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Message-ID: <3FF96823.5080608@neomagic.com>
Date: Mon, 05 Jan 2004 19:05:31 +0530
From: B Prashanth <bprashanth@neomagic.com>
Reply-To: B Prashanth <bprashanth@neomagic.com>
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Subject: [AVT] AMR Profile
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Dear All,
Can anyone please tell me that if RTP stack is sending AMR speech data 
then does the encoded AMR payload, i.e. the AMR speech frame, contain 
the AMR frame header (specifying the Frame type) or Does the AMR speech 
frame contain just the raw, encoded speech data in the form of Class A, 
B and C bits?
Please note that the header being talked about here is the AMR speech 
frame's header and not the RTP's AMR profile header (Specified in RFC 3267).

-- 
Prashanth B.P.
Sr. Design Engr.
NeoMagic Design Center.
Polyplex Building, First Floor
B-37 Sector 1
Noida 
INDIA 201301
E-Mail:  bprashanth@neomagic.com
Tel:     +91-120-2536205/6  Extn:4103
Fax:     +91-120-2536207

NeoMagic Corporation, based in Santa Clara, California, provides Applications Processors to enable new generations of handheld systems, offering the lowest power, smallest form-factor, and best multimedia features and performance. The company has pioneered the integration of memory, complex logic, and analog circuits into single-chip solutions. 
URL: http://www.neomagic.com.
--



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Jan  7 16:33:46 2004
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Subject: [AVT] RTP MIDI ...
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Hi everyone,

 From Colin's IETF 58 minutes, on RTP MIDI:

> John Lazzaro plans to ensure he has finished his implementation
> before requesting working group last call to catch any final bugs in
> the specification. However, he expects to have finished this by the
> end of the year.

Came close, but not quite done yet -- on the bright side, running C
now exists for all recovery journal chapters (channel and system),
with most of the bitfield format features exercised (some were
just too painful to fit into sfront's code base), and no major issues
were found.  Many smaller issues were caught --"if you do not test it,
it does not work", as they say in the chip design world.

A few more weeks of continuity work is needed -- going back and
forth between reading 40K lines of C code and 160 pages of I-D, to
see if any issues remain that fell through the cracks ...

---
John Lazzaro
http://www.cs.berkeley.edu/~lazzaro
lazzaro [at] cs [dot] berkeley [dot] edu
---


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Jan  7 18:27:32 2004
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Thread-Topic: I-D ACTION:draft-ash-avt-ecrtp-over-mpls-reqs-00.txt
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From: "Ash, Gerald R (Jerry), ALABS" <gash@att.com>
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Subject: [AVT] RE: I-D ACTION:draft-ash-avt-ecrtp-over-mpls-reqs-00.txt
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All,

Please review and comment on the current draft of 'Requirements for =
ECRTP over MPLS' =
http://www.ietf.org/internet-drafts/draft-ash-avt-ecrtp-over-mpls-reqs-00=
.txt.

As agreed at IETF-58 and on the list, the main changes are as follows:

1. focus on requirements for ECRTP over MPLS (defer other requirements =
until later).
2. eliminate MPLS jargon such as CE, PE, etc., just talk about routers =
R1, R2, etc.
3. eliminate absolute performance objectives and simply state the =
requirement to 'minimize delta degradation'.
4. eliminate absolute scalability requirements, and simply state the =
goal to improve scalability of the number of compressed flows that can =
be handled.

There was considerable discussion of the previous draft at IETF-58 (see =
the AVT meeting minutes at =
http://ietf.org/proceedings/03nov/index.html), but not much discussion =
on the list.

Let's get the discussion going on the list so we can converge on this =
draft prior to IETF-59 (week of March 1), and meet the AVT charter item =
milestone:

Mar 2004 Finish requirements for ECRTP over MPLS; re-charter for =
subsequent work

Thanks,
Jerry

-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: Wednesday, January 07, 2004 4:01 PM
Subject: I-D ACTION:draft-ash-avt-ecrtp-over-mpls-reqs-00.txt

A New Internet-Draft is available from the on-line Internet-Drafts =
directories.


	Title		: Requirements for ECRTP over MPLS
	Author(s)	: J. Ash
	Filename	: draft-ash-avt-ecrtp-over-mpls-reqs-00.txt
	Pages		: 0
	Date		: 2004-1-8
=09
VoIP typically uses the encapsulation voice/RTP/UDP/IP.  When MPLS=20
labels are added, this becomes voice/RTP/UDP/IP/MPLS-labels.  For an=20
MPLS VPN, the packet header is at least 48 bytes, while the voice=20
payload is often no more than 30 bytes, for example.  VoIP header=20
compression can significantly reduce the VoIP overhead through various=20
compression mechanisms, such as enhanced compressed RTP (ECRTP).  This=20
is important on access links where bandwidth is scarce, and can be=20
important on backbone facilities, especially where costs are high (e.g., =

some global cross-sections).  We consider using MPLS to route ECRTP=20
compressed packets over multiple router hops without=20
compression/decompression cycles at each router.  Such an ECRTP over=20
MPLS capability can increase the bandwidth efficiency as well as=20
processing scalability of the maximum number of simultaneous VoIP flows=20
which use header compression at each router.  This draft gives a problem =

statement, goals, requirements, and an example scenario for ECRTP over=20
MPLS.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ash-avt-ecrtp-over-mpls-reqs-00=
.txt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "avt IETF" <avt@ietf.org>, "Toip list" <toip@snowshore.com>
Cc: "Text Telephone Forum" <textphone@lsv.pi.se>
Date: Thu, 8 Jan 2004 00:25:40 +0100
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Subject: [AVT] A new version of the text transport is ready. FW: I-D ACTION:draft-ietf-avt-rfc2793bis-01.txt
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This is a multi-part message in MIME format.

------=_NextPart_000_0385_01C3D57D.F2316EA0
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A new version of the RTP transport specification for real time interactiv=
e
text is available as described below.

Further down is a collection of comments received on the previous version=
,
and the responses from the authors.

I hope this specification will contribute to compatibility between
conversational multimedia applications as well as other applications
including text among available media.

Actions in ITU are depending on progress of this specification, so any
further comment should be submitted to the IETF avt group very soon.

Regards
-------------------------------------------
Gunnar Hellstr=F6m
Omnitor AB
Renathv=E4gen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------


-----Original Message-----
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Subject: I-D ACTION:draft-ietf-avt-rfc2793bis-01.txt


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	Title		: RTP Payload for Text Conversation
	Author(s)	: G. Hellstrom
	Filename	: draft-ietf-avt-rfc2793bis-01.txt
	Pages		: 19
	Date		: 2004-1-7

This memo describes how to carry text conversation session contents
in RTP packets. Text conversation session contents are specified in
ITU-T Recommendation T.140 [1].

Text conversation is used alone or in connection to other
conversational facilities such as video and voice, to form multimedia
conversation services.

This RTP payload description contains an optional possibility to
include redundant text from already transmitted packets in order to
reduce the risk of text loss caused by packet loss. The redundancy
coding follows RFC 2198.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rfc2793bis-01.txt

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------Answers on comments by Magnus Westerlund -------------------------

> 1. The abstract should probably be rewritten, I think the second
> paragraph is not relevant enough for this, while the RTP payload format
> for gateways are not mentioned at all.

Done. The second paragraph is replaced with a short notice about the two =
RTP
payloads.


> 2. I think the TOC is long enough to have it alone on the second page,
> i.e. I would propose that one start the introduction chapter on page 3.

Done.


> 3. The third last paragraph in chapter 1 seems to be a clear motivation
> for using TCP to transport text. I think there need to be some further
> elaboration on the reasons for using RTP, because the second last
> paragraph is not a obvious result of the previous one.

A description of the benefit of using RTP for text transmission is added.


> 4. Chapter 1, last paragraph, last sentence "and introduces a means of
> transporting text
>    interleaved with voice within the same RTP session.". Is " a means"
> correct english?

Changed to "a method".


> 5. In section 2, a reference to the complete information for RFC 2119
> would be nice.

Done.


> 6. Section 3:   "When transport of T.140 text session data in RTP is
> desired, the
>    payloads as described in this specification SHOULD be used." This
> sentence seems a bit strange. I don't think using normative words about
> using this payload format real has any meaning.

Changed to a sentence that describes the contents of the section instead.


> 7. Section 3.1, "This format is primarily used when text is transmitted
> on a separate
>    RTP session dedicated for the transmission of text and not shared
>    with other media, such as audio, DTMF, etc.  IP textphone devices
>    most commonly use this format."
>
> I think the restriction this payload format imposes on the media stream
> should be clearly documented in this section.

One unnecessary restriction has been eliminated. That was that FEC use wa=
s
described as possible for audio/t140 only. Now it is described as possibl=
e
for both formats. Other specific restrictions are not known. The descript=
ion
of ITU-T T.140 gives the application framework and the natural restrictio=
ns
that follow from the intention of carrying text character by character in=
 a
live conversation. We do not see a need to describe any specific restrict=
ion
here.

> 8. Section 3.2: "This T140block counter may be utilized to detect lost
> characters." I think one should clearly indicate that the counter if fo=
r
> detecting not lost characters but lost T140 blocks. I also think the
> comments made in the WG session in Minneapolis that the last block
> before a switch to another audio format can't be detected.

"characters" are now changed to "blocks".

The exact same problem exists with text/t140. If you do not get the last
transmitted UDP packet, you will not know until you get the next one. The
end result is precisely the same, regardless of text/T140 or audio/t140.
That is similar to
the case for other conversational media. You cannot be sure you heard the
last word, seen the last move or read the last character that the other e=
nd
expressed.


> 9. Section 3.6, "Note, also, that incoming text from a PSTN circuit
> might be at a higher bit-rate than can be played out on an egress PSTN
> circuit." The scenario and what is meant in this sentence is not clear.

We do not want to extend the discussion here of characteristics of the ol=
d
PSTN
textphone types, their different speeds, and what can be done in the tran=
sit
gateway application. We find it more appropriate to describe the
transmission
format here and discuss application in depth in the specifications of
gateway
applications that the audio/t140 format is intended for.


> 10. Section 3.7; "For profiles that use dynamic payload type number
> assignment, this payload format is identified by the name "T140" (see
> section 8). If redundancy is used per RFC 2198, the Payload Type MUST
> indicate that payload format ("RED")." I think that one should be
> clearer. We are using MIME type according to the definition in chapter
> 8, and also RED has its MIME type that can be used to identify the name.
> Therefore I would reformulate the sentences to: "For profiles that use
> dynamic payload type number assignment, this payload format can be
> identified by the MIME types "text/T140" and "audio/T140" (see section
> 8). If redundancy is used per RFC 2198, another payload type number
> needs to be provided for the redundancy format. MIME types for
> identifying RFC 2198 are available in RFC 3555 and RFC YYYY." Where RFC
> YYYY is the RFC number for the text/RED registration.

Done.


> 11. Section 3.7, The RTP sequence number: redefining the RTP sequence
> number through normative text is not good. Please provide a reference t=
o
> RFC 3550 and then you can have the informative text.

Done.


> 12. Section 3.7: "Timestamp: The RTP Timestamp encodes the approximate
> instance of entry of the primary text in the packet. A clock frequency
> of 1000
>      Hz MUST be used for text/t140.  The clock frequency may be
> specified for audio/t140 and is generally set to be 8000 Hz, as that is
> most common for audio."
>
> In the last sentence above, it says: The clock frequency may be ... Thi=
s
> is unclear, it needs normative language, making it clear that the rate
> MAY be set to any value, if not specified by out of band mechanism a RT=
P
> TS rate value of 8000 SHALL be used.

Changed to
"For audio/T140, the clock frequency
     MAY be set to any value. If not specified by out of band mechanism,
     the frequency value is generally set to be 8000 Hz, as that is most
     common for audio."


> 12.5. Section 3.7, why is the marker bit not defined?

The marker bit is not used.


> 13. Section 3.8: What is the following text doing in this chapter?
>   "When redundant transmission of the data according to RFC 2198 is
>    desired, the RTP header is followed by one or more redundant data
>    block headers, one for each redundant data block to be included.
>    Each of these headers provides the timestamp offset and length of th=
e
>    corresponding data block plus a payload type number indicating this
>    payload format ("T140").  Redundant data older than 16383 divided by
>    the clock frequency MUST not be transmitted."
>
> It would seem to belong in section describing the usage of RED. I think
> there are more things in regards to usage of RED, like section 3.9,
> that can be collected in a separate chapter clarifying that it concerns
> the usage of RED for t140.

Using RED or other ways to incrrease reliability has influence over many
sections. It
touches the user needs of reliability, the packetisation, the procedures =
to
discover and
recover loss, the marking of unrecoverable loss, the SDP parameters, and =
the
MIME registrations.
We find it more natural to cover these aspects in one chapter each and
within these chapters cover the
transmission variants.


> 14. Section 4.1 "A maximum buffering time of 500 ms is specified." Wher=
e
> is it specified?

The buffering time is specified in T.140. The reference was quite confusi=
ng
as you noticed, it is changed.


> 15. Section 5: Another comment from the WG meeting in Minneapolis, was
> the need to make it clear that although the MIME parameter CPS  the
> instantaneous reception rate might be higher due to jitter.

Added.


> 16. Section 8.1, as text/t140 in RFC 2793 does not have rate, and it is
> not need (due to fixed rate), I would remove it. The mapping to SDP
> chapter can simply explain that one simply puts in the rate.

Still, the rate is a required parameter in the MIME registration. Therefo=
re,
it is not removed.
It is also best to maintain it for backwards compatibility with RFC2793
implementations.


> 17. There should also be an interoperability consideration regarding CP=
S
> parameter.

The action required is described in the text. Further implications if the
parameter is not obeyed,
belongs more to descriptions in gateway specifications where the audio/t1=
40
format is applied.

-----------Answers on comments by Gregg
Vanderheiden--------------------------------------------

>I think we should move to amend RFC2793bis to INCLUDE the redundancy as =
a
>central part of the standard.  (a MUST).

>Can you think of why you would use RFC2793bis without redundancy?     Wi=
ll
>it be used for anything besides text over VoIP?

>I am particularly concerned that in an emergency, an otherwise QOS
>environment could become congested quickly and RFC2793 would fail except=
 if
>it had redundancy required.   And if it will be required, then why not a=
dd
>it to the base definition.

Gregg wanted that the use of text/t140 with 300 ms between transmissions =
and
one original and 3 redundant generations
of text transmitted, would be pronounced as mandatory to ensure sufficien=
t
text reliability in
congested and catastrophy situations.

We have a very clear sharpening from the current RFC2793, by RECOMMENDING
these parameters.
We do not want to limit the application by mandating these parameters, an=
d
believe that we can rely on implementers to be wise enough to follow the
RECOMMENDED values when there are no reasons to do otherwise.

-----------Answers on Frank Shear=B4=B4s
comments-------------------------------------------------------
>I noticed that the RFC 2793bis draft never speaks of byte ordering. Is i=
t
>assumed, since this is usually used in the context of RTP, that the byte
>order of the T140block counter is the byte order defined in RTP, i.e.,
>network order?

>Wouldn't it be slightly better to specifically mention the byte order in
the
>draft, just to avoid any potential confusion?

The following text is inserted:
a 16-bit "t140block counter" (with the most
   significant bit transmitted first)

>A very minor mistake: in section 5, "from section 14 or RFC 3550 apply"
>should be "from section 14 of RFC 3550 apply".

Woops, we missed that one. ( it is now in section 7 ). Hope it can be lef=
t
for final edits.


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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Sun Jan 11 19:23:39 2004
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Date: Mon, 12 Jan 2004 00:19:57 +0000
From: Colin Perkins <csp@csperkins.org>
To: "Ash, Gerald R (Jerry), ALABS" <gash@att.com>
Cc: avt@ietf.org
Subject: Re: [AVT] RE: I-D ACTION:draft-ash-avt-ecrtp-over-mpls-reqs-00.txt
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--> "Ash, Gerald R (Jerry), ALABS" <gash@att.com> writes:
> Please review and comment on the current draft of 'Requirements for ECRTP
> over MPLS'
> http://www.ietf.org/internet-drafts/draft-ash-avt-ecrtp-over-mpls-reqs-00.txt.

To me, this is a considerable improvement on the previous version. A few
comments:

In a number of places, the draft describes "using MPLS to route ECRTP
compressed packets over multiple router hops". Does this mean routing
compressed packets over an MPLS LSP? If so, I recommend clarifying the
draft to prevent confusion with multiple IP hops.

I'm not sure discussion of access links is appropriate, since MPLS is
unlikely to be used in this scenario.

In section 3, the requirements list ECRTP without discussion. I think ECRTP
is an appropriate solution, but it would be useful to motivation the choice
by discussing the requirements for the compression. This is particularly
important, since there are alternative algorithms (e.g. ROHC) that may be
considered.

In section 3, the draft nots that protocol extensions may be required for
ECRTP. Are these changes to ECRTP or a mapping of ECRTP onto a new "link"
layer? If the latter, the draft could do with clarification.

The draft should discuss the type of extension needed for RSVP-TE and LDP,
since this will feed into any new work plan.

Security considerations: Are there no issues with DoS attacks on MPLS
routers, due to the increased number of flows? 

Before publication, the draft will need an IANA Consideratations section
(even if it just says "No IANA actions are required") and the IPR boiler-
plate.

-- 
Colin Perkins
http://csperkins.org/

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Jan 12 10:10:43 2004
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Hi Ivica,

I think Rod manage to repeat my opinions on what need to be done to have 
a solution that works. And there are probably more interesting problems 
buried in the actual application of the proposals. Therefore I think 
that it is necessary that someone who desires this type of solutions 
starts working on the problems. I would think that a test implementation 
combined with writing the necessary documents to propose the necessary 
standard extensions. All to ensure that one has not forgotten things. 
Then there are probably also some issues around how to make the solution 
the right level of generic applicability.

I hope you better understand why I would not call it straightforward.

Cheers

Magnus




_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: B Prashanth <bprashanth@neomagic.com>
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Subject: Re: [AVT] AMR Profile
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Hi,

When AMR is sent over RTP, the speech data in the payload format is as 
specified by RFC 3267. RFC 3267 gives the following information about 
how the speech data shall be arranged. From RFC 3267:

4.3.3. Speech Data

   Speech data of a payload contains one or more speech frames or
   comfort noise frames, as described in the ToC of the payload.

      Note, for ToC entries with FT=14 or 15, there will be no
      corresponding speech frame present in the speech data.

   Each speech frame represents 20 ms of speech encoded with the mode
   indicated in the FT field of the corresponding ToC entry.  The length
   of the speech frame is implicitly defined by the mode indicated in
   the FT field.  The order and numbering notation of the bits are as
   specified for Interface Format 1 (IF1) in [2] for AMR and [4] for
   AMR-WB.  As specified there, the bits of speech frames have been
   rearranged in order of decreasing sensitivity, while the bits of
   comfort noise frames are in the order produced by the encoder.  The
   resulting bit sequence for a frame of length K bits is denoted d(0),
   d(1), ..., d(K-1).

   [2]   3GPP TS 26.101, "AMR Speech Codec Frame Structure", version
         4.1.0 (2001-06), 3rd Generation Partnership Project (3GPP).

   [4]   3GPP TS 26.201 "AMR Wideband speech codec; Frame Structure",
         version 5.0.0 (2001-03), 3rd Generation Partnership Project
         (3GPP).




B Prashanth wrote:

> Dear All,
> Can anyone please tell me that if RTP stack is sending AMR speech data 
> then does the encoded AMR payload, i.e. the AMR speech frame, contain 
> the AMR frame header (specifying the Frame type) or Does the AMR 
> speech frame contain just the raw, encoded speech data in the form of 
> Class A, B and C bits? 


The answer is that the speech frame does only contain the raw speech 
bits, denoted d(0)..d(K-1) as explained by Section 4.2 in TS 26.101. Not 
the the following IF1 fields: Frame Quality Indicator (1 bit), Mode 
Indication (3 bits), Mode Request (3 bits), and Codec CRC (8 bits). 
These fields are instead located in other fields in the RTP payload for 
AMR.

>
> Please note that the header being talked about here is the AMR speech 
> frame's header and not the RTP's AMR profile header (Specified in RFC 
> 3267).

With AMR profile I assume that you are talking about the RTP payload 
format, not profile. A Profile is something else in RTP terms.

Do you think the quoted text is unclear and needs modification?

Cheers

Magnus Westerlund



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Jan 14 10:16:41 2004
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From: "Lars-Erik Jonsson (LU/EAB)" <lars-erik.jonsson@ericsson.com>
To: "'Colin Perkins'" <csp@csperkins.org>,
        "Ash, Gerald R (Jerry), ALABS"
	 <gash@att.com>
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Subject: RE: [AVT] RE: I-D ACTION:draft-ash-avt-ecrtp-over-mpls-reqs-00.tx
	t
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Hi all,

I agree with all Colins comments, and I have further commented on
two of them, as well as providing some additional points below.

/L-E


> In section 3, the requirements list ECRTP without discussion. 
> I think ECRTP is an appropriate solution, but it would be
> useful to motivation the choice by discussing the requirements
> for the compression. This is particularly important, since
> there are alternative algorithms (e.g. ROHC) that may be
> considered.

I would really like to see such a discussion, or even better,
actually cover both ECRTP and ROHC. That should not make the
work much harder, it rather makes sense to do both at the same
time.


> In section 3, the draft notes that protocol extensions may be 
> required for ECRTP. Are these changes to ECRTP or a mapping
> of ECRTP onto a new "link" layer? If the latter, the draft
> could do with clarification.

I guess the latter is the case...


L-E.1:
Is really "Hop-by-hop ECRTP over MPLS" the correct title? I
think this is really about the opposite, i.e. HC over multiple
MPLS hops. I would suggest:
  "Requirements for Header Compression over MPLS",
as that would also cover both ECRTP and ROHC (then we can
decide later whether to cover ROHC or not) and we get rid of
the hop-by-hop confusion.


L-E.2:
The abstract is rather long, can we do something about that?
Since the details are in the intro, we can remove some, I think.


L-E.3:
The following text is confusing, and it is also an example where
I think my previous comment on "hop-by-hop" apply (section 2):
"More important in almost all locations is decreasing the
access bandwidth.  While hop-by-hop ECRTP over MPLS clearly
helps decrease bandwidth requirements, it also avoids
compression-decompression cycles at every router hop.  So
although bandwidth is getting cheaper, the value of compression
does not go away. In addition, IPv6 will increase the size of
headers and therefore increase the importance of header
compression for VoIP flows."

I do not know exactly what the above text wanted to say. The
argument that bandwidth cost is still important has been
addressed in the top of the section and could potentially be
summarized before starting the processing discussion), an
alternative text could be (end of current paragraph plus
2 new paragraphs to follow the current one):

  "So although bandwidth is getting cheaper, the value of
  compression does not go away.

  While hop-by-hop header compression could be applied to
  decrease bandwidth requirements, that would instead imply a
  processing requirement for compression-decompression cycles at
  every router hop. Therefore, end-to-end header compression over
  MPLS seems to be a viable alternative to get the compression
  benefits without introducing costly processing demands on
  intermediate nodes.

  It should further be noted that IPv6 will increase the size
  of headers, and therefore increase the importance of header
  compression for VoIP flows."


L-E.4:
The third paragraph of section 2 is also confusing. Is not 
the idea to use existing HC techniques along with MPLS
labels, in principle???


L-E.5:
Goal a in section 3 could explicitly say MPLS networks, i.e.
"a. provide more efficient voice transport over MPLS networks,"


L-E.6:
Goal c, should it not say "not significantly increase..."???


L-E.7:
The uppercase MUST, SHOULD, etc are not defined to have a
special meaning, so it is meaningless to use them if we do
not give them special meaning. Note that you have to define
their meaning yourself, you can not simply point at 2119, as
those definitions only apply to standards track document.
I would use lower case words.


L-E.8:
Requirement b could simply say "should use existing protocols,
such as ECRTP and/or ROHC,...."


L-E.9:
For requirement d, I would suggest "Should make use of..."


L-E.10:
The example scenario makes me even more convinced that
"hop by hop" is NOT the right terminology for what we are
aiming for, as it is rather the opposite, we want to avoid
hop-by-hop compression.


Overall, I agree with Colin that this is a significant 
improvement from the last document version. Good work!

BR
/L-E

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] Comments on draft-ietf-rmonmib-raqmon-framework-04.txt
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I've copied this list of comments to AVT as some of them relate to the use
of RTCP for reporting purposes.

1. Scope
Although the framework makes general reference to "real time applications",
most of the examples give are VoIP related. It seems that the primary
intended use of RAQMON is VoIP performance monitoring and the umbrella "real
time applications" scope is something of a "red herring".

2. VoIP Media path QoS
The document suggests that RAQMON provides QoS statistics, including media
path stats, but does not refer to or incorporate RFC3611 - the scope of
which is specifically VoIP media path QoS.  The RFC3611 VoIP metrics form
the basis of a number of VoIP QoS reporting protocols, and seem to represent
a reasonable level of cross-industry consensus on what is "reasonable" for
VoIP QoS monitoring.   RAQMON stands alone in trying to argue that RTCP
SR/RR statistics are sufficient for VoIP (or Video) QoS monitoring when the
industry has moved on to other more appropriate metrics.

3. Reporting intervals
RAQMON proposes RTCP as one option for the transport of PDUs. RFC3550
provides guidance on the intervals at which RTCP frames should be sent.  The
RAQMON framework proposes an independent set of rules on RTCP transmission
timing - the AVT group is usually fairly insistent that the rules described
in RFC3550 are applied to RTCP and hence within the RAQMON draft the current
definition of reporting intervals should be replaced by references to the
appropriate parts of RFC3550.
RAQMON encourages the use of multiple consecutive RTCP frame transmissions
to send long RAQMON reports.  This would certainly be in conflict with the
recommendations of RFC3550.

4. Measurement methodology
RAQMON proposed that the measurement methodology used to generate the
reported metrics is "outside the scope" of RAQMON and should be left to the
implementer.  Whilst implementers can "get away with this" in certain closed
system applications it makes any form of open system/ multi-vendor network/
inter-domain QoS management almost impossible.  If the only application of
RAQMON is for use in closed, single-vendor applications then why make it a
standard?
Reference to using "IETF and ITU specificed methodologies" is insufficient,
for example there are many definitions of jitter within ITU and IETF
standards.

Some of the metrics specified are also impractical - for example one way
delay.  This requires an accurate synchronized time source at the two ends
of an IP connection - probes that implement this metric often use GPS to
obtain the level of timing accuracy needed, simply relying on NTP timers
(even with an NTP server) would be generally inadequate.

As another example, Section 5.11 defines Round Trip Delay as potentially
including or not including the end system delay.  Jitter buffers can
sometimes reach several hundred milliseconds in size and hence knowing if a
reported 300 millisecond round trip delay did or didn't include the end
system delay could make a large difference to the user perceived delay.

As another example, Section 5.14 describes the count of the number of
"application packets" as a metric that "could be measured in different ways,
including the methodology described in RFC3550".  Is an application packet a
CODEC frame, an RTP frame, UDP or IP packet?

5. Mis-understanding of loss and jitter
>From section 5.18 "Packet loss tracks persistent congestion while jitter
measures tracks transient congestion. Packet loss metric SHOULD indicate
loss associated to the network environment as well as local device losses
over time. Packet loss of underlying transport network can be measured using
various methodologies e.g. as described in [RFC2680], [RFC 3550] and local
device level packet losses should to be captured by the local device
specific algorithms."
5.a The first sentence is incorrect, also there is no apparent reason for
making the statement
5.b SHOULD leaves us not knowing if the reported loss number includes packet
discards or not.  If I have a call on which 2% of the packets were reported
as (lost, lost+discarded?) then do I know if this was a good call?
Obviously if the 2% were lost and an additional 5% were discarded then the
call would be much worse than if the 2% related to both lost and discarded
packets!!
5.c With reference to comment 4 above, "various methodologies" and "local
device specific algorithms" leaves us completely in the dark on what has
been reported.
6. References
Doesn't refer to RFC3611?
Regards
Alan Clark








_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] secure avt
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Hello,

my question has to do with secure avt. I am a network analyst for a hospital
IS department, and we are implementing a distance learning center. Part of
my responsibility will be securing AV transmissions between our learning
center and other sites, and I suppose this is a very general request for the
status of things like using SRTP for doing this. My concerns are centered
upon HIPAA and how that governing body will affect these types of data
transmissions. Is there any currently accepted standard for doing this type
of thing? Is SRTP a current accepted standard for doing this? Or is this
part of what the AVT task force is developing.

Any input would be appreciated



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Lars-Erik Jonsson (LU/EAB)" <lars-erik.jonsson@ericsson.com>
To: "'Colin Perkins'" <csp@csperkins.org>,
        "Ash, Gerald R (Jerry), ALABS"
	 <gash@att.com>
Cc: avt@ietf.org
Subject: RE: [AVT] RE: I-D ACTION:draft-ash-avt-ecrtp-over-mpls-reqs-00.tx
	 t
Date: Thu, 15 Jan 2004 10:19:42 +0100
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Ash and others,

Unfortunately, Magnus had sent me the draft and I assumed it was
the new official version, but I guess it was a pre-version. The
official one is again an improvement, so you can ignore my
comments L-E.1 and L-E.10 below, but I still believe it would make
sense to generalize ECRTP to "Header Compression", i.e. to include
ROHC in the scope.

Cheers,
/L-E

> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of
> Lars-Erik Jonsson (LU/EAB)
> Sent: den 14 januari 2004 16:13
> To: 'Colin Perkins'; Ash, Gerald R (Jerry), ALABS
> Cc: avt@ietf.org
> Subject: RE: [AVT] RE: I-D
> ACTION:draft-ash-avt-ecrtp-over-mpls-reqs-00.tx t
> 
> 
> Hi all,
> 
> I agree with all Colins comments, and I have further commented on
> two of them, as well as providing some additional points below.
> 
> /L-E
> 
> 
> > In section 3, the requirements list ECRTP without discussion. 
> > I think ECRTP is an appropriate solution, but it would be
> > useful to motivation the choice by discussing the requirements
> > for the compression. This is particularly important, since
> > there are alternative algorithms (e.g. ROHC) that may be
> > considered.
> 
> I would really like to see such a discussion, or even better,
> actually cover both ECRTP and ROHC. That should not make the
> work much harder, it rather makes sense to do both at the same
> time.
> 
> 
> > In section 3, the draft notes that protocol extensions may be 
> > required for ECRTP. Are these changes to ECRTP or a mapping
> > of ECRTP onto a new "link" layer? If the latter, the draft
> > could do with clarification.
> 
> I guess the latter is the case...
> 
> 
> L-E.1:
> Is really "Hop-by-hop ECRTP over MPLS" the correct title? I
> think this is really about the opposite, i.e. HC over multiple
> MPLS hops. I would suggest:
>   "Requirements for Header Compression over MPLS",
> as that would also cover both ECRTP and ROHC (then we can
> decide later whether to cover ROHC or not) and we get rid of
> the hop-by-hop confusion.
> 
> 
> L-E.2:
> The abstract is rather long, can we do something about that?
> Since the details are in the intro, we can remove some, I think.
> 
> 
> L-E.3:
> The following text is confusing, and it is also an example where
> I think my previous comment on "hop-by-hop" apply (section 2):
> "More important in almost all locations is decreasing the
> access bandwidth.  While hop-by-hop ECRTP over MPLS clearly
> helps decrease bandwidth requirements, it also avoids
> compression-decompression cycles at every router hop.  So
> although bandwidth is getting cheaper, the value of compression
> does not go away. In addition, IPv6 will increase the size of
> headers and therefore increase the importance of header
> compression for VoIP flows."
> 
> I do not know exactly what the above text wanted to say. The
> argument that bandwidth cost is still important has been
> addressed in the top of the section and could potentially be
> summarized before starting the processing discussion), an
> alternative text could be (end of current paragraph plus
> 2 new paragraphs to follow the current one):
> 
>   "So although bandwidth is getting cheaper, the value of
>   compression does not go away.
> 
>   While hop-by-hop header compression could be applied to
>   decrease bandwidth requirements, that would instead imply a
>   processing requirement for compression-decompression cycles at
>   every router hop. Therefore, end-to-end header compression over
>   MPLS seems to be a viable alternative to get the compression
>   benefits without introducing costly processing demands on
>   intermediate nodes.
> 
>   It should further be noted that IPv6 will increase the size
>   of headers, and therefore increase the importance of header
>   compression for VoIP flows."
> 
> 
> L-E.4:
> The third paragraph of section 2 is also confusing. Is not 
> the idea to use existing HC techniques along with MPLS
> labels, in principle???
> 
> 
> L-E.5:
> Goal a in section 3 could explicitly say MPLS networks, i.e.
> "a. provide more efficient voice transport over MPLS networks,"
> 
> 
> L-E.6:
> Goal c, should it not say "not significantly increase..."???
> 
> 
> L-E.7:
> The uppercase MUST, SHOULD, etc are not defined to have a
> special meaning, so it is meaningless to use them if we do
> not give them special meaning. Note that you have to define
> their meaning yourself, you can not simply point at 2119, as
> those definitions only apply to standards track document.
> I would use lower case words.
> 
> 
> L-E.8:
> Requirement b could simply say "should use existing protocols,
> such as ECRTP and/or ROHC,...."
> 
> 
> L-E.9:
> For requirement d, I would suggest "Should make use of..."
> 
> 
> L-E.10:
> The example scenario makes me even more convinced that
> "hop by hop" is NOT the right terminology for what we are
> aiming for, as it is rather the opposite, we want to avoid
> hop-by-hop compression.
> 
> 
> Overall, I agree with Colin that this is a significant 
> improvement from the last document version. Good work!
> 
> BR
> /L-E
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Jan 15 11:23:44 2004
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Date: Thu, 15 Jan 2004 08:20:39 -0800
To: "John Cumbo" <john@cumbo.org>
From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [AVT] secure avt
Cc: <avt@ietf.org>
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John,
    SRTP soon will be an Internet Proposed Standard.  It very well may be 
the right choice for securing your application, all things being 
equal.  You will need to have an authorization and key management 
infrastructure to support an SRTP product.

    You should also consider various alternatives.  One alternative is 
IPsec, which secures UDP and has been successful in providing 
confidentiality and/or authentication for media streams for many 
applications.  SRTP is quite new and there are lots of IPsec-based products 
but fewer SRTP-based products at the present time.  You might choose IPsec 
over SRTP if you want to have an IPsec VPN among your sites and don't need 
any of the features that SRTP has over IPsec, such as the following.

   1. SRTP allows compressed RTP and has much less packet overhead than IPsec.
   2. SRTP provides application-layer security, which is more secure for 
some applications particularly when multi-user computers are used as end 
systems.
   3. SRTP is tailored to RTP and supports, for example, encrypted RTP with 
unencrypted RTCP flows.

Mark

At 11:30 AM 1/14/2004, John Cumbo wrote:
>Hello,
>
>my question has to do with secure avt. I am a network analyst for a hospital
>IS department, and we are implementing a distance learning center. Part of
>my responsibility will be securing AV transmissions between our learning
>center and other sites, and I suppose this is a very general request for the
>status of things like using SRTP for doing this. My concerns are centered
>upon HIPAA and how that governing body will affect these types of data
>transmissions. Is there any currently accepted standard for doing this type
>of thing? Is SRTP a current accepted standard for doing this? Or is this
>part of what the AVT task force is developing.
>
>Any input would be appreciated
>
>
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Thu, 15 Jan 2004 21:37:56 +0000
From: Colin Perkins <csp@csperkins.org>
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[Takes working group chair hat off...]

The following may be of interest to the group.
Colin


------- Forwarded message:
>Date: Thu, 15 Jan 2004 16:09:47 -0500
>From: Internet-Drafts@ietf.org
>To: IETF-Announce: ;
>Subject: I-D ACTION:draft-perkins-avt-rtp-reactive-fec-00.txt
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> 
> 	Title		: RTP Retransmission Using Reactive FEC
> 	Author(s)	: C. Perkins
> 	Filename	: draft-perkins-avt-rtp-reactive-fec-00.txt
> 	Pages		: 8
> 	Date		: 2004-1-15
> 	
> This memo describes how the RTP Payload Format for Generic Forward
> Error Correction can be combined with the Extended RTP Profile for
> RTCP-based Feedback to provide a solution for retransmission of lost
> RTP packets. Such a retransmission format is expected to be useful to
> improve the reliability of real-time applications with relaxed delay
> bounds, for example non-interactive streaming audio/video.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-perkins-avt-rtp-reactive-fec-00.txt
...
------- End of Forwarded Message

-- 
Colin Perkins
http://csperkins.org/

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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hi,
I would like to have explaining about SSRC synchronization source could you 
please give me an example about it ,I know that is is different from source 
address, but are there any relation between them?
also I would to ask this question:
how the packets travel fron the source address to destination address is 
there any protocol for choosing the routing,? please give me more 
information about that

thanks for your help

nour

_________________________________________________________________
Express yourself with cool new emoticons http://www.msn.co.uk/specials/myemo


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



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From: Abdelbasset Trad <Abdelbasset.Trad@sophia.inria.fr>
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SSRC is an identification related to a session used for synchronisation 
at the destination, one user (one source address) may have several SSRC 
for instance if he has a video session and audio session being 
transmitted. SSRCs are chosen randomly at every RTP transmission in 
order to be unique and to not have the same SSRC at different users. 
Routing is independent from RTP transmission, it is done at routers 
using protocols such OSPF, RIP...
A. Trad

>
>>
>> hi,
>> I would like to have explaining about SSRC synchronization source 
>> could you please give me an example about it ,I know that is is 
>> different from source address, but are there any relation between them?
>> also I would to ask this question:
>> how the packets travel fron the source address to destination address 
>> is there any protocol for choosing the routing,? please give me more 
>> information about that
>>
>> thanks for your help
>>
>> nour
>>
>> _________________________________________________________________
>> Express yourself with cool new emoticons 
>> http://www.msn.co.uk/specials/myemo
>>
>>
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www1.ietf.org/mailman/listinfo/avt
>
>
>
>
>


-- 

------------------------------------------
   Abdelbasset TRAD
   INRIA Planete Project
   2004 Route des Lucioles, BP-93
   06902 Sophia-Antipolis Cedex, France
   Phone: (+33) 4 92 38 50 22
   Fax: (+33) 4 92 38 79 78
   Email: atrad@sophia.inria.fr
   http://www-sop.inria.fr/planete/atrad
------------------------------------------ 




_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload Format for H.261 Video Streams
	Author(s)	: T. Turletti, C. Huitema, R. Even
	Filename	: draft-ietf-avt-rfc2032-bis-00.txt
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	Title		: RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video (H.263+)
	Author(s)	: J. Ott, G. Sullivan, S. Wenger, C. Zhu, R. Even
	Filename	: draft-ietf-avt-rfc2429-bis-00.txt
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Hi Paul,

Here are some comments on the text/RED mime registration:

1. Section 3: rate parameter: I think the "may" in the last sentence 
should be a normative one.

2. Section 3: The "pt" list is not motivated, one does not understand 
the purpose of it. I think you should add a little text to make it 
clear. Expanding the first sentence like this may be sufficient: "a 
comma-separated list of RTP payload types that MAY occur in the 
redundancy payload."

3. Section 3: Security considerations: Referencing the wrong chapter. I 
think you could also should rewrite it so that it becomes more 
standalone. Proposal: "see section 5 of RFC XXXX." Do also put in a 
RFC-editor note about replacing XXXX with the documents number.

4. Section 4, second bullet. The rate parameter is not explicitly 
stated. I think the second sentence should be: "The rate parameter's 
value goes as rate in the "a=rtpmap" attribute." The MUST statement 
belongs to the MIME type parameter in general and should be moved there 
instead.

5. Section 4, the last bullet for the "pt" list. This is under 
specified, the following problems exist:
-  The removal or not of <"> quoting etc. is not defined.
- How to handle any "dynamic-payload-type "=" content-type" string is 
also not explained.

6. Section 4, the "pt" example. I think it can be improved with stating 
what the 99 is.

7. Section 5: A proposal for how to handle the security problem is 
always good. Proposal: "To prevent modification of the primary, 
secondary or header information, payload integrity protection over at 
least the complete RTP packet is recommend, for example using SRTP []."

8. Section 6, The first reference is an informative one, please move it 
to an informative reference list. Also reference 7 is not used at all, 
please remove.

I expect that the when these things are addressed the document is ready 
for WG last call.

cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi Gunnar and Paul,

Here is my comments on the payload format.

1. Abstract: You need to remove the [1] reference.

2. Section 1: There is a MUST in the introduction. It should be changed 
to lower case must of two reasons. First, this document does not define 
that T.140 elements use ISO 10646-1. Secondly, having normative language 
in an introduction is not appropriate.

3. Section 3.2: Last sentence of second paragraph. "This T140block 
counter may be
   utilized to detect lost blocks." I would propose to change this to: 
"This T140block counter is intended to be
   utilized to detect lost blocks and avoid duplication of blocks."

4. section 3.3, last sentence: Is this SHOULD correct? Why does the 
paragraph implies that CCS SHOULD be place in one block, why not MAY or 
SHALL?

5. The robustness features of the draft. As I have privately commented I 
would prefer to see the robustness features be collected into a single 
chapter. I think by moving sections: 3.4, 3.5,  3.8, 3.9, and 4.4 into a 
new chapter, between 4 and 5 the draft can be made more understandable.

6. Section 3.6: I think one should be more explicit about the fact that 
the rendering of (reasonably) late blocks is recommend and very beneficial.

7. Section 3.6: End of last paragraph: I think I understand the 
motivation behind the recommendation that voice is discarded rather then 
the T.140 when going from IP to PSTN at a gateway. However I think one 
should reflect somewhat on the effects on the audio also.

8. Section 3.7: Timestamp. "For audio/T140, the clock frequency
     MAY be set to any value. If not specified by out of band mechanism,
     the frequency value is generally set to be 8000 Hz, as that is most
     common for audio."
I think the "generally set to be 8000 Hz" is not good enough. There 
needs to be strict rules on what to do, otherwise you are in trouble. I 
think that if not out-of-band signalling, where the interleaved audio 
can be matched in rate, a fixed rate shall be set.

I am also missing the sentence saying: "When using this payload format 
interleaved with an audio codec, the rate SHOULD be chosen to be the 
same as the audio codec(s) to avoid RTP timestamp rate switching."

9. Section 3.7: Missing marker bit definition. Even if not used it MUST 
be defined as being not used and set to 0.

10. Section 3.8: I think there are no reason to have a section called 
"additional headers" and then simply having it state, no headers exist. 
It is mostly about redundancy anyway, please rename.

11. section 3.9, second paragraph. This paragraph is very hard to 
interpret. First it talks about audio/t140, then about text/t140, and 
when you come to the rules you are rather confused about for what it 
applies. Please try to improve the paragraph through restructuring.

12. In regards to the robustness options. As we now have available 
mechanism to report packet losses, RFC 3611, but probably more important 
draft-ietf-avt-rtcp-feedback and its NACK format, there is now 
possibilities to do reactive repairs. I think the possibility should at 
least be mentioned, or if it doesn't work, that should be motivated. 
Please investigate this.

13. Section 5, Second paragraph: "To control the character transmission 
rate, the "fmtp" attribute [7]
   is used with the following syntax:"
I think this sentence should be changed to explain that; first CPS is a 
MIME parameter, secondly that this is how it is mapped according to 
section x to SDP.

14. Section 6.3: The RED SDP examples. Why does on have a=fmtp lines 
like this? To my understanding of the pt list it should only contain a 
single 98. This is either a bug here, or problem in the text/red document.

15. Section 7: I think the security section can be improved:
- Separate the three main issues, confidentiality, integrity, and source 
authentication. They are different, and have different severity and 
attacks, and different counters.
- The section should give recommendations to mechanism that allows one 
to counter these problems. For the two first SRTP works very well. For 
the last it provides some properties that can be used to accomplish 
that, combined with the signalling.
So please expand this.

16. Section 8: Should be titled "IANA Consideration". And the 
introduction to the chapter should request that the two mime types are 
registered.

17. Section 8.1, and 8.2: The registration should mention "This type is 
only defined for transfer via RTP."

18. Section 8.1, and 8.2: Security consideration: Please point at 
section 7 of RFC XXXX.

19. Section 8.2: The rate parameter for audio should contain the rule 
that it SHOULD be matched with interleaved audio codecs.

20. Section 11: Are really all references normative. I would expect that 
5, 6, and 8 are informative. I have however not looked closely at it.

21. As it seems there will be some more RFC editors notes, they may be 
moved into a separate RFC-editor section instead of being spread through 
the document.


Cheers

Magnus Westerlund 

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
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Date: Wed, 21 Jan 2004 20:26:44 +0000
From: Colin Perkins <csp@csperkins.org>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Cc: gunnar.hellstrom@omnitor.com, paulej@packetizer.com, avt@ietf.org
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--> Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
> Hi Gunnar and Paul,
> 
> Here is my comments on the payload format.
...
> 12. In regards to the robustness options. As we now have available 
> mechanism to report packet losses, RFC 3611, but probably more important 
> draft-ietf-avt-rtcp-feedback and its NACK format, there is now 
> possibilities to do reactive repairs. I think the possibility should at 
> least be mentioned, or if it doesn't work, that should be motivated. 
> Please investigate this.

Unless specified otherwise, it should be assumed that the standard, payload
format independent, repair schemes can be used with any new payload format.
I don't see any need to list them specially here.

> 14. Section 6.3: The RED SDP examples. Why does on have a=fmtp lines 
> like this? To my understanding of the pt list it should only contain a 
> single 98. This is either a bug here, or problem in the text/red
> document.

The SDP fragments such as

       m=text 11000 RTP/AVP 98 100
       a=rtpmap:98 t140/1000
       a=rtpmap:100 red/1000
       a=fmtp:100 98/98

appear to be correct. The "a=fmtp:100 98/98" line indicates that both
primary and redundant coding use payload type 98. 

> 15. Section 7: I think the security section can be improved:
> - Separate the three main issues, confidentiality, integrity, and source 
> authentication. They are different, and have different severity and 
> attacks, and different counters.
> - The section should give recommendations to mechanism that allows one 
> to counter these problems. For the two first SRTP works very well. For 
> the last it provides some properties that can be used to accomplish 
> that, combined with the signalling.
> So please expand this.

Note that this is recommendations, rather than "MUST" requirements, since
SRTP is not appropriate in all cases.

-- 
Colin Perkins
http://csperkins.org/

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hello avt groupers (particularly Colin and Magnus),

I have attached an rtp payload format for the ATRAC family of audio 
codecs Sony has been developing.  Some of you may be familiar with the 
ATRAC-X payload we were working on last year.  With a bit of guidance 
from Mangus, we have decided to considerably simplify the payload, 
while at the same time generalize it so that previous ATRAC codecs, 
which are still widely used, could be included.

As I seek comments from listers on what needs to be fixed and 
clarified, I have a question about the naming convention.  Must we 
treat this as a completely new payload format (-00), which would mean 
committing to an earlier deadline, or can we consider this a continued 
revision of previous submissions (making this a -04)?

Please fire away with comments.

Thanks,
Matt



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Internet Engineering Task Force                       Mitsuyuki Hatanaka
Internet Draft                                           Matthew Romaine
January 2004                                            Sony Corporation
Expires: June 2004

                    RTP payload format for ATRAC family
                <draft-hatanaka-avt-rtp-atrac-family-00.txt>

Status of this Memo

This document is an Internet-Draft and is in full conformance with all 
provisions of Section 10 of RFC2026.

Internet-Drafts are working documents of the Internet Engineering Task 
Force (IETF), its areas, and its working groups.  Note that other groups
may also distribute working documents as Internet-Drafts.

Internet-Drafts are draft documents valid for a maximum of six months 
and may be updated, replaced, or obsoleted by other documents at any 
time.  It is inappropriate to use Internet- Drafts as reference
material or to cite them other than as "work in progress." 

The list of current Internet-Drafts can be accessed at 
http://www.ietf.org/ietf/1id-abstracts.txt 

The list of Internet-Draft Shadow Directories can be accessed at 
http://www.ietf.org/shadow.html.

the key words "MUST, "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [1].

Copyright (C) The Internet Society (2004).  All Rights Reserved.

Abstract

This document describes an RTP payload format for efficient and
flexible transporting of audio data encoded with the ATRAC family of
codecs.  Recent enhancements to the ATRAC family of codecs support high
quality audio coding with multiple channels.  The RTP payload format as
presented in this document includes support for data fragmentation and
elementary redundancy measures.















Hatanaka                    Expires June 2004                   [Page 1]
Internet Draft     <draft-hatanaka-avt-rtp-atrac-00.txt>    January 2004

1 Introduction

The ATRAC family of perceptual audio codecs are designed to address
numerous needs for high quality low bit-rate audio transfer.  ATRAC
technology can be found in many consumer and professional products and
applications, including MD players, voice recorders, mobile phones, and
CD players.  The need for real-time streaming of audio data has grown,
and this document details our efforts in increasing the product and
application space for the ATRAC family of codecs.

Recent advances in ATRAC technology allow for multiple channels of
audio to be encoded in customizable groupings.  This should allow for
future expansions in scaled streaming.  To provide the greatest
flexibility in streaming any one of the ATRAC family member codecs
however, this payload format does not distinguish between the codecs on
a packet level.

This simplified payload format contains only the basic information
needed to disassemble a packet of ATRAC audio in order to decode it.
Timestamps are in sample units, with audio data encoded into frames of
512, 1024, or 2048 samples.  There is basic support for fragmentation
and redundancy, so ATRAC frames MAY exceed an MTU size of 1500 octets.


2 Payload Format

2.1 RTP Header

 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1  
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X|  CC   |M|     PT      |       sequence number         |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                          timestamp                            |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|            synchronization source (SSRC) identifier           |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|           contributing source (CSRC) identifiers              |
|                             .....                             |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Extension (X): 1bit
Set to zero as no header extensions are currently being supported.

Marker (M): 1 bit
Set to zero as profiling is currently not used.

Payload Type (PT): 7 bits
The payload type can either be dynamically allocated at the application
level, or an RTP profile for a class of applications is expected to
assign the payload type for this format.  A dynamic allocation should
designate this format as ATRAC-Family.



Hatanaka                    Expires June 2004                   [Page 2]
Internet Draft     <draft-hatanaka-avt-rtp-atrac-00.txt>    January 2004

Sequence number: 16 bits
This field is as defined in [1].

Timestamp: 32 bits 
A timestamp representing the sampling time of the first sample of the
first ATRAC frame in the RTP packet.  The clock frequency MUST be set
to the sample rate of the encoded audio data, and is conveyed
out-of-band.

V/P/CC identifiers:
These three fields, 2 bits, 1 bit, and 4 bits respectively, are as
defined in [1].

SSRC/CSRC identifiers:
These two fields, each 32 bits with one SSRC field and a maximum of 16
CSRC fields, are as defined in [1].  2.2 Payload Header

The ATRAC family payload header is a scant two octets.  This should
make processing very simple.

 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5  
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|C|FrgNo| Rsrvd |NFrames| FrOff |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Continuous flag (C): 1 bit
Set to one if this is a continuation of a fragmented packet.

Fragment Number (FrgNo): 3 bits
In the event of data fragmentation, this value is 1 for the first
packet, and increases sequentially for the remaining fragmented data
packets.

Number of Frames (NFrames): 4 bits
The number of frames in this packet. This allows for a maximum of 16
ATRAC-encoded audio encapsulations per packet, with 0 indicating one
frame.  Keep in mind only the first frame is allowed to be fragmented.
Additionally, this must not be anything other than 0 for subsequent
packets containing the fragmented frame.

Frame Offset (FrOff): 4 bits
The purpose of frame offsets is to provide a basic mechanism for the
transmission of redundant data.  Using the current packet's timestamp
as a reference, the frame offset is converted to units of the
timestamp, which then corresponds to the playback time of the first
frame in the packet.  This field should NOT be used in packets
containing fragmented data.








Hatanaka                    Expires June 2004                   [Page 3]
Internet Draft     <draft-hatanaka-avt-rtp-atrac-00.txt>    January 2004

2.3 Payload Data

ATRAC payload data consists of 4 bits representing the frame ID.  The
following 2 octets then represent the byte-length of encoded audio
data.  After that, the actual audio data follows.

 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|FrameID|      Block Length     |       ATRAC data...             |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Frame ID: 4 bits
A sequential identifier, starting from 0 for each new non-fragmented
packet, identifying the start of each frame in this packet. 

Block length: 12 bits
The byte length of encoded audio data until the end of the current
packet.  This is so that in the case of fragmentation, if only a
subsequent packet is received, decoding can still occur.  12 bits
allows for a maximum block length of 4096 bytes.  In the event data is
larger but would still fit within MTU limits, fragmentation must occur.


3 Frame Packetization

Each RTP packet contains either an integer number of ATRAC encoded
audio frames, with a maximum of 16, or one ATRAC frame fragment.  

As many complete ATRAC frames as can fit in a single path-MTU should be
placed in an RTP packet, with the aforementioned maximum of 16.
However, if an ATRAC frame will not fit into an RTP packet, it must be
fragmented. 

The start of a fragmented frame gets placed in its own RTP packet,
its Continuous bit (C) set to one, and its Fragment Number (FragNo) set
to one.  As the frame must be the only one in the packet, the Number of
Frames field is also one.  Subsequent packets are to contain the
remaining fragmented frame data, with the Fragment Number increasing
sequentially and the Continuous bit (C) consistently set to one.  As
subsequent packets do not contain any new frames, the Number of Frames
field is to be zero.  The last packet of fragmented data must have the
Continuous bit (C) set to zero.  

In the event of fragmentation, the basic redundancy measures should
NOT be used.

3.1 Example Fragmented ATRAC Frame

An example of a fragmented ATRAC frame is presented below.  The encoded
audio data frame is split over three RTP packets.  For brevity, the RTP
packet header details have been excluded.




Hatanaka                    Expires June 2004                   [Page 4]
Internet Draft     <draft-hatanaka-avt-rtp-atrac-00.txt>    January 2004

Packet 1:
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1  
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1|  1  | Rsrvd |   1   |   0   |   1   |      block length     |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                          ATRAC data...                        |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Packet 2:
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1  
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1|  2  | Rsrvd |   0   |   0   |   1   |      block length     |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                     ...more ATRAC data...                     |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Packet 3:
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1  
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0|  3  | Rsrvd |   0   |   0   |   1   |      block length     |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                ...the last of the ATRAC data                  |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The following points highlight important characteristics of the example
above:

-) the transition from one to zero of the Continuous bit (C)
-) a sequential increase in the Fragment Number
-) except for the first packet, a zero for the Number of Frames field


4 SDP Usage

Certain attributes will need to be initiated before content can be
streamed.  For these purposes, the Session Description Protocol [3]
should be used.  Specifics in implementation are outside the scope of
this document.  However, the following attributes may prove useful.

m = audio PortNumber (X-)ATRAC/44100/2
a = rtpmap:PT_ATRAC (X-)ATRAC/44100/2
a = fmtp:PT_ATRAC Profile [profile] FrameSampleSize [fsSize] BitRate
[bps] ChannelConfigIndex [cci]












Hatanaka                    Expires June 2004                   [Page 5]
Internet Draft     <draft-hatanaka-avt-rtp-atrac-00.txt>    January 2004

5 Security Considerations

Certain security precautions may be desired to protect copyrighted
material.  The payload format as described in this document is subject
to the security considerations defined in [1].  These security
considerations imply the protection and confidentiality of the streamed
content through encryption.  Encryption may be performed on the encoded
ATRAC since the compression scheme follows an end-to-end model.


6 References

[1] RTP: A Transport Protocol for Real-Time Applications (RFC 1889)

[2] RTP Payload Format for Vorbis Encoded Audio. 
    Work in progress, draft-kerr-avt-rtp-vorbis-00.txt

[3] Handley, M. and V. Jacobson, "SDP: Session Description Protocol",
    RFC 2327, April 1998


7 Full Copyright Statement

Copyright (C) The Internet Society (2004).  All Rights Reserved.

This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it or
assist in its implementation may be prepared, copied, published and
distributed, in whole or in part, without restriction of any kind,
provided that the above copyright notice and this paragraph are
included on all such copies and derivative works.  However, this
document itself may not be modified in any way, such as by removing the
copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of developing
Internet standards in which case the procedures for copyrights defined
in the Internet Standards process must be followed, or as required to
translate it into languages other than English.

The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.

This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT
NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION HEREIN WILL
NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR
FITNESS FOR A PARTICULAR PROPOSE.








Hatanaka                    Expires June 2004                   [Page 6]
Internet Draft     <draft-hatanaka-avt-rtp-atrac-00.txt>    January 2004

8 Authors' Addresses

Mitsuyuki Hatanaka
Sony Corporation
6-7-35 Kitashinagawa Shinagawa-ku
Tokyo, 141-0001
Japan

Email: hatanaka@av.crl.sony.co.jp

Matthew Romaine
Sony Corporation
6-7-35 Kitashinagawa Shinagawa-ku
Tokyo, 141-0001
Japan

Email: Matthew.Romaine@jp.sony.com

Jun Matsumoto
Sony Corporation
6-7-35 Kitashinagawa Shinagawa-ku
Tokyo, 141-0001
Japan

Email: jun@av.crl.sony.co.jp






























Hatanaka                    Expires June 2004                   [Page 7]

--Apple-Mail-21--201518565
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----------------------------------------
Matthew M.I. Romaine
IMNC Audio Codec Group
Sony Corporation
phone: 03-5448-6065
fax: 03-5448-5617
----------------------------------------
--Apple-Mail-21--201518565--


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Jan 22 02:23:42 2004
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Date: Wed, 21 Jan 2004 23:20:54 -0800 (PST)
From: Greg Herlein <gregh@arkstream.com>
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Subject: [AVT] RFC2250 Profile Implementations?
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Does anyone know of any RFC2250 implementations that are open 
source - particularly that support MPEG transport streams?  

Second, can anyoneconfirm which streaming servers that support 
RTP also support this RFC?

I'm double checking my own research, and just wondering if I 
missed any...

Thanks in advance,

Greg Herlein


_______________________________________________
Audio/Video Transport Working Group
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From exim@www1.ietf.org  Thu Jan 22 02:54:33 2004
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From: "Even, Roni" <roni.even@polycom.co.il>
To: "'Matthew Romaine'" <Matthew.Romaine@jp.sony.com>, avt@ietf.org
Cc: Mitsuyuki Hatanaka <hatanaka@av.crl.sony.co.jp>,
        Jun Matsumoto
	 <jun@av.crl.sony.co.jp>
Subject: RE: [AVT] pre-submission of rtp payload for atrac-family
Date: Thu, 22 Jan 2004 09:51:57 +0200
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Hi,
A small question, I did not see any reference in the draft for the codec
itself, is it a publicly available codec
Thanks
Roni Even

*************************************
Roni Even

Polycom Israel

Tel: +972-3-9251200
Cell: +972-55-481099
email:roni.even@polycom.co.il
*******************************************


-----Original Message-----
From: Matthew Romaine [mailto:Matthew.Romaine@jp.sony.com]
Sent: Thursday, January 22, 2004 5:34 AM
To: avt@ietf.org
Cc: Mitsuyuki Hatanaka; Jun Matsumoto
Subject: [AVT] pre-submission of rtp payload for atrac-family



Hello avt groupers (particularly Colin and Magnus),

I have attached an rtp payload format for the ATRAC family of audio 
codecs Sony has been developing.  Some of you may be familiar with the 
ATRAC-X payload we were working on last year.  With a bit of guidance 
from Mangus, we have decided to considerably simplify the payload, 
while at the same time generalize it so that previous ATRAC codecs, 
which are still widely used, could be included.

As I seek comments from listers on what needs to be fixed and 
clarified, I have a question about the naming convention.  Must we 
treat this as a completely new payload format (-00), which would mean 
committing to an earlier deadline, or can we consider this a continued 
revision of previous submissions (making this a -04)?

Please fire away with comments.

Thanks,
Matt



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Cc: avt@ietf.org, Mitsuyuki Hatanaka <hatanaka@av.crl.sony.co.jp>,
        Jun Matsumoto <jun@av.crl.sony.co.jp>
From: Matthew Romaine <Matthew.Romaine@jp.sony.com>
Subject: Re: [AVT] pre-submission of rtp payload for atrac-family
Date: Thu, 22 Jan 2004 17:33:41 +0900
To: "Even, Roni" <roni.even@polycom.co.il>
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Hi Roni,

The codec itself is proprietary, so details are only available through 
licensing.

Thanks,
Matt


On 2004/01/22, at 16:51, Even, Roni wrote:

> Hi,
> A small question, I did not see any reference in the draft for the 
> codec
> itself, is it a publicly available codec
> Thanks
> Roni Even
>
> *************************************
> Roni Even
>
> Polycom Israel
>
> Tel: +972-3-9251200
> Cell: +972-55-481099
> email:roni.even@polycom.co.il
> *******************************************


_______________________________________________
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Hi,

Okay, some further comments. Sorry for including the wrong address to 
Gunnar (its an .se not .com address).

Colin Perkins wrote:

>--> Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
>  
>
>>Hi Gunnar and Paul,
>>
>>Here is my comments on the payload format.
>>    
>>
>...
>  
>
>>12. In regards to the robustness options. As we now have available 
>>mechanism to report packet losses, RFC 3611, but probably more
>>    
>>
>important 
>  
>
>>draft-ietf-avt-rtcp-feedback and its NACK format, there is now 
>>possibilities to do reactive repairs. I think the possibility should
>>    
>>
>at 
>  
>
>>least be mentioned, or if it doesn't work, that should be motivated. 
>>Please investigate this.
>>    
>>
>
>Unless specified otherwise, it should be assumed that the standard,
>payload
>format independent, repair schemes can be used with any new payload
>format.
>I don't see any need to list them specially here.
>  
>

Yes, your right. Although I like to make people more aware of the 
possibilities, it is probably a less good idea to create unnecessary 
convolution.

>  
>
>>14. Section 6.3: The RED SDP examples. Why does on have a=fmtp lines 
>>like this? To my understanding of the pt list it should only contain a
>>    
>>
>
>  
>
>>single 98. This is either a bug here, or problem in the text/red
>>document.
>>    
>>
>
>The SDP fragments such as
>
>       m=text 11000 RTP/AVP 98 100
>       a=rtpmap:98 t140/1000
>       a=rtpmap:100 red/1000
>       a=fmtp:100 98/98
>
>appear to be correct. The "a=fmtp:100 98/98" line indicates that both
>primary and redundant coding use payload type 98. 
>  
>

I think we have an inconsistency between the MIME type registrations in 
RFC 3555 and draft-ietf-avt-text-red, and whats in RFC 2198. In the RFC 
3555 the following is written about the payload type list paramter:


   Required parameters:
        pt: a comma-separated list of RTP payload types.  Because
        comma is a special character, the list must be a quoted-string
        (enclosed in double quotes).  For static payload types, each
        list element is simply the type number.  For dynamic payload
        types, each list element is a mapping of the dynamic payload
        type number to an embedded MIME content-type specification for
        the payload format corresponding to the dynamic payload type.
        The format of the mapping is:

           dynamic-payload-type "=" content-type

        If the content-type string includes a comma, then the
        content-type string MUST be a quoted-string.  If the content-
        type string does not include a comma, it MAY still be quoted.
        Since it is part of the list which must itself be a quoted-
        string, that means the quotation marks MUST be quoted with
        backslash quoting as specified in RFC 2045.  If the content-
        type string itself contains a quoted-string, then the
        requirement for backslash quoting is recursively applied.  To
        specify the audio/RED payload format in SDP, the pt parameter
        is mapped to an a=fmtp attribute by eliminating the parameter
        name (pt) and changing the commas to slashes.  For example,
        'pt="0,5"' maps to 'a=fmtp:99 0/5'.  A more complicated
        example, with a dynamic payload type, is:

           pt = "0, 103 = \"audio/G729D;annexb=yes\" "

           m=audio 49170 RTP/AVP 99 0 103
           a=rtpmap:99 RED/8000
           a=fmtp:99 0/103
           a=rtpmap:103 G729D/8000
           a=fmtp:103 annexb=yes

In RFC 2198 the following paragraph can be found:

To receive a redundant stream, this is all that is required.  However
   to send a redundant stream, the sender needs to know which codecs are
   recommended for the primary and secondary (and tertiary, etc)
   encodings.  This information is specific to the redundancy format,
   and is specified using an additional attribute "fmtp" which conveys
   format-specific information.  A session directory does not parse the
   values specified in an fmtp attribute but merely hands it to the
   media tool unchanged.  For redundancy, we define the format
   parameters to be a slash "/" separated list of RTP payload types.

   Thus a complete example is:

       m=audio 12345 RTP/AVP 121 0 5
       a=rtpmap:121 red/8000/1
       a=fmtp:121 0/5

   This specifies that the default format for senders is redundancy with
   PCM as the primary encoding and DVI as the secondary encoding.
   Encodings cannot be specified in the fmtp attribute unless they are
   also specified as valid encodings on the media ("m=") line.

So how do we resolve this. I think that the intention of declaring 
primary and secondary codec looks nice but is not really necessary and 
has problems. I think the most important factor is to know which PTs 
that may appear in the RED format so be able to set up the correct 
processing. The reason I think the primary/secondary concept is wrong is 
the case where you have more than two codecs, and the possibility that 
more than one is in fact the primary. An example of this is when having 
congestion control indication that the bandwidht must be reduced. Then 
maybe the best way of doing this is to switch primary encoding while 
still using redundancy. In this case the RFC 2198 language can't express 
it.

This needs to be resolved for the text/RED MIME registration.

Cheers

Magnus Westerlund 

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
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Hi Matthew,

Matthew Romaine wrote:

>As I seek comments from listers on what needs to be fixed and 
>clarified, I have a question about the naming convention.  Must we 
>treat this as a completely new payload format (-00), which would mean 
>committing to an earlier deadline, or can we consider this a continued 
>revision of previous submissions (making this a -04)?
>  
>
The numbering of the version of drafts is in relation to the rest of the 
name. As you have changed it, correctly (in my opinion) as you have 
changed the scope, this draft should be submitted as 00. Please submit 
this version AS IT IS to the internet-drafts@ietf.org to avoid any 
future confusion about the numbering.

Cheers

Magnus Westerlund 

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
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Hi Paul,

[CCing the AVT list as there are issues here that needs open discussion]

Response inline.

Paul E. Jones wrote:

>Magnus,
>
>I'll try to make these changes.  I have questions for clarification:
>
>  
>
>>1. Section 3: rate parameter: I think the "may" in the last sentence
>>should be a normative one.
>>    
>>
>
>What do you mean by "normative"?
>

You should use normative language, in other words the MAY normative word 
that you use through reference to RFC 2119.

>
>  
>
>>5. Section 4, the last bullet for the "pt" list. This is under
>>specified, the following problems exist:
>>-  The removal or not of <"> quoting etc. is not defined.
>>    
>>
>
>Are you suggesting that people would not necessarily know if the quotes
>are
>part of the syntax or not?
>  
>
You are stripping the outer quotation marks, are you leaving the inner 
in place, in there converted form, or do you change to standard quotation?

>  
>
>>- How to handle any "dynamic-payload-type "=" content-type" string is
>>also not explained.
>>    
>>
>
>I do not understand your concern.  Can you explain further?
>  
>

When you are in the SDP domain, you SHOULD NOT / SHALL NOT use the 
pt=content-type structure at all. As the information is available in the 
SDP as defined PT. Therefore you only need to reference these PTs. So 
when having SDP should you in fact remove all of these pt=content-type 
declarations and only use PTs in the list?

I would also like to point out that the this declaration of 
"dynamic-payload-type "=" content-type" is not sufficiently clear. For 
RTP payload formats using parameters to configure this, this string is 
insufficient unless the parameter list can be included in the 
content-type part.

>  
>
>>7. Section 5: A proposal for how to handle the security problem is
>>always good. Proposal: "To prevent modification of the primary,
>>secondary or header information, payload integrity protection over at
>>least the complete RTP packet is recommend, for example using SRTP
>>    
>>
>[]."
>
>I agree with the added text, but concerned about referencing SRTP
>explicitly.  Will the RFC be published before this document?  I do not
>want
>to hold up progress due to a reference.  Would be be OK to leave off
>"for
>example using SRTP"?
>

The SRTP is in the RFC editors queue, so yes, it will be available 
before this document. Put SRTP into the informative reference list and 
write an RFC editors note that request that he replace the draft 
reference with the RFC number when it becomes available.

>
>  
>
>>8. Section 6, The first reference is an informative one, please move
>>    
>>
>it
>  
>
>>to an informative reference list. Also reference 7 is not used at all,
>>please remove.
>>    
>>
>
>I can do that.  What about 7, as well (RFC 3555)?
>
>  
>
>>I expect that the when these things are addressed the document is
>>    
>>
>ready
>  
>
>>for WG last call.
>>    
>>
>
>I've got all changes made, except those that are questions above.
>Looking
>forward to your reply... and thank you very much for your careful
>review.
>  
>

Great!

Magnus Westerlund 

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Magnus,

> >>1. Section 3: rate parameter: I think the "may" in the last sentence
> >>should be a normative one.
> >>
> >>
> >
> >What do you mean by "normative"?
> >
>
> You should use normative language, in other words the MAY normative word
> that you use through reference to RFC 2119.

Got it.. it's now MAY.  As a general question, why would capitalization
suggest a different meaning?

> >>5. Section 4, the last bullet for the "pt" list. This is under
> >>specified, the following problems exist:
> >>-  The removal or not of <"> quoting etc. is not defined.
> >>
> >>
> >
> >Are you suggesting that people would not necessarily know if the quotes
> >are
> >part of the syntax or not?
> >
> >
> You are stripping the outer quotation marks, are you leaving the inner
> in place, in there converted form, or do you change to standard quotation?

Well, this is a good question.  Clearly, I know what the end result is to
be, but how should this be written?  The text I am using was borrowed from
RFC 3555, which uses the same notation.  Please suggest specific changes to
clarify this.


> >>- How to handle any "dynamic-payload-type "=" content-type" string is
> >>also not explained.
> >>
> >>
> >
> >I do not understand your concern.  Can you explain further?
> >
> >
>
> When you are in the SDP domain, you SHOULD NOT / SHALL NOT use the
> pt=content-type structure at all. As the information is available in the
> SDP as defined PT. Therefore you only need to reference these PTs. So
> when having SDP should you in fact remove all of these pt=content-type
> declarations and only use PTs in the list?
>
> I would also like to point out that the this declaration of
> "dynamic-payload-type "=" content-type" is not sufficiently clear. For
> RTP payload formats using parameters to configure this, this string is
> insufficient unless the parameter list can be included in the
> content-type part.

Ah, I understand.  As I mentioned above, this follows the example in RFC
3555, which also does not go into discussion on this (as far as I know).  I
assume the reason for that was simply that the entire focus was on SDP,
anyway, with little focus on MIME, really.

In any case, I can add text in this regard, but I need some guidance as to
what we should have inserted to make this clearer.

> >>7. Section 5: A proposal for how to handle the security problem is
> >>always good. Proposal: "To prevent modification of the primary,
> >>secondary or header information, payload integrity protection over at
> >>least the complete RTP packet is recommend, for example using SRTP
> >>
> >>
> >[]."
> >
> >I agree with the added text, but concerned about referencing SRTP
> >explicitly.  Will the RFC be published before this document?  I do not
> >want
> >to hold up progress due to a reference.  Would be be OK to leave off
> >"for
> >example using SRTP"?
> >
>
> The SRTP is in the RFC editors queue, so yes, it will be available
> before this document. Put SRTP into the informative reference list and
> write an RFC editors note that request that he replace the draft
> reference with the RFC number when it becomes available.

OK.. no problem.

Paul



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Paul,

Comments inline.

Paul E. Jones wrote:

>Got it.. it's now MAY.  As a general question, why would capitalization
>suggest a different meaning?
>  
>
This is due to that you have defined to have a separate meaning when you 
include the following statement in regards to RFC 2119.
      The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL
      NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED",  "MAY", and
      "OPTIONAL" in this document are to be interpreted as described in
      RFC 2119.

The reason why you do this is that in most case you desire to separate 
the cases when something really must be followed, i.e. being normative, 
and when it is only for information and does not carry necessary the 
same strict meaning. You really should read RFC 2119, its only 3 pages.

>Well, this is a good question.  Clearly, I know what the end result is
>to
>be, but how should this be written?  The text I am using was borrowed
>from
>RFC 3555, which uses the same notation.  Please suggest specific changes
>to
>clarify this.
>
>  
>

I think we need to resolve what discrepancy we have between RFC 2198 and 
RFC 3555 before being able to close this issue. However I don't think 
that much text is required, just being explicit about it.

>>>      
>>>
>>When you are in the SDP domain, you SHOULD NOT / SHALL NOT use the
>>pt=content-type structure at all. As the information is available in
>>    
>>
>the
>  
>
>>SDP as defined PT. Therefore you only need to reference these PTs. So
>>when having SDP should you in fact remove all of these pt=content-type
>>declarations and only use PTs in the list?
>>
>>I would also like to point out that the this declaration of
>>"dynamic-payload-type "=" content-type" is not sufficiently clear. For
>>RTP payload formats using parameters to configure this, this string is
>>insufficient unless the parameter list can be included in the
>>content-type part.
>>    
>>
>
>Ah, I understand.  As I mentioned above, this follows the example in RFC
>3555, which also does not go into discussion on this (as far as I know).
>I
>assume the reason for that was simply that the entire focus was on SDP,
>anyway, with little focus on MIME, really.
>
>In any case, I can add text in this regard, but I need some guidance as
>to
>what we should have inserted to make this clearer.
>
>  
>
I would definitely like to have some more input from Colin and others 
about how this problem should be resolved.


Cheers

Magnus Westerlund 

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Colin & Lars-Erik,

Thanks very much for your comments and suggestions.  In addition to all =
of your editorial suggestions, the changes to the updated draft will =
include:

> In a number of places, the draft describes "using MPLS to route ECRTP
> compressed packets over multiple router hops". Does this mean routing
> compressed packets over an MPLS LSP? If so, I recommend clarifying the
> draft to prevent confusion with multiple IP hops.

Changed the language as suggested.
=20
> I'm not sure discussion of access links is appropriate, since MPLS is
> unlikely to be used in this scenario.

We now refer to 'low-speed' links rather than 'access links'.  It's =
debatable how far MPLS will extend.
=20
> In section 3, the requirements list ECRTP without discussion.=20
> I think ECRTP is an appropriate solution, but it would be useful to=20
> motivate the choice by discussing the requirements for the=20
> compression. This is particularly important, since there are=20
> alternative algorithms (e.g. ROHC) that may be considered.

We've added discussion as suggested.  ECRTP should be used to make ECRTP =
over MPLS more tolerant of packet loss, to guard against frequent =
resynchronizations, and to minimize the need for error recovery.  ROHC =
can also be considered, however ROHC does not accommodate packet =
reordering to protect against out-of-sequence packets, which can occur =
on MPLS LSPs.  We didn't change the title of the document, as suggested =
by L-E, since Colin specifically suggested the current title, and that =
also corresponds to the AVT milestone.
=20
> In section 3, the draft notes that protocol extensions may be=20
> required for ECRTP. Are these changes to ECRTP or a mapping of
> ECRTP onto a new "link" layer? If the latter, the draft could
> do with clarification.  The draft should discuss the type of
> extension needed for RSVP-TE and LDP, since this will feed=20
> into any new work plan.

We've added text on protocol extensions required for ECRTP, RSVP-TE, and =
LDP

> Security considerations: Are there no issues with DoS attacks on MPLS
> routers, due to the increased number of flows?

We've added text to discuss DoS attacks in the security section.

> Before publication, the draft will need an IANA Considerations section
> (even if it just says "No IANA actions are required") and the=20
> IPR boiler-plate.

We've added an IANA Considerations Section and Intellectual Property =
Statement Section.

Thanks,
Jerry

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Fri Jan 23 19:17:34 2004
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Date: Fri, 23 Jan 2004 16:14:25 -0800 (PST)
From: Greg Herlein <gregh@arkstream.com>
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Subject: [AVT] RFC2250 (DVB UDP) MPEG2 Encapsulation - Clients?
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Does anyone know of a shipping client that definitely supports 
rfc2250 (or it's big brother, the DVB spec for MPEG2-TS in RTP in 
UDP)?

I'm looking for other clients - software or hardware - that 
actually support this (for real).  

There's a lot of confusion out there regarding what protocols 
folks really mean when they toss of the phrase "MPEG2 in UDP" - 
and I'm trying to nail down firmly which support the right way 
(tm).  <grin>

Greg

_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] Updated RTP MIDI I-Ds ...
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Hi everyone,

         Sent off the updated versions of the RTP MIDI payload
format document (draft-ietf-avt-rtp-midi-format-01.txt) and
implementation guide (draft-ietf-avt-rtp-midi-guidelines-01.txt)
to internet-drafts@ietf.org yesterday.  Pick up copies here:

http://www.cs.berkeley.edu/~lazzaro/sa/pubs/txt/current-rtp-midi.txt
http://www.cs.berkeley.edu/~lazzaro/sa/pubs/txt/current-guide.txt

	All of the bitfields have now been coded into sfront -- code
to send and receive packets, and diagnostic code to print out
packets in flight and to show how the receiver recovers from real
(801.11b) and simulated packet loss.  The change log
describes fixes to bugs that were caught during sfront coding.
Also, the SDP ABNF passes the IETF online validator.

	I think these documents are ready for Last Call --
i.e. this is the "candidate Last Call" release I referenced in
earlier postings to the list.  So, if you've been holding off
reading the documents until I declared them "done" ...
now is a good time to read them.

---
John Lazzaro
http://www.cs.berkeley.edu/~lazzaro
lazzaro [at] cs [dot] berkeley [dot] edu
---


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
CC: "Paul E. Jones" <paulej@packetizer.com>, avt@ietf.org
Subject: Re: [AVT] Comments on draft-ietf-text-red-00.txt
References: <400E9B40.3080206@ericsson.com> <00d901c3e042$cc6a7ca0$e1cd6a9c@cisco.com> <400F914E.9090006@ericsson.com> <00af01c3e0d5$bb3e6430$34cc6a9c@cisco.com> <400FBDAA.1050208@ericsson.com>
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Hi,

I have had some off-line discussion regarding the MIME registration 
problem for text/RED. Know understanding the intention having the 
ordering requirement on RFC 2198 being firm for the "pt" list, I would 
propose the following text change for the "pt" list:

"pt: a comma-separated ordered list enumerating the primary, secondary, 
etc., encoding's RTP payload types in accordance with RFC 2198.  Because 
comma is ..."

Then one addition to the SDP section (4) is also needed in my opinion. 
Please add the following text where appropriate.

"For each redundancy payload type defined the ordering of the primary, 
and redundancy encoding(s) are fixed. If more than one combination of 
primary, and redundancy encoding(s) are desired, multiple redundancy 
payload types needs to be defined."

Then only the last issue below remains to be resolved as I see it.

Cheers

Magnus

Magnus Westerlund wrote:

>
>>>>     
>>>
>>> When you are in the SDP domain, you SHOULD NOT / SHALL NOT use the
>>> pt=content-type structure at all. As the information is available in
>>>   
>>
>> the
>>  
>>
>>> SDP as defined PT. Therefore you only need to reference these PTs. So
>>> when having SDP should you in fact remove all of these pt=content-type
>>> declarations and only use PTs in the list?
>>>
>>> I would also like to point out that the this declaration of
>>> "dynamic-payload-type "=" content-type" is not sufficiently clear. For
>>> RTP payload formats using parameters to configure this, this string is
>>> insufficient unless the parameter list can be included in the
>>> content-type part.
>>>   
>>
>>
>> Ah, I understand.  As I mentioned above, this follows the example in RFC
>> 3555, which also does not go into discussion on this (as far as I know).
>> I
>> assume the reason for that was simply that the entire focus was on SDP,
>> anyway, with little focus on MIME, really.
>>
>> In any case, I can add text in this regard, but I need some guidance as
>> to
>> what we should have inserted to make this clearer.
>>
>>  
>>
> I would definitely like to have some more input from Colin and others 
> about how this problem should be resolved.
>

-- 

Magnus Westerlund 

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Jan 26 20:03:48 2004
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To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
Cc: <avt@ietf.org>
References: <400E9B40.3080206@ericsson.com> <00d901c3e042$cc6a7ca0$e1cd6a9c@cisco.com> <400F914E.9090006@ericsson.com> <00af01c3e0d5$bb3e6430$34cc6a9c@cisco.com> <400FBDAA.1050208@ericsson.com> <4014DABD.3030105@ericsson.com>
Subject: Re: [AVT] Comments on draft-ietf-text-red-00.txt
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Magnus,

> I have had some off-line discussion regarding the MIME registration
> problem for text/RED. Know understanding the intention having the
> ordering requirement on RFC 2198 being firm for the "pt" list, I would
> propose the following text change for the "pt" list:
>
> "pt: a comma-separated ordered list enumerating the primary, secondary,
> etc., encoding's RTP payload types in accordance with RFC 2198.  Because
> comma is ..."

There was a desire early on not to explicitly tie the MIME registration to
RFC 2198.  Certainly, it is the main driver for producing the document, but
it was viewed as not the only potential user of this registration.  Thus, do
we really want this text or just leave it as it is in the current draft?

> Then one addition to the SDP section (4) is also needed in my opinion.
> Please add the following text where appropriate.
>
> "For each redundancy payload type defined the ordering of the primary,
> and redundancy encoding(s) are fixed. If more than one combination of
> primary, and redundancy encoding(s) are desired, multiple redundancy
> payload types needs to be defined."

Perhaps you can educate me here, but why would the order matter?  The
payload format defined in RFC 2198 allows for the PT of the redundant data
to be specified along with the payload.  Thus, it seems that the ordering of
is not so important.  Of course, a distinction must be made between primary
versus redundant, but it is not clear to me why this strict clause is
needed. Please clarify.

> Then only the last issue below remains to be resolved as I see it.

And, how can we close on that issue below?

Paul

>
> Cheers
>
> Magnus
>
> Magnus Westerlund wrote:
>
> >
> >>>>
> >>>
> >>> When you are in the SDP domain, you SHOULD NOT / SHALL NOT use the
> >>> pt=content-type structure at all. As the information is available in
> >>>
> >>
> >> the
> >>
> >>
> >>> SDP as defined PT. Therefore you only need to reference these PTs. So
> >>> when having SDP should you in fact remove all of these pt=content-type
> >>> declarations and only use PTs in the list?
> >>>
> >>> I would also like to point out that the this declaration of
> >>> "dynamic-payload-type "=" content-type" is not sufficiently clear. For
> >>> RTP payload formats using parameters to configure this, this string is
> >>> insufficient unless the parameter list can be included in the
> >>> content-type part.
> >>>
> >>
> >>
> >> Ah, I understand.  As I mentioned above, this follows the example in
RFC
> >> 3555, which also does not go into discussion on this (as far as I
know).
> >> I
> >> assume the reason for that was simply that the entire focus was on SDP,
> >> anyway, with little focus on MIME, really.
> >>
> >> In any case, I can add text in this regard, but I need some guidance as
> >> to
> >> what we should have inserted to make this clearer.
> >>
> >>
> >>
> > I would definitely like to have some more input from Colin and others
> > about how this problem should be resolved.
> >
>
> -- 
>
> Magnus Westerlund
>
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Jan 26 20:03:49 2004
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Subject: [AVT] The need for Offer/Answer sections in RTP payload format specifications
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Hi AVT members,

We AVT chairs would like to make all writers of RTP payload format aware
of one thing. There exist a need for most RTP payload formats to have a
section discussing the usage of the MIME parameters in regards to the
SDP Offer and Answer model defined in RFC 3264. This concerns all RTP
payload formats that use parameters to configure either the payload
format itself or the transported codec.

The reason for these is to achieve the best possible interoperability.
And the interoperability is affected in several ways through the use of
MIME defined parameters.
- Configuration of RTP payload formats and codecs: If one end-point A
offers functionality set SA and end-point B offers another functionality
set SB. Then it is possible that SA has no common subset with SB, thus
preventing interoperability.

- Understanding when offered set SA and supported set SB contains
possibilities for sub-setting. Certain offered sets can be subsetted and
still useful. A MIME specification should be explicit about if
parameters can be downgraded or not. For example, an video profile level
specification can be downgraded to a lower level in the response and
still achieve interoperability. However an AMR payload format
configuration parameter like "octet-align" is binary, i.e. requiring one
to support this configuration.

- Possibility to make recommendations on how to form offers that
maximizes interoperability. However this is application dependent and
should normally be kept in an more open terms. However for example for
the AMR payload format can be configured into several usages. For some
applications it does make sense to offer multiple payload types where
the payload format is configured in different way. For example a
streaming server can offer both a octet-aligned only PT and one with
interleaving if they would use offer answer.

Thus we request that authors write an "Offer-Answer model consideration"
in their SDP usage section. This should cover the following:

- How configurations effect the interoperability.
- What parameters that are possible to subset and which are not.
- Recommendations on how to achieve good interoperability.

Comments on this are solicited.

Best Regards

Magnus & Colin

-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com





_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi Paul,

comments inline:

Paul E. Jones wrote:

>Magnus,
>
>  
>
>>I have had some off-line discussion regarding the MIME registration
>>problem for text/RED. Know understanding the intention having the
>>ordering requirement on RFC 2198 being firm for the "pt" list, I would
>>propose the following text change for the "pt" list:
>>
>>"pt: a comma-separated ordered list enumerating the primary,
>>    
>>
>secondary,
>  
>
>>etc., encoding's RTP payload types in accordance with RFC 2198.
>>    
>>
>Because
>  
>
>>comma is ..."
>>    
>>
>
>There was a desire early on not to explicitly tie the MIME registration
>to
>RFC 2198.  Certainly, it is the main driver for producing the document,
>but
>it was viewed as not the only potential user of this registration.
>Thus, do
>we really want this text or just leave it as it is in the current draft?
>  
>

No, It must be tied to RFC 2198. In my recollection the opinion 
expressed was to not tie it to RFC 2793. I think we should go with the 
new text, because I find the old text confusing as it does not inform 
explicitly that about the semantics of the parameter.

>>Then only the last issue below, remains to be resolved as I see it.
>>    
>>
>
>And, how can we close on that issue below?
>  
>

I don't know really, it seems that we should be compatible with what is 
used for audio/RED. However what is really used here is not totally 
clear. Based on the examples in RFC 2198 one does rely on having dynamic 
PTs to be declared through "a=rtpmap".

What is you others opinion about this?

Cheers

Magnus

>  
>
>>Cheers
>>
>>Magnus
>>
>>Magnus Westerlund wrote:
>>
>>    
>>
>>>>>When you are in the SDP domain, you SHOULD NOT / SHALL NOT use the
>>>>>pt=content-type structure at all. As the information is available
>>>>>          
>>>>>
>in
>  
>
>>>>the
>>>>
>>>>
>>>>        
>>>>
>>>>>SDP as defined PT. Therefore you only need to reference these PTs.
>>>>>          
>>>>>
>So
>  
>
>>>>>when having SDP should you in fact remove all of these
>>>>>          
>>>>>
>pt=content-type
>  
>
>>>>>declarations and only use PTs in the list?
>>>>>
>>>>>I would also like to point out that the this declaration of
>>>>>"dynamic-payload-type "=" content-type" is not sufficiently clear.
>>>>>          
>>>>>
>For
>  
>
>>>>>RTP payload formats using parameters to configure this, this
>>>>>          
>>>>>
>string is
>  
>
>>>>>insufficient unless the parameter list can be included in the
>>>>>content-type part.
>>>>>
>>>>>          
>>>>>
>>>>Ah, I understand.  As I mentioned above, this follows the example
>>>>        
>>>>
>in
>RFC
>  
>
>>>>3555, which also does not go into discussion on this (as far as I
>>>>        
>>>>
>know).
>  
>
>>>>I
>>>>assume the reason for that was simply that the entire focus was on
>>>>        
>>>>
>SDP,
>  
>
>>>>anyway, with little focus on MIME, really.
>>>>
>>>>In any case, I can add text in this regard, but I need some
>>>>        
>>>>
>guidance as
>  
>
>>>>to
>>>>what we should have inserted to make this clearer.
>>>>
>>>>
>>>>
>>>>        
>>>>
>>>I would definitely like to have some more input from Colin and
>>>      
>>>
>others
>  
>
>>>about how this problem should be resolved.
>>>
>>>      
>>>
>>-- 
>>
>>Magnus Westerlund
>>
>>Multimedia Technologies, Ericsson Research EAB/TVA/A
>>----------------------------------------------------------------------
>>Ericsson AB                | Phone +46 8 4048287
>>Torshamsgatan 23           | Fax   +46 8 7575550
>>S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>>
>>
>>    
>>
>
>  
>

-- 

Magnus Westerlund 

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Jan 27 04:17:36 2004
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Subject: [AVT] FW: draft-burmeister-avt-rtcp-feedback-sim-03.txt
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Dear all,

the Last Call period for the RTCP Feedback Sims draft has ended. ADs had
minor comments addressed by this update. The revised draft
(draft-burmeister-avt-rtcp-feedback-sim-03.txt) was sent yesterday to the=
 ID
manager.

Meanwhile you can download it from:
http://www.pel.panasonic.de/ietf/docs/draft-burmeister-avt-rtcp-feedback-=
sim
-03.txt



cheers,

Jos=E9

> -----Original Message-----
> From: Jose Rey [mailto:rey@panasonic.de]
> Sent: Monday, January 26, 2004 5:46 PM
> To: internet-drafts@ietf.org
> Cc: magnus.westerlund@era.ericsson.se; Colin Perkins
> Subject: draft-burmeister-avt-rtcp-feedback-sim-03.txt
>
>
>
> Dear I-D editors,
>
>   this is to submit the following Internet Draft for continued
>   discussion in the AVT Working Group:
>
>        Title:    Extended RTP Profile for RTCP-based Feedback
>                  - Results of the Timing Rule Simulations -
>        Authors:   Burmeister, et al.
>        Date:      January, 26th 2004
>        File name: draft-burmeister-avt-rtcp-feedback-sim-03.txt
>
>
>
>   cheers,
>  Jos=E9


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Tue, 27 Jan 2004 09:21:13 +0000
From: Colin Perkins <csp@csperkins.org>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Cc: paulej@packetizer.com, avt@ietf.org
Subject: Re: [AVT] Comments on draft-ietf-text-red-00.txt
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--> Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
... 
> I don't know really, it seems that we should be compatible with what is 
> used for audio/RED. However what is really used here is not totally 
> clear. Based on the examples in RFC 2198 one does rely on having dynamic 
> PTs to be declared through "a=rtpmap".
> 
> What is you others opinion about this?

RFC 2198 requires strict ordering so the sender knows which is the primary
encoding, and which is the redundant. 

Colin

_______________________________________________
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From: "Jose Rey" <rey@panasonic.de>
To: <avt@ietf.org>
Cc: "Colin Perkins" <csp@csperkins.org>,
        "Magnus Westerlund" <magnus.westerlund@era.ericsson.se>,
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Date: Tue, 27 Jan 2004 10:26:17 +0100
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Subject: [AVT] FW: draft-ietf-avt-rtp-retransmission-10.txt
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Dear all,


the last revision of the RTP Retransmission Payload Format was already se=
nt
to the ID manager.

Meanwhile, you can download it under:

http://www.pel.panasonic.de/ietf/docs/draft-ietf-avt-rtp-retransmission-1=
0.t
xt


Changes:
+ some comments regarding the clarification of the IPR issues have been
addressed in the introduction.
+ additionally, some normative language nits were corrected in section 4.
+ no changes in the contents

Colin, I think it is appropriate to re-issue Last Call at this point

cheers,

Jos=E9

> -----Original Message-----
> From: Jose Rey [mailto:rey@panasonic.de]
> Sent: Monday, January 26, 2004 5:46 PM
> To: IETF I-D Submission
> Cc: Colin Perkins; Magnus Westerlund; David Leon
> Subject: draft-ietf-avt-rtp-retransmission-10.txt
>
>
>
>  Dear I-D editors,
>
>  this is to submit the following Internet Draft for continued
>  discussion in the AVT Working Group:
>
>       Title:     RTP Retransmission Payload Format
>       Authors:   Rey, et al.
>       Date:      January, 26th 2004
>       File name: draft-ietf-avt-rtp-retransmission-10.txt
>
>
>
>  cheers,
>
> Jose


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Belling Thomas <thomas.belling@siemens.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, avt@ietf.org
Cc: Colin Perkins <csp@csperkins.org>
Subject: RE: [AVT] The need for Offer/Answer sections in RTP payload forma
	t specifications
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Dear Magnus and Collin,

thank you for this initiative.
We also had some internal dicussions about the AMR RTP payload format, =
RFC 3267, with that respect.
Is there any intention to remmedy this, e.g. by updating the RFC or =
writing an additional one?

Best regards, Thomas


----------------------------------
Dr. Thomas Belling
Siemens AG          =20
Sankt-Martin-Str. 76=20
D-81541 M=FCnchen Germany
ICM N PG SP ST N1
MCH M 34 307
Tel     +49 89 636 75207
Fax     +49 89 636 75577
Mobile  +49 172 2974678
Email   Thomas.Belling@siemens.com




-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On Behalf Of =
Magnus Westerlund
Sent: Monday, January 26, 2004 2:54 PM
To: avt@ietf.org
Cc: Colin Perkins
Subject: [AVT] The need for Offer/Answer sections in RTP payload format =
specifications


Hi AVT members,

We AVT chairs would like to make all writers of RTP payload format =
aware
of one thing. There exist a need for most RTP payload formats to have a
section discussing the usage of the MIME parameters in regards to the
SDP Offer and Answer model defined in RFC 3264. This concerns all RTP
payload formats that use parameters to configure either the payload
format itself or the transported codec.

The reason for these is to achieve the best possible interoperability.
And the interoperability is affected in several ways through the use of
MIME defined parameters.
- Configuration of RTP payload formats and codecs: If one end-point A
offers functionality set SA and end-point B offers another =
functionality
set SB. Then it is possible that SA has no common subset with SB, thus
preventing interoperability.

- Understanding when offered set SA and supported set SB contains
possibilities for sub-setting. Certain offered sets can be subsetted =
and
still useful. A MIME specification should be explicit about if
parameters can be downgraded or not. For example, an video profile =
level
specification can be downgraded to a lower level in the response and
still achieve interoperability. However an AMR payload format
configuration parameter like "octet-align" is binary, i.e. requiring =
one
to support this configuration.

- Possibility to make recommendations on how to form offers that
maximizes interoperability. However this is application dependent and
should normally be kept in an more open terms. However for example for
the AMR payload format can be configured into several usages. For some
applications it does make sense to offer multiple payload types where
the payload format is configured in different way. For example a
streaming server can offer both a octet-aligned only PT and one with
interleaving if they would use offer answer.

Thus we request that authors write an "Offer-Answer model =
consideration"
in their SDP usage section. This should cover the following:

- How configurations effect the interoperability.
- What parameters that are possible to subset and which are not.
- Recommendations on how to achieve good interoperability.

Comments on this are solicited.

Best Regards

Magnus & Colin

--=20

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com





_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi Thomas,

Belling Thomas wrote:

>Dear Magnus and Collin,
>
>thank you for this initiative.
>We also had some internal dicussions about the AMR RTP payload format,
>RFC 3267, with that respect.
>Is there any intention to remmedy this, e.g. by updating the RFC or
>writing an additional one?
>
>Best regards, Thomas
>
>  
>

We authors are planning on updating RFC 3267 to go draft standard, then 
we can address this.. We will also fix some small issues with the text, 
however the current errata for RFC 3267 is very short.

I have written an initial proposal for this text which I include below, 
however please remember that it is a first version. Comments are welcome.

Cheers

Magnus

---------- Text Proposal for new chapter 8.3 -------------
8.3. Mapping MIME Parameters into SDP

  The information carried in the MIME media type specification has a
  specific mapping to fields in the Session Description Protocol (SDP)
  [11], which is commonly used to describe RTP sessions.  When SDP is
  used to specify sessions employing the AMR or AMR-WB codec, the
  mapping is as follows:

     -  The MIME type ("audio") goes in SDP "m=" as the media name.

     -  The MIME subtype (payload format name) goes in SDP "a=rtpmap"
        as the encoding name.  The RTP clock rate in "a=rtpmap" MUST be
        8000 for AMR and 16000 for AMR-WB, and the encoding parameters
        (number of channels) MUST either be explicitly set to N or
        omitted, implying a default value of 1.  The values of N that
        are allowed is specified in Section 4.1 in [24].

     -  The parameters "ptime" and "maxptime" go in the SDP "a=ptime"
        and "a=maxptime" attributes, respectively.

     -  Any remaining parameters go in the SDP "a=fmtp" attribute by
        copying them directly from the MIME media type string as a
        semicolon separated list of parameter=value pairs.

8.3.1 Offer-Answer Model Considerations

To achieve good interoperability with the AMR or the AMR-WB RTP payload
in an Offer-Answer usage in SDP the following considerations should be made:

- Each combination of the RTP payload configuration parameters
(octet-align, crc, robust-sorting, interleaving, and channels) is unique
in its bit-pattern and not compatible with any other combination. Due to
the application dependent nature of any configuration and they being
optionally to implement, care must be taken. When creating an offer in
an application desiring to use the more advance features (crc,
robust-sorting, interleaving, or more than one channel), the offerer is
RECOMMENDED to also offer an payload type containing only the
octet-align configuration with a single channel. If multiple
configurations are of interest to the application they may all be
offered, however care should be taken to not offer too many payload types.

- The parameters "mode-set", "mode-change-period", and
"mode-change-neighbor" SHOULD only be used when really needed due to
gateway scenarios, however their nature are such that they must either
be supported or no interoperability exist.

- The parameters "maxptime" and "ptime" should in most cases not effect
the interoperability, however the setting of the parameters can effect
the performance of the application.

8.3.2 Examples

  Some example SDP session descriptions utilizing AMR and AMR-WB
  encodings follow.  In these examples, long a=fmtp lines are folded to
  meet the column width constraints of this document; the backslash
  ("\") at the end of a line and the carriage return that follows it
  should be ignored.

  Example of usage of AMR in a possible GSM gateway scenario:

   m=audio 49120 RTP/AVP 97
   a=rtpmap:97 AMR/8000/1
   a=fmtp:97 mode-set=0,2,5,7; mode-change-period=2; \
     mode-change-neighbor=1
   a=maxptime:20

  Example of usage of AMR-WB in a possible VoIP scenario where UEP may
be used (99) and a fallback declaration (98):

   m=audio 49120 RTP/AVP 99 98
   a=rtpmap:99 AMR-WB/16000
   a=fmtp:99 octet-align=1; crc=1
   a=rtpmap:98 AMR-WB/16000
   a=fmtp:98 octet-align=1

  Example of usage of AMR-WB in a possible streaming scenario (two
  channel stereo):

   m=audio 49120 RTP/AVP 99 100
   a=rtpmap:99 AMR-WB/16000/2
   a=fmtp:99 interleaving=30
   a=maxptime:100

  Note that the payload format (encoding) names are commonly shown in
  upper case.  MIME subtypes are commonly shown in lower case.  These
  names are case-insensitive in both places.  Similarly, parameter
  names are case-insensitive both in MIME types and in the default
  mapping to the SDP a=fmtp attribute.



-- 

Magnus Westerlund 

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Tue, 27 Jan 2004 12:13:08 +0100
From: Franceschini Guido <Guido.Franceschini@TILAB.COM>
Subject: AMR PT (was RE: [AVT] The need for Offer/Answer sections in RTP
 payload forma t specifications)
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Cc: avt@ietf.org
Message-id: <3737D9839ED3D3408C73611BDA907A043522CF@EXC2K05A.cselt.it>
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Dear all,

since I have just read that you are considering to update RFC3267, here =
is another possible issue.

I am recently tracking an interoperability problem with my =
implementation of RFC3267, which uses the Dynamic PT.
The problem is that the other equipment uses the static PT=3D3, assigned =
to GSM according to RFC3551.

I have only found this statement on PT in RFC3267 :

"The assignment of an RTP payload type for this new packet format is
outside the scope of this document, and will not be specified here.
It is expected that the RTP profile under which this payload format
is being used will assign a payload type for this encoding or specify
that the payload type is to be bound dynamically."

Therefore my question is: shall we use the static PT=3D3 or the dynamic =
PT ?

Thanks in advance
Guido Franceschini

TILAB - Multimedia Division
Via G.Reiss Romoli 274
I-10148 Torino, Italy
tel + 39 011 228 6137
fax + 39 011 228 6299



-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Magnus
Westerlund
Sent: Tuesday, January 27, 2004 11:57 AM
To: Belling Thomas
Cc: avt@ietf.org
Subject: Re: [AVT] The need for Offer/Answer sections in RTP payload
forma t specifications


Hi Thomas,

Belling Thomas wrote:

>Dear Magnus and Collin,
>
>thank you for this initiative.
>We also had some internal dicussions about the AMR RTP payload format,
>RFC 3267, with that respect.
>Is there any intention to remmedy this, e.g. by updating the RFC or
>writing an additional one?
>
>Best regards, Thomas
>
> =20
>

We authors are planning on updating RFC 3267 to go draft standard, then=20
we can address this.. We will also fix some small issues with the text,=20
however the current errata for RFC 3267 is very short.

I have written an initial proposal for this text which I include below,=20
however please remember that it is a first version. Comments are =
welcome.

Cheers

Magnus

---------- Text Proposal for new chapter 8.3 -------------
8.3. Mapping MIME Parameters into SDP

  The information carried in the MIME media type specification has a
  specific mapping to fields in the Session Description Protocol (SDP)
  [11], which is commonly used to describe RTP sessions.  When SDP is
  used to specify sessions employing the AMR or AMR-WB codec, the
  mapping is as follows:

     -  The MIME type ("audio") goes in SDP "m=3D" as the media name.

     -  The MIME subtype (payload format name) goes in SDP "a=3Drtpmap"
        as the encoding name.  The RTP clock rate in "a=3Drtpmap" MUST =
be
        8000 for AMR and 16000 for AMR-WB, and the encoding parameters
        (number of channels) MUST either be explicitly set to N or
        omitted, implying a default value of 1.  The values of N that
        are allowed is specified in Section 4.1 in [24].

     -  The parameters "ptime" and "maxptime" go in the SDP "a=3Dptime"
        and "a=3Dmaxptime" attributes, respectively.

     -  Any remaining parameters go in the SDP "a=3Dfmtp" attribute by
        copying them directly from the MIME media type string as a
        semicolon separated list of parameter=3Dvalue pairs.

8.3.1 Offer-Answer Model Considerations

To achieve good interoperability with the AMR or the AMR-WB RTP payload
in an Offer-Answer usage in SDP the following considerations should be =
made:

- Each combination of the RTP payload configuration parameters
(octet-align, crc, robust-sorting, interleaving, and channels) is unique
in its bit-pattern and not compatible with any other combination. Due to
the application dependent nature of any configuration and they being
optionally to implement, care must be taken. When creating an offer in
an application desiring to use the more advance features (crc,
robust-sorting, interleaving, or more than one channel), the offerer is
RECOMMENDED to also offer an payload type containing only the
octet-align configuration with a single channel. If multiple
configurations are of interest to the application they may all be
offered, however care should be taken to not offer too many payload =
types.

- The parameters "mode-set", "mode-change-period", and
"mode-change-neighbor" SHOULD only be used when really needed due to
gateway scenarios, however their nature are such that they must either
be supported or no interoperability exist.

- The parameters "maxptime" and "ptime" should in most cases not effect
the interoperability, however the setting of the parameters can effect
the performance of the application.

8.3.2 Examples

  Some example SDP session descriptions utilizing AMR and AMR-WB
  encodings follow.  In these examples, long a=3Dfmtp lines are folded =
to
  meet the column width constraints of this document; the backslash
  ("\") at the end of a line and the carriage return that follows it
  should be ignored.

  Example of usage of AMR in a possible GSM gateway scenario:

   m=3Daudio 49120 RTP/AVP 97
   a=3Drtpmap:97 AMR/8000/1
   a=3Dfmtp:97 mode-set=3D0,2,5,7; mode-change-period=3D2; \
     mode-change-neighbor=3D1
   a=3Dmaxptime:20

  Example of usage of AMR-WB in a possible VoIP scenario where UEP may
be used (99) and a fallback declaration (98):

   m=3Daudio 49120 RTP/AVP 99 98
   a=3Drtpmap:99 AMR-WB/16000
   a=3Dfmtp:99 octet-align=3D1; crc=3D1
   a=3Drtpmap:98 AMR-WB/16000
   a=3Dfmtp:98 octet-align=3D1

  Example of usage of AMR-WB in a possible streaming scenario (two
  channel stereo):

   m=3Daudio 49120 RTP/AVP 99 100
   a=3Drtpmap:99 AMR-WB/16000/2
   a=3Dfmtp:99 interleaving=3D30
   a=3Dmaxptime:100

  Note that the payload format (encoding) names are commonly shown in
  upper case.  MIME subtypes are commonly shown in lower case.  These
  names are case-insensitive in both places.  Similarly, parameter
  names are case-insensitive both in MIME types and in the default
  mapping to the SDP a=3Dfmtp attribute.



--=20

Magnus Westerlund=20

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt


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Date: Tue, 27 Jan 2004 13:24:22 +0200
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Franceschini Guido <Guido.Franceschini@TILAB.COM> writes:
>The problem is that the other equipment uses the static PT=3,
>assigned to GSM according to RFC3551.

That is GSM-FR codec, completely different beast.

>Therefore my question is: shall we use the static PT=3 or the dynamic PT ?

Use dynamic PT.

--Pekka

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



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To: avt@ietf.org
From: "John Lazzaro" <lazzaro@eecs.berkeley.edu>
Subject: Re: [AVT] The need for Offer/Answer sections in RTP payload format specifications
Date: Tue, 27 Jan 2004 11:27:06 -0800
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> Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
>
> Thus we request that authors write an "Offer-Answer model 
> consideration"
> in their SDP usage section. This should cover the following:
>
> - How configurations effect the interoperability.
> - What parameters that are possible to subset and which are not.
> - Recommendations on how to achieve good interoperability.

In regards to RTP MIDI, I think it may be better for the readers
of the "candidate Last Call" document to judge if the SDP
sections as written (pages 20-24, and pages 68-103) handle
Offer/Answer OK, and if the best solution is a new separate
Offer/Answer section or a rewrite of the existing structure to
discuss Offer/Answer in more depth.  This way, I can handle
those criticisms along with all of the other Last Call feedback
at one time, in the hopes of reducing the total number of
rewrite cycles before "candidate Last Call" becomes "Last Call".

The alternative (going off and doing -02.txt versions to become
the new "candidate Last Call", with a separate Offer/Answer section)
probably knocks a month off the schedule.  Not good ...

My own intuition is that the RTP MIDI document is OK as it
stands in regard to interop -- the world of RTP MIDI applications
is big, and interoperability is going to require a framework
document for each class of applications, written by application
experts.  The purpose of the SDP section in the payload document
is to define a common set of SDP tools that these frameworks
can subset.

---
John Lazzaro
http://www.cs.berkeley.edu/~lazzaro
lazzaro [at] cs [dot] berkeley [dot] edu
---


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
Cc: <avt@ietf.org>
References: <400E9B40.3080206@ericsson.com> <00d901c3e042$cc6a7ca0$e1cd6a9c@cisco.com> <400F914E.9090006@ericsson.com> <00af01c3e0d5$bb3e6430$34cc6a9c@cisco.com> <400FBDAA.1050208@ericsson.com> <4014DABD.3030105@ericsson.com> <00a001c3e40e$7c61b8c0$e1cd6a9c@cisco.com> <40162693.4040805@ericsson.com>
Subject: Re: [AVT] Comments on draft-ietf-text-red-00.txt
Date: Wed, 28 Jan 2004 04:13:59 -0500
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Magnus,

> >>"pt: a comma-separated ordered list enumerating the primary,
> >secondary,
> >>etc., encoding's RTP payload types in accordance with RFC 2198.
> >Because
> >>comma is ..."
> >>
> >>
> >
> >There was a desire early on not to explicitly tie the MIME registration
> >to
> >RFC 2198.  Certainly, it is the main driver for producing the document,
> >but
> >it was viewed as not the only potential user of this registration.
> >Thus, do
> >we really want this text or just leave it as it is in the current draft?
> >
> >
>
> No, It must be tied to RFC 2198. In my recollection the opinion
> expressed was to not tie it to RFC 2793. I think we should go with the
> new text, because I find the old text confusing as it does not inform
> explicitly that about the semantics of the parameter.

Sorry... I had the RFC numbers mixed up when I was responding.  Yes, you're
correct-- it was RFC 2793 that was the concern... no problem with your
proposed text.  I'll make those changes.

> >>Then only the last issue below, remains to be resolved as I see it.
> >And, how can we close on that issue below?
>
> I don't know really, it seems that we should be compatible with what is
> used for audio/RED. However what is really used here is not totally
> clear. Based on the examples in RFC 2198 one does rely on having dynamic
> PTs to be declared through "a=rtpmap".
>
> What is you others opinion about this?

Perhaps pointing out a concrete problem that might result in SDP might help
lead to text that resolves the concerns.

Paul

> Cheers
>
> Magnus
>
> >
> >
> >>Cheers
> >>
> >>Magnus
> >>
> >>Magnus Westerlund wrote:
> >>
> >>
> >>
> >>>>>When you are in the SDP domain, you SHOULD NOT / SHALL NOT use the
> >>>>>pt=content-type structure at all. As the information is available
> >>>>>
> >>>>>
> >in
> >
> >
> >>>>the
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>>SDP as defined PT. Therefore you only need to reference these PTs.
> >>>>>
> >>>>>
> >So
> >
> >
> >>>>>when having SDP should you in fact remove all of these
> >>>>>
> >>>>>
> >pt=content-type
> >
> >
> >>>>>declarations and only use PTs in the list?
> >>>>>
> >>>>>I would also like to point out that the this declaration of
> >>>>>"dynamic-payload-type "=" content-type" is not sufficiently clear.
> >>>>>
> >>>>>
> >For
> >
> >
> >>>>>RTP payload formats using parameters to configure this, this
> >>>>>
> >>>>>
> >string is
> >
> >
> >>>>>insufficient unless the parameter list can be included in the
> >>>>>content-type part.
> >>>>>
> >>>>>
> >>>>>
> >>>>Ah, I understand.  As I mentioned above, this follows the example
> >>>>
> >>>>
> >in
> >RFC
> >
> >
> >>>>3555, which also does not go into discussion on this (as far as I
> >>>>
> >>>>
> >know).
> >
> >
> >>>>I
> >>>>assume the reason for that was simply that the entire focus was on
> >>>>
> >>>>
> >SDP,
> >
> >
> >>>>anyway, with little focus on MIME, really.
> >>>>
> >>>>In any case, I can add text in this regard, but I need some
> >>>>
> >>>>
> >guidance as
> >
> >
> >>>>to
> >>>>what we should have inserted to make this clearer.
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>I would definitely like to have some more input from Colin and
> >>>
> >>>
> >others
> >
> >
> >>>about how this problem should be resolved.
> >>>
> >>>
> >>>
> >>-- 
> >>
> >>Magnus Westerlund
> >>
> >>Multimedia Technologies, Ericsson Research EAB/TVA/A
> >>----------------------------------------------------------------------
> >>Ericsson AB                | Phone +46 8 4048287
> >>Torshamsgatan 23           | Fax   +46 8 7575550
> >>S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> >>
> >>
> >>
> >>
> >
> >
> >
>
> -- 
>
> Magnus Westerlund
>
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Jan 28 04:57:45 2004
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From: "Jose Rey" <rey@panasonic.de>
To: <avt@ietf.org>
Date: Wed, 28 Jan 2004 10:54:05 +0100
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Subject: [AVT] FW: draft-burmeister-avt-rtcp-feedback-sim-04.txt
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Hi all,

FYI: the feedback sims draft -04 (with corrected email addresses) has bee=
n
submitted the ID manager, except for that it is the same exact document
as -03.


http://www.pel.panasonic.de/ietf/docs/draft-burmeister-avt-rtcp-feedback-=
sim
-04.txt

cheers,

Jos=E9

> -----Original Message-----
> From: Jose Rey [mailto:rey@panasonic.de]
> Sent: Wednesday, January 28, 2004 10:46 AM
> To: internet-drafts@ietf.org
> Cc: magnus.westerlund@era.ericsson.se; Colin Perkins
> Subject: draft-burmeister-avt-rtcp-feedback-sim-04.txt
>
>
>
> Dear I-D editors,
>
>   this is to submit the following Internet Draft for continued
>   discussion in the AVT Working Group:
>
>        Title:    Extended RTP Profile for RTCP-based Feedback
>                  - Results of the Timing Rule Simulations -
>        Authors:   Burmeister, et al.
>        Date:      January, 28th 2004
>        File name: draft-burmeister-avt-rtcp-feedback-sim-04.txt
>
>
>
>   cheers,
>  Jos=E9


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi Paul,

Good, that resolves all but the last issue. Which I will try to 
recapitulate here:

The MIME type parameter declaration declares that any dynamic payload 
type is to be expressed in the form "PT=content-type". However when 
mapping this into a SDP it seems strange to use this form, as any 
dynamic PT is already declared in the SDP using "a=rtpmap". So I think 
the mapping to SDP text should explain how this should be done in the 
best way.

My proposal would be to add something like the following into the bullet 
about how to handle the "pt" parameter in section 4.

"Any dynamic payload type in the list, SHALL NOT include its 
content-type, only the payload type number. The mapping of payload types 
to the content-type is done using the normal SDP procedures with 
"a=rtpmap". "

Would this be satisfactory?

Cheers

Magnus Westerlund 

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
Cc: <avt@ietf.org>
References: <400E9B40.3080206@ericsson.com> <00d901c3e042$cc6a7ca0$e1cd6a9c@cisco.com> <400F914E.9090006@ericsson.com> <00af01c3e0d5$bb3e6430$34cc6a9c@cisco.com> <400FBDAA.1050208@ericsson.com> <4014DABD.3030105@ericsson.com> <00a001c3e40e$7c61b8c0$e1cd6a9c@cisco.com> <40162693.4040805@ericsson.com> <002101c3e57f$11904260$e1cd6a9c@cisco.com> <40179E9F.3000909@ericsson.com>
Subject: Re: [AVT] Comments on draft-ietf-text-red-00.txt
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Magnus,

That sounds fine to me.  I'll insert that and make the complete text
available again as a TD within a week or so.  Of course, further input is
certainly welcome.

Paul

----- Original Message ----- 
From: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
To: "Paul E. Jones" <paulej@packetizer.com>
Cc: <avt@ietf.org>
Sent: Wednesday, January 28, 2004 6:35 AM
Subject: Re: [AVT] Comments on draft-ietf-text-red-00.txt


> Hi Paul,
>
> Good, that resolves all but the last issue. Which I will try to
> recapitulate here:
>
> The MIME type parameter declaration declares that any dynamic payload
> type is to be expressed in the form "PT=content-type". However when
> mapping this into a SDP it seems strange to use this form, as any
> dynamic PT is already declared in the SDP using "a=rtpmap". So I think
> the mapping to SDP text should explain how this should be done in the
> best way.
>
> My proposal would be to add something like the following into the bullet
> about how to handle the "pt" parameter in section 4.
>
> "Any dynamic payload type in the list, SHALL NOT include its
> content-type, only the payload type number. The mapping of payload types
> to the content-type is done using the normal SDP procedures with
> "a=rtpmap". "
>
> Would this be satisfactory?
>
> Cheers
>
> Magnus Westerlund
>
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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--NextPart

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

	Title		: An Implementation Guide for RTP MIDI
	Author(s)	: J. Lazzaro
	Filename	: draft-ietf-avt-rtp-midi-guidelines-01.txt
	Pages		: 35
	Date		: 2004-1-27
	
This memo offers non-normative implementation guidance for the RTP
     MIDI payload format.  The memo presents its advice in the context
     of a network musical performance application.  In this application
     two musicians, located in different physical locations, interact
     over a network to perform as they would if located in the same
     room.  Underlying the performances are RTP MIDI sessions over
     unicast UDP.  Algorithms for sending and receiving recovery
     journals (the resiliency structure for the payload format) are
     described in detail.  Although the memo focuses on network musical
     performance, the presented implementation advice is relevant to
     other RTP MIDI applications.

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

	Title		: RTP Payload Format for MIDI
	Author(s)	: J. Lazzaro
	Filename	: draft-ietf-avt-rtp-midi-format-01.txt
	Pages		: 119
	Date		: 2004-1-27
	
This memo describes an RTP payload format for the MIDI command
language.  The format encodes all commands that may legally appear
on a MIDI 1.0 DIN cable.  The format is suitable for interactive
applications (such as the remote operation of musical instruments)
and content-delivery applications (such as file streaming).  The
format may be used over unicast and multicast UDP as well as TCP,
and defines tools for graceful recovery from packet loss.  Stream
behavior, including the MIDI rendering method, may be customized
during session setup.  The format also serves as a mode for the
mpeg4-generic format, to support the MPEG 4 Audio Object Types for
General MIDI, Downloadable Sounds Level 2, and Structured Audio.

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From: "Ash, Gerald R (Jerry), ALABS" <gash@att.com>
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Cc: "Ash, Gerald R (Jerry), ALABS" <gash@att.com>
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Subject: [AVT] I-D ACTION:draft-ash-avt-ecrtp-over-mpls-reqs-01.txt
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All,

Please review and comment on the current draft of 'Requirements for =
ECRTP over MPLS' =
http://www.ietf.org/internet-drafts/draft-ash-avt-ecrtp-over-mpls-reqs-01=
.txt.

The draft has been updated according to the comments received (see =
attached email).  Let's keep the discussion going on the list so that we =
can converge on this draft prior to IETF-59 (week of March 1), and meet =
the AVT charter item milestone:

Mar 2004 Finish requirements for ECRTP over MPLS; re-charter for =
subsequent work

Thanks,
Jerry

-----Original Message-----
From: Ash, Gerald R (Jerry), ALABS=20
Sent: Thursday, January 22, 2004 9:24 AM
To: Lars-Erik Jonsson (LU/EAB); Colin Perkins; avt@ietf.org
Cc: Ash, Gerald R (Jerry), ALABS
Subject: [AVT] RE: I-D ACTION:draft-ash-avt-ecrtp-over-mpls-reqs-00.txt

Colin & Lars-Erik,

Thanks very much for your comments and suggestions.  In addition to all =
of your editorial suggestions, the changes to the updated draft will =
include:

> In a number of places, the draft describes "using MPLS to route ECRTP
> compressed packets over multiple router hops". Does this mean routing
> compressed packets over an MPLS LSP? If so, I recommend clarifying the
> draft to prevent confusion with multiple IP hops.

Changed the language as suggested.
=20
> I'm not sure discussion of access links is appropriate, since MPLS is
> unlikely to be used in this scenario.

We now refer to 'low-speed' links rather than 'access links'.  It's =
debatable how far MPLS will extend.
=20
> In section 3, the requirements list ECRTP without discussion.=20
> I think ECRTP is an appropriate solution, but it would be useful to=20
> motivate the choice by discussing the requirements for the=20
> compression. This is particularly important, since there are=20
> alternative algorithms (e.g. ROHC) that may be considered.

We've added discussion as suggested.  ECRTP should be used to make ECRTP =
over MPLS more tolerant of packet loss, to guard against frequent =
resynchronizations, and to minimize the need for error recovery.  ROHC =
can also be considered, however ROHC does not accommodate packet =
reordering to protect against out-of-sequence packets, which can occur =
on MPLS LSPs.  We didn't change the title of the document, as suggested =
by L-E, since Colin specifically suggested the current title, and that =
also corresponds to the AVT milestone.
=20
> In section 3, the draft notes that protocol extensions may be=20
> required for ECRTP. Are these changes to ECRTP or a mapping of
> ECRTP onto a new "link" layer? If the latter, the draft could
> do with clarification.  The draft should discuss the type of
> extension needed for RSVP-TE and LDP, since this will feed=20
> into any new work plan.

We've added text on protocol extensions required for ECRTP, RSVP-TE, and =
LDP

> Security considerations: Are there no issues with DoS attacks on MPLS
> routers, due to the increased number of flows?

We've added text to discuss DoS attacks in the security section.

> Before publication, the draft will need an IANA Considerations section
> (even if it just says "No IANA actions are required") and the=20
> IPR boiler-plate.

We've added an IANA Considerations Section and Intellectual Property =
Statement Section.

Thanks,
Jerry

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Magnus,

This is a good thing. One point I would add to it is this. The 
offer/answer model has two ways you can interpret parameters that 
occur in SDP. Either they are *negotiated*, such that the aim of 
offer/answer is to achieve a common value(s) for the parameters that 
are shared between both sides, and the other is simply a statement of 
"this is what I am going to use", and both sides can have totally 
different values. This is called "declarative". It is important that 
the SDP usage section be very clear on what the intended semantic is.

As examples, the set of codecs is negotiated, but the packetization 
intervals (ptime) are declarative.

-Jonathan R.

Magnus Westerlund wrote:

> Hi AVT members,
> 
> We AVT chairs would like to make all writers of RTP payload format aware
> of one thing. There exist a need for most RTP payload formats to have a
> section discussing the usage of the MIME parameters in regards to the
> SDP Offer and Answer model defined in RFC 3264. This concerns all RTP
> payload formats that use parameters to configure either the payload
> format itself or the transported codec.
> 
> The reason for these is to achieve the best possible interoperability.
> And the interoperability is affected in several ways through the use of
> MIME defined parameters.
> - Configuration of RTP payload formats and codecs: If one end-point A
> offers functionality set SA and end-point B offers another functionality
> set SB. Then it is possible that SA has no common subset with SB, thus
> preventing interoperability.
> 
> - Understanding when offered set SA and supported set SB contains
> possibilities for sub-setting. Certain offered sets can be subsetted and
> still useful. A MIME specification should be explicit about if
> parameters can be downgraded or not. For example, an video profile level
> specification can be downgraded to a lower level in the response and
> still achieve interoperability. However an AMR payload format
> configuration parameter like "octet-align" is binary, i.e. requiring one
> to support this configuration.
> 
> - Possibility to make recommendations on how to form offers that
> maximizes interoperability. However this is application dependent and
> should normally be kept in an more open terms. However for example for
> the AMR payload format can be configured into several usages. For some
> applications it does make sense to offer multiple payload types where
> the payload format is configured in different way. For example a
> streaming server can offer both a octet-aligned only PT and one with
> interleaving if they would use offer answer.
> 
> Thus we request that authors write an "Offer-Answer model consideration"
> in their SDP usage section. This should cover the following:
> 
> - How configurations effect the interoperability.
> - What parameters that are possible to subset and which are not.
> - Recommendations on how to achieve good interoperability.
> 
> Comments on this are solicited.
> 
> Best Regards
> 
> Magnus & Colin
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Colin Perkins <csp@csperkins.org>
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Subject: [AVT] Advancing RTP spec & profile to Full Standard
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Folks,

The RTP specification (RFC 3550) and the RTP Audio/Video Profile (RFC 3551)
have remained at the Draft Standard level for more than four months, and
past the occurrence of one IETF meeting.  This exceeds the requirements of
RFC 2026 for advancement to full Internet Standard status.

Consequently, with the concurrence of the AVT working group, we would like
to send a request to the IESG that these RFCs be advanced unchanged to Full
Internet Standard.  If you have any comments on this step, either agreement
or objections, please reply to the mailing list.

Colin, Magnus and Steve

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Cc: avt@ietf.org, John Gildred <john@pioneer-pra.com>
From: John Gildred <john@pioneer-pra.com>
Subject: Re: [AVT] Advancing RTP spec & profile to Full Standard
Date: Wed, 28 Jan 2004 17:48:40 -0800
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Agreed.
On Jan 28, 2004, at 5:24 PM, Colin Perkins wrote:

> Folks,
>
> The RTP specification (RFC 3550) and the RTP Audio/Video Profile (RFC 
> 3551)
> have remained at the Draft Standard level for more than four months, 
> and
> past the occurrence of one IETF meeting.  This exceeds the 
> requirements of
> RFC 2026 for advancement to full Internet Standard status.
>
> Consequently, with the concurrence of the AVT working group, we would 
> like
> to send a request to the IESG that these RFCs be advanced unchanged to 
> Full
> Internet Standard.  If you have any comments on this step, either 
> agreement
> or objections, please reply to the mailing list.
>
> Colin, Magnus and Steve
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>

-John Gildred
Vice President of Engineering
Pioneer Research Center USA, Inc.
A Division of Pioneer Electronics
101 Metro Drive, Suite 264
San Jose, California 95110
Office (408) 437-1800 Ext. 105
Mobile (510) 295-7770


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] Advancing RTP spec & profile to Full Standard
Date: Wed, 28 Jan 2004 17:59:46 -0800
Message-ID: <B8561865DB141248943E2376D0E852153B2774@dhost001-17.intermedia.net>
Thread-Topic: [AVT] Advancing RTP spec & profile to Full Standard
Thread-Index: AcPmBuIqu+FQdCtlQNmDtd2lt8ZjCwABK9+A
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To: "Colin Perkins" <csp@csperkins.org>, <avt@ietf.org>
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Agreed.=20

> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]=20
> Sent: Wednesday, January 28, 2004 5:24 PM
> To: avt@ietf.org
> Subject: [AVT] Advancing RTP spec & profile to Full Standard
>=20
> Folks,
>=20
> The RTP specification (RFC 3550) and the RTP Audio/Video=20
> Profile (RFC 3551) have remained at the Draft Standard level=20
> for more than four months, and past the occurrence of one=20
> IETF meeting.  This exceeds the requirements of RFC 2026 for=20
> advancement to full Internet Standard status.
>=20
> Consequently, with the concurrence of the AVT working group,=20
> we would like to send a request to the IESG that these RFCs=20
> be advanced unchanged to Full Internet Standard.  If you have=20
> any comments on this step, either agreement or objections,=20
> please reply to the mailing list.
>=20
> Colin, Magnus and Steve
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>=20

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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

	Title		: RTP Retransmission Payload Format
	Author(s)	: J. Rey
	Filename	: draft-ietf-avt-rtp-retransmission-10.txt
	Pages		: 27
	Date		: 2004-1-28
	
RTP retransmission is an effective packet loss recovery technique 
for real-time applications with relaxed delay bounds.  This 
document describes an RTP payload format for performing 
retransmissions.  Retransmitted RTP packets are sent in a separate 
stream from the original RTP stream.  It is assumed that feedback 
from receivers to senders is available.  In particular, it is 
assumed that RTCP feedback as defined in the extended RTP profile 
for RTCP-based feedback (denoted RTP/AVPF), is available in this 
memo.

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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Jan 29 12:24:49 2004
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Date: Thu, 29 Jan 2004 08:26:07 -0800
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I have a question regarding RFC 3555 media types. Is it allowed to 
include non-registered parameters with an existing IETF registered 
media type? For example: video/mp2t; my-profile=p1

If this is not allowed, how would I best provide this extra format info 
in the content type without creating a new type? I am not so concerned 
about SDP description, just the content type definition.

-John


_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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--NextPart

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

	Title		: RTP Payload Format for MIDI
	Author(s)	: J. Lazzaro
	Filename	: draft-ietf-avt-rtp-midi-format-01.txt
	Pages		: 119
	Date		: 2004-1-27
	
This memo describes an RTP payload format for the MIDI command
language.  The format encodes all commands that may legally appear
on a MIDI 1.0 DIN cable.  The format is suitable for interactive
applications (such as the remote operation of musical instruments)
and content-delivery applications (such as file streaming).  The
format may be used over unicast and multicast UDP as well as TCP,
and defines tools for graceful recovery from packet loss.  Stream
behavior, including the MIDI rendering method, may be customized
during session setup.  The format also serves as a mode for the
mpeg4-generic format, to support the MPEG 4 Audio Object Types for
General MIDI, Downloadable Sounds Level 2, and Structured Audio.

A URL for this Internet-Draft is:
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_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Jan 29 13:56:34 2004
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--NextPart

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

	Title		: An Implementation Guide for RTP MIDI
	Author(s)	: J. Lazzaro
	Filename	: draft-ietf-avt-rtp-midi-guidelines-01.txt
	Pages		: 35
	Date		: 2004-1-27
	
This memo offers non-normative implementation guidance for the RTP
     MIDI payload format.  The memo presents its advice in the context
     of a network musical performance application.  In this application
     two musicians, located in different physical locations, interact
     over a network to perform as they would if located in the same
     room.  Underlying the performances are RTP MIDI sessions over
     unicast UDP.  Algorithms for sending and receiving recovery
     journals (the resiliency structure for the payload format) are
     described in detail.  Although the memo focuses on network musical
     performance, the presented implementation advice is relevant to
     other RTP MIDI applications.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-midi-guidelines-01.txt

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_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Fri Jan 30 03:23:39 2004
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Taking into account several requests for postponing the submission
deadline for ICT2004, the organising committee has decided to
extend it to February 14 so that the participation could be as large
as possible. So take your time and visit the ICT2004 web site
at www.ict2004.com.br.




_______________________________________________
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From exim@www1.ietf.org  Fri Jan 30 04:21:35 2004
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Subject: [AVT] Deadlines for upcomming 59th IETF meeting
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Hi,

The AVT WG is planning to have two sessions during the next IETF 
meeting, for a total of 4 or 4.5 hours. If you have issues or are going 
to submit drafts for the WG in Korea, please make note of the following 
deadlines.

The deadline for submission of -00 internet-drafts is 9 am US EDT (GMT 
-04:00) on Monday the 9th of February. Individual submissions intended 
for eventual working group discussion should be named 
draft-<authorname>-avt-<shortsubject>-00.txt. Initial working group 
drafts, which have previously been approved by the chairs, may use the 
draft-ietf-avt- prefix.

The deadline for all other internet-draft submissions is Monday 16th of 
February at 9 am US EDT. This day is also the last day to pre-register 
to the meeting, the cut-off is at 12.00 EDT.

Any one desiring to have a presentation and discussion during the AVT 
working group's slots in Minneapolis MUST request time on the agenda. A 
submission of a draft is not sufficient for receiving agenda time. Send 
the request before the 16th of February, 2003 at 24.00 GMT to BOTH 
chairs. In the request include the following information:

A. Presentation Title:

B. Name of presenter:

C. Draft Reference(s):

D. Requested amount of time in minutes:


Cheers

Magnus Westerlund 

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Fri Jan 30 06:00:44 2004
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Subject: [AVT] Comments on draft-ietf-avt-rfc2032-bis-00.txt
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Hi,

I have read this draft and have some comments.

1. All pages are missing the right page break codes.

2. Section 1. In the introduction I think the document should explain 
its status as being an update to RFC 2032. Isn't the intention to go to 
draft standard. I think this should be explained.

3. After Section 1. I am missing a "Conventions used in this document" 
section. This should mention RFC 2119 and also other conventions, like 
abbreviations, and that any picture is called frames that also is 
written somewhere else. The text in the document must also be updated 
using the normative language.

4. Section 2.1: Section 2.1, first paragraph: "Each MB carries
   information on a group of 16x16 pixels: luminance information is
   specified for 4 blocks of 8x8 pixels, while chrominance information
   is given by two "red" and "blue" colour difference components at a
   resolution of only 8x8 pixels." This sentence is strange. I think it 
means that within each MB of 16x16 pixels the luminance information is 
encoded for each pixel, sub divided into four 8x8 groups. While the 
chromiance information is subsampled, for 2x2 pixels, resulting in color 
resolution of 8x8 within the MB.

5. Section 2.1, The start code is specified as being 15 zeros and a 
single 1, I think one should clarify if this is bits or bytes.

6. Section 2.1. The start code bit-order is not obvious, as it is 
sometimes later in shortened form is represented as 10000000.

7. Section 3.1: The payload type specification. I think we should use 
some of the more explicit text, referring to profiles or dynamic 
allocation. Also the reference is wrong, it should be RFC 3551.

8. I haven't found any place requiring video data from different frames 
to be in separate RTP packets, however it seems a necessary restriction.

9. Section 3.1, The payload header I and V bits: Are these parameters 
not more suitable to be in a MIME parameter? I understand that we can't 
remove them.

10. Section 3.1: I am missing a more explicit section about what is 
allowed to place in the data part of the payload.

11. Section 4: I think this section should be brought up to current 
level by referencing the relevant techniques in published or on its way 
RFCs, like the FIR in the feedback specification. Also NACK formats, etc.

12. Section 4.1: The relation for these with the RTCP feedback document 
should be spelled out.

13. Section 4.1, second paragraph:
"   The H.261-specific control packets differ from normal RTCP packets in
   that they are not transmitted to the normal RTCP destination
   transport address for the RTP session (which is often a multicast
   address).  Instead, these control packets are sent directly via
   unicast from the decoder to the coder.  The destination port for
   these control packets is the same port that the coder uses as a
   source port for transmitting RTP (data) packets.  Therefore, these
   packets may be considered "reverse" control packets."

should this usage be allowed to continue, or should it be deprecated? It 
seems to have issues in relation to signalling, DOS, congestion control 
that will have to be explained in the updated version. Does 
implementations exist of this?

14. Section 5.1 the RTCP packet types registration, should probably be 
updated and included in this document?

15. Section 5: The parameters should be included in a semicolon 
separated list, not space separated. See RFC 3555.

16. Section 5.1.1: All the ANNEX values need to have values, otherwise 
they do not adhere to syntax. Therefore as they are binary, I can see 
two alternatives, either all get a 0 or 1 value, with a default of 0, 
i.e. not used. Or we create one parameter "annexes" which contain a list 
of the used annexes.

17. The document is missing clear RFC editor statements when it at least 
uses a replacement rule for the RFC number.

18. Section 5.1.1: The "author/Change controller" string has a stray ">".

19. Section 5.2: Again the space separated list, needs to be a semicolon 
separated list.

20. Section 5.2: Please write up offer/answer considerations.

21. Section 6. I think the security consideration needs to be expanded 
on in regards to the following topics:
- The non-uniform decoder processing based on input, making DOS attacks 
possible.
- The FIR and NACK issues, see rtcp feedback draft.
- Congestion control, especially in regards to FIR and NACK
- Integrity protection
- Authentication

22. If this document is intended to go to draft, which I hope, the 
interoperability should be investigated. However it must not necessary 
go into the draft.

23. Section 8, Should become section 2, with another name.

24. Section 9. The second bullet is missing numbering. Also in regards 
to the optional new parameters, how does they effect interoperability.

25. Shouldn't there be at least an informative Reference to RFC 2032?

26. Authors list. you needs to dig up contact information for the other 
authors, Christian can't be a problem. If you fail you will have to 
remove them from the author list, and instead credit them somehow else. 
They must also agree to be ready to sign of the document when it is time 
for the authors 48 hours.






-- 

Magnus Westerlund 

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Fri Jan 30 07:56:41 2004
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From: Colin Perkins <csp@csperkins.org>
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--> Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
> Hi,
> 
> The AVT WG is planning to have two sessions during the next IETF 
> meeting, for a total of 4 or 4.5 hours. If you have issues or are going 
> to submit drafts for the WG in Korea, please make note of the following 
> deadlines.
> 
> The deadline for submission of -00 internet-drafts is 9 am US EDT (GMT 
> -04:00) on Monday the 9th of February. Individual submissions intended 
> for eventual working group discussion should be named 
> draft-<authorname>-avt-<shortsubject>-00.txt. Initial working group 
> drafts, which have previously been approved by the chairs, may use the 
> draft-ietf-avt- prefix.
> 
> The deadline for all other internet-draft submissions is Monday 16th of 
> February at 9 am US EDT. This day is also the last day to pre-register 
> to the meeting, the cut-off is at 12.00 EDT.

A minor correction: both cutoff dates are at 9:00am EST (GMT -05:00).

Colin

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi

Here is my comments on the update of RFC 2429. A lot of comments are 
similar to the ones for draft-ietf-avt-rfc2032-bis-00.txt.

1. Please try to submit documents with correct document dates, this says 
november.

2. In section 1, declare the intentions that this will update RFC 2429.

3. Conventions used in this document is missing. RFC 2119 pointer, and 
probably some other things should be here.

4. Section 2: fifth paragraph: Here I think one today should put in 
pointers at RTCP feedback.

5. Please work through and upgrade where appropriate to normative language.

6. Section 3: There should somewhere in the format specification be a 
note about that the usage of the timestamp make the RTCP jitter 
calculation to be less useful as it introduces jitter before transmission.

7. Section 3. Payload type: Please update the text with reference to 
profiles and usage of dynamic payload types.

8. Section 5.1: Reserved bits: I think it is appropriate to add: "MUST 
be ignored by receivers."

9. Section 5.1: PLEN: "If PLEN>0, the extra picture header
   is attached immediately following the rest of the payload header."
I don't think it is clear enough what attached immediately following the 
rest of payload header. Is this before or after the other video 
redundancy coding header extension?

10. Section 6.1: What is the start code bit order?

11. Section 8.1, This registration updates the previous registered 
version in RFC 3555. please spell this out.

12. Section 8: Again the space separated list needs to be replace with 
semicolon separated.

13. Section 8.1: all parameters needs values, and default 
interpretations. Some parameters could be changed into values for a new 
aggregating parameter.

14. There is need for RFC editor consideration.

15. Section 8.1.1: In the Author/Change controller a stray ">"

16. Section 8.1.1: PAR: It does not define how the values are separated, 
colon maybe.

17. Section 8.1.2: I don't know if you can make a parameter registration 
though reference.

18. Section 8.2.1: Please change the title of  "... options with SIP" to 
Offer/Answer.

19. Section 8.2.1: The order of the pictures size is not defined within 
the MIME parameter space at all. I think it belongs there if it is needed.

20. Section 8.2.1: The example using picture sizes: The definition of 
custom does not declare that space is allowed in the comma separated list.

21. Section 8.2.1: MaxBR and BPP: "MaxBitRate is video
   decoder property, hence it differs from SDP b : bandwidth-value
   attribute which refers more to application's total bandwidth (an
   application consists often of both audio and video)."

What the b= SDP parameter refers to depends on the modifier, CT, AS, RS, 
RR. So I think you need to be a bit more specific here.

22. Section 9. I thought that most video codecs contains some 
non-uniform processing depending on input. Is this not at least a small 
threat of DOS.

23. Section 9: I am missing the following topics: Integrity, and 
Authentication. Congestion control on best effort networks is also a 
consideration as the codec can after all produce a rather high bandwidth.

24. Section 11, belongs in the conventions section in the beginning.

25. Should RFC 2190 be declared historic?

26. What MIME type should be recommended to support in this new version.

27. What is the interoperability status on all parts of this specification?

28. The new parameters and possibly some other changes needs 
interoperability statements.

29. Section 12: The numbering of the changes are strange.

30. Please get the correct contact information for the previous authors. 
And contact them to in regards to there possible interaction with this 
specification in the future.

31. Please get correct page breaks.

Cheers

Magnus Westerlund 

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Fri Jan 30 11:49:37 2004
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--NextPart

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

	Title		: RTP Payload Format for Uncompressed Video
	Author(s)	: L. Gharai, C. Perkins
	Filename	: draft-ietf-avt-uncomp-video-05.txt
	Pages		: 19
	Date		: 2004-1-29
	
This memo specifies a packetization scheme for encapsulating
uncompressed video into a payload format for the Real-time
Transport Protocol, RTP. It supports a range of standard- and
high-definition video formats, including common television
formats such as ITU BT.601, and standards from the Society of
Motion Picture and Television Engineers (SMPTE), such as SMPTE
274M and SMPTE 296M. The format is designed to be applicable
and extensible to new video formats as they are developed.

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



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 > Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
 >
 > The deadline for all other internet-draft submissions is Monday 16th
 > of February at 9 am US EDT.

Tentatively, the plan for the RTP MIDI documents is to wait until
the 12th of February for comments, then spend the next four
days updating the documents to submit -02.txt versions in time
for the Feb 16th deadline.  At the very least, -02.txt will be adjusted
for the new Offer/Answer requirements.

So, it would be ideal if folks who were interested in RTP
MIDI could find the time to do a read-through and send along
comments by Feb 12.  If that's not possible, it would be good to
know that there were folks who were planning to send in
comments but needed more time to read them ... thanks,

---
John Lazzaro
http://www.cs.berkeley.edu/~lazzaro
lazzaro [at] cs [dot] berkeley [dot] edu
---


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Sat, 31 Jan 2004 14:23:52 +0000
From: Colin Perkins <csp@csperkins.org>
To: John Gildred <john@pioneer-pra.com>
Cc: avt@ietf.org
Subject: Re: [AVT] Extending Parameters
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--> John Gildred <john@pioneer-pra.com> writes:
> I have a question regarding RFC 3555 media types. Is it allowed to 
> include non-registered parameters with an existing IETF registered 
> media type? For example: video/mp2t; my-profile=p1

Such use does not conform to the standard.

> If this is not allowed, how would I best provide this extra format info 
> in the content type without creating a new type? I am not so concerned 
> about SDP description, just the content type definition.

If the MIME parameter is generally useful, the payload format RFC could be
revised (as is happening for H.261 and H.263).

-- 
Colin Perkins
http://csperkins.org/

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