[Standards] XEP-0301 0.5 comments [Sections 6 and beyond]

Mark Rejhon markybox at gmail.com
Mon Jul 23 23:37:24 UTC 2012


[Part 2 of 2, continued, ultra-long discussion regarding Kevin's comments]
Note: Due to the large number of comments from Kevin, I'm focussing on
addressing Kevin's concerns for now.
I'd love to hear comments from others (Gunnar, Peter, Matt, etc) on the
discussions between me and Kevin.


On Mon, Jul 23, 2012 at 10:32 AM, Kevin Smith <kevin at kismith.co.uk> wrote:
>
> 6.1.4 - "it is acceptable for the transmission interval of <rtt/> to
> vary" - yet earlier there was a SHOULD saying it doesn't vary, wasn't
> there?
>

[Comment]
http://xmpp.org/extensions/xep-0301.html#transmission_interval
In the "Transmission Interval" section, I said "approximately 0.7 second"
and it refers to "continuously-changing message"

Meaning this is valid: You could transmit varying intervals and stay in
spec, like 0.5s, 1s, 0.7s, 0.73s, 1.1s, 5s (typing paused then resumes),
0.7s, 0.3s, 0.8s, etc.   You can technically  optimizes for surges of
faster and slower typing, as long as the default average interval is
approximately 0.7 seconds.   Some clients will use strict calculations, and
send at exactly 0.7 seconds whenever the message has changed.  That
behavior acceptable too.  (RealJabber does this)  ....
Is there a minor wording that I can do, in "Transmission Interval" to make
this clear that it is allowed to vary, while satisfying people who ask "is
that a default value?" (that's why I added the word 'default')



> 6.2.1 - I suspect this should be more prominent than buried inside
> Implementation Notes
>

[Comment & Question]
I'm glad you think this section is important enough to be part of the
Protocol.  Activation Methods is quite an important inclusion in the
specification, even though some people may disagree (Gunnar prefers real
time text to be activated at all times, for example -- and technically I
agree -- but realistcally, implementers want to choose their own activation
mechanisms).

Perhaps I could split it into a "5.1. Business Rules" section (ala
XEP-0085) but I'm not sure that this is appropriate.
Alternatively, I can add it as a "6. Business Rules" (bumping
Implementation Notes as a section 7)
Peter, David, et cetra from XSF, any comments?



> 6.2.1 - I think that presence decloaking is probably a better approach
> to this than sending init.
>

[Change Made & Comment]
*"Signalling first (by transmitting <rtt [[[event='init'/> as the first
<rtt/> element.)"*

-- The primary purpose of <rtt event='init'/> is not for disco. Therefore,
decloaking has nothing to do with init (unless init is used for disco)
-- That includes any reason, such as activating before typing.  Some
implementers want an activation feature (e.g. button, menu, preferences,
etc)
-- Timing of activation can be separate of the timing of sender beginning
to compose text.   The existence of <rtt event='init'/> allows decoupling
activation timing from actual transmission of real-time text.
-- Activation/Deactivation may occur multiple times during the same chat
session.  It's useful for signalling re-activation of real-time text after
<rtt event='cancel'/> because some implementations might otherwise ignore
real-time text for the remainder of the chat session after  receiving <rtt
event='cancel'/>.
-- Clients can send <rtt event='init'/> even if they have some real-time
text to begin immediately. (i.e. <rtt event='init'/> immediately followed
by <rtt event='new'/>)  ... Thus, it is acceptable for implementers to
always send <rtt event='init'/>
-- Theoretically <rtt event='init'/> could be made REQUIRED, but that's not
a good idea, especially because recipients can come online after the sender
has already started composing a message (includes MUC and simultaneous
login situations).

Even if we eliminate the implicit discovery requirement and "Determining
Support" is always followed, the use of 'init' is still a requirement for
some implementers for activation/deactivation, to decouple the timing of
beginning of real-time text, from the timing of actual creation of a
real-time message.   So, sending init is still useful even you follow
Determining Support.

Can you provide any suggestions of any further clarifications for <rtt
event='init'/>?



> 6.2.1 - That said, if people disagree and want another 85-ish
> non-disco mess, I think this can be clarified a bit - at the moment it
> sounds like disco and init discovery are alternatives, rather than
> init only being a fallback for when disco isn't available. Perhaps
> something like:
> """
> Activation of real-time text in a chat session (immediate or
> user-initiated) can be done by:
> * Immediately transmitting real-time text (if the feature is
> advertised in by the recipient, as described in Determining Support);
> or
> * Where Disco knowledge isn't available (e.g. sending to an entity for
> which presence information isn't available, and thus the full JID
> isn't known and can't be queried) by sending a <message/> stanza
> containing only a "<rtt event='init'/>". In this case there MUST be no
> further transmission of RTT elements until the recipient indicates
> support - either by exposing information necessary to use service
> discovery, or by replying with a (non-cancel event) RTT element of its
> own.
>

[Comment]
One observation, section 5.1 of XEP-0085:
http://xmpp.org/extensions/xep-0085.html#support
"Before generating chat state notifications, a User SHOULD explicitly
discover whether the Contact supports the protocol defined herein ..."

Likewise, XEP-0301 already allows implicit negotiation simply by sending
<rtt event='init'/> which is also valid anyway even if you do "Determining
Support".  The primary purpose of <rtt event='init'/> is not for disco, but
to decouple activation timing from creation of a new real-time message.  It
just happens to be the best "first rtt element" to use.  (It happens to be
a conveniently valid element to use with or without a sending client
determining disco first, in the same style as XEP-0085)

As XEP-0301 was also designed to also behave as an extended chat state, I'd
rather keep it synchronous with XEP-0085 requirements
(I should have studied it more closely months ago, and synced up
requirements much earlier -- to keep the debate simpler).
I'll stay in sync with future changes to XEP-0085, too (if that's ever
done).
[If discussion is needed on this point, let's split reply to a separate
thread again -- it's an a topic meriting its own separate thread, since
this could distract from all the other good minor changes in this thread]

Technically, that <rtt event='init'/> isn't "messy" because its purpose is
not designed for disco --
it simply permits decoupling of the activation moment from the moment of
creating a new real-time message.


6.3 - "All action elements only have absolute positioning, and
> positioning does not depend on previous action elements" - this isn't
> true, positioning is dependent upon processing of previous action
> elements - a deletion will effect a change of index in all subsequent
> code points.
>

[Change Made]
*"Action elements only use absolute positioning (relative positions are not
used by this standard), so clients do not need to remember the position
value from previous action elements."*

You do not need to keep track of the state of the previous cursor position.
 People with an ANSI escape code mindset (VT100, VT102, from the old
communications terminals days etc) were asking me if cursor positions can
be relative to the previous cursor position. (answer: no). ... Relative
cursor positions are never used.

Bottom line: Clients don't need to remember cursor position state
information between action elements.  They only need it for display
purposes after processing an action element -- *the cursor position is
completely reinitialized after every action element*.   (except <w/>
elements which has no effect on the real-time message text or cursor itself)
The only dependancy is the length, if p is not defined, then p defaults to
the message length.
Comments?  I have to keep the RFC4103 and VT100 "mindset" people happy too.



> 6.4.1 - It might be useful to reference some method of calculating
> this. It's not immediately obvious to me that it's trivial to work out
> edits without resorting to something that ends up polynomial in the
> worst case (or oversimplifying the edit), so some guidance would be
> handy here.
>

[Comment]
-- It's actually simpler than it looks
-- It is a linear calculation (CPU expense linearly proportional to message
length), not polynomial.
-- Text change event occurs every key press, so most of the time, message
change is only 1 character between text change events!.
-- In almost all cases, text change events will generate only 1 character
of change.   Except for things like autotext, autocorrect, and pastes --
then it's a single block event.
-- Therefore, I don't bother to compute more than one edit per change
event.   It's not worth optimizing for this edge case (it'll just look like
one larger text change).

Becaues of this, I don't bother to worry about two separate text changes in
two parts of the message -- I'm only worried about the first and last
changed character.  Then I create ONE or TWO action element (either a text
insert and/or delete event) Pretty simple.  Programming algorithm is found
here (line 592):
http://code.google.com/p/realjabber/source/browse/trunk/Java/src/RealTimeText.java?r=24#592


Note: There are several simplifications that can be made for implementers
that choose to not implement a Remote Cursor (just define position as -1),
and less state information is necessary (in simplest implementations, you
really only need to remember the real-time message string.  This is the
minimum state info to keep for JavaScript state preservation while waiting
between calls, other than whether or not real-time text is active or not)


6.4.3 - this says that implementations "may" do this, and I suspect
> that it really is discouraged rather than truly optional (indeed, the
> language elsewhere says as much).
>

[Change Made]
Beginning now says *"It is possible for sender clients to implement
[[[Message Reset]]] as the only method of transmitting changes to a
real-time message."*
Although it already explains why it's discouraged, I've now removed the
word "may" to reduce the permissive-sounding tone.



> 6.4.4 - this looks like something discouraged, too, but this isn't
> mentioned that I can see.
>

[Comment]
6.4.4 is useful if it's not humans generating real-time text.  For example
transcription bots, gateways, etc.  So it's quite simple/useful to have
append-only real-time text (and you can still do key press intervals, if
needed, unless you're outputting fully-transcribed words one full word at a
time)


6.5 - "Upon receiving Action Elements in incoming <rtt/> elements,
> they are added to a queue in the order they are received. This
> provides immunity to variable network conditions, since the queueing
> action smooth out the latency fluctuations of incoming transmission."
> - it's not clear to me that it's the queuing that does anything to the
> latency. Also 'action *will* smooth out'.
>

[Change Made]
*"This provides immunity to variable network conditions, since the queueing
action will smooth out incoming transmission (e.g. receiving new <rtt/>
while still processing a delayed <rtt/>).*

Network issues can cause huge variability in transmission interval.
-- The sender may be sending <rtt/> elements on a 0.7s, 0.7s, 0.7s, 0.7s,
0.7s
-- The recipient may be receiving <rtt/> elements delayed 0.9s, 1.3s, 1.5s,
0.0s, 0.0s  (due to network conditions)
This results in stall-surge behaviours of real-time text that need to be
smoothed out during playback, for the best user experience while preserving
key press intervals.

Due to various network bufferings that occurs in intermediate servers, and
potentially reception issues (loss of wireless reception for a fraction of
a second), and other reasons of huge ping variability, the recipient could
get the messages in 0.9s, 1.3s, 1.5s, 0.0s, 0.0s  .... That means the
<rtt/> received at T+1.3s would be received only 0.4s after the one at
T+0.9s .... therefore, the recipient is still playing the previous <rtt/>
while it received the next <rtt/>.   It is therefore, necessary to use the
queueing/buffering action for improved user experience.    Also observe
after a lag (0.9s, 1.3s, 1.5s), the the final 3 <rtt/> is received
simultaneously (1.5s, 0.0s, 0.0s), causing a whopping total 2.1s of <rtt/>
actoin elements to be buffered up for playback to the recipient's display.
 That's also why I included the last paragraph about speeding up playback
if there's a situation of excess amount of buffering of action elements
from a sudden incoming surge of <rtt/> elements.



>  6.5 - " In addition, it is best to process <w/> elements using
> non-blocking programming techniques." - I don't really know what this
> is doing here.
>

[Change Made]
*"In addition, it is best to process <w/> elements asynchronously, to avoid
interfering with client operation."*
*
*
This is simply a generic comment that indirectly refers to timers and
multithreading, rather than inserting a "Sleep" statement in the middle of
a single-threaded program.  This causes freezing in user interfaces,
especially with long <w/> elements (e.g. <w n='500'/>) could cause a 1/2
second program freeze while it's processing that action element, which is
bad.   If you're doing MUC, or multiple windows, and you have lots of <w/>
elements simultaneously, they all need to be processed asynchronously on
their respectively real-time messages.


6.6 - "There are other special basic considerations" - isn't that
> nearly oxymoronic?
>

[Change Made] -- removed sentence.  The heading "Other Guidelines" is
self-explanatory


6.6.1 - "For specialized clients that send continuous real-time text
> (e.g. news ticker, captioning, transcription, TTY gateway), a Body
> Element can be automatically sent when messages reach a certain
> length. This allows continuous real-time text without real-time
> messages becoming excessively large." - Is this true? Sending a body
> means you reset the state to the content of the body and terminate
> that RTT message, which doesn't seem consistent with continuing RTT.
>

[Change Made]
The change made:* "...a [[[Body Element(link)]]] can be sent and then a new
real-time message started immediately after, every time a message reaches a
reasonable size"*

It was meant to explain you just simply begin a new real-time message in
order to prevent real-time messages from becoming excessively large.
 Observe that in
http://www.marky.com/realjabber/anim/real_time_text_demo.html you can begin
a new real-time message immediately after sending a body element.



> 6.6.3.1 - This doesn't seem like the wrong approach if RTT is wanted
> in a MUC (at least until we have per-MUC disco stuff), but I'm
> somewhat worried about the effect this has as an amplification attack.
> I don't know what we should say here, but if people can have a think
> it'd be good.
>

[Comment]
I know MUC is a loaded can of worms, isn't it -- so a lot of things are
beyond scope of XEP-0301.
So, for this reason, I even say that implementers can choose to implement
real-time text only for one-on-one conversation, and avoid the MUC issues
altogether.

However, MUC is a requirement for some implementers.
Technically, there are a lot of things that can be done:
-- (within spec) Methods found in [[[Congestion Considerations]]]
-- (beyond scope) Server-based methods of controlling this (i.e.
rate-limiting, bandwidth-optimizing such as merger of <rtt/> elements
(action elements can safely be merged in adjacent <rtt/> transmissions if
it's part of the same message), server-side disco for commanding
everybody's transmission interval, server-side signalling of changed
transmission intervals during changing server loads, etc)
-- There are potential ideas beyond scope of XEP-0301 to improve the MUC
situation.

...MUC is a requirement of XEP-0301 but I wanted to include at least
minimal coverage to MUC that is reasonable.  It is a part of a Next
Generation 911 demo (InDigital Inc.  Although they rarely publicly comment,
search for "indigital.com" recipients in standards at xmpp.org for their
comments) for XEP-0301 that was shown to FCC earlier (reasoning: allows
other PSdAP people joining to "monitor" a conversation between a caller and
an emergency operator.  Also technically provides a convenient mechanism
for transferring real-time text conversation from one person to another).
 Such an emergency optimized server would essentially be intranet-optimized
(i.e. not open to public XMPP).
....Also, as many of us already are familiar with, 2-person chats can be
turned into 3-person chats by inviting someone, and I needed to include at
least basic MUC info, for implementers that want to do that in a reasonable
seamless manner.  I feel that any MUC improvements such as server-based
interval control and other improvements, can be specified in the future,
perhaps as an extended XEP.   There are not many implementations of MUC an
XEP-0301 yet, and we need more field experience without removing MUC from
XEP-0301.   I do not foresee that future MUC-specific optimizations would
necessarily impact protocol



> 6.6.3.2 - this seems inconsistent with an earlier section that (I
> think) was recommending or mandating support for multiple full JIDs.
>

[Comment]
I made comments earlier.
Even when senders send to the full JID, recipients can just process
real-time messages based on bare JID.
This makes it simpler for implementers of clients to implement only a
single real-time message per chat window.
It is a significant user interface complexity concern to gain the
capability of multiple simultaneous real-time messages in the same chat
window user interface.
Also, it is intuitive behaviour because of "Keeping Real-Time Text
Synchronized" so a simultaneous login user switching computers, the
recipient would simply see their copy of the real-time message switch
instantly from the partially-composed message from the old system to the
partially-composed message from the active system.  (thanks to the Message
Reset feature of "Keeping Real-Time Text Synchronized".   Even further
enhanced if good resource locking is done, too.  This is acceptable UX
behaviour, as a login is meant (in 99%+ of cases) to only have one typist.



> 6.6.5 - seems somewhat out of place. How many systems are there these
> days that can't keep up with a human typist? And telling people that
> they need to make their applications flicker-free just seems odd.


[Comment]
When retrofitting real-time text to an existing chat program, some tend to
make their software cause a repaint every key press, so it's meritworthy to
make a brief mention, although I agree it is quite borderline from the
perspective of a "specification".   At 10 key presses per second for a
120WPM typist, the real-time message can be repainted 10 times per second,
and if the repaint is not done efficiently, it can flicker or consume CPU,
etc.
Suggestions of a better wording is welcome?



> 6.6.6 seems redundant.
>

[Comment]
It might be, but "Total Conversation" is quite significant among
accessibility circles in Europe, it's not used as much in North America,
but I must satisfy this audience, too.  Improved wording is welcome though,
but I don't think anything in 6.6.6 affects protocol


7 - these examples seem to be to a bare JID, and therefore can't have
> had caps already indicate support, but lack support discovery. It'd be
> good to note this.
>

[Change Made]
*"For simplicity, these examples use a bare JID, even in situations where a
full JID might be more appropriate."*
Good point, senders still really generally ought to send to full JID, even
though recipients don't have to keep track of real-time messages per full
JID (recipients can just track per bare JID).
Also, when the resource is not locked yet (i.e. recipient hasn't replied
yet) it is fine to send real-time text only to the bare JID.  This makes
the real-time text show up on all concurrently-logged in resources
simultaneously.  XEP-0085 Chat States tend to work this way too in most
software.  Also, I've considered the Activation/Deactivation scenarios in a
Simultaneous Login scenario, in both resource-locked and resource-unlocked
states.  All of the scenario combinations check out to intuitive and/or
acceptable UX.

(Once resource is locked, the real-time message pauses on unused resources
since the unused resources are no longer receiving any real-time text, and
eventually gets timed out via Stale Message handling)



> 7.4.2 - this includes an RTT including a wait in the element with the
> body - but once the body is received the RTT state is discarded and
> the body replaces it, if I remember earlier in the XEP correctly (and
> it was quite a while ago now).
>

[Comment]
That's right.
-- However, it's important to finish sending <rtt/> for the whole message,
because it assists in verifying the integrity of real-time text in some
mission-critical implementations.   Basically, the text in body can be
compared to the real-time message, to make sure that the message is
identical, and if it isn't identical, it's possible to display a warning
indicator that the <rtt/> final text disagrees with <body/> text.
-- Technically, I could recommend that <rtt/> always be transmitted in
separate <message/> than <rtt/>, although I don't think it is a good idea
to make this REQUIRED.
-- Also, some implementations may choose to finish playing back <rtt/>
before displaying the <body/>, although it's true that I generally
recommend catching up the message immediately to prevent lagging, but I
don't strictly require that:
http://xmpp.org/extensions/xep-0301.html#receiving_realtime_text
Comments appreciated, in the light of my reasoning?



> 8 - Why are we picking out Google Talk as an XMPP exemplar?
>

[Change Made] -- 3 votes were received to this date, so I've now removed
even though I wanted to keep it as an example.  There are so many
programmers on Google Talk who doesn't quite realize that Google Talk is
XMPP.  That's why I wanted to keep at least one mention.


8 - Why are we telling SIP clients what specs to use?
>

[Change Made]
Those are "examples" only, so I've made a change to make that clear.
New text: *"For example, clients that use XMPP can utilize this XEP-0301
specification, and clients that use SIP might utilize IETF RFC4103, RFC5194
and ITU-T T.140."*


8 - All of this section seems somewhat out of place in a XEP.
>

[Comment]
I've managed to reduce the size of Interoperability Considerations
significantly (to the best of my ability), but there are several people
including actual implementers (outside the XMPP umbrella) that are
demanding this text be bigger than it is now.  Gunnar made a gateway for
SIP-to-XMPP interoperable real-time text, and it is a big raison d'etre of
keeping Section 8, he is also sharing his experiences as well.  I'm also
debating with people against making this size bigger by not adding too much
information to this section, since I also agree that it's mostly out of
scope of this specification.   Gunnar also repeatedly told me I should not
make this section even smaller too.  Edward Tie wants me to add more TTY
info to it. (I slipstreamed a small sentence "This can include TTY and
textphones" after the gateway servers sentences.  There are other
implementers outside of XMPP raising a big fuss about how to interoperate
with XMPP, so I think some *semblance* of section 8 is extremely critical
to satisfying a particularly vocal and important audience of accessibility
advocates.
The current Section 8 a compromise between what XMPP wants and what
accessibility implementers want, as XEP-0301 is of interest to
accessibility vendors, moreso than other specifications, and there are
special reasons to make XEP-0301 intereoperate with other standards used in
accessible communications.


10.1 - "It is important for implementers of real-time text to educate
> users about real-time text. " - this doesn't really seem right.


[Change Made] -- Good catch, I see the redundancy.
*"It is important for implementors to educate users about real-time text".*



> 10.1 - I think a sensible Privacy note would be to make RTT opt-in.


[Comment]
That depends on the market.  Mainstream client? (opt-in)
 Accessibiltiy-market client? (opt-out)   Emergency mode?
I am in contact with different implementers who will pounce on me if I
suggest either direction (opt-in versus opt-out).


10.2 - "also needs to also "

10.2 - "(e.g. deferred XEP-0200)" - just XEP-0200, I'd have thought.
>

[Change Made] for both



> 10.2 - I think blaming encryption for the increased number of stanzas
> RTT generates is a little disingenuous.
>

[Change Made]
There was a debate on this mailing list about turning off <rtt/> when
stanza-level encryption is done, because of the extra overhead.  I instead
opted to add an explanation sentence, rather than recommending turning off
<rtt/> when stanza-level encryption is done.
I will re-word the sentence to avoid sounding 'disingenious'.
*"It is noted that real-time text can have a higher rate of message
stanzas, contributing to additional overhead. See [[[Congestion
Considerations(link)]]]"*


10.3 - "The nature of real-time text result in"
>

[Change Made]
*"The nature of real-time text can result in"*



> 10.3 - "than may otherwise happen in a non-real-time text
> conversation. This may lead to increased" s/may/would/ s/may/will/
> respectively will remove normative language.
>

[Change Made]



> 10.3 - "including stanzas dropped by an overloaded server" - I think
> "including stanzas dropped during a network or server error" would be
> more appropriate.
>

[Change Made]
*"including stanzas dropped during a network issue or server error".*
Your suggestion is good.

Side note: It is a catch-all for error and non-error situations ("network
issue"), including DoS protection that kicks in, or even plain server
non-compliance (that doesn't return any error conditions).   If networks
would work perfectly, we would not even need language like this.   Issues
can even be obscure.  An example is the HTTP layer of a BOSH connection
(where the BOSH itself works flawlessly, but a DoS situation cause HTTP to
drop a few request, which looks like dropped stanzas to the end software).
  Or routers in certain countries dropping packets due to a disallowed
phrase, etc.  (this can manifest itself as dropped message stanzas, until a
new message is created or the offending text is removed)     Anyway,
avoiding politics -- but clearly, there's all sorts of weird reasons for
dropped message stanzas, regardless of reason of dropped stanza.  Server
non-compliance issues.  DoS protection kicking in.  All conveniently caught
by the catchall phrase "network issue and server errors" without needing
further explanation.


10.3 - "Use of this specification in the recommended way will cause a
> load that is only marginally higher than a user communicating without
> this specification." - do you have numbers for this? It seems quite
> counterintuitive, I'd expect it to increase the server load due to
> message routing roughly by a factor of the number of RTT transmitted
> between each typical <body/>.
>

[Comment]
Not always necessarily true -- The average instant message is short, often
1 to 5 words. (under 40 chars)
Most people on chat networks don't type large messages.  (Programmer types
like me do, though)

Hi!
How are you?
Good. You?
What's up?
Wanna go to movies?
Sure.
Star Wars XVIII?
No, Jaws IX.
Hey, have u heard? He did it!
OMG, tell me!
etc.

Although an extreme example, you get the idea.  Someone else (Gunnar?
Gregg?) cited a paper that the average instant message length is less than
40 characters.  Also, XMPP message transmissions are sent anyway such as
XEP-0085 chat states, and <rtt/> can be slipstreamed into some of those
same messages that are being transmitted anyway.   It is true that RTT will
result in some people hitting Enter less often, but that's just replacing
earlier behaviour anyway.

Therefore, it's frequently about 2 to 3x more stanzas than what would have
happened without real-time text.  On top of this, for an optimized server,
additional messages sent shortly after the previous message, have only a
small additional 'cost' in resources.
So the statement has truth to it --
(Anyone, can someone remind me of the link to the university paper?  It was
also posted here over a year ago.  Maybe I should cite it, to solidify this
statement.)



> 10.3 - "Bandwidth overhead of real-time text is very low compared to
> many other activities possible on XMPP networks including in-band file
> transfers and audio" - This is a little disingenuous where IBB is a
> fallback, and audio never travels over the XMPP network. I'd remove
> the line completely.
>

[Change Made]
*"Bandwidth overhead of real-time text is very low compared to many other
activities possible on XMPP networks."*
It is more generic.  I actually get questions of how much bandwidth
XEP-0301 uses, at least as a relative basis to other XMPP technologies.  To
go into further detail, I could insert some details from the document that
I made for Darren Sturman who said bandwidth considerations make or break a
standard -- and I could insert the bandwidth-estimation formula I developed
-- or go into generalities such as "average typing speed consumes about X
bytes per second".    But I think this sentence should be sufficient;
questions can be addressed separately from the spec.   Comments?


14 - (I appreciate the acknowledgement, thank you)
>

[Comment]
You're welcome!
Any other contributors that I have forgotten?


14 - It's usual in XEPs that acknowledgements are done personally
> rather than by affiliation, so I think it'd be sensible to just leave
> the names in and remove affiliations.
>

[Comment]
OK, thanks -- I will make some inquiries if that is OK.  Some people are
used to RFC's where affilation is often mentioned.  In all probability,
I'll be making this change.


14 - I find the comment acknowledging the invention a bit odd. It's
> assumed that the XEP is your own work, and "invention" is a term I've
> more commonly come across in relation to patents - I assume there
> isn't a patent associated with this that you're assigning to the XSF?
>

[Comment & Question]
There is no patent.  Many told me I should patent it, but I've instead
open-sourced the idea out into the open.  I believe this is the world's
first real-time text standard that preserves key press intervals
independently of transmission intervals.  A comparison animation at
http://www.marky.com/realjabber/anim/real_time_text_demo.html )
 -- Ideally, it would be nice to be acknowledged for this idea somehow
*somewhere*, one way or another, even if it just generically says *"Mark
Rejhon came up with the method of preserving key press intervals, which is
called "Natural Typing" at R3TF"*.   (The technique is called "Natural
Typing" within all of us at R3TF)
Comments?


Appendix B - it's usual to just have author name, email and JID here.
> We don't generally link out to the authors' websites.
>

[Change Made]


Thanks!
Mark Rejhon
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