cd51b40bb3b1804a960944e--

From tobias.oberstein@tavendo.de  Sun Jul 31 10:13:08 2011
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From: Tobias Oberstein <tobias.oberstein@tavendo.de>
To: Patrick McManus <pmcmanus@mozilla.com>, "hybi@ietf.org" <hybi@ietf.org>
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Subject: Re: [hybi] draft-10 questions
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thanks for your explanations .. very helpful.

1. - 3.: ok, got it. nice example with 1.;)

> > 4.
> > The spec defines 2 new URI schemes for WebSockets: ws and wss.
> >
> > On the other hand,
> > http://tools.ietf.org/html/draft-ietf-websec-origin-02
> >
> > "If the two origins are scheme/host/port triples, the two origins are
> > the same if, and only if, they have identical schemes, hosts, and
> > ports."
> >
> > Does that mean that with browsers, a JS script coming from (http,
> > somehost, 80) is not allowed to contact (ws, somehost, 80), because htt=
p
> !=3D ws?
> >
>=20
> loosely - the js can open ws to any host (not just somehost) because ws
> builds in an origin enforcement opportunity for the server and this
> mechanism dates back to day 1 of websockets. (as opposed to http which
> needs to worry about hosts that don't participate in cross origin enforce=
ment
> algorithms).

just to be sure to get that right:

A js served to a browser from http://somehost:80 can contact _any_ ws(s)://=
XXX:YYY ?

It's at the sole discretion of that other host to accept (depending e.g. on=
 ws-origin) the ws ?

so for WebSockets in browsers, the "same origin policy" does not apply at a=
ll?

but the ws-origin header _will_ be set reliably by browser .. e.g. also for=
 background WebWorkers stuff
spawned by the original served js?

ok.

How does that translate to cookie handling (in browsers)?

e.g. under what circumstances will a cookie A set by the original js servin=
g http://somehost:80 be
delivered in the headers of an ws outgoing from that js?

or, is that browser implementation dependent?

sorry for that trail of Qs, but the answer was somehow completely unexpecte=
d for me ..


From diogo.pereira@ist.utl.pt  Sun Jul 31 10:49:39 2011
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From: Diogo Pereira <diogo.pereira@ist.utl.pt>
Date: Sun, 31 Jul 2011 18:49:02 +0100
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Subject: Re: [hybi] max frame size. was: consensus call on ability to announce max frame size
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On Sun, Jul 31, 2011 at 01:39, Greg Wilkins <gregw@intalio.com> wrote:
> I think it is more than just MUX that needs this. =C2=A0Because of the
> datagram style API of websockets, applications will not be stream
> based (blocked reading content), but will be called back when frames
> and/or messages are received. =C2=A0Thus there will be implementations th=
at
> have fixed buffer sizes and will not work with large frames and/or
> messages. =C2=A0 =C2=A0 =C2=A0For large messages, there is not much the
> implementations can do to influence the applications, so I don't see a
> limit being worthwhile. =C2=A0But for a max frame size, it is pretty simp=
le
> for implementations to fragment messages to respect a limit.


Can't the receiver just fragment large frames into smaller ones and
pass those to the application? That way it only needs to buffer as
much as it wants to.


--=20
Diogo

From gregw@intalio.com  Sun Jul 31 16:04:43 2011
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Subject: Re: [hybi] max frame size. was: consensus call on ability to announce max frame size
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On 1 August 2011 03:49, Diogo Pereira <diogo.pereira@ist.utl.pt> wrote:
> Can't the receiver just fragment large frames into smaller ones and
> pass those to the application? That way it only needs to buffer as
> much as it wants to.


Sure it can.    But semantically websocket endpoints are meant to
handle entire messages and are not set up to stream messages or even
to handle them frame by frame.   Specifically the APIs offered to
application are message based and not stream based, so there is little
point calling such and application with a frame (what is it going to
do with 2 bytes of a 6 bytes UTF-8 character?).

So implementations that receive large frames normally have to: a)
buffer them in memory; b) buffer them somewhere else eg disk; c) send
a frame-to-large error.

Solution a) can be asking for a DOS attack, so even with flexible
buffers, eventually most implementations will implement some limit and
go to c) Solution b) is much the same, except the limit is likely to
be much larger. Going directly to solution c) is the simplest and we
should expect simple implementation to exist.

In essence the limit is really a max message size limit, but there is
little good pre-warning what that limit is, because message size is
governed by applications and many/most applications will not know in
advance how large their message are or can be.   So telling them a
limit is not very helpful as they will not be able to act on it.  If
an app that needs 128k messages is run on infrastructure that has a
64k message limit, then it will just be an error (probably a 1003?).

If however, there is a frame size limit, then there is good reason to
advertise that, because implementations can act on it and do the
fragmentation before sending.  Note that the reason for a small frame
size limit is might be because of a desired to allow efficient MUX or
control frame interleaving rather than buffer limits, so false
fragmentation by the receiver is not that useful.

cheers

From gregw@intalio.com  Sun Jul 31 16:45:06 2011
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Subject: [hybi] Mux back pressure
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The thread about the max message size has made me think about the back
pressure mechanism proposed in the x-google-mux extension.   Each
channel has a maximum number of bytes that can be sent without
receiving an explicit flow control message allowing more bytes to be
sent.   I'm now wondering if byte count is the right flow control, and
perhaps message count would be better?

With stream based mechanisms like HTTP, byte counts work fine because
with the server receives a message it typically will have (or start)
an application thread blocked reading for the content.  The server
reads the content only as fast as it can give it to the waiting
application thread.  If the application consumer is running slowing
that the producer+network, then data is not read from the connection,
back pressure is generated and eventually the producer in the client
will block as it tries to write data.

But with message based mechanism, the data is not passed to the
application until an entire message is received.  Thus the
server/client will buffer the data until the end of the message is
received.   Thus if we use bytes as the measure of websocket mux back
pressure,  the OK to send more will be based on the bytes being placed
in a buffer and not passed to the application.   The problem with this
is that it does not measure the capability of the real consumer - only
the capability of the buffers.    This is somewhat akin to the buffer
bloat problem that is currently being seen in TCP/IP networks.  A
producer might send many thousands of small websocket messages,
thinking that the consumer is receiving them because flow control
messages are being received, when in reality those messages are just
being buffered in the receiver and the consumer might be many hundreds
or thousands of messages behind.

Note that without Mux, the backpressure will depend somewhat on the
implementation, but I think many implementations will not begin
reading/parsing the next ws message until the previous has been
processed by the application.  Thus a slow consumer will also
eventually result in TCP/IP back pressure to the producer.   So I'm
wondering if we want Mux connections to mimic this, then the back
pressure measure for mux should be number of messages instead of
number of bytes - or perhaps both?

cheers

From diogo.pereira@ist.utl.pt  Sun Jul 31 17:03:56 2011
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Subject: Re: [hybi] max frame size. was: consensus call on ability to announce max frame size
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On Mon, Aug 1, 2011 at 00:04, Greg Wilkins <gregw@intalio.com> wrote:
> Sure it can. =C2=A0 =C2=A0But semantically websocket endpoints are meant =
to
> handle entire messages and are not set up to stream messages or even
> to handle them frame by frame. =C2=A0 Specifically the APIs offered to
> application are message based and not stream based, so there is little
> point calling such and application with a frame (what is it going to
> do with 2 bytes of a 6 bytes UTF-8 character?).

Then the problem is the message size, not the frame size.
Maximum message size negotiation would solve this problem, as would a
stream-based or frame-based API. Maximum frame size negotiation would
achieve nothing, since one could still send e.g. a 1 GB message in
64-byte frames.


--
Diogo

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
To: Gabriel Montenegro <Gabriel.Montenegro@microsoft.com>
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Subject: Re: [hybi] consensus call on not specifying DNS SRV as mandatory to implement
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2011/7/30 Gabriel Montenegro <Gabriel.Montenegro@microsoft.com>:
> At the meeting and on the mailing list leading up to it, the chairs did n=
ot
> see consensus for the following action:
>
> =C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0 Specify DNS SRV as mandatory to implement in the websock=
et
> protocol specification.
>
> We=E2=80=99re confirming on the mailing list, and will declare this conse=
nsus call
> final by next Friday Aug 5.


I was proposing it, but after long discussions in the maillist I've
understood that mandating DNS SRV in WS clients would break too much
assumptions in HTTP world (which commonly just sees above HTTP layer
and not below).

The existence of HTTP proxies is also a big handicap since those
proxies should be upgraded/modified in order to perform DNS SRV
resolution *just* in case the HTTP request is a WebSocket handshake.
This last argument is enough to not mandate SRV resolution.

So I see two options (for the future):

1) As Greg says: consider again DNS SRV if we ever have a non HTTP handshak=
e.

2) Introduce SRV (optional of course) in HTTP protocol (WS would just
inherit it automatically).


Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [hybi] Mux back pressure
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On Sun, Jul 31, 2011 at 4:45 PM, Greg Wilkins <gregw@intalio.com> wrote:

> The thread about the max message size has made me think about the back
> pressure mechanism proposed in the x-google-mux extension.   Each
> channel has a maximum number of bytes that can be sent without
> receiving an explicit flow control message allowing more bytes to be
> sent.   I'm now wondering if byte count is the right flow control, and
> perhaps message count would be better?
>
> With stream based mechanisms like HTTP, byte counts work fine because
> with the server receives a message it typically will have (or start)
> an application thread blocked reading for the content.

This is a bit of simplification (for Java servlet based servers maybe).
Regardless, TCP based flow-control serves HTTP pretty well because what's
exposed to the application (i.e. RPC) and the underlying transport (i.e. TCP
connection) have a 1:1 mapping. Still, things may start to break when the
application tries to stream data asynchronously over a long-lived
connection. The mux protocol for websocket will likely have to deal with
end-to-end flow-control in a more complicated way. I tend to agree that
message-based flow-control is necessary, given the nature of the messaging
semantics that gets exposed to the application.



 The server
> reads the content only as fast as it can give it to the waiting
> application thread.  If the application consumer is running slowing
> that the producer+network, then data is not read from the connection,
> back pressure is generated and eventually the producer in the client
> will block as it tries to write data.
>
> But with message based mechanism, the data is not passed to the
> application until an entire message is received.  Thus the
> server/client will buffer the data until the end of the message is
> received.   Thus if we use bytes as the measure of websocket mux back
> pressure,  the OK to send more will be based on the bytes being placed
> in a buffer and not passed to the application.   The problem with this
> is that it does not measure the capability of the real consumer - only
> the capability of the buffers.    This is somewhat akin to the buffer
> bloat problem that is currently being seen in TCP/IP networks.  A
> producer might send many thousands of small websocket messages,
> thinking that the consumer is receiving them because flow control
> messages are being received, when in reality those messages are just
> being buffered in the receiver and the consumer might be many hundreds
> or thousands of messages behind.
>
> Note that without Mux, the backpressure will depend somewhat on the
> implementation, but I think many implementations will not begin
> reading/parsing the next ws message until the previous has been
> processed by the application.  Thus a slow consumer will also
> eventually result in TCP/IP back pressure to the producer.   So I'm
> wondering if we want Mux connections to mimic this, then the back
> pressure measure for mux should be number of messages instead of
> number of bytes - or perhaps both?
>
> cheers
> _______________________________________________
> hybi mailing list
> hybi@ietf.org
> https://www.ietf.org/mailman/listinfo/hybi
>

--000e0cd47cc2be4c2b04a9667820
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<br><br><div class=3D"gmail_quote">On Sun, Jul 31, 2011 at 4:45 PM, Greg Wi=
lkins <span dir=3D"ltr">&lt;<a href=3D"mailto:gregw@intalio.com">gregw@inta=
lio.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D=
"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
The thread about the max message size has made me think about the back<br>
pressure mechanism proposed in the x-google-mux extension. =A0 Each<br>
channel has a maximum number of bytes that can be sent without<br>
receiving an explicit flow control message allowing more bytes to be<br>
sent. =A0 I&#39;m now wondering if byte count is the right flow control, an=
d<br>
perhaps message count would be better?<br>
<br>
With stream based mechanisms like HTTP, byte counts work fine because<br>
with the server receives a message it typically will have (or start)<br>
an application thread blocked reading for the content. </blockquote><div>Th=
is is a bit of=A0simplification=A0(for Java servlet based servers maybe). R=
egardless, TCP based flow-control serves HTTP pretty well because what&#39;=
s exposed to the application (i.e. RPC) and the underlying transport (i.e. =
TCP connection) have a 1:1 mapping. Still, things may start to break when t=
he application tries to stream data asynchronously over a long-lived connec=
tion. The mux protocol for websocket will likely have to deal with end-to-e=
nd flow-control in a more complicated way. I tend to agree that message-bas=
ed flow-control is necessary, given the nature of the messaging semantics t=
hat gets exposed to the application.</div>
<div><br></div><div><br></div><div><br></div><blockquote class=3D"gmail_quo=
te" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;=
">=A0The server<br>
reads the content only as fast as it can give it to the waiting<br>
application thread. =A0If the application consumer is running slowing<br>
that the producer+network, then data is not read from the connection,<br>
back pressure is generated and eventually the producer in the client<br>
will block as it tries to write data.<br>
<br>
But with message based mechanism, the data is not passed to the<br>
application until an entire message is received. =A0Thus the<br>
server/client will buffer the data until the end of the message is<br>
received. =A0 Thus if we use bytes as the measure of websocket mux back<br>
pressure, =A0the OK to send more will be based on the bytes being placed<br=
>
in a buffer and not passed to the application. =A0 The problem with this<br=
>
is that it does not measure the capability of the real consumer - only<br>
the capability of the buffers. =A0 =A0This is somewhat akin to the buffer<b=
r>
bloat problem that is currently being seen in TCP/IP networks. =A0A<br>
producer might send many thousands of small websocket messages,<br>
thinking that the consumer is receiving them because flow control<br>
messages are being received, when in reality those messages are just<br>
being buffered in the receiver and the consumer might be many hundreds<br>
or thousands of messages behind.<br>
<br>
Note that without Mux, the backpressure will depend somewhat on the<br>
implementation, but I think many implementations will not begin<br>
reading/parsing the next ws message until the previous has been<br>
processed by the application. =A0Thus a slow consumer will also<br>
eventually result in TCP/IP back pressure to the producer. =A0 So I&#39;m<b=
r>
wondering if we want Mux connections to mimic this, then the back<br>
pressure measure for mux should be number of messages instead of<br>
number of bytes - or perhaps both?<br>
<br>
cheers<br>
_______________________________________________<br>
hybi mailing list<br>
<a href=3D"mailto:hybi@ietf.org">hybi@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/hybi" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/hybi</a><br>
</blockquote></div><br>

--000e0cd47cc2be4c2b04a9667820--

From brofield@gmail.com  Sun Jul 31 17:23:51 2011
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From: Brodie Thiesfield <brodie@jellycan.com>
To: hybi@ietf.org
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Subject: [hybi] extension params (was  draft-10 questions)
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All,

During the recent IETF meeting the format of the websocket extension
header parameters was brought up. The issue being should the header
follow the rules of HTTP (allowing quoted strings) or the simplified
rules of websocket draft (allowing only tokens).

On Mon, Aug 1, 2011 at 1:02 AM, Patrick McManus <pmcmanus@mozilla.com> wrote:
> On Sun, 2011-07-31 at 06:13 -0700, Tobias Oberstein wrote:
>> 2.
>> In the context of a persistent http connection, what is allowed _before_ the WebSocket handshake
>> (that is before client sends GET with upgrade header)?
>
> prior to the 101 the connection is http. So yes, you can send the
> bootstrap http request on a reused http connection. You can do anything
> HTTP allows you to do, because the handshake is carried out over HTTP.
> That's the point of reusing the infrastructure.

Since a websocket connection is plain HTTP prior to upgrade, then a
server supporting normal HTTP requests on a connection (i.e. not a WS
specific server) will already be using its full HTTP parser.

It is not possible to determine what is a WS upgrade message until the
headers have already been parsed, e.g. the following message is a
valid upgrade but that is not known until the last byte is parsed.
Additionally, take away the "Upgrade" header from the end and it is a
plain HTTP GET with some unnecessary headers.

        GET /chat HTTP/1.1
        Host: server.example.com
        Sec-WebSocket-Key: dGhlIHNhbXBsZSBub25jZQ==
        Sec-WebSocket-Origin: http://example.com
        Sec-WebSocket-Protocol: chat, superchat
        Sec-WebSocket-Version: 8
        Connection: Upgrade
        Upgrade: websocket

The WS headers have already been parsed by the HTTP parser. Unless the
special parsing of the Sec-Websocket-Extension depends solely on the
header key, there can be no special parsing. If it does depend on the
header key, then won't this require a change to the HTTP spec to
introduce the rule that this header key is parsed specially? If it was
so defined, then every header key of every HTTP message will need to
be examined to see if it requires the simplified parsing.

If a WS only handshake is ever defined, then all limitations can be
decided here, but since the current handshake is using HTTP for
bootstrapping, I don't think that we can specify any format for any
part of the upgrade message that differs from HTTP.

Of course, a server implementation may not implement a full HTTP
parser and return a 400 error on quoted string, but for the spec I
believe that we are bound by their spec until the protocol has been
switched to WS.


Regards,
Brodie

From gregw@intalio.com  Sun Jul 31 21:53:25 2011
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From: Greg Wilkins <gregw@intalio.com>
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Can't this be an instance of being forgiving in what you parse and
strict in what you generate?

We should allow ws servers to use existing HTTP parsers and to accept
quoted strings for any of the header values.
However we could also say that ws clients MUST NOT generate any header
values that need to be quoted.

cheers


On 1 August 2011 10:23, Brodie Thiesfield <brodie@jellycan.com> wrote:
> All,
>
> During the recent IETF meeting the format of the websocket extension
> header parameters was brought up. The issue being should the header
> follow the rules of HTTP (allowing quoted strings) or the simplified
> rules of websocket draft (allowing only tokens).
>
> On Mon, Aug 1, 2011 at 1:02 AM, Patrick McManus <pmcmanus@mozilla.com> wr=
ote:
>> On Sun, 2011-07-31 at 06:13 -0700, Tobias Oberstein wrote:
>>> 2.
>>> In the context of a persistent http connection, what is allowed _before=
_ the WebSocket handshake
>>> (that is before client sends GET with upgrade header)?
>>
>> prior to the 101 the connection is http. So yes, you can send the
>> bootstrap http request on a reused http connection. You can do anything
>> HTTP allows you to do, because the handshake is carried out over HTTP.
>> That's the point of reusing the infrastructure.
>
> Since a websocket connection is plain HTTP prior to upgrade, then a
> server supporting normal HTTP requests on a connection (i.e. not a WS
> specific server) will already be using its full HTTP parser.
>
> It is not possible to determine what is a WS upgrade message until the
> headers have already been parsed, e.g. the following message is a
> valid upgrade but that is not known until the last byte is parsed.
> Additionally, take away the "Upgrade" header from the end and it is a
> plain HTTP GET with some unnecessary headers.
>
> =A0 =A0 =A0 =A0GET /chat HTTP/1.1
> =A0 =A0 =A0 =A0Host: server.example.com
> =A0 =A0 =A0 =A0Sec-WebSocket-Key: dGhlIHNhbXBsZSBub25jZQ=3D=3D
> =A0 =A0 =A0 =A0Sec-WebSocket-Origin: http://example.com
> =A0 =A0 =A0 =A0Sec-WebSocket-Protocol: chat, superchat
> =A0 =A0 =A0 =A0Sec-WebSocket-Version: 8
> =A0 =A0 =A0 =A0Connection: Upgrade
> =A0 =A0 =A0 =A0Upgrade: websocket
>
> The WS headers have already been parsed by the HTTP parser. Unless the
> special parsing of the Sec-Websocket-Extension depends solely on the
> header key, there can be no special parsing. If it does depend on the
> header key, then won't this require a change to the HTTP spec to
> introduce the rule that this header key is parsed specially? If it was
> so defined, then every header key of every HTTP message will need to
> be examined to see if it requires the simplified parsing.
>
> If a WS only handshake is ever defined, then all limitations can be
> decided here, but since the current handshake is using HTTP for
> bootstrapping, I don't think that we can specify any format for any
> part of the upgrade message that differs from HTTP.
>
> Of course, a server implementation may not implement a full HTTP
> parser and return a 400 error on quoted string, but for the spec I
> believe that we are bound by their spec until the protocol has been
> switched to WS.
>
>
> Regards,
> Brodie
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