Internet-Drafts | 28 Dec 2006 21:50
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I-D ACTION:draft-ietf-ieprep-domain-frame-08.txt

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

	Title		: A Framework for Supporting Emergency Telecommunications Services (ETS) Within a Single
Administrative Domain
	Author(s)	: K. Carlberg
	Filename	: draft-ietf-ieprep-domain-frame-08.txt
	Pages		: 18
	Date		: 2006-12-28
	
This document presents a framework discussing the role of various
   protocols and mechanisms that could be considered candidates for
   supporting Emergency Telecommunication Services (ETS) within a single
   administrative domain.  Comments about their potential usage as well
   as their current deployment are provided to the reader.  Specific
   solutions are not presented.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-ieprep-domain-frame-08.txt

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(Continue reading)

Janet P Gunn | 2 Feb 2004 22:43
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RE: I-D ACTION:draft-ietf-ieprep-domain-req-00.txt


Picky is fine.  I have been known to be picky!

Remaining comments in line.

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                      "Ken Carlberg"                                                                                             
                      <carlberg                To:      "'Janet P Gunn'" <jgunn6 <at> csc.com>                                        
                       <at> g11.org.uk>             cc:      <ieprep <at> ietf.org>                                                        
                                               Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt              
                      01/28/2004 01:40                                                                                           
                      PM                                                                                                         

Janet,

The following response may be a bit picky on some issues/comments, but I
feel it will be helpful to clarify things a bit.

> I think that, under "normal" circumstances, the QoS mechanisms for
> conventional VoIP will be more than adequate to support ETS, and
(Continue reading)

Ken Carlberg | 3 Feb 2004 13:58
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RE: I-D ACTION:draft-ietf-ieprep-domain-req-00.txt


> [jpg] I guess my unwritten question here is "Are you planning to have
a
> document which DOES address this."  I agree that it is somewhat
> out-of-scpoe for the two new documents.

My plans with respect to potential IEPREP documents is to work within
the scope of the charter.  Lets discuss this offline to make sure I know
exactly what you are looking for and I'll see if I can provide
assistance.

-ken
Eagan, Christopher | 30 Jan 2004 16:44
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RE: I-D ACTION:draft-ietf-ieprep-domain-req-00.txt

> We have demonstrated in lab tests to DISA and various other 
> organizations 
> that, a 1%, 5% and higher packet loss has significant impact on Voice 
> Quality (VQ) of all codecs, while at the same time - having some 
> organizations maintain the requirement that VQ cannot drop 
> below a 4.0 MOS.

I don't think the 4.0 MOS requirement is applicable to all (perhaps
most) voip carrying networks that could benefit from MLEF or another
packet level marking scheme.  This is cited as a requirement for some
DoD switched networks and does not necessarily extend to ALL voip
carrying networks, or all DoD voip carrying networks, or even all DoD
switched voice networks.  Actually, providing a 3.0 MOS would be a
significant improvement to some current voice networks - especially
wireless, mobile, low bandwidth ones.  Given this, 30% packet loss is
sustainable and acceptable in some situations.  Furthermore, MOS is not
necessarily the best measure of voice quality for all voice networks.
For example, in tactical networks, intelligibility is important while
fidelity is not as important.

Regardless, I realize if someone said that some arbitrary X quality was
required, you could come with the same argument with different packet
loss numbers.  The point is, each network and even each link may have
different requirements.  Providing multiple tools (e.g. RSVP based CAC +
MLEF) to engineer each network, or in some cases each link, for the
specific requirements is needed.

Chris
Janet P Gunn | 27 Jan 2004 23:32
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Re: I-D ACTION:draft-ietf-ieprep-domain-req-00.txt


I think that, under "normal" circumstances, the QoS mechanisms for
conventional VoIP will be more than adequate to support ETS, and "special
treatment" in Call Admission Control is what will be needed for ETS.

But when  network congestion and/ or damage are severe, the "conventional"
mechanisms may no longer be adequate (especially when network failure
initiates the re-routing of Label Switched Paths, and significantly reduces
the available capacity).

As Fred Baker says in draft-baker-tsvwg-mlpp-that-works.00.txt, if call
admission is working properly, traffic marking is not needed in normal
operations.  It is when there are route changes - typically due to network
failures - that marking is needed.  When there are network failures, the
capacity and route assignments on which call admission was based no longer
apply, the QoS cannot be guaranteed until the new routes are stabilized,
and the call admission process adapts to the new capacity and routes.  This
may be a very local effect, of quite short duration.  But it has a
significant impact on the call quality of the calls that are affected.
While that paper addresses an IP implementation of MLPP, many of the same
concerns apply to ETS.

Network failures, and/or  massive and sustained statistical anomalies
(large, local bursts), and the associated route changes, are the problems
we need to focus on.

It is in this context that ETS calls must be given a high probability of
meeting the QoS criteria, even when network congestion and or damage are
severe.

(Continue reading)

Ken Carlberg | 28 Jan 2004 19:40
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RE: I-D ACTION:draft-ietf-ieprep-domain-req-00.txt

Janet,

The following response may be a bit picky on some issues/comments, but I
feel it will be helpful to clarify things a bit.

> I think that, under "normal" circumstances, the QoS mechanisms for
> conventional VoIP will be more than adequate to support ETS, and
"special
> treatment" in Call Admission Control is what will be needed for ETS.

Within the context of what we have written in the past in the IETF, ETS
is an umbrella term that is meant to cover a wide variety of
applications.  Some of these applications (eg, I-Am-Alive) do not
support the concept of CAC. So to say that "special treatment" (whatever
that is) of CAC is what will be needed for ETS is rather too specific a
statement to make.  

I think we can agree that in those applications used to support ETS and
dependent on CAC, an ability to distinguish and perform CAC is of high
interest and needs to be supported.

> As Fred Baker says in draft-baker-tsvwg-mlpp-that-works.00.txt, if
call
> admission is working properly, traffic marking is not needed in normal
> operations.  It is when there are route changes - typically due to
network
> failures - that marking is needed.  

Keep in mind that Fred's draft is in respect to MLPP.  Arguments from
the draft may be applied to other circumstances, but I would refrain
(Continue reading)

Mpierce1 | 9 Oct 2003 20:29
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Re: I-D ACTION:draft-ietf-ieprep-framework-06.txt

In a message dated 10/8/2003 6:54:38 PM Eastern Standard Time, sob <at> harvard.edu writes:


I'm not sure that jitter per se is a big issue - the distribution of
delay is - most of the packets need to get to the other end within
a reasonable limit - that can be because the basic delay is near the
limit and there is no jitter or the basic latency is less and the jitter
brings the delay to thye limit.


Problem seems partly to be "What is your definition of jitter?" I believe jitter is the same as "the distribution of delay".

In any case, "jitter" (or packet delay variation, as I'd rather call it, as ITU-T SG13 does) is an important issue. Delay is an important isses. Packet loss is an important issue. It's kind of hard to say that one is "more important" than the other.

Mike
Gunn, Janet | 9 Oct 2003 17:58

RE: I-D ACTION:draft-ietf-ieprep-framework-06.txt


"Bridging" as in James's topology paper.  Where the "call" originates and
terminates in the CSN, but passes through a single IP domain in between.  SO
not a "stub" because it is neither source nor destination.  But possessing
the other "single domain" characteristics.

-----Original Message-----
From: Ken Carlberg [mailto:carlberg <at> g11.org.uk]

> 4.4

> Third para
> Do you mean "stub domain" or "single domain"? The statement would seem
to
> also apply to a single domain "bridging" scenario.

I meant stub domains.  I'm not exactly sure what you mean by a 'single
domain "bridging" scenario'.  What is being bridged?
Gunn, Janet | 9 Oct 2003 16:26

RE: I-D ACTION:draft-ietf-ieprep-framework-06.txt

Drop rates.

It all depends on where you are looking.

I agree that from an end-user or end-application perspective, you can say
that VoIP needs a lower drop rate than say email or web access (using TCP).
The dropped VoIP packets are gone forever, but the dropped TCP packets will
be retransmitted, and will get there "eventually".

But from a network congestion perspective, the "ranking" is reversed.  The
dropped VoIP packets  are gone and do not contribute to additional
congestion.  The retransmitted TCP packets add to the traffic volume, and
make the congestion worse.  From a simplified perspective, you could say
that a 5% drop rate for TCP packets results is at least (because you have to
"compound" that 5% on each retransmission) 5% additional TCP traffic offered
to the network.  Not something that is desirable in an already congested
network.  

I think my concerns can be addressed by minor rewording, or a parenthetical
comment. 

Janet 

-----Original Message-----
From: Andrew Thiessen [mailto:andrew <at> its.bldrdoc.gov]
Sent: Wednesday, October 08, 2003 7:24 PM
To: 'Scott Bradner'; carlberg <at> g11.org.uk; ieprep <at> ietf.org; Gunn, Janet
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-06.txt

Right...

Jitter is important in that the size of the jitter buffer adds to the
overall mouth-to-ear delay and that an accurate jitter buffer size
selection is critical. If the jitter buffer size is smaller than the
delay variation on the network, voice traffic will necessarily be
dropped. If the jitter buffer is appropriately sized, the only impact
jitter should have is the delay that the buffer adds.

Whether or not a voice application can sustain a higher drop rate is
fairly dependent on the codec that is used. For instance, a G-711 codec
is more resilient to loss than a G-729 codec. The drop rate of a cloud
has a direct impact on the quality of the voice traffic. For instance,
we have lab tests that show with certain codecs that the estimated MOS
score drops by 1 for every 5% increase in loss.

Drop rates in elastic applications are typically more tolerant of drop
due somewhat to the protocols that are used. For instance, TCP has a
built in mechanism to retransmit lost packets, where UDP does not. Email
and web browsing would therefore be much more resilient to loss in that
a session with loss is easier to maintain with TCP than UDP. Would it
take forever to get email and see a web page with high loss? Yes, but
you would see it eventually. With VoIP, you may be able to maintain some
type of conversation, but the quality could be so bad that there
wouldn't be any point.

Andy

-----Original Message-----
From: ieprep-admin <at> ietf.org [mailto:ieprep-admin <at> ietf.org] On Behalf Of
Scott Bradner
Sent: Wednesday, October 08, 2003 3:54 PM
To: carlberg <at> g11.org.uk; ieprep <at> ietf.org; Janet.Gunn <at> DynCorp.com
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-06.txt

> Substance
> 1.2 third para
> I have always understood that jitter was more important than delay for
VoIP,
> but you do not mention it.

I'm not sure that jitter per se is a big issue - the distribution of 
delay is - most of the packets need to get to the other end within 
a reasonable limit - that can be because the basic delay is near the
limit and there is no jitter or the basic latency is less and the jitter
brings the delay to thye limit.

Scott

_______________________________________________
Ieprep mailing list
Ieprep <at> ietf.org
https://www1.ietf.org/mailman/listinfo/ieprep
Gunn, Janet | 9 Oct 2003 16:00

RE: I-D ACTION:draft-ietf-ieprep-framework-06.txt

To clarify what I mean.

If you have a long, but consistent delay (say of the order of .5 seconds
(500 milliseconds) in an intercontinental satellite call) it is very
irritating from a conversation point of view- figuring out when it is "your
turn to talk", and avoiding talking over each other.  But the speech quality
is perfectly intelligible.

But if you have a highly variable delay, ranging from, say, 100 milliseconds
to 500 milliseconds, with an average delay of  300 milliseconds, you have a
high probability of "gaps" in the middle of words, even if there is no
packet loss.  You can, of course, buffer all the packets to the equivalent
of 500 milliseconds delay, in which case the speech is intelligible, but
then you gain no advantage from the fact that the average delay is 300
milliseconds instead of 500 milliseconds.

You could thus say that what is important is the "maximum delay" (rather
than the combination of "average delay" and "jitter").  But in the real
world, there is no such thing as an upper bound on delay (except as it is
enforced by dropping "late" packets), but there is a distribution.  So what
you really want to say is that, say, "95% of the packets must have a delay
of less than 300 milliseconds", recognizing that 5% of the packets may get
dropped if the buffer mechanism at the receiver imposes a consistent 300
millisecond delay.

This is what I mean by "jitter is important".  I would be equally satisfied
by saying "distribution of delay" is important,  But neither "average delay"
nor "maximum delay", by themselves, are sufficient criteria. 

-----Original Message-----
From: Scott Bradner [mailto:sob <at> harvard.edu]
Sent: Wednesday, October 08, 2003 6:54 PM
To: carlberg <at> g11.org.uk; ieprep <at> ietf.org; Gunn, Janet
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-06.txt

> Substance
> 1.2 third para
> I have always understood that jitter was more important than delay for
VoIP,
> but you do not mention it.

I'm not sure that jitter per se is a big issue - the distribution of 
delay is - most of the packets need to get to the other end within 
a reasonable limit - that can be because the basic delay is near the
limit and there is no jitter or the basic latency is less and the jitter
brings the delay to thye limit.

Scott
Scott Bradner | 9 Oct 2003 16:27
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RE: I-D ACTION:draft-ietf-ieprep-framework-06.txt

> So what
> you really want to say is that, say, "95% of the packets must have a delay
> of less than 300 milliseconds", recognizing that 5% of the packets may get
> dropped if the buffer mechanism at the receiver imposes a consistent 300
> millisecond delay.

that is just the right idea

Scott
Dvorak, Charles A (Chuck | 9 Oct 2003 07:37
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RE: I-D ACTION:draft-ietf-ieprep-framework-06.txt

Are you talking about random loss or bursty loss? (The former is a curiosity but only the latter is realistic)

-----Original Message-----
From: Andrew Thiessen [mailto:andrew <at> its.bldrdoc.gov]
Sent: Wednesday, October 08, 2003 7:24 PM
To: 'Scott Bradner'; carlberg <at> g11.org.uk; ieprep <at> ietf.org;
Janet.Gunn <at> DynCorp.com
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-06.txt

Right...

Jitter is important in that the size of the jitter buffer adds to the
overall mouth-to-ear delay and that an accurate jitter buffer size
selection is critical. If the jitter buffer size is smaller than the
delay variation on the network, voice traffic will necessarily be
dropped. If the jitter buffer is appropriately sized, the only impact
jitter should have is the delay that the buffer adds.

Whether or not a voice application can sustain a higher drop rate is
fairly dependent on the codec that is used. For instance, a G-711 codec
is more resilient to loss than a G-729 codec. The drop rate of a cloud
has a direct impact on the quality of the voice traffic. For instance,
we have lab tests that show with certain codecs that the estimated MOS
score drops by 1 for every 5% increase in loss.

Drop rates in elastic applications are typically more tolerant of drop
due somewhat to the protocols that are used. For instance, TCP has a
built in mechanism to retransmit lost packets, where UDP does not. Email
and web browsing would therefore be much more resilient to loss in that
a session with loss is easier to maintain with TCP than UDP. Would it
take forever to get email and see a web page with high loss? Yes, but
you would see it eventually. With VoIP, you may be able to maintain some
type of conversation, but the quality could be so bad that there
wouldn't be any point.

Andy
Gunn, Janet | 8 Oct 2003 17:02

RE: I-D ACTION:draft-ietf-ieprep-framework-06.txt

Comments on draft-ietf-ieprep-framework-06.txt
Comments on substance first, then editorial.

Substance
1.2 third para
I have always understood that jitter was more important than delay for VoIP,
but you do not mention it.

1.2 fourth para
This is stated as the opposite of what I have always understood.  I have
always seen it that VoIP applications can tolerate a higher drop rate, and
even a moderate delay, as long as jitter is very small.  Conversely elastic
applications such as email and web browsing can tolerate high jitter and
delay, but need a low drop rate (because in email and web browsing the
dropped packets need to be retransmitted, but in VoIP, if they are dropped
you forget about them).  Even a drop rate of "5%" is considered a very high
drop rate in a (data) world where the starting point for drop rates is 10 to
the minus 6.

4.4 
first para

The statement that "the user authenticates himself... using a PIN" is true
for GETS but not for WPS, nor for line authentication, nor for other related
programs, and therefore not for "PSTN" in general.  If you said "the user
authenticates himself as usual to the network" it would be a valid
statement.

Editorial

Section 1
First para
" Best effort ...infers..." should be "Best effort ... implies ..."
("infers" = "deduces")

Third para
Do you mean "stub domain" or "single domain"? The statement would seem to
also apply to a single domain "bridging" scenario.

1.2
first para
It would be clearer if you said "... though not necessarily Voice over IP
which is not IP telephony" , instead of just "... though not necessarily
Voice over IP

Section 3
Last line
Should be "violate", not "violet".

Section 4.1.5
MEGACO (RFC 3015) has been "obsoleted" by Gateway Control Protocol Version 1
(RFC 3525)  The statement that priority 0 is the lowest and priority 15 is
the highest is in 3525 but not in 3015. 

4.4 last para 
says "IEPS" rather than "ETS".  Is this intentional?

4.5 last para
"meetings" should be "meeting"

Section 5, text after figure 1
Do you mean "Grade of Service" (= blocking probability based on Erlang's
formulae) or "Quality of Service" (= jitter, delay, drop rate)?

Janet

-----Original Message-----
From: Ken Carlberg [mailto:carlberg <at> g11.org.uk]
Sent: Wednesday, October 08, 2003 8:31 AM
To: ieprep <at> ietf.org
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-06.txt

Barring objections from the group, I'd like to request that the chairs
announce a last call for this Telephony Framework draft.  I've purposely
delayed this request so that the requirements drafts could be advanced
up the standardization food chain.  With that process winding down, I'd
like to finish off the Framework draft.

Thanks,

-ken

_______________________________________________
Ieprep mailing list
Ieprep <at> ietf.org
https://www1.ietf.org/mailman/listinfo/ieprep
Ken Carlberg | 9 Oct 2003 16:00
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RE: I-D ACTION:draft-ietf-ieprep-framework-06.txt

> Substance
> 1.2 third para
> I have always understood that jitter was more important than delay for
> VoIP, but you do not mention it.

The section you refer to focuses on published material relating to VoIP,
neither of which focuses on jitter.  So its not so much a case of my
choosing to not discuss jitter, but that those papers did not.  If there
is a paper with some stats that makes an argument for what you state
above, please send it to me and I'll fold it in the section.

In the meantime, I can add a comment later on in the diff-serv section
that mentions the use of EF as a mechanism for addressing jitter.

> 1.2 fourth para
> This is stated as the opposite of what I have always understood.  I
have
> always seen it that VoIP applications can tolerate a higher drop rate,
and
> even a moderate delay, as long as jitter is very small.  Conversely
> elastic
> applications such as email and web browsing can tolerate high jitter
and
> delay, but need a low drop rate (because in email and web browsing the
> dropped packets need to be retransmitted, but in VoIP, if they are
dropped
> you forget about them).  Even a drop rate of "5%" is considered a very
> high drop rate in a (data) world where the starting point for drop
rates
> is 10 to the minus 6.

Again, if there is a paper/publication that you can cite, that would be
very helpful.  

Many times there are caveats to each position taken.  The use of FEC,
transcoding (see the NRL work that Ran mentioned awhile back),
interleaving (purposely placing packets out of order), etc., can improve
the perceived quality of a VoIP flow.  The lack of additional mechanisms
or functions in those citations is probably an obvious thing that I
should state in that section.

As for elastic applications, there are two things to focus on:
congestion collapse and buffer overflow.  I've seen stats of TCP flows
operating (barely) at sustained loss rates of ~40%.  Buffer overflow is
application specific and can cause connections to break at even modest
levels of loss (ie, before congestion collapse).  But to be fair, if I
can't find suitable published stats, I will alter/remove the 4'th
paragraph.

> 4.4
> first para
> 
> The statement that "the user authenticates himself... using a PIN" is
true
> for GETS but not for WPS, nor for line authentication, nor for other
> related
> programs, and therefore not for "PSTN" in general.  If you said "the
user
> authenticates himself as usual to the network" it would be a valid
> statement.

Ok, I'll remove the word "PIN".

> Editorial
> 
> Section 1
> First para
> " Best effort ...infers..." should be "Best effort ... implies ..."
> ("infers" = "deduces")

ok

> Third para
> Do you mean "stub domain" or "single domain"? The statement would seem
to
> also apply to a single domain "bridging" scenario.

I meant stub domains.  I'm not exactly sure what you mean by a 'single
domain "bridging" scenario'.  What is being bridged?

> 1.2
> first para
> It would be clearer if you said "... though not necessarily Voice over
IP
> which is not IP telephony" , instead of just "... though not
necessarily
> Voice over IP

I'll clarify the sentence

> Section 3
> Last line
> Should be "violate", not "violet".

Ok

> Section 4.1.5
> MEGACO (RFC 3015) has been "obsoleted" by Gateway Control Protocol
Version
> 1
> (RFC 3525)  The statement that priority 0 is the lowest and priority
15 is
> the highest is in 3525 but not in 3015.

ok

> 4.4 last para
> says "IEPS" rather than "ETS".  Is this intentional?

I'll switch it to ETS

> 4.5 last para
> "meetings" should be "meeting"

ok

> Section 5, text after figure 1
> Do you mean "Grade of Service" (= blocking probability based on
Erlang's
> formulae) or "Quality of Service" (= jitter, delay, drop rate)?

Should be QoS

-ken
Scott Bradner | 9 Oct 2003 00:53
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RE: I-D ACTION:draft-ietf-ieprep-framework-06.txt

> Substance
> 1.2 third para
> I have always understood that jitter was more important than delay for VoIP,
> but you do not mention it.

I'm not sure that jitter per se is a big issue - the distribution of 
delay is - most of the packets need to get to the other end within 
a reasonable limit - that can be because the basic delay is near the
limit and there is no jitter or the basic latency is less and the jitter
brings the delay to thye limit.

Scott
Andrew Thiessen | 9 Oct 2003 01:23

RE: I-D ACTION:draft-ietf-ieprep-framework-06.txt

Right...

Jitter is important in that the size of the jitter buffer adds to the
overall mouth-to-ear delay and that an accurate jitter buffer size
selection is critical. If the jitter buffer size is smaller than the
delay variation on the network, voice traffic will necessarily be
dropped. If the jitter buffer is appropriately sized, the only impact
jitter should have is the delay that the buffer adds.

Whether or not a voice application can sustain a higher drop rate is
fairly dependent on the codec that is used. For instance, a G-711 codec
is more resilient to loss than a G-729 codec. The drop rate of a cloud
has a direct impact on the quality of the voice traffic. For instance,
we have lab tests that show with certain codecs that the estimated MOS
score drops by 1 for every 5% increase in loss.

Drop rates in elastic applications are typically more tolerant of drop
due somewhat to the protocols that are used. For instance, TCP has a
built in mechanism to retransmit lost packets, where UDP does not. Email
and web browsing would therefore be much more resilient to loss in that
a session with loss is easier to maintain with TCP than UDP. Would it
take forever to get email and see a web page with high loss? Yes, but
you would see it eventually. With VoIP, you may be able to maintain some
type of conversation, but the quality could be so bad that there
wouldn't be any point.

Andy

-----Original Message-----
From: ieprep-admin <at> ietf.org [mailto:ieprep-admin <at> ietf.org] On Behalf Of
Scott Bradner
Sent: Wednesday, October 08, 2003 3:54 PM
To: carlberg <at> g11.org.uk; ieprep <at> ietf.org; Janet.Gunn <at> DynCorp.com
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-06.txt

> Substance
> 1.2 third para
> I have always understood that jitter was more important than delay for
VoIP,
> but you do not mention it.

I'm not sure that jitter per se is a big issue - the distribution of 
delay is - most of the packets need to get to the other end within 
a reasonable limit - that can be because the basic delay is near the
limit and there is no jitter or the basic latency is less and the jitter
brings the delay to thye limit.

Scott

_______________________________________________
Ieprep mailing list
Ieprep <at> ietf.org
https://www1.ietf.org/mailman/listinfo/ieprep
Gunn, Janet | 28 Apr 2003 20:04

RE: I-D ACTION:draft-ietf-ieprep-ets-telephony-03.txt

Two very minor comments.

In section 3, item 5, the first sentence says "best available" but the last
sentence says "better than best effort".  Is this intended, or is the second
one also supposed to be "best available"?

In section 5, the first sentence appears to be missing a ")" at the end.

-----Original Message-----
From: Internet-Drafts <at> ietf.org [mailto:Internet-Drafts <at> ietf.org]
Sent: Monday, April 21, 2003 7:32 AM
Cc: ieprep <at> ietf.org
Subject: [Ieprep] I-D ACTION:draft-ietf-ieprep-ets-telephony-03.txt

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

	Title		: IP Telephony Requirements for Emergency 
                          Telecommunication Service
	Author(s)	: K. Carlberg, R. Atkinson
	Filename	: draft-ietf-ieprep-ets-telephony-03.txt
	Pages		: 6
	Date		: 2003-4-18
	
This document presents a list of requirements in support of Emergency
Telecommunications Service (ETS) within the context of IP telephony.
It is an extension to the general requirements presented in [3].
Solutions to these requirements are not presented in this document.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-ieprep-ets-telephony-03.txt

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Ken Carlberg | 28 Apr 2003 20:47
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Re: I-D ACTION:draft-ietf-ieprep-ets-telephony-03.txt


> In section 3, item 5, the first sentence says "best available" but the last
> sentence says "better than best effort".  Is this intended, or is the second
> one also supposed to be "best available"?

yes, the sentence as it is written in version 3 is as intended.

> In section 5, the first sentence appears to be missing a ")" at the end.

it has been fixed.

regards,

-ken

ps, I've included the I-D editor on this email to close the loop, but
    I don't believe the address should be included again
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RE: I-D ACTION:draft-ietf-ieprep-framework-03.txt


> > What Henning didn't say is that he and James have ANOTHER document
> > (currently in the SIP working group) which DOES address the Resource
> > Priority Header, and should probably be referenced in the document.
> >
> > http://www.ietf.org/internet-drafts/draft-polk-sip-resource-02.txt
> 
> we don't want to reference the above draft because its still a work in
> progress.  

but, your reference cites many other work-in-progress drafts ( some even
expired.) For example, the Classifier Extension Header for RTP. I don't 
know which WG is discussing that draft. So, I see no harm in referring
both the requirements (RFC3487) and the actual priority header draft.
Ken Carlberg | 20 Mar 2003 01:34
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Re: I-D ACTION:draft-ietf-ieprep-framework-03.txt


> > we don't want to reference the above draft because its still a work in
> > progress.
> 
> but, your reference cites many other work-in-progress drafts ( some even
> expired.) For example, the Classifier Extension Header for RTP. I don't
> know which WG is discussing that draft. So, I see no harm in referring
> both the requirements (RFC3487) and the actual priority header draft.

my mistake for not finishing the original sentence above.  I should have
added that RFC-3487 is a sufficient reference for the original passage
in the Framework document (though i'll probably slightly alter that
passage shortly to make sure of this).  given a choice between a draft
and an RFC as a reference point, its best to choose the RFC.

as for the other drafts referenced, one is allowed to cite them within
constraints.  I'd like to replace those in the draft where possible, and
in fact two will be replaced by new RFCs (eg, 3487 :-) that came out
right after I submitted version of the Framework draft.  hopefully, in
short period, two others will also be replaced by RFCs.

-ken
Gunn, Janet | 19 Mar 2003 22:51

RE: I-D ACTION:draft-ietf-ieprep-framework-03.txt

What Henning didn't say is that he and James have ANOTHER document
(currently in the SIP working group) which DOES address the Resource
Priority Header, and should probably be referenced in the document.

http://www.ietf.org/internet-drafts/draft-polk-sip-resource-02.txt

 Communications Resource Priority for the Session Initiation
                             Protocol (SIP)

-----Original Message-----
From: Henning Schulzrinne [mailto:hgs <at> cs.columbia.edu]
Sent: Monday, February 17, 2003 1:51 PM
To: Gunn, Janet
Cc: 'Ken Carlberg'; ieprep <at> ietf.org
Subject: Re: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-03.txt

Indeed. The requirements draft takes great pains to be solution-neutral 
and thus does not address a specific mechanism. In reality, the number 
of possibilities aren't infinite, given the constraints of the protocol. 
There will shortly be a solution-oriented draft that does indeed propose 
a new header field, after trying to exhaustively go through all the 
design options and identifying why no other solution satisfies the 
requirements stated. As always, I might have missed possibilities or 
solutions, but I'm sure that participants in this and SIP-related 
working groups won't be shy in pointing out any such omission.

In short, Janet is correct.

Gunn, Janet wrote:
> Maybe Henning can clarify, but the current version of
> draft-ietf-ieprep-sip-reqs-03.txt says
> "This document describes requirements rather than possible existing or
>    new protocol features. Although it is scoped to deal with SIP-based
>    applications, this should not be taken to imply that mechanisms have
>    to be SIP protocol features such as header fields, methods or URI
>    parameters."
> 
> It presents requirements for which a new header field is the "obvious"
> solution, but no longer states it as a requirement FRO a header field.
> 
> -----Original Message-----
> From: Ken Carlberg [mailto:K.Carlberg <at> cs.ucl.ac.uk]
> 
> 
>>Pg 11 reference to Henning's document - it no longer refers explicitly
>>to headers.
> 
> 
> the original passage is:
> 
>    [15] is a (soon to be) RFC that defines the requirements for a new
>    header field for SIP in reference to resource priority.  This new
>    header field is meant to provide an additional measure of distinction
>    that can influence the behavior of gateways and SIP proxies.
> 
> my understanding is that it pertains to requirements for a header *field*.

> so yes, we are in agreement.
> _______________________________________________
> Ieprep mailing list
> Ieprep <at> ietf.org
> https://www1.ietf.org/mailman/listinfo/ieprep
Ken Carlberg | 20 Mar 2003 00:53
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Re: I-D ACTION:draft-ietf-ieprep-framework-03.txt


> What Henning didn't say is that he and James have ANOTHER document
> (currently in the SIP working group) which DOES address the Resource
> Priority Header, and should probably be referenced in the document.
>
> http://www.ietf.org/internet-drafts/draft-polk-sip-resource-02.txt

we don't want to reference the above draft because its still a work in
progress.  its best to focus on RFC3487, "Requirements for Resource
Priority Mechanisms for the Session Initiation Protocol (SIP)"

-ken
Gunn, Janet | 19 Feb 2003 16:08

RE: I-D ACTION:draft-ietf-ieprep-framework-03.txt

That is fine with me.  I just don't want to require it.

-----Original Message-----
From: Nguyen, An [mailto:nguyena <at> ncs.gov]
Sent: Tuesday, February 18, 2003 9:23 PM
To: Gunn, Janet; 'Ian Brown'
Cc: 'ieprep <at> ietf.org'
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-03.txt

Janet,

I think centralized authentication is needed for authorized emergency
services. It is not applicable to GETS and WPS now. 
But it should be in the future. Anyhow, the wordings in the draft should
also allow room for centralized authentication implementation when needed.

An

-----Original Message-----
From: Gunn, Janet [mailto:Janet.Gunn <at> DynCorp.com]
Sent: Tuesday, February 18, 2003 5:23 PM
To: 'Nguyen, An'; 'Ian Brown'
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-03.txt

Nothing wrong with it- I just don't think should be assumed.

-----Original Message-----
From: Nguyen, An [mailto:nguyena <at> ncs.gov]
Sent: Tuesday, February 18, 2003 11:57 AM
To: 'Ian Brown'; Gunn, Janet
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-03.txt

Janet,

I am curious ... What is the wrong with centralized authentication? We will
have a scenario where SIP calls are from one end to other. GETS and WPS
might not be part of those calls at all.

An

-----Original Message-----
From: Ian Brown [mailto:I.Brown <at> cs.ucl.ac.uk]
Sent: Tuesday, February 18, 2003 8:20 AM
To: 'Gunn, Janet'
Cc: ieprep <at> ietf.org
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-03.txt

Janet wrote:
> The usage "authentication center" implies to me that 
> authentication is centralized.

I'll update this to remove the confusion.

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Ieprep <at> ietf.org
https://www1.ietf.org/mailman/listinfo/ieprep
Nguyen, An | 19 Feb 2003 03:23

RE: I-D ACTION:draft-ietf-ieprep-framework-03.txt

Janet,

I think centralized authentication is needed for authorized emergency
services. It is not applicable to GETS and WPS now. 
But it should be in the future. Anyhow, the wordings in the draft should
also allow room for centralized authentication implementation when needed.

An

-----Original Message-----
From: Gunn, Janet [mailto:Janet.Gunn <at> DynCorp.com]
Sent: Tuesday, February 18, 2003 5:23 PM
To: 'Nguyen, An'; 'Ian Brown'
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-03.txt

Nothing wrong with it- I just don't think should be assumed.

-----Original Message-----
From: Nguyen, An [mailto:nguyena <at> ncs.gov]
Sent: Tuesday, February 18, 2003 11:57 AM
To: 'Ian Brown'; Gunn, Janet
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-03.txt

Janet,

I am curious ... What is the wrong with centralized authentication? We will
have a scenario where SIP calls are from one end to other. GETS and WPS
might not be part of those calls at all.

An

-----Original Message-----
From: Ian Brown [mailto:I.Brown <at> cs.ucl.ac.uk]
Sent: Tuesday, February 18, 2003 8:20 AM
To: 'Gunn, Janet'
Cc: ieprep <at> ietf.org
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-03.txt

Janet wrote:
> The usage "authentication center" implies to me that 
> authentication is centralized.

I'll update this to remove the confusion.

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Ieprep <at> ietf.org
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Scott Bradner | 19 Feb 2003 17:07
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RE: I-D ACTION:draft-ietf-ieprep-framework-03.txt

> I think centralized authentication is needed for authorized emergency
> services.

I find it hard to imagine a single centeralized authentication system
to deal with emergency services on the international Internet

Scott
Gunn, Janet | 17 Feb 2003 20:16

RE: I-D ACTION:draft-ietf-ieprep-framework-03.txt

That's OK with me.  It's usage, not content.

-----Original Message-----
From: Ken Carlberg [mailto:K.Carlberg <at> cs.ucl.ac.uk]
Sent: Monday, February 17, 2003 2:03 PM
To: Gunn, Janet
Cc: ieprep <at> ietf.org
Subject: Re: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-03.txt

> My point is that "probability" is an INHERENTLY quantitative concept...

i think we talking about an ant hill in the scheme of things.  i'll just
reword the paragraph.  and i'll wait until tomorrow for any additional
messages on the same subject.

-ken
Gunn, Janet | 17 Feb 2003 19:48

RE: I-D ACTION:draft-ietf-ieprep-framework-03.txt

I was referring to this sentence
"In an
   IP network, the authentication center will need to securely signal
   back to the IP ingress point that a given user is authorized to send
   ETS related flows. "

The usage "authentication center" implies to me that authentication is
centralized.

-----Original Message-----
From: Ken Carlberg [mailto:K.Carlberg <at> cs.ucl.ac.uk]

> Pg 8
> Authentication is not necessarily centralized.

page 8 doesn't have any word on authentication. is there a specific
phrase or sentence that you were concerned about?  your statement is
a bit general and I'd rather not read things into that may not be
there.

>
Ian Brown | 18 Feb 2003 14:19
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RE: I-D ACTION:draft-ietf-ieprep-framework-03.txt

Janet wrote:
> The usage "authentication center" implies to me that 
> authentication is centralized.

I'll update this to remove the confusion.
Ken Carlberg | 17 Feb 2003 20:03
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Re: I-D ACTION:draft-ietf-ieprep-framework-03.txt


> My point is that "probability" is an INHERENTLY quantitative concept...

i think we talking about an ant hill in the scheme of things.  i'll just
reword the paragraph.  and i'll wait until tomorrow for any additional
messages on the same subject.

-ken
Gunn, Janet | 17 Feb 2003 19:43

RE: I-D ACTION:draft-ietf-ieprep-framework-03.txt

My point is that "probability" is an INHERENTLY quantitative concept.  Maybe
it is "where I am coming from" but, to me, saying that "probability cannot
be easily quantified" is as grating as saying "number cannot easily be
quantified".  It may be very difficult to know, or even estimate, for
instance, the NUMBER of grains of sand on the beach.  But it is still
INHERENTLY quantifiable.

I think part of the problem may be in applying "probability" to delay, loss,
and jitter.

The probability of loss is certainly well defined and measurable.  But when
you get to jitter and delay, you can only talk about the probability that it
is within defined bounds. It is awkward to talk about "probability of delay"
or "probability of jitter" because that probability is 1, there is ALWAYS
some delay and some jitter.

I will remain uncomfortable with this usage, but it IS a usage issue, not a
content issue.
-----Original Message-----
From: Ken Carlberg [mailto:K.Carlberg <at> cs.ucl.ac.uk]

> I don't agree with the statement "... probability ... cannot easily be
> quantified."  I quantify probability all the time.

the meaning statement is within a larger context as stated below:

   The critical word in this objective
   is "probability", as opposed to assurance or guarantee -- the latter
   two placing a higher burden on the network.  It stands to reason,
   though, that the word "probability" is a less tangible description
   that cannot be easily quantified.  It is relative in relation to
   other traffic transiting the same network.

I'm not sure its fair to cut out so much of what is stated and expect
the parts to stand on its own.  as I stated in the comments on the
General Requirements, aspects such as delay, loss, and jitter can be
(within limitations) addressed by the network.  so if one is going to
focus on probability, which of those three does one concentrate on?  to
me, that not an easy question to make a definitive statement about. 
Gunn, Janet | 17 Feb 2003 19:31

RE: I-D ACTION:draft-ietf-ieprep-framework-03.txt

Maybe Henning can clarify, but the current version of
draft-ietf-ieprep-sip-reqs-03.txt says
"This document describes requirements rather than possible existing or
   new protocol features. Although it is scoped to deal with SIP-based
   applications, this should not be taken to imply that mechanisms have
   to be SIP protocol features such as header fields, methods or URI
   parameters."

It presents requirements for which a new header field is the "obvious"
solution, but no longer states it as a requirement FRO a header field.

-----Original Message-----
From: Ken Carlberg [mailto:K.Carlberg <at> cs.ucl.ac.uk]

> Pg 11 reference to Henning's document - it no longer refers explicitly
> to headers.

the original passage is:

   [15] is a (soon to be) RFC that defines the requirements for a new
   header field for SIP in reference to resource priority.  This new
   header field is meant to provide an additional measure of distinction
   that can influence the behavior of gateways and SIP proxies.

my understanding is that it pertains to requirements for a header *field*.  
so yes, we are in agreement.
Henning Schulzrinne | 17 Feb 2003 19:51

Re: I-D ACTION:draft-ietf-ieprep-framework-03.txt

Indeed. The requirements draft takes great pains to be solution-neutral 
and thus does not address a specific mechanism. In reality, the number 
of possibilities aren't infinite, given the constraints of the protocol. 
There will shortly be a solution-oriented draft that does indeed propose 
a new header field, after trying to exhaustively go through all the 
design options and identifying why no other solution satisfies the 
requirements stated. As always, I might have missed possibilities or 
solutions, but I'm sure that participants in this and SIP-related 
working groups won't be shy in pointing out any such omission.

In short, Janet is correct.

Gunn, Janet wrote:
> Maybe Henning can clarify, but the current version of
> draft-ietf-ieprep-sip-reqs-03.txt says
> "This document describes requirements rather than possible existing or
>    new protocol features. Although it is scoped to deal with SIP-based
>    applications, this should not be taken to imply that mechanisms have
>    to be SIP protocol features such as header fields, methods or URI
>    parameters."
> 
> It presents requirements for which a new header field is the "obvious"
> solution, but no longer states it as a requirement FRO a header field.
> 
> -----Original Message-----
> From: Ken Carlberg [mailto:K.Carlberg <at> cs.ucl.ac.uk]
> 
> 
>>Pg 11 reference to Henning's document - it no longer refers explicitly
>>to headers.
> 
> 
> the original passage is:
> 
>    [15] is a (soon to be) RFC that defines the requirements for a new
>    header field for SIP in reference to resource priority.  This new
>    header field is meant to provide an additional measure of distinction
>    that can influence the behavior of gateways and SIP proxies.
> 
> my understanding is that it pertains to requirements for a header *field*.  
> so yes, we are in agreement.
> _______________________________________________
> Ieprep mailing list
> Ieprep <at> ietf.org
> https://www1.ietf.org/mailman/listinfo/ieprep
Gunn, Janet | 17 Feb 2003 19:25

RE: I-D ACTION:draft-ietf-ieprep-framework-03.txt

Point taken- but the current wording "...interactive voice has a lower
threshold of loss than
>> elastic applications..."implies that ALL interactive voice is elastic.

-----Original Message-----
From: RJ Atkinson [mailto:rja <at> extremenetworks.com]
Sent: Sunday, February 16, 2003 10:44 AM
To: Ken Carlberg
Cc: ieprep
Subject: Re: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-03.txt

On Saturday, Feb 15, 2003, at 15:27 America/Montreal, Ken Carlberg 
wrote:
>> 1.2
>> Should say "...interactive voice has a lower threshold of loss than
>> elastic applications..." NOT "...than other elastic applications..."
>> Interactive voice is NOT an elastic application.
>
> the change will be made.

I disagree with the claim that "interactive voice is NOT an elastic
application".  There are deployed VoIP systems (e.g. in production use
by US DoD) that provide an existence proof that VoIP can be an elastic
application.  Those systems shift from higher-bandwidth codecs to
lower-bandwidth codecs in response to detected congestion on access 
links.
That doesn't mean that all VoIP deployments have that property,
but clearly *some* definitely do today (and have done for a few years 
now).

>> What do you mean by "carriers"?
>
> telco carriers.  i've not come across the term carriers in association
> with ISPs.  however, i shall extend the term so that it will read as
> "telecom carriers"

Would be better talk about "telephony carriers" or "IP carriers"
to reduce ambiguity.  "telecom carriers" is an ambiguous term,
unfortunately.

Ran

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Nguyen, An | 15 Feb 2003 00:58

RE: I-D ACTION:draft-ietf-ieprep-framework-03.txt


-----Original Message-----
From: Gunn, Janet [mailto:Janet.Gunn <at> DynCorp.com]
Sent: Friday, February 14, 2003 4:06 PM
To: ieprep <at> ietf.org
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-03.txt

>1.1.1	
>NCS is now part of the Department of Homeland Security.
>
Not officially until March 1. The cut-off date for WG last call is Feb 24. 
Technically, it should still be NCS. But I think we should add a sentence in
the paragraph to link the NCS to the Department of Homeland security. 

An
Gunn, Janet | 14 Feb 2003 22:05

RE: I-D ACTION:draft-ietf-ieprep-framework-03.txt


Most of these comments are editorial rather than content-based.

Abstract
Is there a definition of "stub domain"?

1.1.1	
NCS is now part of the Department of Homeland Security.

The non governmental participants in GETS are a LOT more than "a select set
of individuals"  I THINK the GETS users are split roughly 50/50 between
government and non government users.

1.2
Should say "...interactive voice has a lower threshold of loss than elastic
applications..."
NOT
"...than other elastic applications..."  Interactive voice is NOT an elastic
application.

What do you mean by "carriers"?

"...some of the protocols of discussed in section 4..." 
should be
"...some of the protocols discussed in section 4..." 

Section 2, pg 6
When talking about "higher probability" it would be helpful to define
"higher probability than WHAT".  I am assuming  it means "higher probability
than for an equivalent  non-emergency call".

I don't agree with the statement  "... probability ... cannot easily be
quantified."  I quantify probability all the time.

Some kind of weird formatting "=A0" before "Hence, ..."

Pg 7
"More
   specifically, we associate this objective in the context of IP
   telephony acting as part of the Public Telephone Network (PTN).
   This, as opposed to the use of IP telephony within a private or stub
   network."
Might want to put this in the scope section.

"a higher probability
   of call completion than that of normal IP telephony call traffic." Might
want to say this earlier.

Pg 8
Authentication is not necessarily centralized.

Section 3- First sentence needs to be reworked to reflect that you now have
TWO (not just ONE) objectives.

Pg 11 reference to Henning's document - it no longer refers explicitly to
headers.

Pg 12 Here you introduce MLPP.  Would it make sense to also discuss it in
section 1.1?

Pg 15.  This statement is NOT true:
"As a point of reference, existing SLAs established by the NCS
   for GETS service tend to focus on a maximum allocation of (e.g., 1%)
   of calls allowed to be established through a given LEC using HPC.
   Once this limit is reached, all other GETS calls experience the same
   probability of call completion as the general public.  "

GETS is ENGINEERED for 1% GETS calls, and some of the SLAs refer to such a
limit, but there is NO real time MECHANISM to track GETS usage, and NO real
time MECHANISM to change the treatment of GETS calls.

In WPS (the wireless extension to GETS) there IS a traffic threshold (25%
WPS in a single cell) beyond which the priority treatment for radio channels
changes.  But NOT in GETS.

Under "traffic engineering", might it make sense to discuss the Sprint and
Genuity access link traffic engineering approaches described in Atlanta.

Pg 16.

In general, PSTN authentication does NOT need to be PIN based.  WPS
authentication is not PIN based, and, for GSM, not centralized.  In
discussing generic PSTN based priority systems, it would be helpful to
describe them in a way which encompasses both WPS AND GETS. (WPS is now
operational, at least as "IOC phase 1".)

Section 5 Key Scenarios

"imply", not "infer"

Pg 18 
"The fact that
   each administrative domain peers with each other as part of the PSTN,
   means that existing security, in the form of Personal Identification
   Number (PIN) authentication (under the context of telephony
infrastructure protection), is the default scope of security. "

No longer true.  WPS authentication "exists" (went live Dec 22), and is NOT
PIN based.  To describe PINs as the "default mechanism" WOULD place
"additional requirements on existing authorized emergency telephony
systems."

Page 22

Under standards work, you might want to make reference to the changes to the
CDMA standards to support emergency traffic.  Like the CPC, there MAY be a
need to translate these new parameters into SIP, etc.

Janet
-----Original Message-----
From: Internet-Drafts <at> ietf.org [mailto:Internet-Drafts <at> ietf.org]
Sent: Tuesday, January 21, 2003 7:52 AM
Cc: ieprep <at> ietf.org
Subject: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-03.txt

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

	Title		: Framework for Supporting IEPS in IP Telephony
	Author(s)	: K. Carlberg, I. Brown
	Filename	: draft-ietf-ieprep-framework-03.txt
	Pages		: 27
	Date		: 2003-1-20
	
This document presents a framework for supporting authorized
emergency related communication within the context of IP telephony.
We present a series of objectives that reflect a general view of how
authorized emergency service, in line with the Emergency
Telecommunications Service (ETS), should be realized within today's
IP architecture and service models.  From these objectives, we
present a corresponding set of protocols and capabilities, which
provide a more specific set of recommendations regarding existing
IETF protocols.  Finally, we present two scenarios that act as
guiding models for the objectives and functions listed in this
document.  These, models, coupled with an example of an existing
service in the PSTN, contribute to a constrained solution space.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-ieprep-framework-03.txt

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Ken Carlberg | 15 Feb 2003 21:27
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Re: I-D ACTION:draft-ietf-ieprep-framework-03.txt


> Abstract
> Is there a definition of "stub domain"?

definitions and references can be found in rfc 1631, 3022, 1322, 1364,
1477, 1478, 1745, and some others

> 1.1.1	
> NCS is now part of the Department of Homeland Security.

I'll note this in the next version of the draft

> The non governmental participants in GETS are a LOT more than "a select
> set of individuals" I THINK the GETS users are split roughly 50/50
> between government and non government users.

i think "select set of individuals" is fine.  nothing breaks with the
phrase and it doesn't relate to size.  but the point is taken that the
system is not just a collections of users from the govt.

> 1.2
> Should say "...interactive voice has a lower threshold of loss than
> elastic applications..." NOT "...than other elastic applications..."
> Interactive voice is NOT an elastic application.

the change will be made.

> What do you mean by "carriers"?

telco carriers.  i've not come across the term carriers in association
with ISPs.  however, i shall extend the term so that it will read as
"telecom carriers"

> "...some of the protocols of discussed in section 4..." 
> should be
> "...some of the protocols discussed in section 4..." 

the change will be made.

> Section 2, pg 6
> 
> When talking about "higher probability" it would be helpful to define
> "higher probability than WHAT".  I am assuming it means "higher
> probability than for an equivalent non-emergency call".

ok.  additional verbage will be added to provide a point of comparison.

> I don't agree with the statement "... probability ... cannot easily be
> quantified."  I quantify probability all the time.

the meaning statement is within a larger context as stated below:

   The critical word in this objective
   is "probability", as opposed to assurance or guarantee -- the latter
   two placing a higher burden on the network.  It stands to reason,
   though, that the word "probability" is a less tangible description
   that cannot be easily quantified.  It is relative in relation to
   other traffic transiting the same network.

I'm not sure its fair to cut out so much of what is stated and expect
the parts to stand on its own.  as I stated in the comments on the
General Requirements, aspects such as delay, loss, and jitter can be
(within limitations) addressed by the network.  so if one is going to
focus on probability, which of those three does one concentrate on?  to
me, that not an easy question to make a definitive statement about. 

> Some kind of weird formatting "=A0" before "Hence, ..."

its a problem i have with NROFF.  i'll fix it in the next release.

> Pg 7
> "More
>   specifically, we associate this objective in the context of IP
>   telephony acting as part of the Public Telephone Network (PTN).
>   This, as opposed to the use of IP telephony within a private or stub
>   network."
>
> Might want to put this in the scope section.

noted.

> "a higher probability
>    of call completion than that of normal IP telephony call traffic." 
> 
> Might want to say this earlier.

noted

> Pg 8
> Authentication is not necessarily centralized.

page 8 doesn't have any word on authentication. is there a specific
phrase or sentence that you were concerned about?  your statement is
a bit general and I'd rather not read things into that may not be
there.

> Section 3- First sentence needs to be reworked to reflect that you now
> have TWO (not just ONE) objectives.

ok, the change will be made.

> Pg 11 reference to Henning's document - it no longer refers explicitly
> to headers.

the original passage is:

   [15] is a (soon to be) RFC that defines the requirements for a new
   header field for SIP in reference to resource priority.  This new
   header field is meant to provide an additional measure of distinction
   that can influence the behavior of gateways and SIP proxies.

my understanding is that it pertains to requirements for a header *field*.  
so yes, we are in agreement.

> Pg 12 Here you introduce MLPP.  Would it make sense to also discuss it
> in section 1.1?

i'll give it some thought. 

> Pg 15.  This statement is NOT true:
> "As a point of reference, existing SLAs established by the NCS
>   for GETS service tend to focus on a maximum allocation of (e.g., 1%)
>   of calls allowed to be established through a given LEC using HPC.
>   Once this limit is reached, all other GETS calls experience the same
>   probability of call completion as the general public.  "
>
> GETS is ENGINEERED for 1% GETS calls, and some of the SLAs refer to such
> a limit, but there is NO real time MECHANISM to track GETS usage, and NO
> real time MECHANISM to change the treatment of GETS calls.

you've totally lost me.  if you claim that my statement of "SLAs...focus
on a maximum allocation" is incorrect, how can you then state that "SLAs
refer to such a limit"?  are the words "focus" and "refer" that different?

we have also had private discussions of how some vendors provide limits
on trunk queuing used for GETS calls.  is that not a real-time
functionality?  usage of a service, and in particular its individual
components, can be in several forms.  my original passage does not refer
to a specific form nor where limits exist.

> In WPS (the wireless extension to GETS) there IS a traffic threshold
> (25% WPS in a single cell) beyond which the priority treatment for radio
> channels changes.  But NOT in GETS.

ok.  but as stated above, i did not refer to a specific form of
threshold or mechanism used to achieve it int he original passage.

> Under "traffic engineering", might it make sense to discuss the Sprint
> and Genuity access link traffic engineering approaches described in
> Atlanta.

I'll review what was presented to see what can be added to the TE section.

> Pg 16.
> 
> In general, PSTN authentication does NOT need to be PIN based.  WPS
> authentication is not PIN based, and, for GSM, not centralized.  In
> discussing generic PSTN based priority systems, it would be helpful to
> describe them in a way which encompasses both WPS AND GETS. (WPS is now
> operational, at least as "IOC phase 1".)

i'll defer my response to Ian on this.

> Section 5 Key Scenarios
> 
> "imply", not "infer"

ok

> Pg 18 
> "The fact that
>   each administrative domain peers with each other as part of the PSTN,
>   means that existing security, in the form of Personal Identification
>   Number (PIN) authentication (under the context of telephony 
> infrastructure protection), is the default scope of security. "
> 
> No longer true.  WPS authentication "exists" (went live Dec 22), and is
> NOT PIN based.  To describe PINs as the "default mechanism" WOULD place
> "additional requirements on existing authorized emergency telephony
> systems."

understood about WPS and its use of the unique SIM identifier.  again,
i'll defer to Ian on the subject of security.  however, i have a
question about the subject of PINs.  Is there a general term that
encompasses the use of PINs (the numbers that people use to identify use
of service) and unique identifiers used to phone SIMs?  in my ignorance,
I thought that these two unique identifiers could be termed as PINs.

> Page 22
> 
> Under standards work, you might want to make reference to the changes to
> the CDMA standards to support emergency traffic.  Like the CPC, there
> MAY be a need to translate these new parameters into SIP, etc.

if we add info into the body of the document refering to CDMA standards,
then we'll certainly add the corresponding reference.

regards,

-ken
RJ Atkinson | 16 Feb 2003 16:44
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Re: I-D ACTION:draft-ietf-ieprep-framework-03.txt


On Saturday, Feb 15, 2003, at 15:27 America/Montreal, Ken Carlberg 
wrote:
>> 1.2
>> Should say "...interactive voice has a lower threshold of loss than
>> elastic applications..." NOT "...than other elastic applications..."
>> Interactive voice is NOT an elastic application.
>
> the change will be made.

I disagree with the claim that "interactive voice is NOT an elastic
application".  There are deployed VoIP systems (e.g. in production use
by US DoD) that provide an existence proof that VoIP can be an elastic
application.  Those systems shift from higher-bandwidth codecs to
lower-bandwidth codecs in response to detected congestion on access 
links.
That doesn't mean that all VoIP deployments have that property,
but clearly *some* definitely do today (and have done for a few years 
now).

>> What do you mean by "carriers"?
>
> telco carriers.  i've not come across the term carriers in association
> with ISPs.  however, i shall extend the term so that it will read as
> "telecom carriers"

Would be better talk about "telephony carriers" or "IP carriers"
to reduce ambiguity.  "telecom carriers" is an ambiguous term,
unfortunately.

Ran
Ken Carlberg | 17 Feb 2003 14:27
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Re: I-D ACTION:draft-ietf-ieprep-framework-03.txt


> I disagree with the claim that "interactive voice is NOT an elastic
> application".  There are deployed VoIP systems (e.g. in production use
> by US DoD) that provide an existence proof that VoIP can be an elastic
> application.  Those systems shift from higher-bandwidth codecs to
> lower-bandwidth codecs in response to detected congestion on access
> links.
>
> That doesn't mean that all VoIP deployments have that property, but
> clearly *some* definitely do today (and have done for a few years now).

yes, good point.  and we also have potential feedback from RTCP (when
its not being blocked).  its ia case of all TCP apps are elastic, and
UDP based apps have a tendency not to be -- its up to the developer to
add it.  I'll make this distinction clearer in the next draft.

> Would be better talk about "telephony carriers" or "IP carriers" to
> reduce ambiguity.  "telecom carriers" is an ambiguous term,
> unfortunately.

ok, i'll expand the respective terms for clarity

-ken
Ian Brown | 16 Feb 2003 14:55
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RE: I-D ACTION:draft-ietf-ieprep-framework-03.txt

Janet wrote:
>In general, PSTN authentication does NOT need to be PIN based.  WPS 
>authentication is not PIN based, and, for GSM, not centralized.  In 
>discussing generic PSTN based priority systems, it would be helpful to 
>describe them in a way which encompasses both WPS AND GETS. (WPS is 
>now operational, at least as "IOC phase 1".)

OK -- I'll make a couple of updates in the next version.

Thanks,
Ian.
RJ Atkinson | 15 Feb 2003 01:11
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Re: I-D ACTION:draft-ietf-ieprep-framework-03.txt


On Friday, Feb 14, 2003, at 16:05 America/Montreal, Gunn, Janet wrote:
> 1.1.1	
> NCS is now part of the Department of Homeland Security.
>
> The non governmental participants in GETS are a LOT more than "a 
> select set
> of individuals"  I THINK the GETS users are split roughly 50/50 between
> government and non government users.

Relative to the total US telephone-accessible population, the total
number of authorised GETS users is *TINY*.

So "select set of individuals" seems indisputably accurate, regardless
of what percentage are govt/non-govt.

Ran

Gmane