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To: "Hager, Todd" <THH@dolby.com>
cc: avt@ietf.org
Subject: Re: [AVT] question: mapping dynamic RTP formats using SDP 
In-Reply-To: Your message of "Fri, 28 Feb 2003 17:17:54 PST."
             <5FCCC03CAF7C5C4C8E2F995E3D72E1334BBB5C@platinum.dolby.net> 
Date: Sat, 01 Mar 2003 19:20:33 -0500
From: Colin Perkins <csp@csperkins.org>
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--> "Hager, Todd" writes:
>What is the best method to define a valid encoding name for a new
>encoding format?
> 
>Regarding <draft-flaks-avt-rtp-ac3-03.txt>, I'm trying to determine the
>best method of choosing a value for the payload type for AC-3 in the RTP
>header.
> 
>I am proceeding under the assumption that the best course is to randomly
>choose a value from the dynamic range (96-127) described in <RTP Profile
>for Audio and Video Conferences with Minimal Control.txt> (RFC 1890),
>and use SDP to somehow map this value to the AC-3 encoding type. I have
>questions regarding the SDP mapping process.

Each application using the AC-3 payload format picks a payload type number
from the dynamic range each time it starts.  The chosen value is exchanged
in SDP, as part of the call setup signalling (e.g. using SIP or RTSP).

>Pages 21 and 22 in the SDP specification (RFC 2327) indicate the
>following mechanism to map a dynamic format ID :
> 
>m=<media> <port>/<number of ports> <transport> <fmt list>
> 
>and 
> 
>a=rtpmap:<payload type> <encoding name>/<clock rate>[/<encoding
>parameters>]
> 
>Therefore,  an SDP description ought to be able to incorporate the
>following two lines:
> 
>m=audio 49232 RTP/AVP 98
>a=rtpmap:98 <AC3_ENCODING_NAME>/48000

Correct. The actual payload type number that is assigned will be different
each time an application is started, and the "a=rtpmap:" line is used to
convey the mapping from the fixed encoding name to the ephemeral payload
type number.

>How does the AVT group suggest I define <AC3_ENCODING_NAME>? Should I
>submit a new Internet Draft describing a full profile specifically for
>this format, or can I simply use the AC-3 RTP Payload document itself as
>the vehicle that registers a valid AV Profile encoding name?
> 
>Alternatively, though the current AC-3 RTP Payload draft document does
>not currrently define an AV RTP Profile encoding name, it does define a
>MIME type description for AC-3. The SDP spec allows MIME types to
>describe media encodings, but the wording of the SDP spec (page 22)
>implies that this is only intended to be used for non-RTP streams. 

RTP originally used a separate namespace for payload format names, but 
now uses the MIME namespace.  The registration procedure is defined in 
draft-ietf-avt-rtp-mime-06.txt.

>Which is the most appropriate course of action:
> 
>1) Define a new RTP profile, 
>2) Use the AC-3 RTP payload document to define a valid encoding name for
>the AC-3 format, or
>3) Use the defined MIME type to map a dynamic AV Profile payload type to
>AC-3?

The third option is correct. The MIME type defined in the AC-3 payload
format is used as the SDP encoding name (The MIME type registration in
Section 4 of draft-flaks-avt-rtp-ac3-03.txt looks good, I don't think
you need to add anything).

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



From mailnull@www1.ietf.org  Sat Mar  1 22:06:12 2003
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From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
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Subject: [AVT] MWPP I-Ds ...
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Hi everyone,

	Sent the San Francisco versions of the MWPP normative and
guidelines documents off to internet-drafts today, you can pick
up early copies at:

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

	The heavy-duty work was done on the normative document
(another 3-page change log), the guidelines document changes are a
refresh to keep it in sync with the normative document.

	The original hope was that the normative document would be a
"candidate Last Call", but alas, it was not meant to be. The original
"open issues" list is actually (mostly) addressed by the new document
-- the problem is that the "open issues" list is growing again ...

	On the bright side, the list is growing for a good reason; the
MMA (the MIDI Manufacturers Association, <http://www.midi.org/> ),
which works together with the AMEI ( <http://www.amei.or.jp/> ), has
been passing along MWPP concerns to me, via Jim Wright.

	Jim and I have worked out a set of MMA requirements for MWPP,
and I'm going to spend March doing design work to meet them. If those
changes are non-controversial for AVT folks, then we may finally be
near Last Call in early April.

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Sat, 1 Mar 2003 22:11:05 -0800 (PST)
From: Stephen Casner <casner@acm.org>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
cc: AVT WG <avt@ietf.org>
Subject: Re: [AVT] RTP: SSRC multiplexing for same medium
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Delayed response as I finally get to finishing the editing:

On Thu, 16 Jan 2003, Jonathan Rosenberg wrote:
> Stephen Casner wrote:
> >>2. Does this text imply that an RTP receiver MUST be prepared to receive
> >>multiple multiplexed streams on a single session? I expect most
> >>implementations today using unicast are not prepared for that (although
> >>I believe they should be)
> >
> > No, it would be inappropriate to make that a MUST.  We do say, in
> > Section 6:
> >
> >    RTP application designers SHOULD avoid mechanisms that can only
> >    work in unicast mode and will not scale to larger numbers.
>
> I have a hard time seeing how this implies that receiving multiple
> streams on a single session is not a MUST.

Certainly it is a good idea, but as you can see, the word in the
sentence I quoted from Section 6 is SHOULD, not MUST.

> Indeed, receiving many
> streams on a single session is how multicast works, so one would imagine
> that the direct translation of this to unicast, receiving multiple
> streams on a single port for multi-unicast, would be the recommended
> approach.

Yes, I agree.

> If RTP does not mandate that a user be prepared to receive multiple
> streams on a single session for unicast (even though it does for
> multicast), I think it needs to say that its up to signaling mechanisms
> to indicate such a limitation. Of course, SDP has no such mechanism....

I've added the following sentence in the definition of "RTP session"
to address that:

   The RTP framework allows the variations defined here, but a
   particular control protocol or application design will usually
   impose constraints on these variations.

                                                        -- Steve

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



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From: Timur Friedman <timur.friedman@lip6.fr>
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--------------Boundary-00=_WD65OVMBFJDCRLLQ6Y7F
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  charset="iso-8859-1"
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Hello,

Please find attached the latest version of the RTP Extended
Reports (RTP XR) draft, draft-ietf-avt-rtcp-report-extns-03.txt, which ha=
s=20
just been sent to internet-drafts@ietf.org.

We request that this draft be advanced to working group final call at thi=
s=20
stage, on the proposed standard RFC track.

Here are the changes since the previous version:

* The 02 draft specified signal and noise levels in terms of dBm0 with
  respect to overload conditions - this has been changed to dBm, while
  retaining the definitions for talkspurts vs silence periods.  The
  signal and noise level fields are now adjacent to each other.

* In Section 4, changed the text "An implementation MAY ignore
  incoming blocks with types either not relevant or unknown" to be "An
  implementation SHOULD ignore...".  Thank you Colin for spotting
  this.

* Reverted the definition of reserved fields to the way it was in the
  01 draft.  A receiver that does not know of a definition for the
  field simply ignores the field, not the entire block or report.

* Changed CODEC to codec.  This term appears in Webster's New World
  Dictionary, Third College Edition as a common noun.  Corrected some
  spelling errors.

A PDF version of this draft can be found at the following location:

http://www-rp.lip6.fr/site_rp/_publications/378-draft-ietf-avt-rtcp-repor=
t-extns-03.pdf

Regards,

Timur Friedman

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Content-Transfer-Encoding: quoted-printable







INTERNET-DRAFT                                              2 March 2003
Internet Engineering Task Force                Expires: 2 September 2003
Audio/Video Transport Working Group

                                                 Timur Friedman, Paris 6
                                                 Ramon Caceres, ShieldIP
                                                 Alan Clark, Telchemy
                                                 Editors

                     RTP Extended Reports (RTP XR)

                draft-ietf-avt-rtcp-report-extns-03.txt


Status of this Memo

   This document is an Internet-Draft and is subject to 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/1id-abstracts.html

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

Copyright Notice

   Copyright (C) The Internet Society (2003).  All Rights Reserved.

Abstract

   This document defines the extended report (XR) packet type for the
   RTP control protocol (RTCP).  XR packets are composed of report
   blocks, and seven block types are defined here.  The purpose of the
   extended reporting format is to convey information that supplements
   the six statistics that are contained in the report blocks used by
   RTCP's sender report (SR) and receiver report (RR) packets.  Some
   applications, such as multicast inference of network characteristics



Friedman, Caceres, and Clark, eds.                              [Page 1]
=0C
draft-ietf-avt-rtcp-report-extns-03.txt                     2 March 2003


   (MINC) or voice over IP (VoIP) monitoring, require other and more
   detailed statistics.  In addition to the block types defined here,
   additional block types may be defined in the future by adhering to
   the framework that this document provides.

Table of Contents

   1.     Introduction ..............................................  2
   1.1    Terminology ...............................................  3
   2.     XR Packet Format ..........................................  4
   3.     Extended Report Block Framework ...........................  5
   4.     Extended Report Blocks ....................................  6
   4.1    Loss RLE Report Block .....................................  6
   4.1.1  Run Length Chunk .......................................... 12
   4.1.2  Bit Vector Chunk .......................................... 12
   4.1.3  Terminating Null Chunk .................................... 12
   4.2    Duplicate RLE Report Block ................................ 13
   4.3    Timestamp Report Block .................................... 14
   4.4    Statistics Summary Report Block ........................... 16
   4.5    Receiver Timestamp Report Block ........................... 19
   4.6    DLRR Report Block ......................................... 20
   4.7    VoIP Metrics Report Block ................................. 21
   4.7.1  Packet Loss and Discard Metrics ........................... 22
   4.7.2  Burst Metrics ............................................. 23
   4.7.3  Delay Metrics ............................................. 25
   4.7.4  Signal Related Metrics .................................... 26
   4.7.5  Call Quality or Transmission Quality Metrics .............. 29
   4.7.6  Configuration Parameters .................................. 30
   4.7.7  Jitter Buffer Parameters .................................. 31
   5.     IANA Considerations ....................................... 32
   5.1    XR Packet Type ............................................ 32
   5.2    RTP XR Block Type Registry ................................ 32
   6.     Security Considerations ................................... 33
   A.     Algorithms ................................................ 34
   A.1    Sequence Number Interpretation ............................ 34
   A.2    Example Burst Packet Loss Calculation ..................... 35
          Intellectual Property ..................................... 37
          Full Copyright Statement .................................. 37
          Acknowledgments ........................................... 38
          Contributors .............................................. 38
          Authors' Addresses ........................................ 39
          References ................................................ 40
          Normative References ...................................... 40
          Non-Normative References .................................. 41

1. Introduction

   This document defines the extended report (XR) packet type for the



Friedman, Caceres, and Clark, eds.                              [Page 2]
=0C
draft-ietf-avt-rtcp-report-extns-03.txt                     2 March 2003


   RTP control protocol (RTCP) [7].  XR packets convey information
   beyond that already contained in the reception report blocks of
   RTCP's sender report (SR) or receiver report (RR) packets.  The
   information is of use across RTP profiles, and so is not
   appropriately carried in SR or RR profile-specific extensions.
   Information used for network management falls into this category, for
   instance.

   The definition is broken out over the three following sections of
   this document, starting with a general framework and finishing with
   the specific information conveyed.  The framework defined by Section
   2 contains common header information followed by a series of
   components called report blocks.  Section 3 defines the format common
   to such blocks.  Section 4 defines seven block types.

   Seven report block formats are defined by this document:

   - Loss RLE Report Block (Section 4.1): Run length encoding of RTP
   packet loss reports.

   - Duplicate RLE Report Block (Section 4.2): Run length encoding of
   reports of RTP packet duplicates.

   - Timestamp Report Block (Section 4.3): A list of timestamps of
   received RTP packets.

   - Statistics Summary Report Block (Section 4.4): Statistics on RTP
   packet sequence numbers, losses, duplicates, jitter, and TTL values.

   - Receiver Timestamp Report Block (Section 4.5): Receiver-end
   timestamps that complement the sender-end timestamps already defined
   for RTCP.

   - DLRR Report Block (Section 4.6): The delay since the last Receiver
   Timestamp Report Block was received, allowing non-senders to
   calculate round-trip times.

   - VoIP Metrics Report Block (Section 4.7): Metrics for monitoring
   Voice over IP (VoIP) calls.

   These blocks are defined within a minimal framework: a type field and
   a length field are common to all XR blocks.  The purpose is to
   maintain flexibility and to keep overhead low.  Other block formats,
   beyond the seven defined here, may be defined within this framework
   as the need arises.

1.1 Terminology




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   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] and
   indicate requirement levels for compliance with this specification.

2. XR Packet Format

   The XR packet consists of a header of two 32-bit words, followed by a
   number, possibly zero, of extended report blocks.


    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=3D2|P|reserved |   PT=3DXP=3D207   |             length            =
|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           SSRC/CSRC                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :                         report blocks                         :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   version (V): 2 bits
        Identifies the version of RTP.  This specification applies to
        RTP version two.

   padding (P): 1 bit
        If the padding bit is set, this XR packet contains some
        additional padding octets at the end.  The semantics of this
        field are identical to the semantics of the padding field in the
        SR packet, as defined by the RTP specification.

   reserved: 5 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   packet type (PT): 8 bits
        Contains the constant 207 to identify this as an RTCP XR packet.
        This value is registered with the Internet Assigned Numbers
        Authority (IANA), as described in Section 5.1.

   length: 16 bits
        As described for the RTP sender report (SR) packet (see Section
        6.3.1 of the RTP specification [7]).  Briefly, the length of
        this XR packet in 32-bit words minus one, including the header
        and any padding.




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   SSRC: 32 bits
        The synchronization source identifier for the originator of this
        XR packet.

   report blocks: variable length.
        Zero or more extended report blocks.  In keeping with the
        extended report block framework defined below, each block MUST
        consist of one or more 32-bit words.

3. Extended Report Block Framework

   Extended report blocks are stacked, one after the other, at the end
   of an XR packet.  An individual block's length is a multiple of 4
   octets.  The XR header's length field describes the total length of
   the packet, including these extended report blocks.

   Each block has block type and length fields that facilitate parsing.
   A receiving application can demultiplex the blocks based upon their
   type, and can use the length information to locate each successive
   block, even in the presence of block types it does not recognize.

   An extended report block has the following format:


    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |      BT       | type-specific |         block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :                      type-specific data                       :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        Identifies the block format.  Seven block types are defined in
        Section 4.  Additional block types may be defined in future
        specifications.  This field's name space is managed by the
        Internet Assigned Numbers Authority (IANA), as described in
        Section 5.2.

   type-specific: 8 bits
        The use of these bits is determined by the block type
        definition.

   block length: 16 bits
        The length of this report block including the header, in 32-bit
        words minus one.  If the block type definition permits, zero is
        an acceptable value, signifying a block that consists of only



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        the BT, type-specific, and block length fields, with a null
        type-specific data field.

   type-specific data: variable length
        The use of this field is defined by the particular block type,
        subject to the constraint that it MUST be a multiple of 32 bits
        long.  If the block type definition permits, It MAY be zero bits
        long.

4. Extended Report Blocks

   This section defines seven extended report blocks: block types for
   losses, duplicates, packet reception timestamps, detailed reception
   statistics, receiver timestamps, receiver inter-report delays, and
   voice over IP (VoIP) metrics.  An implementation SHOULD ignore
   incoming blocks with types either not relevant or unknown to it.
   Additional block types MUST be registered with the Internet Assigned
   Numbers Authority (IANA) [5], as described in Section 5.2.

4.1 Loss RLE Report Block

   This block type permits detailed reporting upon individual packet
   receipt and loss events.  Such reports can be used, for example, for
   multicast inference of network characteristics (MINC) [8].  With
   MINC, one can discover the topology of the multicast tree used for
   distributing a source's RTP packets, and of the loss rates along
   links within that tree.  Or they could be used to provide raw data to
   a network management application.

   Since a Boolean trace of lost and received RTP packets is potentially
   lengthy, this block type permits the trace to be compressed through
   run length encoding.  To further reduce block size, loss event
   reports can be systematically dropped from the trace in a mechanism
   called thinning that is described below and that is studied in [9].

   A participant that generates a Loss RLE Report Block should favor
   accuracy in reporting on observed events over interpretation of those
   events whenever possible.  Interpretation should be left to those who
   observe the report blocks.  Following this approach implies that
   accounting for Loss RLE Report Blocks will differ from the accounting
   for the generation of the SR and RR packets described in the RTP
   specification [7] in the following two areas: per-sender accounting
   and per-packet accounting.

   In its per-sender accounting, an RTP session participant SHOULD NOT
   make the receipt of a threshold minimum number of RTP packets a
   condition for reporting upon the sender of those packets.  This
   accounting technique differs from the technique described in Section



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   6.2.1 and Appendix A.1 of the RTP specification that allows a
   threshold to determine whether a sender is considered valid.

   In its per-packet accounting, an RTP session participant SHOULD treat
   all sequence numbers as valid.  This accounting technique differs
   from the technique described in Appendix A.1 of the RTP specification
   that suggests ruling a sequence number valid or invalid on the basis
   of its contiguity with the sequence numbers of previously received
   packets.

   Sender validity and sequence number validity are interpretations of
   the raw data.  Such interpretations are justified in the interest,
   for example, of excluding the stray old packet from an unrelated
   session from having an effect upon the calculation of the RTCP
   transmission interval.  The presence of stray packets might, on the
   other hand, be of interest to a network monitoring application.

   One accounting interpretation that is still necessary is for a
   participant to decide whether the 16 bit sequence number has rolled
   over.  Under ordinary circumstances this is not a difficult task.
   For example, if packet number 65,535 (the highest possible sequence
   number) is followed shortly by packet number 0, it is reasonable to
   assume that there has been a rollover.  However it is possible that
   the packet is an earlier one (from 65,535 packets earlier).  It is
   also possible that the sequence numbers have rolled over multiple
   times, either forward or backward.  The interpretation becomes more
   difficult when there are large gaps between the sequence numbers,
   even accounting for rollover, and when there are long intervals
   between received packets.

   The per-packet accounting technique mandated here is for a
   participant to keep track of the sequence number of the packet most
   recently received from a sender.  For the next packet that arrives
   from that sender, the sequence number MUST be judged to fall no more
   than 32,768 packets ahead or behind the most recent one, whichever
   choice places it closer.  In the event that both choices are equally
   distant (only possible when the distance is 32,768), the choice MUST
   be the one that does not require a rollover.  Appendix A.1 presents
   an algorithm that implements this technique.

   Each block reports on a single source, identified by its SSRC.  The
   receiver that is supplying the report is identified in the header of
   the RTCP packet.

   Choice of beginning and ending RTP packet sequence numbers for the
   trace is left to the application.  These values are reported in the
   block.  The last sequence number in the trace MAY differ from the
   sequence number reported on in any accompanying SR or RR report.



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   Note that because of sequence number wraparound the ending sequence
   number MAY be less than the beginning sequence number.  A Loss RLE
   Report Block MUST NOT be used to report upon a range of 65,534 or
   greater in the sequence number space, as there is no means to
   identify multiple wraparounds.

   The trace described by a Loss RLE report consists of a sequence of
   Boolean values, one for each sequence number of the trace.  A value
   of one represents a packet receipt, meaning that one or more packets
   having that sequence number have been received since the most recent
   wraparound of sequence numbers (or since the beginning of the RTP
   session if no wraparound has been judged to have occurred).  A value
   of zero represents a packet loss, meaning that there has been no
   packet receipt for that sequence number as of the time of the report.
   If a packet with a given sequence number is received after a report
   of a loss for that sequence number, a later Loss RLE report MAY
   report a packet receipt for that sequence number.

   The encoding itself consists of a series of 16 bit units called
   chunks that describe sequences of packet receipts or losses in the
   trace.  Each chunk either specifies a run length or a bit vector, or
   is a null chunk.  A run length describes between 1 and 16,383 events
   that are all the same (either all receipts or all losses).  A bit
   vector describes 15 events that may be mixed receipts and losses.  A
   null chunk describes no events, and is used to round out the block to
   a 32 bit word boundary.

   The mapping from a sequence of lost and received packets into a
   sequence of chunks is not necessarily unique.  For example, the
   following trace covers 45 packets, of which the 22nd and 24th have
   been lost and the others received:


   1111 1111 1111 1111 1111 1010 1111 1111 1111 1111 1111 1


   One way to encode this would be:


   bit vector 1111 1111 1111 111
   bit vector 1111 1101 0111 111
   bit vector 1111 1111 1111 111
   null chunk


   Another way to encode this would be:





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   run of 21 receipts
   bit vector 0101 1111 1111 111
   run of 9 receipts
   null chunk


   The choice of encoding is left to the application.  As part of this
   freedom of choice, applications MAY terminate a series of run length
   and bit vector chunks with a bit vector chunk that runs beyond the
   sequence number space described by the report block.  For example, if
   the 44th packet in the same sequence were lost:


   1111 1111 1111 1111 1111 1010 1111 1111 1111 1111 1110 1


   This could be encoded as:


   run of 21 receipts
   bit vector 0101 1111 1111 111
   bit vector 1111 1110 1000 000
   null chunk


   In this example, the last five bits of the second bit vector describe
   a part of the sequence number space that extends beyond the last
   sequence number in the trace.  These bits have been set to zero.

   All bits in a bit vector chunk that describe a part of the sequence
   number space that extends beyond the last sequence number in the
   trace MUST be set to zero, and MUST be ignored by the receiver.

   A null packet MUST appear at the end of a Loss RLE Report Block if
   the number of run length plus bit vector chunks is odd.  The null
   chunk MUST NOT appear in any other context.

   Caution should be used in sending Loss RLE Report Blocks because,
   even with the compression provided by run length encoding, they can
   easily consume bandwidth out of proportion with normal RTCP packets.
   The block type includes a mechanism, called thinning, that allows an
   application to limit report sizes.

   A thinning value, T, selects a subset of packets within the sequence
   number space: those with sequence numbers that are multiples of 2^T.
   Packet reception and loss reports apply only to those packets.  T can
   vary between 0 and 15.  If T is zero then every packet in the
   sequence number space is reported upon.  If T is fifteen then one in



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   every 32,768 packets is reported upon.

   Suppose that the trace just described begins at sequence number
   13,821.  The last sequence number in the trace is 13,865.  If the
   trace were to be thinned with a thinning value of T=3D2, then the
   following sequence numbers would be reported upon: 13,824, 13,828,
   13,832, 13,836, 13,840, 13,844, 13,848, 13,852, 13,856, 13,860,
   13,864.  The thinned trace would be as follows:


      1    1    1    1    1    0    1    1    1    1    0


   This could be encoded as follows:


   bit vector 1111 1011 1100 000
   null chunk


   The last four bits in the bit vector, representing sequence numbers
   13,868, 13,872, 13,876, and 13,880, extend beyond the trace and are
   thus set to zero and are ignored by the receiver.  With thinning, the
   loss of the 22nd packet goes unreported because its sequence number,
   13,842, is not a multiple of four.  Packet receipts for all sequence
   numbers that are not multiples of four also go unreported.  However,
   in this example thinning has permitted the Loss RLE Report Block to
   be shortened by one 32 bit word.

   Choice of the thinning value is left to the application.

   The Loss RLE Report Block has the following format:



















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    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=3D1      | rsvd. |   T   |         block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          chunk 1              |             chunk 2           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :                              ...                              :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          chunk n-1            |             chunk n           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A Loss RLE Report Block is identified by the constant 1.

   rsvd.: 4 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   thinning (T): 4 bits
        The amount of thinning performed on the sequence number space.
        Only those packets with sequence numbers 0 mod 2^T are reported
        on by this block.  A value of 0 indicates that there is no
        thinning, and all packets are reported on.  The maximum thinning
        is one packet in every 32,768 (amounting to two packets within
        each 16-bit sequence space).

   block length: 16 bits
        Defined in Section 3.

   begin_seq: 16 bits
        The first sequence number that this block reports on.

   end_seq: 16 bits
        The last sequence number that this block reports on plus one.

   chunk i: 16 bits
        There are three chunk types: run length, bit vector, and
        terminating null, defined in the following sections.  If the
        chunk is all zeroes then it is a terminating null chunk.
        Otherwise, the leftmost bit of the chunk determines its type: 0
        for run length and 1 for bit vector.



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4.1.1 Run Length Chunk


    0                   1
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |C|R|        run length         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   chunk type (C): 1 bit
        A zero identifies this as a run length chunk.

   run type (R): 1 bit
        Zero indicates a run of 0s.  One indicates a run of 1s.

   run length: 14 bits
        A value between 1 and 16,383.  The value MUST not be zero for a
        run length chunk (zeroes in both the run type and run length
        fields would make the chunk a terminating null chunk).  Run
        lengths of 15 or less MAY be described with a run length chunk
        despite the fact that they could also be described as part of a
        bit vector chunk.

4.1.2 Bit Vector Chunk


    0                   1
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |C|        bit vector           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   chunk type (C): 1 bit
        A one identifies this as a bit vector chunk.

   bit vector: 15 bits
        The vector is read from left to right, in order of increasing
        sequence number (with the appropriate allowance for wraparound).

4.1.3 Terminating Null Chunk

   This chunk is all zeroes.







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    0                   1
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


4.2 Duplicate RLE Report Block

   This block type permits per-sequence-number reports on duplicates in
   a source's RTP packet stream.  Such information can be used for
   network diagnosis, and provide an alternative to packet losses as a
   basis for multicast tree topology inference.

   The Duplicate RLE Report Block format is identical to the Loss RLE
   Report Block format.  Only the interpretation is different, in that
   the information concerns packet duplicates rather than packet losses.
   The trace to be encoded in this case also consists of zeros and ones,
   but a zero here indicates the presence of duplicate packets for a
   given sequence number, whereas a one indicates that no duplicates
   were received.

   The existence of a duplicate for a given sequence number is
   determined over the entire reporting period.  For example, if packet
   number 12,593 arrives, followed by other packets with other sequence
   numbers, the arrival later in the reporting period of another packet
   numbered 12,593 counts as a duplicate for that sequence number.  The
   duplicate does not need to follow immediately upon the first packet
   of that number.  Care must be taken that a report does not cover a
   range of 65,534 or greater in the sequence number space.

   No distinction is made between the existence of a single duplicate
   packet and multiple duplicate packets for a given sequence number.
   Note also that since there is no duplicate for a lost packet, a loss
   is encoded as a one in a Duplicate RLE Report Block.

   The Duplicate RLE Report Block has the following format:














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    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=3D2      | rsvd. |   T   |         block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          chunk 1              |             chunk 2           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :                              ...                              :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          chunk n-1            |             chunk n           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A Duplicate RLE Report Block is identified by the constant 2.

   rsvd.: 4 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   thinning (T): 4 bits
        As defined in Section 4.1.

   block length: 16 bits
        Defined in Section 3.

   begin_seq: 16 bits
        As defined in Section 4.1.

   end_seq: 16 bits
        As defined in Section 4.1.

   chunk i: 16 bits
        As defined in Section 4.1.

4.3 Timestamp Report Block

   This block type permits per-sequence-number reports on packet receipt
   timestamps for a given source's RTP packet stream.  Such information
   can be used for MINC inference of the topology of the multicast tree
   used to distribute the source's RTP packets, and of the delays along
   the links within that tree.  It can also be used to measure partial
   path characteristics and to model distributions for packet jitter.



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   At least one packet MUST have been received for each sequence number
   reported upon in this block.  If this block type is used to report
   timestamps for a series of sequence numbers that includes lost
   packets, several blocks are required.  If duplicate packets have been
   received for a given sequence number, and those packets differ in
   their receiver timestamps, any timestamp other than the earliest MUST
   NOT be reported.  This is to ensure consistency among reports.

   Timestamps consume more bits than loss or duplicate information, and
   do not lend themselves to run length encoding.  The use of thinning
   is encouraged to limit the size of Timestamp Report Blocks.

   The Timestamp Report Block has the following format:


    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=3D3      | rsvd. |   T   |         block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        RTP timestamp 1                        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        RTP timestamp 2                        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :                              ...                              :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        RTP timestamp n                        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A Timestamp Report Block is identified by the constant 3.

   rsvd.: 4 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   thinning (T): 4 bits
        As defined in Section 4.1.

   block length: 16 bits
        Defined in Section 3.




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   begin_seq: 16 bits
        As defined in Section 4.1.

   end_seq: 16 bits
        As defined in Section 4.1.

   RTP timestamp i: 32 bits
        The timestamp reflects the packet arrival time at the receiver.
        It is preferable for the timestamp to be established at the link
        layer interface, or in any case as close as possible to the wire
        arrival time.  Units and format are the same as for the
        timestamp in RTP data packets.  As opposed to RTP data packet
        timestamps, in which nominal values may be used instead of
        system clock values in order to convey information useful for
        periodic playout, the receiver timestamps should reflect the
        actual time as closely as possible.  The initial value of the
        timestamp is random, and subsequent timestamps are offset from
        this value.

4.4 Statistics Summary Report Block

   This block reports statistics beyond the information carried in the
   standard RTCP packet format, but not as fine grained as that carried
   in the report blocks previously described.  Information is recorded
   about lost packets, duplicate packets, jitter measurements, and TTL
   values (TTL values being taken from the TTL field of IPv4 packets, if
   the data packets are carried over IPv4).  Such information can be
   useful for network management.

   The report block contents are dependent upon a bit vector carried in
   the first part of the header.  Not all parameters need to be reported
   in each block.  Flags indicate which are and which are not reported.
   The fields corresponding to unreported parameters MUST be set to
   zero. The receiver MUST ignore any Statistics Summary Report Block
   with a non-zero value in any field flagged as unreported.

   The Statistics Summary Report Block has the following format:














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    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=3D4      |L|D|J|T| rsvd. |       block length =3D 9        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        lost_packets                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        dup_packets                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         min_jitter                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         max_jitter                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         avg_jitter                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         dev_jitter                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   min_ttl     |   max_ttl     |   avg_ttl     |     dev_ttl   |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A Statistics Summary Report Block is identified by the constant
        4.

   loss report flag (L): 1 bit
        Bit set to 1 if the lost_packets field contains a report, 0
        otherwise.

   duplicate report flag (D): 1 bit
        Bit set to 1 if the dup_packets field contains a report, 0
        otherwise.

   jitter flag (J): 1 bit
        Bit set to 1 if the min_jitter, max_jitter, avg_jitter, and
        dev_jitter fields all contain reports, 0 if none of them do.

   TTL flag (T): 1 bit
        Bit set to 1 if the min_ttl, max_ttl, avg_ttl, and dev_ttl
        fields all contain reports, 0 if none of them do.

   rsvd.: 4 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and



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        MUST be ignored by the receiver.

   block length: 16 bits
        The constant 9, in accordance with the definition of this field
        in Section 3.

   begin_seq: 16 bits
        As defined in Section 4.1.

   end_seq: 16 bits
        As defined in Section 4.1.

   lost_packets: 32 bits
        Number of lost packets in the above sequence number interval.

   dup_packets: 32 bits
        Number of duplicate packets in the above sequence number
        interval.

   min_jitter: 32 bits
        The minimum relative transit time between two packets in the
        above sequence number interval.  All jitter values are measured
        as the difference between a packet's RTP timestamp and the
        reporter's clock at the time of arrival, measured in the same
        units.

   max_jitter: 32 bits
        The maximum relative transit time between two packets in the
        above sequence number interval.

   avg_jitter: 32 bits
        The average relative transit time between each two packet series
        in the above sequence number interval.

   dev_jitter: 32 bits
        The standard deviation of the relative transit time between each
        two packet series in the above sequence number interval.

   min_ttl: 8 bits
        The minimum TTL value of data packets in sequence number range.

   max_ttl: 8 bits
        The maximum TTL value of data packets in sequence number range.

   avg_ttl: 8 bits
        The average TTL value of data packets in sequence number range.

   dev_ttl: 8 bits



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        The standard deviation of TTL values of data packets in sequence
        number range.

4.5 Receiver Timestamp Report Block

   This block extends RTCP's timestamp reporting so that non-senders may
   also send timestamps.  It recapitulates the NTP timestamp fields from
   the RTCP Sender Report [7, Sec. 6.3.1].  A non-sender may estimate
   its RTT to other participants, as proposed in [11], by sending this
   report block and receiving DLRR Report Blocks (see next section) in
   reply.


    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=3D5      |   reserved    |       block length =3D 2        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |              NTP timestamp, most significant word             |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |             NTP timestamp, least significant word             |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A Receiver Timestamp Report Block is identified by the constant
        5.

   reserved: 8 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   block length: 16 bits
        The constant 2, in accordance with the definition of this field
        in Section 3.

   NTP timestamp: 64 bits
        Indicates the wallclock time when this block was sent so that it
        may be used in combination with timestamps returned in DLRR
        Report Blocks (see next section) from other receivers to measure
        round-trip propagation to those receivers.  Receivers should
        expect that the measurement accuracy of the timestamp may be
        limited to far less than the resolution of the NTP timestamp.
        The measurement uncertainty of the timestamp is not indicated as
        it may not be known. A report block sender that can keep track
        of elapsed time but has no notion of wallclock time may use the
        elapsed time since joining the session instead. This is assumed



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        to be less than 68 years, so the high bit will be zero.  It is
        permissible to use the sampling clock to estimate elapsed
        wallclock time. A report sender that has no notion of wallclock
        or elapsed time may set the NTP timestamp to zero.

4.6 DLRR Report Block

   This block extends RTCP's delay since last sender report (DLSR)
   mechanism [7, Sec. 6.3.1] so that non-senders may also calculate
   round trip times, as proposed in [11].  It is termed DLRR for delay
   since last receiver report, and may be sent in response to a Receiver
   Timestamp Report Block (see previous section) from a receiver to
   allow that receiver to calculate its round trip time to the
   respondent.  The report consists of one or more 3 word sub-blocks:
   one sub-block per receiver report.


    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=3D6      |   reserved    |         block length          |
   +=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D=
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
   |                 SSRC_1 (SSRC of first receiver)               | sub-
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ bloc=
k
   |                         last RR (LRR)                         |   1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                   delay since last RR (DLRR)                  |
   +=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D=
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
   |                 SSRC_2 (SSRC of second receiver)              | sub-
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ bloc=
k
   :                               ...                             :   2
   +=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D=
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+


   block type (BT): 8 bits
        A DLRR Report Block is identified by the constant 6.

   reserved: 8 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   block length: 16 bits
        Defined in Section 3.

   last RR timestamp (LRR): 32 bits
        The middle 32 bits out of 64 in the NTP timestamp (as explained
        in the previous section) received as part of a Receiver



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        Timestamp Report Block from participant SSRC_n. If no such block
        has been received, the field is set to zero.

   delay since last RR (DLRR): 32 bits
        The delay, expressed in units of 1/65536 seconds, between
        receiving the last Receiver Timestamp Report Block from
        participant SSRC_n and sending this DLRR Report Block.  If no
        Receiver Timestamp Report Block has been received yet from
        SSRC_n, the DLRR field is set to zero (or the DLRR is omitted
        entirely). Let SSRC_r denote the receiver issuing this DLRR
        Report Block. Participant SSRC_n can compute the round-trip
        propagation delay to SSRC_r by recording the time A when this
        Receiver Timestamp Report Block is received.  It calculates the
        total round-trip time A-LSR using the last SR timestamp (LSR)
        field, and then subtracting this field to leave the round-trip
        propagation delay as A-LSR-DLSR. This is illustrated in [7, Fig.
        2].

4.7 VoIP Metrics Report Block

   The VoIP Metrics Report Block provides metrics for monitoring voice
   over IP (VoIP) calls.  These metrics include packet loss and discard
   metrics, delay metrics, analog metrics, and voice quality metrics.
   The block reports separately on packets lost on the IP channel, and
   those that have been received but then discarded by the receiving
   jitter buffer.  It also reports on the combined effect of losses and
   discards, as both have equal effect on call quality.

   In order to properly assess the quality of a Voice over IP call it is
   desirable to consider the degree of burstiness of packet loss [10].
   Following a Gilbert-Elliott model [2], a period of time, bounded by
   lost and/or discarded packets, with a high rate of losses and/or
   discards is a "burst," and a period of time between two bursts is a
   "gap."  Bursts correspond to periods of time during which the packet
   loss rate is high enough to produce noticeable degradation in audio
   quality.  Gaps correspond to periods of time during which only
   isolated lost packets may occur, and in general these can be masked
   by packet loss concealment.  Delay reports include the transit delay
   between RTCP end points and the VoIP end system processing delays,
   both of which contribute to the user perceived delay.  Additional
   metrics include signal, echo, noise, and distortion levels.  Call
   quality metrics include R factors (as described by the E Model
   defined in [2]) and mean opinion scores (MOS scores).

   An implementation that sends these blocks SHOULD send at least one
   every ten seconds for the duration of the call, SHOULD send one
   whenever a codec type change or other significant change occurs,
   SHOULD send one when significant call quality degradation is detected



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   and SHOULD send one upon call termination.  Implementations MUST
   provide values for all the fields defined here.  For certain metrics,
   if the value is undefined or unknown, then the specified default or
   unknown field value MUST be provided.

   The block is encoded as seven 32-bit words:


    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=3D7      |   reserved    |       block length =3D 6        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   loss rate   | discard rate  | burst density |  gap density  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |       burst duration          |         gap duration          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     round trip delay          |       end system delay        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | signal level  |  noise level  |     RERL      |     Gmin      |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   R factor    | ext. R factor |    MOS-LQ     |    MOS-CQ     |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   RX config   |  JB nominal   |  JB maximum   |  JB abs max   |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A VoIP Metrics Report Block is identified by the constant 7.

   reserved: 8 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   block length: 16 bits
        The constant 6, in accordance with the definition of this field
        in Section 3.

   The remaining fields are described in the following six sections:
   Packet Loss and Discard Metrics, Delay Metrics, Signal Related
   Metrics, Call Quality or Transmission Quality Metrics, Configuration
   Metrics, and Jitter Buffer Parameters.

4.7.1 Packet Loss and Discard Metrics

   It is very useful to distinguish between packets lost by the network
   and those discarded due to jitter. Both have equal effect on the



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   quality of the voice stream however having separate counts helps
   identify the source of quality degradation. These fields MUST be
   populated.

   loss rate: 8 bits
        The fraction of RTP data packets from the source lost since the
        beginning of reception, expressed as a fixed point number with
        the binary point at the left edge of the field.  This value is
        calculated by dividing the total number of packets lost (after
        the effects of applying any error protection such as FEC) by the
        total number of packets expected, multiplying the result of the
        division by 256, limiting the maximum value to 255 (to avoid
        overflow), and taking the integer part.  The numbers of
        duplicated packets and discarded packets do not enter into this
        calculation.  Since receivers cannot be required to maintain
        unlimited buffers, a receiver MAY categorize late-arriving
        packets as lost.  The degree of lateness that triggers a loss
        SHOULD be significantly greater than that which triggers a
        discard.

   discard rate: 8 bits
        The fraction of RTP data packets from the source that have been
        discarded since the beginning of reception, due to late or early
        arrival, under-run or overflow at the receiving jitter buffer.
        This value is expressed as a fixed point number with the binary
        point at the left edge of the field.  It is calculated by
        dividing the total number of packets discarded (excluding
        duplicate packet discards) by the total number of packets
        expected, multiplying the result of the division by 256,
        limiting the maximum value to 255 (to avoid overflow), and
        taking the integer part.

4.7.2 Burst Metrics

   A burst, informally, is a period of high packet losses and/or
   discards.  Formally, a burst is defined as a longest sequence of
   packets bounded by lost or discarded packets with the constraint that
   within a burst any sequence of successive packets that were received,
   and not discarded due to delay variation, is of length less than a
   value Gmin.

   A gap, informally, is a period of low packet losses and/or discards.
   Formally, a gap is defined as any of the following: (a) the period
   from the start of an RTP session to the receipt time of the last
   received packet before the first burst, (b) the period from the end
   of the last burst to either the time of the report or the end of the
   RTP session, whichever comes first, or (c) the period of time between
   two bursts.



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   For the purpose of determining if a lost or discarded packet near the
   start or end of an RTP session is within a gap or a burst it is
   assumed that the RTP session is preceded and followed by at least
   Gmin received packets, and that the time of the report is followed by
   at least Gmin received packets.

   A gap has the property that any lost or discarded packets within the
   gap must be preceded and followed by at least Gmin packets that were
   received and not discarded.  This gives a maximum loss/discard
   density within a gap of: 1 / (Gmin + 1).

   A Gmin value of 16 is RECOMMENDED as it results in gap
   characteristics that correspond to good quality (i.e. low packet loss
   rate, a minimum distance of 16 received packets between lost packets)
   and hence differentiates nicely between good and poor quality
   periods.

   For example, a 1 denotes a received, 0 a lost, and X a discarded
   packet in the following pattern covering 64 packets:


   11110111111111111111111X111X1011110111111111111111111X111111111
   |---------gap----------|--burst---|------------gap------------|


   The burst consists of the twelve packets indicated above, starting at
   a discarded packet and ending at a lost packet.  The first gap starts
   at the beginning of the session and the second gap ends at the time
   of the report.

   If the packet spacing is 10 ms and the Gmin value is the recommended
   value of 16, the burst duration is 120 ms, the burst density 0.33,
   the gap duration 230 ms + 290 ms =3D 520 ms, and the gap density 0.04.

   This would result in reported values as follows (see field
   descriptions for semantics and details on how these are calculated):

   loss density          12, which corresponds to 5%
   discard density       12, which corresponds to 5%
   burst density         84, which corresponds to 33%
   gap density           10, which corresponds to 4%
   burst duration       120, value in milliseconds
   gap duration         520, value in milliseconds

   burst density: 8 bits
        The fraction of RTP data packets within burst periods since the
        beginning of reception that were either lost or discarded.  This
        value is expressed as a fixed point number with the binary point



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        at the left edge of the field.  It is calculated by dividing the
        total number of packets lost or discarded (excluding duplicate
        packet discards) within burst periods by the total number of
        packets expected within the burst periods, multiplying the
        result of the division by 256, limiting the maximum value to 255
        (to avoid overflow), and taking the integer part.

   gap density: 8 bits
        The fraction of RTP data packets within inter-burst gaps since
        the beginning of reception that were either lost or discarded.
        The value is expressed as a fixed point number with the binary
        point at the left edge of the field.  It is calculated by
        dividing the total number of packets lost or discarded
        (excluding duplicate packet discards) within gap periods by the
        total number of packets expected within the gap periods,
        multiplying the result of the division by 256, limiting the
        maximum value to 255 (to avoid overflow), and taking the integer
        part.

   burst duration: 16 bits
        The mean duration, expressed in milliseconds, of the burst
        periods that have occurred since the beginning of reception.
        The duration of each period is calculated based upon the packets
        that mark the beginning and end of that period.  It is equal to
        the timestamp of the end packet, plus the duration of the end
        packet, minus the timestamp of the beginning packet.  If the
        actual values are not available, estimated values MUST be used.
        If there have been no burst periods, the burst duration value
        MUST be zero.

   gap duration: 16 bits
        The mean duration, expressed in milliseconds, of the gap periods
        that have occurred since the beginning of reception.  The
        duration of each period is calculated based upon the packet that
        marks the end of the prior burst and the packet that marks the
        beginning of the subsequent burst. It is equal to the timestamp
        of the subsequent burst packet, minus the timestamp of the prior
        burst packet, plus the duration of the prior burst packet.  If
        the actual values are not available, estimated values MUST be
        used.  In the case of a gap that occurs at the beginning of
        reception, the sum of the timestamp of the prior burst packet
        and the duration of the prior burst packet are replaced by the
        reception start time.  In the case of a gap that occurs at the
        end of reception, the timestamp of the subsequent burst packet
        is replaced by the reception end time.  If there have been no
        gap periods, the gap duration value MUST be zero.

4.7.3 Delay Metrics



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   For the purpose of the following definitions, the RTP interface is
   the interface between the RTP instance and the voice application
   (i.e.  FEC, de-interleaving, de-multiplexing, jitter buffer). For
   example, the time delay due to RTP payload multiplexing would be
   considered to be part of the voice application or end-system delay
   whereas delay due to multiplexing RTP frames within a UDP frame would
   be considered part of the RTP reported delay.  This distinction is
   consistent with the use of RTCP for delay measurements.

   round trip delay: 16 bits
        The most recently calculated round trip time between RTP
        interfaces, expressed in milliseconds. This value is the time of
        receipt of the most recent RTCP packet from source SSRC, minus
        the LSR (last SR) time reported in its SR (sender report), minus
        the DLSR (delay since last SR) reported in its SR.  A non-zero
        LSR value is REQUIRED in order to calculate round trip delay. A
        value of 0 is permissible during the first two or three RTCP
        exchanges as insufficient data may be available to determine
        delay however MUST be populated as soon as a delay estimate is
        available.

   end system delay: 16 bits
        The most recently estimated end system delay, expressed in
        milliseconds.  End system delay is defined as the total
        encoding, decoding and jitter buffer delay determined at the
        reporting endpoint.  This is the time required for an RTP frame
        to be buffered, decoded, converted to analog form, looped back
        at the local analog interface, encoded, and made available for
        transmission as an RTP frame.  The manner in which this value is
        estimated is implementation dependent.  This parameter MUST be
        provided in all VoIP metrics reports.

   Note that the one way symmetric VoIP segment delay may be calculated
   from the round trip and end system delays as follows.  If the round
   trip delay is denoted RTD and the end system delays associated with
   the two endpoints are ESD(A) and ESD(B) then:


   one way symmetric voice path delay  =3D  ( RTD + ESD(A) + ESD(B) ) / 2


4.7.4 Signal Related Metrics

   The following metrics are intended to provide real time information
   related to the non-packet elements of the voice over IP system to
   assist with the identification of problems affecting call quality.
   The values identified below must be determined for the received audio
   signal. The information required to populate these fields may not be



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   available in all systems, although it is strongly recommended that
   this data SHOULD be provided to support problem diagnosis.

   signal level: 8 bits
        The voice signal relative level is defined as the ratio of the
        signal level to a reference digital milliwatt, expressed in
        decibels as a signed integer in two's complement form.  This is
        measured only for packets containing speech energy.  The intent
        of this metric is not to provide a precise measurement of the
        signal level but to provide a real time indication that the
        signal level may be excessively high or low.

        signal level =3D 10 log 10 ( rms talkspurt power (mW) )

        A value of 127 indicates that this parameter is unavailable.
        Typical values should be generally in the -15 to -20 dBm range.

   noise level: 8 bits
        The noise level is defined as the ratio of the silent period
        background noise level to a reference digital milliwatt,
        expressed in decibels as a signed integer in two's complement
        form.

        noise level =3D 10 log10 ( rms silence power (mW) )

        A value of 127 indicates that this parameter is unavailable.

   residual echo return loss (RERL): 8 bits
        The residual echo return loss is defined as the sum of the
        measured echo return loss (ERL) and the echo return loss
        enhancement (ERLE) expressed in dB as a signed integer in two's
        complement form.  It defines the ratio of a transmitted voice
        signal that is reflected back to the talker.  A low level of
        echo return loss (say less than 20 dB) in conjunction with some
        delay can lead to hollowness or audible echo.  A high level of
        echo return loss (say over 40 dB) is preferable.

   The ERL and ERLE parameters are often available directly from the
   echo canceler contained within the VoIP end system. They relate to
   the echo on the external network attached to the end point.

   In the case of a VoIP gateway this would be line echo that typically
   occurs at 2-4 wire conversion points in the network.  Echo return
   loss from typical 2-4 wire conversions can be in the 8-12 dB range.
   A line echo canceler can provide an ERLE of 30 dB or more and hence
   reduce this to 40-50 dB.  In the case of an IP phone this could be
   residual acoustic echo from speakerphone operation, and may more
   correctly be termed terminal coupling loss (TCL).  A typical handset



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   would result in 40-50 dB of echo due to acoustic feedback.

   Typical values for RERL are as follows:

   (i) IP gateway connected to circuit switched network with 2 wire loop
   Without echo cancellation, typical 2-4 wire converter ERL of 12 dB
   RERL =3D ERL + ERLE =3D 12 + 0 =3D 12 dB

   (ii) IP gateway connected to circuit switched network with 2 wire loop
   With echo canceler that improves echo by 30 dB
   RERL =3D ERL + ERLE =3D 12 + 30 =3D 42 dB

   (iii) IP phone with conventional handset
   Acoustic coupling from handset speaker to microphone 40 dB
   Residual echo return loss =3D TCL =3D 40 dB

   If we denote the "local" end of the VoIP path as A and the remote end
   as B and if the sender loudness rating (SLR) and receiver loudness
   rating (RLR) are known for A (default values 8 dB and 2 dB
   respectively), then the echo loudness level at end A (talker echo
   loudness rating or TELR) is given by:

   TELR(A) =3D SRL(A) + ERL(B) + ERLE(B) + RLR(A)

   TELR(B) =3D SRL(B) + ERL(A) + ERLE(A) + RLR(B)

   Hence in order to incorporate echo into a voice quality estimate at
   the A end of a VoIP connection it is desirable to send the ERL + ERLE
   value from B to A.

   For an IP phone with handset this metric MUST be set to the designed
   or measured terminal coupling loss, which would typically be 40-50
   dB.

   For a PC softphone or speakerphone this metric MUST be set to either
   the value reported by the acoustic echo canceler or to 127 to
   indicate an undefined value.

   For an IP gateway with ERL and ERLE measurements this metric MUST be
   reported as ERL + ERLE.

   For an IP gateway without ERL and ERLE measurement capability then
   this metric MUST be reported as 12 dB if line echo cancellation is
   disabled and 40 dB if line echo cancellation is enabled.

   Gmin
        See Configuration Parameters (Section 4.7.6, below).




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4.7.5 Call Quality or Transmission Quality Metrics

   The following metrics are direct measures of the call quality or
   transmission quality, and incorporate the effects of codec type,
   packet loss, discard, burstiness, delay etc.  These metrics may not
   be available in all systems however SHOULD be provided in order to
   support problem diagnosis.

   R factor: 8 bits
        The R factor is a voice quality metric describing the segment of
        the call that is carried over this RTP session.  It is expressed
        as an integer in the range 0 to 100, with a value of 94
        corresponding to "toll quality" and values of 50 or less
        regarded as unusable.  This metric is defined as including the
        effects of delay, consistent with ITU-T G.107 [4] and ETSI TS
        101 329-5 [2].

        A value of 127 indicates that this parameter is unavailable.

   ext. R factor: 8 bits
        The external R factor is a voice quality metric describing the
        segment of the call that is carried over a network segment
        external to the RTP segment, for example a cellular network. Its
        values are interpreted in the same manner as for the RTP R
        factor.  This metric is defined as including the effects of
        delay, consistent with ITU-T G.107 [4] and ETSI TS 101 329-5
        [2], and relates to the outward voice path from the Voice over
        IP termination for which this metrics block applies.

        A value of 127 indicates that this parameter is unavailable.

   Note that an overall R factor may be estimated from the RTP segment R
   factor and the external R factor, as follows:

   R total =3D RTP R factor + ext. R factor - 94

   MOS-LQ: 8 bits
        The estimated mean opinion score for listening quality (MOS-LQ)
        is a voice quality metric on a scale from 1 to 5, in which 5
        represents excellent and 1 represents unacceptable.  This metric
        is defined as not including the effects of delay and can be
        compared to MOS scores obtained from listening quality (ACR)
        tests. It is expressed as an integer in the range 10 to 50,
        corresponding to MOS x 10.  For example, a value of 35 would
        correspond to an estimated MOS score of 3.5.

        A value of 127 indicates that this parameter is unavailable.




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   MOS-CQ: 8 bits
        The estimated mean opinion score for conversational quality
        (MOS-CQ) is defined as including the effects of delay and other
        effects that would affect conversational quality.  The metric
        may be calculated by converting an R factor determined according
        to ITU-T G.107 [4] or ETSI TS 101 329-5 [2] into an estimated
        MOS using the equation specified in G.107.  It is expressed as
        an integer in the range 10 to 50, corresponding to MOS x 10, as
        for MOS-LQ.

        A value of 127 indicates that this parameter is unavailable.

4.7.6 Configuration Parameters

   Gmin: 8 bits
        The gap threshold.  This field contains the value used for this
        report block to determine if a gap exists.  The recommended
        value of 16 corresponds to a burst period having a minimum
        density of 6.25% of lost or discarded packets, which may cause
        noticeable degradation in call quality; during gap periods, if
        packet loss or discard occurs, each lost or discarded packet
        would be preceded by and followed by a sequence of at least 16
        received non-discarded packets.  Note that lost or discarded
        packets that occur within Gmin packets of a report being
        generated may be reclassified as being part of a burst or gap in
        later reports.  ETSI TS 101 329-5 [2] defines a computationally
        efficient algorithm for measuring burst and gap density using a
        packet loss/discard event driven approach.  This algorithm is
        reproduced in Appendix A.2 of the present document.  Gmin MUST
        not be zero and MUST be provided.

receiver configuration byte (RX config): 8 bits
        This byte consists of the following fields:


         0 1 2 3 4 5 6 7
        +-+-+-+-+-+-+-+-+
        |PLC|JBA|JB rate|
        +-+-+-+-+-+-+-+-+


        packet loss concealment (PLC): 2 bits
             Standard (11) / enhanced (10) / disabled (01) / unspecified
             (00).  When PLC =3D 11 then a simple replay or interpolation
             algorithm is being used to fill-in the missing packet.
             This is typically able to conceal isolated lost packets
             with loss rates under 3%.  When PLC =3D 10 then an enhanced
             interpolation algorithm is being used.  This would



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             typically be able to conceal lost packets for loss rates of
             10% or more.  When PLC =3D 01 then silence is inserted in
             place of lost packets.  When PLC =3D 00 then no information
             is available concerning the use of PLC however for some
             codecs this may be inferred.

        jitter buffer adaptive (JBA): 2 bits
             Adaptive (11) / non-adaptive (10) / reserved (01)/ unknown
             (00).  When the jitter buffer is adaptive then its size is
             being dynamically adjusted to deal with varying levels of
             jitter.  When non-adaptive, the jitter buffer size is
             maintained at a fixed level.  When either adaptive or non-
             adaptive modes are specified then the jitter buffer size
             parameters below MUST be specified.

        jitter buffer rate (JB rate): 4 bits
             J =3D adjustment rate (0-15). This represents the
             implementation specific adjustment rate of a jitter buffer
             in adaptive mode. This parameter is defined in terms of the
             approximate time taken to fully adjust to a step change in
             peak to peak jitter from 30 ms to 100 ms such that:

             adjustment time =3D 2 * J * frame size (ms)

             This parameter is intended only to provide a guide to the
             degree of "aggressiveness" of a an adaptive jitter buffer
             and may be estimated. A value of 0 indicates that the
             adjustment time is unknown for this implementation.

4.7.7 Jitter Buffer Parameters

   jitter buffer nominal size in frames (JB nominal): 8 bits
        This is the current nominal fill point within the jitter buffer,
        which corresponds to the nominal jitter buffer delay for packets
        that arrive exactly on time.  This parameter MUST be provided
        for both fixed and adaptive jitter buffer implementations.

   jitter buffer maximum size in frames (JB maximum): 8 bits
        This is the current maximum jitter buffer level corresponding to
        the earliest arriving packet that would not be discarded.  In
        simple queue implementations this may correspond to the nominal
        size. In adaptive jitter buffer implementations this value may
        dynamically vary up to JB abs max (see below).  This parameter
        MUST be provided for both fixed and adaptive jitter buffer
        implementations.

   jitter buffer absolute maximum size in frames (JB abs max): 8 bits
        This is the absolute maximum size that the adaptive jitter



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        buffer can reach under worst case jitter conditions.  This
        parameter MUST be provided for adaptive jitter buffer
        implementations and its value MUST be set to JB maximum for
        fixed jitter buffer implementations.

5. IANA Considerations

   This document defines a new RTP packet type, the extended report (XR)
   type, within the existing Internet Assigned Numbers Authority (IANA)
   registry of RTP RTCP Control Packet Types.  This document also
   defines a new IANA registry: the registry of RTP XR Block Types.
   Within this new registry, this document defines an initial set of
   seven block types and describes how the remaining types are to be
   allocated.

5.1 XR Packet Type

   The IANA SHALL register the RTP extended report (XR) packet defined
   by this document as packet type 207 in the registry of RTP RTCP
   Control Packet types (PT).

5.2 RTP XR Block Type Registry

   The IANA SHALL create the RTP XR Block Type Registry to cover the
   name space of the extended report block type (BT) field specified in
   Section 3 of this document.  The BT field contains eight bits,
   allowing 256 values.  The IANA SHALL manage the RTP XR Block Type
   Registry according to the Specification Required policy of RFC 2434
   [6].  Future specifications SHOULD attribute block type values in
   strict numeric order following the values attributed in this
   document:


   BT  name
   --  ----
    1  Loss RLE Report Block
    2  Duplicate RLE Report Block
    3  Timestamp Report Block
    4  Statistics Summary Report Block
    5  Receiver Timestamp Report Block
    6  DLRR Report Block
    7  VoIP Metrics Report Block


   Furthermore, future specifications SHOULD avoid the values 0 and 255.
   Doing so facilitates packet validity checking, since all-zeros and
   all-ones are values that might commonly be found in ill-formed
   packets.



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6. Security Considerations

   This document extends the RTCP reporting mechanism.  The security
   considerations that apply to RTCP reports also apply to XR reports.
   This section details the additional security considerations that
   apply to the extensions.

   The extensions introduce heightened confidentiality concerns.
   Standard RTCP reports contain a limited number of summary statistics.
   The information contained in XR reports is both more detailed and
   more extensive (covering a larger number of parameters).  The per-
   packet information contained in Loss RLE, Duplicate RLE, and
   Timestamp Report Blocks facilitates MINC inference of multicast
   distribution trees for RTP data packets, and inference of link
   characteristics such as loss and delay.  This inference reveals
   information that might otherwise be considered confidential to
   autonomous system administrators.  The VoIP Metrics Report Block
   provides information on the quality of ongoing voice calls.  Though
   such information might be carried in application specific format in
   standard RTP sessions, making it available in a standard format here
   makes it more available to potential eavesdroppers.

   No new mechanisms are introduced in this document to ensure
   confidentiality.  Standard encryption procedures can be used when
   confidentiality is a concern to end hosts.  Autonomous system
   administrators concerned about the loss of confidentiality regarding
   their networks can encrypt traffic, or filter it to exclude RTCP
   packets containing the XR report blocks concerned.

   Any encryption or filtering of XR report blocks entails a loss of
   monitoring information to third parties.  For example, a network that
   establishes a tunnel to encrypt VoIP Report Blocks denies that
   information to the service providers traversed by the tunnel.  The
   service providers cannot then monitor or respond to the quality of
   the VoIP calls that they carry, potentially creating problems for the
   network's users.  As a default, XR packets SHOULD NOT be encrypted or
   filtered.

   The extensions also make certain denial of service attacks easier.
   This is because of the potential to create RTCP packets much larger
   than average with the per packet reporting capabilities of the Loss
   RLE, Duplicate RLE, and Timestamp Report Blocks.  Because of the
   automatic bandwidth adjustment mechanisms in RTCP, if some session
   participants are sending large RTCP packets, all participants will
   see their RTCP reporting intervals lengthened, meaning they will be
   able to report less frequently.  To limit the effects of large
   packets, even in the absence of denial of service attacks,
   applications SHOULD limit the size of XR report blocks and employ the



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   thinning techniques described in this document in order to fit
   reports into the space available.

A. Algorithms

A.1 Sequence Number Interpretation

   This the algorithm suggested by Section 4.1 for keeping track of the
   sequence numbers from a given sender.  It implements the accounting
   practice required for the generation of Loss RLE Report Blocks.

   This algorithm keeps track of 16 bit sequence numbers by translating
   them into a 32 bit sequence number space.  The first packet received
   from a source is considered to have arrived roughly in the middle of
   that space.  Each packet that follows is placed either ahead or
   behind the prior one in this 32 bit space, depending upon which
   choice would place it closer (or, in the event of a tie, which choice
   would not require a rollover in the 16 bit sequence number).

   // The reference sequence number is an extended sequence number
   // that serves as the basis for determining whether a new 16 bit
   // sequence number comes earlier or later in the 32 bit sequence
   // space.
   u_int32 _src_ref_seq;
   bool    _uninitialized_src_ref_seq;

   // Place seq into a 32-bit sequence number space based upon a
   // heuristic for its most likely location.
   u_int32 extend_seq(const u_int16 seq) {

           u_int32 extended_seq, seq_a, seq_b, diff_a, diff_b;
           if(_uninitialized_src_ref_seq) {

                   // This is the first sequence number received.  Place
                   // it in the middle of the extended sequence number
                   // space.
                   _src_ref_seq                =3D seq | 0x80000000u;
                   _uninitialized_src_ref_seq  =3D false;
                   extended_seq                =3D _src_ref_seq;
           }
           else {

                   // Prior sequence numbers have been received.
                   // Propose two candidates for the extended sequence
                   // number: seq_a is without wraparound, seq_b with
                   // wraparound.
                   seq_a =3D seq | (_src_ref_seq & 0xFFFF0000u);
                   if(_src_ref_seq < seq_a) {



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                           seq_b  =3D seq_a - 0x00010000u;
                           diff_a =3D seq_a - _src_ref_seq;
                           diff_b =3D _src_ref_seq - seq_b;
                   }
                   else {
                           seq_b  =3D seq_a + 0x00010000u;
                           diff_a =3D _src_ref_seq - seq_a;
                           diff_b =3D seq_b - _src_ref_seq;
                   }

                   // Choose the closer candidate.  If they are equally
                   // close, the choice is somewhat arbitrary: we choose
                   // the candidate for which no rollover is necessary.
                   if(diff_a < diff_b) {
                           extended_seq =3D seq_a;
                   }
                   else {
                           extended_seq =3D seq_b;
                   }

                   // Set the reference sequence number to be this most
                   // recently-received sequence number.
                   _src_ref_seq =3D extended_seq;
           }

           // Return our best guess for a 32-bit sequence number that
           // corresponds to the 16-bit number we were given.
           return extended_seq;
   }

   A.2 Example Burst Packet Loss Calculation.

   This is an algorithm for measuring the burst characteristics for the
   VoIP Metrics Report Block (Section 4.7).  It is reproduced from ETSI
   TS 101 329-5 [2].  The algorithm is event driven and hence extremely
   computationally efficient.

   Given the following definition of states:


   state 1 =3D received a packet during a gap
   state 2 =3D received a packet during a burst
   state 3 =3D lost a packet during a burst
   state 4 =3D lost an isolated packet during a gap


   The "c" variables below correspond to state transition counts, i.e.
   c14 is the transition from state 1 to state 4. It is possible to



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   infer one of a pair of state transition counts to an accuracy of 1
   which is generally sufficient for this application.

   "pkt" is the count of packets received since the last packet was
   declared lost or discarded and "lost" is the number of packets lost
   within the current burst.  "packet_lost" and "packet_discarded" are
   Boolean variables that indicate if the event that resulted in this
   function being invoked was a lost or discarded packet.

   if(packet_lost) {
           loss_count++;
   }
   if(packet_discarded) {
           discard_count++;
   }
   if(!packet_lost && !packet_discarded) {
           pkt++;
   }
   else {
           if(pkt >=3D gmin) {
                   if(lost =3D=3D 1) {
                           c14++;
                   }
                   else {
                           c13++;
                   }
                   lost =3D 1;
                   c11 +=3D pkt;
           }
           else {
                   lost++;
                   if(pkt =3D=3D 0) {
                           c33++;
                   }
                   else {
                           c23++;
                           c22 +=3D (pkt - 1);
                   }
           }
           pkt =3D 0;
   }

   At each reporting interval the burst and gap metrics can be
   calculated as follows.

   // Calculate additional transition counts.
   c31 =3D c13;
   c32 =3D c23;



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   ctotal =3D c11 + c14 + c13 + c22 + c23 + c31 + c32 + c33;

   // Calculate burst and densities.
   p32 =3D c32 / (c31 + c32 + c33);
   if((c22 + c23) < 1) {
           p23 =3D 1;
   }
   else {
           p23 =3D 1 - c22/(c22 + c23);
   }
   burst_density =3D 256 * p23 / (p23 + p32);
   gap_density =3D 256 * c14 / (c11 + c14);

   // Calculate burst and gap durations in ms
   m =3D frameDuration_in_ms * framesPerRTPPkt;
   gap_length =3D (c11 + c14 + c13) * m / c13;
   burst_length =3D ctotal * m / c13 - lgap;

   /* calculate loss and discard densities */
   loss_density =3D 256 * loss_count / ctotal;
   discard_density =3D 256 * discard_count / ctotal;

Intellectual Property

   The IETF takes no position regarding the validity or scope of any
   intellectual property or other rights that might be claimed to
   pertain to the implementation or use of the technology described in
   this document or the extent to which any license under such rights
   might or might not be available; neither does it represent that it
   has made any effort to identify any such rights.  Information on the
   IETF's procedures with respect to rights in standards-track and
   standards-related documentation can be found in BCP 11 [3].  Copies
   of claims of rights made available for publication and any assurances
   of licenses to be made available, or the result of an attempt made to
   obtain a general license or permission for the use of such
   proprietary rights by implementors or users of this specification can
   be obtained from the IETF Secretariat.

   The IETF invites any interested party to bring to its attention any
   copyrights, patents or patent applications, or other proprietary
   rights which may cover technology that may be required to practice
   this standard.  Please address the information to the IETF Executive
   Director.

Full Copyright Statement

   Copyright (C) The Internet Society (2003). All Rights Reserved.




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   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 PURPOSE.

Acknowledgments

   We thank the following people: Colin Perkins, Steve Casner, and
   Henning Schulzrinne for their considered guidance; Sue Moon for
   helping foster collaboration between the authors; Magnus Westerlund
   for his detailed comments; Mounir Benzaid for drawing our attention
   to the reporting needs of MLDA; and Dorgham Sisalem and Adam Wolisz
   for encouraging us to incorporate MLDA block types.

Contributors

   The following people are the authors of this document:

   Kevin Almeroth, UCSB
   Ramon Caceres, ShieldIP
   Alan Clark, Telchemy
   Robert Cole, AT&T Labs
   Nick Duffield, AT&T Labs-Research
   Timur Friedman, Paris 6
   Kaynam Hedayat, Brix Networks
   Kamil Sarac, UT Dallas

   The principal people to contact regarding the individual report
   blocks described in this document are as follows:



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   sec. report block                     principal contributors
   ---- ------------                     ----------------------
   4.1  Loss RLE Report Block            Friedman, Caceres, Duffield
   4.2  Duplicate RLE Report Block       Friedman, Caceres, Duffield
   4.3  Timestamp Report Block           Friedman, Caceres, Duffield
   4.4  Statistics Summary Report Block  Almeroth, Sarac
   4.5  Receiver Timestamp Report Block  Friedman
   4.6  DLRR Report Block                Friedman
   4.7  VoIP Metrics Report Block        Clark, Cole, Hedayat


Authors' Addresses

   Kevin Almeroth <almeroth@cs.ucsb.edu>
   Department of Computer Science
   University of California
   Santa Barbara, CA 93106 USA

   Ramon Caceres <ramon@shieldip.com>
   ShieldIP, Inc.
   11 West 42nd Street, 31st Floor
   New York, NY 10036 USA

   Alan Clark <alan@telchemy.com>
   Telchemy Incorporated
   3360 Martins Farm Road, Suite 200
   Suwanee, GA 30024 USA
   Tel: +1 770 614 6944
   Fax: +1 770 614 3951

   Robert Cole <rgcole@att.com>
   AT&T Labs
   330 Saint Johns Street,
   2nd Floor
   Havre de Grace, MD 21078 USA
   Tel: +1 410 939 8732
   Fax: +1 410 939 8732

   Nick Duffield <duffield@research.att.com>
   AT&T Labs-Research
   180 Park Avenue, P.O. Box 971
   Florham Park, NJ 07932-0971 USA
   Tel: +1 973 360 8726
   Fax: +1 973 360 8050







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   Timur Friedman <timur.friedman@lip6.fr>
   Universite Pierre et Marie Curie (Paris 6)
   Laboratoire LiP6-CNRS
   8, rue du Capitaine Scott
   75015 PARIS France
   Tel: +33 1 44 27 71 06
   Fax: +33 1 44 27 74 95

   Kaynam Hedayat <khedayat@brixnet.com>
   Brix Networks
   285 Mill Road
   Chelmsford, MA 01824 USA
   Tel: +1 978 367 5600
   Fax: +1 978 367 5700

   Kamil Sarac <ksarac@utdallas.edu>
   Department of Computer Science (ES 4.207)
   Eric Jonsson School of Engineering & Computer Science
   University of Texas at Dallas
   Richardson, TX 75083-0688 USA
   Tel: +1 972 883 2337
   Fax: +1 972 883 2349

References

   The references are divided into normative references and non-
   normative references.

Normative References

   [1] S. Bradner, "Key words for use in RFCs to indicate requirement
   levels," BCP 14, RFC 2119, IETF, March 1997.

   [2] ETSI, "Quality of Service (QoS) measurement methodologies," ETSI
   TS 101 329-5 V1.1.1 (2000-11), November 2000.

   [3] R. Hovey and S. Bradner, "The Organizations Involved in the IETF
   Standards Process," best current practice BCP 11, RFC 2028, IETF,
   October 1996.

   [4] ITU-T, "The E-Model, a computational model for use in
   transmission planning," Recommendation G.107 (05/00), May 2000.

   [5] J. Reynolds and J. Postel, "Assigned Numbers," standard STD 2,
   RFC 1700, IETF, October 1994.

   [6] T. Narten and H. Alvestrand, "Guidelines for Writing an IANA
   Considerations Section in RFCs," best current practice BCP 26, RFC



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   2434, IETF, October 1998.

   [7] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A
   transport protocol for real-time applications," RFC 1889, IETF,
   February 1996.

Non-Normative References

   [8] A. Adams, T. Bu, R. Caceres, N. G. Duffield, T. Friedman, J.
   Horowitz, F. Lo Presti, S. B. Moon, V. Paxson, and D. Towsley, "The
   Use of End-to-End Multicast Measurements for Characterizing Internal
   Network Behavior," IEEE Communications Magazine, May 2000.

   [9] R. Caceres, N. G. Duffield, and T. Friedman, "Impromptu
   measurement infrastructures using RTP," Proc. IEEE Infocom 2002.

   [10] A. D. Clark, "Modeling the Effects of Burst Packet Loss and
   Recency on Subjective Voice Quality," Proc. IP Telephony Workshop
   2001.

   [11] D. Sisalem and A. Wolisz, "MLDA: A TCP-friendly Congestion
   Control Framework for Heterogeneous Multicast Environments", Proc.
   IWQoS 2000.




























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Hi all,

I just submitted the newest revision of the AVPF timing rules
simulations draft. This document shall be regarded as a companion
document to AVPF (currently draft-ietf-avt-rtcp-feedback-05.txt).

Until publication, the draft can be reached under:

http://www.pel.panasonic.de/ietf/docs/draft-ietf-avt-rtcp-feedback-sim-0
1.txt

Thanks,


Jose Rey
Mobile Multimedia Team -  MCOM Group
Panasonic European Laboratories GmbH
Langen, Germany
-
rey@panasonic.de
www.pel.panasonic.de
-
Tel: +49(0)6103-766-134
Fax: +49(0)6103-766-166
--

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



From mailnull@www1.ietf.org  Mon Mar  3 07:11:57 2003
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From: "Michael A. Ramalho" <mramalho@cisco.com>
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Subject: [AVT] Updated RGL Codec drafts
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Dear all,

Updated RGL codec and RGL RTP Payload format drafts
will soon be available via the internet-drafts directory.

Until they appear there, you may view them at the following URLs.

http://www.winlab.rutgers.edu/~ramalho/rgl_codec_01.txt
http://www.winlab.rutgers.edu/~ramalho/rgl_codec_01.html

http://www.winlab.rutgers.edu/~ramalho/rgl_rtp_01.txt
http://www.winlab.rutgers.edu/~ramalho/rgl_rtp_01.html

Comments and feedback welcomed.

See you in San Francisco,

Michael Ramalho

Michael A. Ramalho, Ph.D.
Office email: mramalho@cisco.com
Personal email: mar42@cornell.edu
Office: +1.941.708.4650
Cell: +1.941.544.2844

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



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To: Jose Rey <rey@panasonic.de>
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        "Akihiro@Isl. Mei. Co. Jp" <akihiro@isl.mei.co.jp>,
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In-Reply-To: Your message of "Mon, 03 Mar 2003 11:52:02 +0100."
             <NGBBJIBNJOKHDHFKEOMHGECAEKAA.rey@panasonic.de> 
Date: Mon, 03 Mar 2003 09:37:13 -0500
From: Colin Perkins <csp@csperkins.org>
Subject: [AVT] Re: draft-ietf-avt-rtcp-feedback-sim-01.txt
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Thanks. Is there a need to discuss AVPF at the meeting, or should we just
consider the last call?

Colin



--> Jose Rey writes:
>[ Mail was delivered through VPN PEL->MEI/AVCDC! ]
>
>Hi all,
>
>I just submitted the newest revision of the AVPF timing rules
>simulations draft. This document shall be regarded as a companion
>document to AVPF (currently draft-ietf-avt-rtcp-feedback-05.txt).
>
>Until publication, the draft can be reached under:
>
>http://www.pel.panasonic.de/ietf/docs/draft-ietf-avt-rtcp-feedback-sim-0
>1.txt
>
>Thanks,
>
>
>Jose Rey
>Mobile Multimedia Team -  MCOM Group
>Panasonic European Laboratories GmbH
>Langen, Germany
>-
>rey@panasonic.de
>www.pel.panasonic.de
>-
>Tel: +49(0)6103-766-134
>Fax: +49(0)6103-766-166
>--
>
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Mon, 03 Mar 2003 16:01:41 +0100
From: Jose Rey <rey@panasonic.de>
Subject: RE: [AVT] Re: draft-ietf-avt-rtcp-feedback-sim-01.txt
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[ Mail was delivered through VPN PEL->MEI/AVCDC! ]


I think both drafts are ready for last call.

José
> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Colin
> Perkins
> Sent: Monday, March 03, 2003 3:37 PM
> To: Jose Rey
> Cc: Avt@Ietf. Org; Akihiro@Isl. Mei. Co. Jp; Carsten Burmeister; Rolf
> Hakenberg; Casner@Acm. Org; Fukunaga444@Oki. Com; Noriyuki Sato; Joerg
> Ott
> Subject: [AVT] Re: draft-ietf-avt-rtcp-feedback-sim-01.txt
>
>
> Thanks. Is there a need to discuss AVPF at the meeting, or
> should we just
> consider the last call?
>
> Colin
>
>
>
> --> Jose Rey writes:
> >[ Mail was delivered through VPN PEL->MEI/AVCDC! ]
> >
> >Hi all,
> >
> >I just submitted the newest revision of the AVPF timing rules
> >simulations draft. This document shall be regarded as a companion
> >document to AVPF (currently draft-ietf-avt-rtcp-feedback-05.txt).
> >
> >Until publication, the draft can be reached under:
> >
> >http://www.pel.panasonic.de/ietf/docs/draft-ietf-avt-rtcp-fee
> dback-sim-0
> >1.txt
> >
> >Thanks,
> >
> >
> >Jose Rey
> >Mobile Multimedia Team -  MCOM Group
> >Panasonic European Laboratories GmbH
> >Langen, Germany
> >-
> >rey@panasonic.de
> >www.pel.panasonic.de
> >-
> >Tel: +49(0)6103-766-134
> >Fax: +49(0)6103-766-166
> >--
> >
> _______________________________________________
> 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 mailnull@www1.ietf.org  Mon Mar  3 18:01:57 2003
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Date: Mon, 03 Mar 2003 17:56:59 -0500
Subject: [AVT] REVISED: Protocol Action: RTP Payload Format for ETSI ES
 201 108 Distributed Speech Recognition Encoding to Proposed
 Standard
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The IESG has approved the Internet-Draft 'RTP Payload Format for 
ETSI ES 201 108 Distributed Speech Recognition Encoding' 
<draft-ietf-avt-dsr-05.txt> as a Proposed Standard.  This document is 
the product of the Audio/Video Transport Working Group.  The IESG 
contact persons are Allison Mankin and Scott Bradner.
 
 
Technical Summary

This document specifies an RTP payload format for encapsulating ETSI
Standard ES 201 108 front-end signal processing feature streams for
distributed speech recognition (DSR) systems. The use of RTP enables
the distributed systems scenarios to be broader. The payload consists
of concatenated Frame Pairs (FP) of the ETSI-defined inputs, and 
guidance is provided on the size and timing issues of placing the voice 
recognition signals in the RTP payloads.

Besides defining the payload, the document specifies and registers the
MIME subtype audio/dsr-es201108.

Working Group Summary

The Audio-Video Transport Working Group supported advancement
of this specification. In addition, there has been coordination
with respect to the Speech Services Control Working Group.

Protocol Quality

The protocol was reviewed for the IESG by Allison Mankin.

RFC Editor Note:

RFC Editor, please add after the References a section titled
"Copyright Notice," containing RFC 2026, Section 10.4 (C), followed
by a section titled "IPR Notices", containing RFC 2026, Sections
10.4 (A) and 10.4(B)
_______________________________________________
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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		: Enhanced Compressed RTP (CRTP) for links with high 
                          delay,packet loss and reordering
	Author(s)	: T. Koren, S. Casner et al.
	Filename	: draft-ietf-avt-crtp-enhance-07.txt
	Pages		: 0
	Date		: 2003-3-3
	
This document describes a header compression scheme for point to 
point links with packet loss and long delays. It is based on 
Compressed Real-time Transport Protocol (CRTP), the IP/UDP/RTP 
header compression described in RFC 2508. CRTP does not perform well 
on such links: packet loss results in context corruption and due to 
the long delay, many more packets are discarded before the context 
is repaired. To correct the behavior of CRTP over such links, a few 
extensions to the protocol are specified here. The extensions aim to 
reduce context corruption by changing the way the compressor updates 
the context at the decompressor: updates are repeated and include 
updates to full and differential context parameters. With these 
extensions, CRTP performs well over links with packet loss, packet 
reordering and long delays.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-crtp-enhance-07.txt

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_______________________________________________
Audio/Video Transport Working Group
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From: "Balasubramanian Pitchandi" <bs@utstar.com>
To: <avt@ietf.org>
Date: Tue, 4 Mar 2003 11:01:50 -0500
Organization: UTStarcom Inc
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Subject: [AVT] Status of draft-ietf-avt-profile-new-12.txt
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Hello,

 

The AVT Profile draft draft-ietf-avt-profile-new-12.txt has expired on May
2002. Is there a plan to come out with a new draft replacing this?

 

Thanks,

Bala

______________________

Bala Pitchandi

UTStarcom Inc.


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<p class=3DMsoNormal><font size=3D1 face=3DVerdana><span =
style=3D'font-size:9.0pt;
font-family:Verdana'>Hello,</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DVerdana><span =
style=3D'font-size:9.0pt;
font-family:Verdana'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DVerdana><span =
style=3D'font-size:9.0pt;
font-family:Verdana'>The AVT Profile draft =
draft-ietf-avt-profile-new-12.txt
has expired on May 2002. Is there a plan to come out with a new draft =
replacing
this?</span></font></p>

<p class=3DMsoAutoSig><font size=3D2 face=3DTahoma><span =
style=3D'font-size:11.0pt;
font-family:Tahoma'>&nbsp;</span></font></p>

<p class=3DMsoAutoSig><font size=3D1 face=3DVerdana><span =
style=3D'font-size:7.5pt;
font-family:Verdana'>Thanks,</span></font></p>

<p class=3DMsoAutoSig><font size=3D1 face=3DVerdana><span =
style=3D'font-size:7.5pt;
 font-family:Verdana'>Bala</span></font></p>

<p class=3DMsoAutoSig><font size=3D1 face=3DVerdana><span =
style=3D'font-size:7.5pt;
font-family:Verdana'>______________________</span></font></p>

<p class=3DMsoAutoSig><strong><b><font size=3D1 face=3DVerdana><span
 =
style=3D'font-size:7.5pt;font-family:Verdana'>Bala</span></font></b></str=
ong><strong><b><font
size=3D1 face=3DVerdana><span =
style=3D'font-size:7.5pt;font-family:Verdana'>
Pitchandi</span></font></b></strong></p>

<p class=3DMsoAutoSig><strong><b><font size=3D1 face=3DVerdana><span
style=3D'font-size:7.5pt;font-family:Verdana'>UTStarcom =
Inc.</span></font></b></strong></p>

</div>

</body>

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



From mailnull@www1.ietf.org  Tue Mar  4 11:11:01 2003
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Hi,

draft-ietf-avt-profile-new-12.txt is not expired as it is tentatively 
approved by IESG. It awaits a few small clarifications on 
draft-ietf-avt-rtp-new-11.txt. When these are performed the both of them 
will go to publication, if IESG approves these changes.

Cheers

Magnus


Balasubramanian Pitchandi wrote:

> Hello,
>
>  
>
> The AVT Profile draft draft-ietf-avt-profile-new-12.txt has expired on 
> May 2002. Is there a plan to come out with a new draft replacing this?
>
>  
>
> Thanks,
>
> Bala
>
> ______________________
>
> Bala Pitchandi
>
> UTStarcom Inc.
>

-- 

Magnus Westerlund 

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



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



From mailnull@www1.ietf.org  Tue Mar  4 11:13:59 2003
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	(envelope-from sato652@oki.com)
Reply-To: <sato652@oki.com>
From: "Noriyuki SATO" <sato652@oki.com>
To: "Jose Rey" <rey@panasonic.de>, "Colin Perkins" <csp@csperkins.org>
Cc: "Avt@Ietf. Org" <avt@ietf.org>,
        "Akihiro@Isl. Mei. Co. Jp" <akihiro@isl.mei.co.jp>,
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        "Noriyuki Sato" <sato652@oki.co.jp>,
        "Joerg Ott" <jo@tzi.uni-bremen.de>
Subject: RE: [AVT] Re: draft-ietf-avt-rtcp-feedback-sim-01.txt
Date: Tue, 4 Mar 2003 14:04:31 +0900
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Dear Colin,

I agree with Jose.
It would be appreciated if you would start the procedure of WGLC.

Best regards,
Noriyuki Sato

> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Jose Rey
> Sent: Tuesday, March 04, 2003 12:02 AM
> To: Colin Perkins
> Cc: Avt@Ietf. Org; Akihiro@Isl. Mei. Co. Jp; Carsten Burmeister; Rolf Hakenberg; Casner@Acm. Org; Fukunaga444@Oki. Com;
> Noriyuki Sato; Joerg Ott
> Subject: RE: [AVT] Re: draft-ietf-avt-rtcp-feedback-sim-01.txt
>
>
> [ Mail was delivered through VPN PEL->MEI/AVCDC! ]
>
>
> José
> > -----Original Message-----
> > From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Colin
> > Perkins
> > Sent: Monday, March 03, 2003 3:37 PM
> > To: Jose Rey
> > Cc: Avt@Ietf. Org; Akihiro@Isl. Mei. Co. Jp; Carsten Burmeister; Rolf
> > Hakenberg; Casner@Acm. Org; Fukunaga444@Oki. Com; Noriyuki Sato; Joerg
> > Ott
> > Subject: [AVT] Re: draft-ietf-avt-rtcp-feedback-sim-01.txt
> >
> >
> > Thanks. Is there a need to discuss AVPF at the meeting, or
> > should we just
> > consider the last call?
> >
> > Colin
> >
> >
> >
> > --> Jose Rey writes:
> > >[ Mail was delivered through VPN PEL->MEI/AVCDC! ]
> > >
> > >Hi all,
> > >
> > >I just submitted the newest revision of the AVPF timing rules
> > >simulations draft. This document shall be regarded as a companion
> > >document to AVPF (currently draft-ietf-avt-rtcp-feedback-05.txt).
> > >
> > >Until publication, the draft can be reached under:
> > >
> > >http://www.pel.panasonic.de/ietf/docs/draft-ietf-avt-rtcp-fee
> > dback-sim-0
> > >1.txt
> > >
> > >Thanks,
> > >
> > >
> > >Jose Rey
> > >Mobile Multimedia Team -  MCOM Group
> > >Panasonic European Laboratories GmbH
> > >Langen, Germany
> > >-
> > >rey@panasonic.de
> > >www.pel.panasonic.de
> > >-
> > >Tel: +49(0)6103-766-134
> > >Fax: +49(0)6103-766-166
> > >--
> > >
> > _______________________________________________
> > 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
>

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



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Subject: [AVT] New version of draft-ietf-avt-rtcpssm-03.txt
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An updated version of draft-ietf-avt-rtcpssm was submitted on monday. While it 
is in the draft submission queue, you can download a copy from:

http://irg.attlabs.net/rtcp_ssm/draft-ietf-avt-rtcpssm-03.txt

Main features of the latest draft:

- Packet formats aligned with the Reporting Extensions draft.

- More concise rules for application of reflection vs summarisation and SDP 
syntax for initial signalling of parameters.

- IANA considerations

- Example applicability of security requirements to RTSP and SIP sessions.

We've requested a slot to discuss progress with the working group in San 
Francisco.

Thanks,
Julian Chesterfield





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



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Subject: [AVT] Last Call: Enhanced Compressed RTP (CRTP) for links with high
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The IESG has received a request from the Audio/Video Transport Working 
Group to consider Enhanced Compressed RTP (CRTP) for links with high 
delay, packet loss and reordering <draft-ietf-avt-crtp-enhance-07.txt> 
as a Proposed Standard.  

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the 
iesg@ietf.org or ietf@ietf.org mailing lists by 2003-3-19.

Files can be obtained via http://www.ietf.org/internet-drafts/draft-ietf-avt-crtp-enhance-07.txt



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



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	(envelope-from casner@acm.org)
Date: Tue, 4 Mar 2003 10:55:32 -0800 (PST)
From: Stephen Casner <casner@acm.org>
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
cc: bs@utstar.com, <avt@ietf.org>
Subject: Re: [AVT] Status of draft-ietf-avt-profile-new-12.txt
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On Tue, 4 Mar 2003, Magnus Westerlund wrote:

> draft-ietf-avt-profile-new-12.txt is not expired as it is tentatively
> approved by IESG. It awaits a few small clarifications on
> draft-ietf-avt-rtp-new-11.txt. When these are performed the both of them
> will go to publication, if IESG approves these changes.

What Magnus says is true.  But in addition, I have submitted updated
drafts rtp-new-12 and profile-new-13 as the means to convey the
clarifications to the IESG and the RFC Editor.  I did this after
receiving permission from our AD, Allison Mankin, since normally a
draft should not be updated after it is reviewed and approved by the
IESG.  The clarifications we have discussed are substantial enough,
especially when combined with the things I fixed to comply with the
ID-nits requirements (like fixing two spaces at the end of sentences,
and changing the order of some of the sections) that I decided trying
to convey all of that to the RFC Editor in the form of edit notes to
be applied to the previous drafts would not be efficient, effective,
or pleasant.  I am preparing a set of files for the IESG to explain
the significant clarifications and to make it easy to verify that the
other changes are insignificant.

Having refreshed expiration dates should also help folks like Bala who
have been concerned and asked about it.

The new drafts will work their way through the I-D administrator's
queue.  In the meantime, you can get them at:

ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-profile-new-13.ps
ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-profile-new-13.txt
ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-rtp-new-12.ps
ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-rtp-new-12.txt

                                                        -- Steve

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



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From: "Balasubramanian Pitchandi" <bs@utstar.com>
To: "'Stephen Casner'" <casner@acm.org>,
        "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>
Cc: <avt@ietf.org>
Subject: RE: [AVT] Status of draft-ietf-avt-profile-new-12.txt
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Thanks Magnus and Steve. Updating the expiration date and issuing a new
draft would be the ideal solution that would help folks who may not be
subscribing to this mailing list but use the document as a reference.

-- Bala
 

-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On Behalf Of Stephen
Casner
Sent: Tuesday, March 04, 2003 1:56 PM
To: Magnus Westerlund
Cc: bs@utstar.com; avt@ietf.org
Subject: Re: [AVT] Status of draft-ietf-avt-profile-new-12.txt

On Tue, 4 Mar 2003, Magnus Westerlund wrote:

> draft-ietf-avt-profile-new-12.txt is not expired as it is tentatively
> approved by IESG. It awaits a few small clarifications on
> draft-ietf-avt-rtp-new-11.txt. When these are performed the both of them
> will go to publication, if IESG approves these changes.

What Magnus says is true.  But in addition, I have submitted updated
drafts rtp-new-12 and profile-new-13 as the means to convey the
clarifications to the IESG and the RFC Editor.  I did this after
receiving permission from our AD, Allison Mankin, since normally a
draft should not be updated after it is reviewed and approved by the
IESG.  The clarifications we have discussed are substantial enough,
especially when combined with the things I fixed to comply with the
ID-nits requirements (like fixing two spaces at the end of sentences,
and changing the order of some of the sections) that I decided trying
to convey all of that to the RFC Editor in the form of edit notes to
be applied to the previous drafts would not be efficient, effective,
or pleasant.  I am preparing a set of files for the IESG to explain
the significant clarifications and to make it easy to verify that the
other changes are insignificant.

Having refreshed expiration dates should also help folks like Bala who
have been concerned and asked about it.

The new drafts will work their way through the I-D administrator's
queue.  In the meantime, you can get them at:

ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-profile-new-13.ps
ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-profile-new-13.txt
ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-rtp-new-12.ps
ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-rtp-new-12.txt

                                                        -- Steve

_______________________________________________
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 mailnull@www1.ietf.org  Tue Mar  4 16:03:58 2003
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Reply-To: <alan.duric@globalipsound.com>
From: "Alan Duric" <alan.duric@globalipsound.com>
To: "'Stephen Casner'" <casner@acm.org>
Cc: <bs@utstar.com>, <avt@ietf.org>
Subject: RE: [AVT] Status of draft-ietf-avt-profile-new-12.txt
Date: Tue, 4 Mar 2003 22:03:45 +0100
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Hi Steve,

one further comment, in section 4.5.9:

"4.5.9 GSM-EFR

   ... The packing is specified in ETSI Technical Specification TS 101
318."

packing is referring to document that is made historical (since
Dec-2001, see
http://webapp.etsi.org/workprogram/Report_WorkItem.asp?WKI_ID=5874), and
I'm not sure how good practice is to use historical document as
reference.

Regards,
Alan
 




> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On 
> Behalf Of Stephen Casner
> Sent: Tuesday, March 04, 2003 7:56 PM
> To: Magnus Westerlund
> Cc: bs@utstar.com; avt@ietf.org
> Subject: Re: [AVT] Status of draft-ietf-avt-profile-new-12.txt
> 
> 
> On Tue, 4 Mar 2003, Magnus Westerlund wrote:
> 
> > draft-ietf-avt-profile-new-12.txt is not expired as it is 
> tentatively 
> > approved by IESG. It awaits a few small clarifications on 
> > draft-ietf-avt-rtp-new-11.txt. When these are performed the both of 
> > them will go to publication, if IESG approves these changes.
> 
> What Magnus says is true.  But in addition, I have submitted 
> updated drafts rtp-new-12 and profile-new-13 as the means to 
> convey the clarifications to the IESG and the RFC Editor.  I 
> did this after receiving permission from our AD, Allison 
> Mankin, since normally a draft should not be updated after it 
> is reviewed and approved by the IESG.  The clarifications we 
> have discussed are substantial enough, especially when 
> combined with the things I fixed to comply with the ID-nits 
> requirements (like fixing two spaces at the end of sentences, 
> and changing the order of some of the sections) that I 
> decided trying to convey all of that to the RFC Editor in the 
> form of edit notes to be applied to the previous drafts would 
> not be efficient, effective, or pleasant.  I am preparing a 
> set of files for the IESG to explain the significant 
> clarifications and to make it easy to verify that the other 
> changes are insignificant.
> 
> Having refreshed expiration dates should also help folks like 
> Bala who have been concerned and asked about it.
> 
> The new drafts will work their way through the I-D 
> administrator's queue.  In the meantime, you can get them at:
> 
ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-profile-new-13
.ps
ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-profile-new-13
.txt
ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-rtp-new-12.ps
ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-rtp-new-12.txt

                                                        -- Steve

_______________________________________________
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 mailnull@www1.ietf.org  Tue Mar  4 16:04:43 2003
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From: "Alan Duric" <alan.duric@globalipsound.com>
To: "'Stephen Casner'" <casner@acm.org>
Cc: <bs@utstar.com>, <avt@ietf.org>
Subject: RE: [AVT] Status of draft-ietf-avt-profile-new-12.txt
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 yHi Steve,

one further comment, in section 4.5.9:

"4.5.9 GSM-EFR

   ... The packing is specified in ETSI Technical Specification TS 101
318."

packing is referring to document that is made historical (since
Dec-2001, see
http://webapp.etsi.org/workprogram/Report_WorkItem.asp?WKI_ID=5874), and
I'm not sure how good practice is to use historical document as
reference.

Regards,
Alan
 




> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On
> Behalf Of Stephen Casner
> Sent: Tuesday, March 04, 2003 7:56 PM
> To: Magnus Westerlund
> Cc: bs@utstar.com; avt@ietf.org
> Subject: Re: [AVT] Status of draft-ietf-avt-profile-new-12.txt
> 
> 
> On Tue, 4 Mar 2003, Magnus Westerlund wrote:
> 
> > draft-ietf-avt-profile-new-12.txt is not expired as it is
> tentatively
> > approved by IESG. It awaits a few small clarifications on
> > draft-ietf-avt-rtp-new-11.txt. When these are performed the both of 
> > them will go to publication, if IESG approves these changes.
> 
> What Magnus says is true.  But in addition, I have submitted
> updated drafts rtp-new-12 and profile-new-13 as the means to 
> convey the clarifications to the IESG and the RFC Editor.  I 
> did this after receiving permission from our AD, Allison 
> Mankin, since normally a draft should not be updated after it 
> is reviewed and approved by the IESG.  The clarifications we 
> have discussed are substantial enough, especially when 
> combined with the things I fixed to comply with the ID-nits 
> requirements (like fixing two spaces at the end of sentences, 
> and changing the order of some of the sections) that I 
> decided trying to convey all of that to the RFC Editor in the 
> form of edit notes to be applied to the previous drafts would 
> not be efficient, effective, or pleasant.  I am preparing a 
> set of files for the IESG to explain the significant 
> clarifications and to make it easy to verify that the other 
> changes are insignificant.
> 
> Having refreshed expiration dates should also help folks like
> Bala who have been concerned and asked about it.
> 
> The new drafts will work their way through the I-D
> administrator's queue.  In the meantime, you can get them at:
> 
ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-profile-new-13
.ps
ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-profile-new-13
.txt
ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-rtp-new-12.ps
ftp://ftp.packetdesign.com/outgoing/casner/draft-ietf-avt-rtp-new-12.txt

                                                        -- Steve

_______________________________________________
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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From: Stephen Casner <casner@acm.org>
To: Alan Duric <alan.duric@globalipsound.com>
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Subject: RE: [AVT] Status of draft-ietf-avt-profile-new-12.txt
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Alan,

> "4.5.9 GSM-EFR
>
>    ... The packing is specified in ETSI Technical Specification TS 101
> 318."
>
> packing is referring to document that is made historical (since
> Dec-2001, see
> http://webapp.etsi.org/workprogram/Report_WorkItem.asp?WKI_ID=5874), and
> I'm not sure how good practice is to use historical document as
> reference.

Does that mean GSM-EFR encoding is also historic?  Or just the
packing?

                                                        -- Steve

_______________________________________________
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Hi Steve,

> Does that mean GSM-EFR encoding is also historic?  Or just 
> the packing?

ETSI Technical Specification TS 101 318 refers just to packing of GSM
EFR (as well as for packing of GSM FR - 06.10 and GSM HR - 06.20). 
Possible "work around" could be if packing for GSM EFR is explained in a
same way as packing for GSM FR in section 4.5.8

Regards,
Alan
 

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



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Subject: [AVT] Gibson Magic over RTP?
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Hi,

Does anyone know of a plan to map Gibson Magic, which
is a layer-2 Ethernet audio/video LAN protocol, into RTP for
extension into the WAN environment?

http://www.gibsonmagic.com/specification.html

Cheers,
  Chuck Harrison
  Far Field Associates, LLC
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Tue Mar  4 19:59:08 2003
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From: Colin Perkins <csp@isi.edu>
Subject: [AVT] Agenda for San Francisco meeting
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Folks,

Enclosed is the agenda for the AVT meeting in San Francisco. Please send
any comments or corrections to the chairs. 

Those who have agenda time scheduled are requested to prepare a summary of
the issues they wish to discuss, and to send this to the mailing list ahead
of the meeting. This gives those who won't be attending the opportunity to
post their comments and concerns, and potentially even reach some consensus,
before the meeting time is consumed. It also gives other participants the
opportunity to consider the issue in more depth. Summary messages SHOULD be
sent to the mailing list by Monday 10th March.

We also request that all presenters send us their slides ahead of time,
preferably in PDF format. This has multiple purposes:

  - It streamlines the presentation process by having the presentations on
    a single laptop.

  - It avoids the delay that sometimes occurs in getting the presentations
    sent to us after the meeting, for inclusion in the minutes.

  - It allows us to preview the presentations to see if they are following
    the guidelines for what should be covered during the meeting.

Presentations SHOULD be submitted before 12 noon EST (GMT -05:00) on Friday
14th March, so we can include them on the IETF website for online viewing 
during the session. 

Agenda time is conditional on sending both a summary of open issues and 
a copy of your slides ahead of the meeting, and 5 minutes ahead doesn't
count!

Those presenting new drafts: briefly describe the basic problem and your
proposed solution, discuss known issues, limitations and problems, outline
where you want input from the working group, and how you wish us to
proceed.  When presenting work which has been previously discussed:
describe only the changes from the previous draft, highlight unresolved
problems, and propose future directions. Presenters are reminded of the
requirements for disclosure of intellectual property relating to
contributions, in section 10 of RFC 2026.

We strongly encourage all participants to read all relevent drafts before
the meeting, and to comment on the open issues raised on the mailing list.

Thanks for your assistance to try to conduct an effective meeting.

Colin & Steve


===============================================================================
Audio/Video Transport Working Group

CHAIRS: Steve Casner <casner@acm.org>
        Colin Perkins <csp@csperkins.org>


Wednesday, 19th March at 13:00-15:00
====================================

Introduction and status update				(Casner/Perkins)   15
	draft-ietf-avt-rtp-new-12.txt
	draft-ietf-avt-profile-new-13.txt
	draft-ietf-avt-uxp-05.txt
	draft-ietf-avt-ulp-07.txt          (Under revision)
	draft-ietf-avt-tcrtp-07.txt
	draft-ietf-avt-rtcp-feedback-05.txt
	draft-ietf-avt-rtcp-feedback-sim-01.txt
	draft-ietf-avt-crtp-enhance-07.txt (IETF last call)
	draft-ietf-avt-rtcp-bw-05.txt      (RFC Editor)
	draft-ietf-avt-rtp-mime-06.txt     (RFC Editor)
	draft-ietf-avt-dsr-05.txt          (RFC Editor)
	draft-ietf-avt-evrc-smv-03.txt     (RFC Editor)
		See also draft-garudadri-qcp-00.txt
	RFC 3497

The Secure Real-time Transport Protocol                                    15
	draft-ietf-avt-srtp-05.txt

Framing RTP over Connection-Oriented Transport 		(Lazzaro)          15
	draft-lazzaro-avt-rtp-framing-contrans-00.txt

RTP Extended Reports (RTP XR) 				(Friedman/Clark)   15
	draft-ietf-avt-rtcp-report-extns-03.txt

RTCP Extensions for SSM with unicast feedback		(Chesterfield)     15
	draft-ietf-avt-rtcpssm-03.txt

RTP Payload Format for JVT Video 			(Wenger)	   15
 	draft-ietf-avt-rtp-h264-01.txt

RTP Payload Format for uncompressed video 		(Gharai)	   15
	draft-ietf-avt-uncomp-video-02.txt
	

Thursday, 20th March at 09:00-11:30
===================================

RTP Payload Format for a 64 kbit/s transparent call	(Kreuter)          15
	draft-kreuter-avt-rtp-clearmode-01.txt
	See also the TDM emulation work in the PWE3 working group

The MIDI Wire Protocol Packetization (MWPP) 		(Lazzaro)          15
	draft-ietf-avt-mwpp-midi-rtp-06.txt
	draft-lazzaro-avt-mwpp-coding-guidelines-02.txt

RTP Payload Format for ATRAC-X				(Romaine)	   15
	draft-hatanaka-avt-rtp-atracx-01.txt

RTP Payload Format for iLBC Speech 			(Duric)            15
	draft-ietf-avt-rtp-ilbc-01.txt
	draft-ietf-avt-ilbc-codec-01.txt

RTP Payload Format for the Speex Codec 			(Herlein)          15
	draft-herlein-speex-rtp-profile-00.txt

RTP Payload Format for RGL Codec			(Ramalho)          15
	draft-ramalho-rgl-desc-01.txt 
	draft-ramalho-rgl-rtpformat-01.txt

RTP Payload Format for ETSI ES 202 050 DSR		(Xie)	           15
	draft-xie-avt-dsr-es202050-00.txt

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



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Subject: [AVT] ATRAC-X pre-presentation summary
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(apologies for those who already received this; mistyped initial 
address)

Hello all,

Summary of issues that will be covered during my ATRAC-X presentation 
update:

Clarifications and Additions:
- Timestamp and sample rate clarification
- Redundant data framework modified
- New method for multi-channel data decomposition introduced
- Multiplexing/QoS reasoning solidified

Time Stamp
- corresponds to presentation time in milliseconds

Sample Rate
- all ATRAC-X streams during single session must be identical

Revision of Redundant Data Framework
- Comment from Atlanta session : "For redundancy to be useful the 
redundant copy may need to be separated further than one slot."
- TimeStampeOffset added: 14bit unsigned offset value
- What's wrong with RFC 2198?
- RFC 2198 requires payload headers for primary and redundant data; 
creates unnecessary overhead
- Block length field defined at 10bits; insufficient for ATRAC-X's high 
bit-rate data
- Creates complexity in managing ATRAC-X's fragmentation data framework

Multi-channel decomposition
- Introduction of Block Coupling Pattern (BCP); 5bits
- Introduction of Total Channel Configuration (TCC); 3 bits
(table/figure to be presented; also in draft)

Further examples and and figures to be presented.

Looking forward to it.

Matt

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



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Subject: [AVT] Re: New I-Ds for Ogg technologies (Vorbis over RTP, Ogg file format,
 Mimetypes)
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Dear Allison, all,

thank you very much for getting the application/ogg MIME type on its 
way. The accompanying I-D still needed some work and after extensive 
discussions in the Xiph/Ogg community, we have now submitted an updated 
version. It should become available soon at 
http://www.ietf.org/internet-drafts/draft-pfeiffer-ogg-fileformat-02.txt
and may be put in front of the IESG again to be moved to an 
informational RFC.

Best Regards,

Silvia Pfeiffer.

Here's the list of major changes:

1) The paragraph of trouble changed to (Section 4):

Each Ogg page contains only one type of data as it belongs to one 
logical bitstream only.  Pages are of variable size and have a page 
header containing encapsulation and error recovery information.  Each 
logical bitstream in a physical Ogg bitstream starts with a special 
start page (bos=beginning of stream) and ends with a special page 
(eos=end of stream).  The bos page contains information to uniquely 
identify the codec type and MAY contain information to set up the 
decoding process.  The bos page SHOULD also contain information about 
the encoded media - for example, for audio, it should contain sample 
rate and number of channels.  By convention, the first bytes of the bos 
page contain magic data that uniquely identify the required codec.  It 
is the responsibility of anyone fielding a new codec to make sure it is 
possible to reliably distinguish his/her codec from all other codecs in 
use.  There is no fixed way to detect the end of the codec-identifying 
marker.  The format of the bos page is dependent on the codec and 
therefore MUST be given in the encapsulation specification of that 
logical bitstream type.  An example for such a media mapping is "Ogg 
Vorbis", which provides the name and revision of the Vorbis codec, the 
audio rate and the audio quality on the Ogg Vorbis bos page.  The vorbis 
bos page starts with the byte 0x01, followed by "vorbis" (a total of 7 
bytes of identifier).


2) Another major change was made to the paragraph on "Theora", the video 
codec (Section 4):

Another example is the upcoming "Ogg Theora" specification which 
encapsulates Theora-encoded video data and usually comes multiplexed 
with a Vorbis stream for an Ogg containing synchronised audio and video. 
  As Ogg does not specify temporal relationships between the 
encapsulated concurrently multiplexed bitstreams, the temporal 
synchronisation between the audio and video stream will be specified in 
this media mapping.

3) All the other changes were minor editorial ones.



Allison Mankin wrote:
> Silvia, AVT WG, Chairs,
> 
> We IESG passed draft-walleij-ogg-mediatype-08.txt as a Proposed
> Standard today.  It had had a four week Last Call some time back and
> was not changed, other than now having the internet-draft available to
> document the fileformat.  Congratulations!
> 
> The accompanying i-d, draft-pfeiffer-ogg-fileformat, was on the agenda
> as well and Harald Alvestrand raised a technical question that needs
> to be resolved before we can progress it as Informational.  Harald's
> written question:
> 
>   Technical omission (I think): The document says that the BOS page
>   "contains information to identify the codec type and any additional
>   information to set up the decoding process.  The format of that page
>   is therefore dependent on the codec and therefore MUST be given in
>   the encoding specification of that logical bitstream type."
> 
>   It's pretty clear that OGG players can identify which codec is being
>   used. But this document does not say which bytes of the BOS page can
>   be used to identify the codec. (first 4 bytes? first 16 bytes? up to
>   the first NUL?)
> 
> Thanks,
> 
> Allison
> 

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



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Thank you, and thanks for taking the time to understand my worry and 
addressing it!

Based on the info below, I should have no problems letting this progress!

--On onsdag, mars 05, 2003 16:26:33 +1100 Silvia.Pfeiffer@csiro.au wrote:

> Dear Allison, all,
>
> thank you very much for getting the application/ogg MIME type on its way.
> The accompanying I-D still needed some work and after extensive
> discussions in the Xiph/Ogg community, we have now submitted an updated
> version. It should become available soon at
> http://www.ietf.org/internet-drafts/draft-pfeiffer-ogg-fileformat-02.txt
> and may be put in front of the IESG again to be moved to an informational
> RFC.


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



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Subject: RE: [AVT] draft-ietf-avt-rtp-retransmission-06.txt
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Hi Steve & Colin,

I haven't seen the retransmission draft in the status update. We would
like you to consider it for WGLC as solicited below.

thanks,

José



> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Jose
> Rey
> Sent: Friday, February 28, 2003 2:41 PM
> To: Avt@Ietf. Org
> Cc: Colin Perkins; Casner@Acm. Org
> Subject: [AVT] draft-ietf-avt-rtp-retransmission-06.txt
>
>
> Hi all,
>
> we just submitted the hopefully final revision of the retransmission
> draft. Since the -05 draft there are only minor editorial changes, as
> commented on the list.
>
> We would like to ask the WG chairs to consider it for WGLC and you all
> to give it a final check,
>
> Until publication, the draft can be reached under:
>
> http://www.pel.panasonic.de/ietf/docs/draft-ietf-avt-rtp-retra
> nsmission-
> 06.txt
>
> Thanks,
>
>
> Jose Rey
> Mobile Multimedia Team -  MCOM Group
> Panasonic European Laboratories GmbH
> Langen, Germany
> -
> rey@panasonic.de
> www.pel.panasonic.de
> -
> Tel: +49(0)6103-766-134
> Fax: +49(0)6103-766-166
> --
>
>
> _______________________________________________
> 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		: Extended RTP Profile for RTCP-based Feedback(RTP/AVPF)
	Author(s)	: J. Ott, S. Wenger, N. Sato
	Filename	: draft-ietf-avt-rtcp-feedback-05.txt
	Pages		: 50
	Date		: 2003-3-4
	
Real-time media streams that use RTP are not resilient against 
packet losses.  Receivers may use the base mechanisms of RTCP to 
report packet reception statistics and thus allow a sender to adapt 
its transmission behavior in the mid-term as sole means for 
feedback and feedback-based error repair (besides a few codec-
specific mechanisms).  This document defines an extension to the 
Audio-visual Profile (AVP) that enables receivers to provide, 
statistically, more immediate feedback to the senders and thus 
allow for short-term adaptation and efficient feedback-based repair 
mechanisms to be implemented.  This early feedback profile (AVPF)
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	Title		: RTP Payload Format for Uncompressed Video
	Author(s)	: L. Gharai, C. Perkins
	Filename	: draft-ietf-avt-uncomp-video-02.txt
	Pages		: 17
	Date		: 2003-3-4
	
This memo specifies a packetization scheme for encapsulating
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Subject: [AVT] I-D ACTION:draft-ietf-avt-rtp-retransmission-06.txt
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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 Retransmission Payload Format
	Author(s)	: J. Rey et al.
	Filename	: draft-ietf-avt-rtp-retransmission-06.txt
	Pages		: 27
	Date		: 2003-3-4
	
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 [1] ( denoted 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 mailnull@www1.ietf.org  Wed Mar  5 13:28:36 2003
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Date: Wed, 05 Mar 2003 10:19:36 -0800
From: Chuck Harrison <cfharr@erols.com>
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Hi, all,

A few comments on this draft, which is looking very good so far.

---
In uncomp-video-02 we have:

4.3.  Payload Data

Depending on the video format, each RTP packet can include either a
single complete scan line, a single fragment of a scan line, or one
(or
more) complete scan lines plus a fragment of a scan line. [...]


I think there are additional logical possibilities allowed by
the document as it stands: a packet may begin with the tail
fragment of a line, then continue with 0 or more full lines, and
possibly the head fragment of another line.

If the intent is to exclude these possibilities, we need normative
language to say so. If these operational modes are permitted, the
quoted text above is misleading and should be amended.

---
On page 3 of the draft, we have

[...] This payload format
requires that applications using such synchronized ancillary data MUST
deliver it in separate RTP sessions which operate concurrently with
the
video session.  The normal RTP mechanisms SHOULD be used to
synchronize
the media.

I believe the MUST is inappropriate and SHOULD is preferable.

The idea that frame-accurate timecode will be carried as a
parallel RTP session is a laudable but forward-looking conjecture.
IMHO reliable use of this payload format in the near future
will depend on carriage of vertical interval timecode (VITC)
in an unused scan line which is part of the vertical blanking
interval. (This is normal video operational practice, as
mentioned in RFC 2431.)

At the top of page 2 of the draft, we have "...the contents of
horizontal and vertical blanking SHOULD NOT be transported." This
does not bother me as SHOULD NOT language does not prevent me from
carrying VITC when I need it, until the multisession RTP sync
issues get ironed out. On the other hand the MUST language of
page 3 is burdensome.

---
<nitpick>
A minor MUST/SHOULD nit at sec. 4.3 regarding filling to a pgroup
boundary: since a receiver MUST ignore the fill data, it seems
unneccessary that the sender MUST fill with zeroes, specifically.
The sender MUST fill, and SHOULD fill with zeroes.
</nitpick>

---

The MIME and SDP parameters do not include the video frame rate.
There are many plausible scenarios in which it is important
to distinguish alternative frame rates. For example, choosing
the right VOD stream during RTSP setup would need something like
this.

---

Cheers,
  Chuck Harrison
  Far Field Associates, LLC
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Wed Mar  5 16:48:56 2003
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From: "Adam Li" <adamli@divx.com>
To: "Ietf-Avt" <avt@ietf.org>, "3GPP2 TSG-C" <tsgc@3gpp2.org>
Cc: <keith.miller@nokia.com>, <randy@qualcomm.com>, <mccap@lucent.com>,
        <mdturner@lucent.com>, <smathai@lucent.com>, <lioy@qualcomm.com>,
        <zeng@packetvideo.com>, <sherwood@packetvideo.com>,
        <villa@icsl.ucla.edu>, <yllee@samsung.com>, <jeonghoon@samsung.com>,
        <tom.hiller@lucent.com>, <David.Leon@nokia.com>, <nleung@qualcomm.com>,
        <dgal@lucent.com>, <ajayrajkumar@lucent.com>,
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        <craig.greer@nokia.com>, <magda@qualcomm.com>, <casner@acm.org>,
        <ned.freed@mrochek.com>, <mankin@psg.com>, <hgarudad@qualcomm.com>,
        "'Colin Perkins'" <csp@isi.edu>
Date: Wed, 5 Mar 2003 11:58:44 -0800
Organization: DivXNetworks, Inc.
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Subject: [AVT] File format for EVRC/SMV vocoder
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Hi folks,

One issue just recently been raised about the file format for EVRC and
SMV data, which is specified the MIME registration for them. We very
much would like to hear your ideas and comments on this issue. 

As you may have known, we are working the IETF specification on the RTP
Payload Format for EVRC/SMV. It specifies how the EVRC/SMV payload is
carried in RTP sessions, and it also registered the MIME format for the
EVRC and SMV vocoders. The draft (draft-ietf-avt-evrc-smv) is currently
at the final editing stage at the IETF FRC Editor's queue.

There has been a file format defined in the draft-ietf-avt-evrc-smv at
the MIME registration for EVRC/SMV. The file format specified there
hasn't been changed much since the first version of the draft in 2000.

There is a new Internet Draft, draft-garudadri-qcp-00.txt, specifies a
different way of storage using the QCP file format, which has been used
for QCEPL-13K and EVRC by some applications.

Currently, it has been proposed to remove the file format specification
in the draft-ietf-avt-evrc-smv, and to expand the draft-garudadri-qcp to
be the only formal specification for the file format for EVRC/SMV. On
the other hand, there are also suggestions to keep the file format at it
is specified in the draft-ietf-avt-evrc-smv.

We would like to hear your suggestions, and hear from you how you think
would be the best way to proceed. Your comments are very much
appreciated.

Adam


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



From mailnull@www1.ietf.org  Wed Mar  5 16:49:03 2003
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From: "Adam Li" <adamli@icsl.ucla.edu>
To: "Ietf-Avt" <avt@ietf.org>, "3GPP2 TSG-C" <tsgc@3gpp2.org>
Cc: <keith.miller@nokia.com>, <randy@qualcomm.com>, <mccap@lucent.com>,
        <mdturner@lucent.com>, <smathai@lucent.com>, <lioy@qualcomm.com>,
        <zeng@packetvideo.com>, <sherwood@packetvideo.com>,
        <villa@icsl.ucla.edu>, <yllee@samsung.com>, <jeonghoon@samsung.com>,
        <tom.hiller@lucent.com>, <David.Leon@nokia.com>, <nleung@qualcomm.com>,
        <dgal@lucent.com>, <ajayrajkumar@lucent.com>,
        <Lars-Erik.Jonsson@epl.ericsson.se>,
        <magnus.westerlund@era-t.ericsson.se>, <vbharga@cisco.com>,
        <craig.greer@nokia.com>, <magda@qualcomm.com>, <casner@acm.org>,
        <ned.freed@mrochek.com>, <mankin@psg.com>, <hgarudad@qualcomm.com>,
        "'Colin Perkins'" <csp@isi.edu>
Date: Wed, 5 Mar 2003 12:13:02 -0800
Organization: UCLA
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Subject: [AVT] File format for EVRC/SMV vocoder
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Hi folks,

One issue just recently been raised about the file format for EVRC and
SMV data, which is specified the MIME registration for them. We very
much would like to hear your ideas and comments on this issue. 

As you may have known, we are working the IETF specification on the RTP
Payload Format for EVRC/SMV. It specifies how the EVRC/SMV payload is
carried in RTP sessions, and it also registered the MIME format for the
EVRC and SMV vocoders. The draft (draft-ietf-avt-evrc-smv) is currently
at the final editing stage at the IETF FRC Editor's queue.

There has been a file format defined in the draft-ietf-avt-evrc-smv at
the MIME registration for EVRC/SMV. The file format specified there
hasn't been changed much since the first version of the draft in 2000.

There is a new Internet Draft, draft-garudadri-qcp-00.txt, specifies a
different way of storage using the QCP file format, which has been used
for QCEPL-13K and EVRC by some applications.

Currently, it has been proposed to remove the file format specification
in the draft-ietf-avt-evrc-smv, and to expand the draft-garudadri-qcp to
be the only formal specification for the file format for EVRC/SMV. On
the other hand, there are also suggestions to keep the file format at it
is specified in the draft-ietf-avt-evrc-smv.

We would like to hear your suggestions, and hear from you how you think
would be the best way to proceed. Your comments are very much
appreciated.

Adam


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



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Thread-Topic: File format for EVRC/SMV vocoder
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To: <adamli@divx.com>, <avt@ietf.org>, <tsgc@3gpp2.org>
Cc: <randy@qualcomm.com>, <mccap@lucent.com>, <mdturner@lucent.com>,
        <smathai@lucent.com>, <lioy@qualcomm.com>, <zeng@packetvideo.com>,
        <sherwood@packetvideo.com>, <villa@icsl.ucla.edu>, <yllee@samsung.com>,
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        <magnus.westerlund@era-t.ericsson.se>, <vbharga@cisco.com>,
        <craig.greer@nokia.com>, <magda@qualcomm.com>, <casner@acm.org>,
        <ned.freed@mrochek.com>, <mankin@psg.com>, <hgarudad@qualcomm.com>,
        <csp@isi.edu>, <jlee@nextreaming.com>, <sakazawa@kddilabs.jp>
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Subject: [AVT] RE: File format for EVRC/SMV vocoder
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Adam,

Please include Jae-Yong Lee and Shigeyuki Sakazawa to this discussion. The are co-editors of the 3GPP2 file format specification along with Hari and myself.

regards,
Keith

-----Original Message-----
From: ext Adam Li [mailto:adamli@divx.com]
Sent: 05 March, 2003 1:59 PM
To: Ietf-Avt; 3GPP2 TSG-C
Cc: Miller Keith (NRC/Dallas); randy@qualcomm.com; mccap@lucent.com;
mdturner@lucent.com; smathai@lucent.com; lioy@qualcomm.com;
zeng@packetvideo.com; sherwood@packetvideo.com; villa@icsl.ucla.edu;
yllee@samsung.com; jeonghoon@samsung.com; tom.hiller@lucent.com; Leon
David (NRC/Dallas); nleung@qualcomm.com; dgal@lucent.com;
ajayrajkumar@lucent.com; Lars-Erik.Jonsson@epl.ericsson.se;
magnus.westerlund@era-t.ericsson.se; vbharga@cisco.com; Greer Craig
(Nokia/Dallas); magda@qualcomm.com; casner@acm.org;
ned.freed@mrochek.com; mankin@psg.com; hgarudad@qualcomm.com; 'Colin
Perkins'
Subject: File format for EVRC/SMV vocoder


Hi folks,

One issue just recently been raised about the file format for EVRC and
SMV data, which is specified the MIME registration for them. We very
much would like to hear your ideas and comments on this issue. 

As you may have known, we are working the IETF specification on the RTP
Payload Format for EVRC/SMV. It specifies how the EVRC/SMV payload is
carried in RTP sessions, and it also registered the MIME format for the
EVRC and SMV vocoders. The draft (draft-ietf-avt-evrc-smv) is currently
at the final editing stage at the IETF FRC Editor's queue.

There has been a file format defined in the draft-ietf-avt-evrc-smv at
the MIME registration for EVRC/SMV. The file format specified there
hasn't been changed much since the first version of the draft in 2000.

There is a new Internet Draft, draft-garudadri-qcp-00.txt, specifies a
different way of storage using the QCP file format, which has been used
for QCEPL-13K and EVRC by some applications.

Currently, it has been proposed to remove the file format specification
in the draft-ietf-avt-evrc-smv, and to expand the draft-garudadri-qcp to
be the only formal specification for the file format for EVRC/SMV. On
the other hand, there are also suggestions to keep the file format at it
is specified in the draft-ietf-avt-evrc-smv.

We would like to hear your suggestions, and hear from you how you think
would be the best way to proceed. Your comments are very much
appreciated.

Adam


_______________________________________________
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From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
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Subject: [AVT] SF Issue Summary: draft-lazzaro-avt-rtp-framing-contrans-00.txt
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Hi everyone,

This is the summary message for the Wednesday, 19th March AVT WG
presentation:

Framing RTP over Connection-Oriented Transport      (Lazzaro)   15
       draft-lazzaro-avt-rtp-framing-contrans-00.txt

I'm CC'ing this to MMUSIC, given the topical overlap of the I-D.

This summary is somewhat long, because I describe the decisions that
will move the document forward, so that WG members who won't be in San
Francisco can voice their views on the lists by responding to this
post. My presentation in San Francisco will put the 3 open issues
described below up on the screen, to see if at least within the room a
consensus can be found.

---

This I-D began with an extended discussion on the WG:

http://www1.ietf.org/mail-archive/working-groups/avt/current/msg01847.html

Three requirements arose from this discussion:

  [1] A memo was needed to document the "classical" framing of RTP over
  TCP, to facilitate interoperability with applications that use it
  (several are mentioned in the linked WG thread).

  [2] There was an MMUSIC I-D that defined SDP conventions for setting
  up a TCP media session (draft-ietf-mmusic-sdp-comedia-04.txt, soon to
  be draft-ietf-mmusic-sdp-comedia-05.txt), but comedia defines neither
  the SDP fmt token to specify an RTP/AVP stream, nor the TCP framing 
  such a stream should use.

  [3] RTSP has support for interleaving RTP and RTCP media streams into
  the TCP stream used for RTSP control method traffic. However, although
  RTSP also has the syntactic power to specify non-interleaved TCP session
  in the SETUP method, there is (apparently) no semantics in common use
  for this syntax.

Requirement [1] is simple to address: write a new I-D to document
classical RTP TCP framing. Sections 2-3 of
draft-lazzaro-avt-rtp-framing-contrans-00.txt attempt to do this.

#####> Open Issue #1

Does draft-lazzaro-avt-rtp-framing-contrans-00.txt accurately capture
classical TCP framing, as used in practice? In specific, are the
details in Section 3 correct?

----

The second requirement is more controversial. Given that we try to
avoid "there's more than one way to do it", it would be best if there
was one RTP/AVP TCP framing method for use with
draft-ietf-mmusic-sdp-comedia-05.txt. The discussion in the WG thread
linked above did not reach consensus, but instead ended with two
different options:

   A. Define classical TCP framing as the framing method for RTP/AVP
      to use with draft-ietf-mmusic-sdp-comedia-05.txt.

   B. Write an AVT I-D that describes RTSP interleaved-mode
      RTP/AVP framing, as run over a stand-alone TCP stream.
      Define this framing method as the framing method to use
      with draft-ietf-mmusic-sdp-comedia-05.txt.

Section 4 of draft-lazzaro-avt-rtp-framing-contrans-00.txt implements
"A". Two more open issues:

#####> Open Issue #2

Is A or B above the right path to take? Or, should we define SDP fmt
codes for both A and B?

----

and

#####> Open Issue #3

If A is the right option to take, are the SDP details in Section 4 of
the draft-lazzaro-avt-rtp-framing-contrans-00.txt acceptable to
everyone?

----

The final requirement involves RTSP support for non-interleaved TCP.
Section 5 of draft-lazzaro-avt-rtp-framing-contrans-00.txt defines
candidate RTSP semantics to use the classical TCP framing in RTSP.

However, recent discussions in the Rtspspec-bugs transcripts seem to
indicate that the editors of the core RTSP draft would prefer to
handle this issue as a part of the RTSP draft document. So, unless I
hear otherwise, I would think that deleting Section 5, and all
mentions of RTSP, from draft-lazzaro-avt-rtp-framing-contrans-00.txt,
is appropriate.

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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In-Reply-To: Your message of "Wed, 05 Mar 2003 10:19:36 PST."
             <3E663FB8.BC127026@erols.com> 
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From: Colin Perkins <csp@csperkins.org>
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Hi,

Thanks for the comments. A couple of replies inline.

--> Chuck Harrison writes:
>
>A few comments on this draft, which is looking very good so far.
>
>---
>In uncomp-video-02 we have:
>
>4.3.  Payload Data
>
>Depending on the video format, each RTP packet can include either a
>single complete scan line, a single fragment of a scan line, or one
>(or
>more) complete scan lines plus a fragment of a scan line. [...]
>
>
>I think there are additional logical possibilities allowed by
>the document as it stands: a packet may begin with the tail
>fragment of a line, then continue with 0 or more full lines, and
>possibly the head fragment of another line.
>
>If the intent is to exclude these possibilities, we need normative
>language to say so. If these operational modes are permitted, the
>quoted text above is misleading and should be amended.

Those modes are permitted, so I guess we need to clarify the text. Can you
suggest some wording?

>---
>On page 3 of the draft, we have
>
>[...] This payload format
>requires that applications using such synchronized ancillary data MUST
>deliver it in separate RTP sessions which operate concurrently with
>the
>video session.  The normal RTP mechanisms SHOULD be used to
>synchronize
>the media.
>
>I believe the MUST is inappropriate and SHOULD is preferable.
>
>The idea that frame-accurate timecode will be carried as a
>parallel RTP session is a laudable but forward-looking conjecture.
>IMHO reliable use of this payload format in the near future
>will depend on carriage of vertical interval timecode (VITC)
>in an unused scan line which is part of the vertical blanking
>interval. (This is normal video operational practice, as
>mentioned in RFC 2431.)
>
>At the top of page 2 of the draft, we have "...the contents of
>horizontal and vertical blanking SHOULD NOT be transported." This
>does not bother me as SHOULD NOT language does not prevent me from
>carrying VITC when I need it, until the multisession RTP sync
>issues get ironed out. On the other hand the MUST language of
>page 3 is burdensome.

I'm still not convinced that an additional timecode is necessary: it should
be derivable from the RTP timestamp and RTCP SR packets. Accordingly if you
wish to synchronize the video to another source, you should use the normal
RTCP method, rather than sending an additional timecode stream (indeed, if
you use RTSP, I _think_ there is already a mapping function between the RTP
timestamp and SMPTE timecode).

>---
><nitpick>
>A minor MUST/SHOULD nit at sec. 4.3 regarding filling to a pgroup
>boundary: since a receiver MUST ignore the fill data, it seems
>unneccessary that the sender MUST fill with zeroes, specifically.
>The sender MUST fill, and SHOULD fill with zeroes.
></nitpick>

Okay. Is there a reason why zero fill doesn't work?

>---
>
>The MIME and SDP parameters do not include the video frame rate.
>There are many plausible scenarios in which it is important
>to distinguish alternative frame rates. For example, choosing
>the right VOD stream during RTSP setup would need something like
>this.

You can use the SDP "a=framerate:" attribute, defined in RFC 2327.

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



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To: avt@ietf.org, mmusic@ietf.org
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Subject: [AVT] new I-D on temporal URI addresses
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Hi all,

we have submitted a proposal to specify temporal URI fragments on 
time-continuous Web resources (such as audio and video files/streams). I 
would like to ask for your brains on whether this is a problem that 
you've stumbled across, too, and whether you like the solution that we 
are proposing.

http://www.ietf.org/internet-drafts/draft-pfeiffer-temporal-fragments-00.txt

THE PROBLEM:
------------
We can see a need for the direct specification of points of interest in 
a URI for a/v. There are several uses for it:

1) Representation of search results:
When typing a search term into your favourite search engine, you would 
like it to search on a/v and bring up the link to an offset in a video 
where the search term matches. e.g. goal in last week's soccer matches

2) Favourite links:
When sending a link to a video to a friend, you'd really like to be able 
to specify the main part of your interest in it. e.g. look at my new 
cool sunglasses in my friend's wedding video


THE SOLUTION:
-------------
What we want is to address fragment offsets into videos. Therefore, the 
"#fragment" part of a URI naturally lends itself for including this 
address specification. For specification of the time points, we were 
inspired by the timestamp specification in RTSP. Therefore, we are 
proposing the following pattern:

#@npt=14.5

-> the "#" part signifies a fragment specification
-> the "@" tells us to look "at" that time specification
-> the "npt=" part is a time scheme just like in RTSP and we are also 
proposing npt, smpte, smpte-30-drop, smpte-25, and clock
-> the time format itself depends on the time scheme, this one 
signifying a 14.5 seconds offset

[for more details refer to the I-D]


THE USE:
--------
Here are examples for temporal URI fragment offsets:

http://foo/bar.mov#@smpte=00:00:14:15
(start downloading at 14 seconds and 15 frames into bar.mov)

rtsp://foo/bar.avi#@utc=20030308T143000.00Z
(start streaming at 8th March 2003 at half past 2pm)

http://foo/bar.mpg#@start
(start downloading from the start)

rtsp://foo/bar.rm#@now
(start streaming now)

---
We wanted to make it simple, intuitive and human readable and have 
therefore come up with this solution. Please let me know what you think.

Cheers,

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



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From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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To: Silvia.Pfeiffer@csiro.au
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Subject: Re: [AVT] new I-D on temporal URI addresses
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Hi, Silvia.

I see the possibility of this proposal. However I have a number of 
comments on this proposal.

First is the possibility to specify this as a general mechanism. It 
seems that you want to have this to apply to all URI schemes. Is that 
really possible? How do you know that the use of @ will not collide with 
any usage? For example the rtsp scheme does not define how to interpret 
the fragment. So this is up to the server responding to a request. So 
this can potentially already be used by some implementations. Also does 
it make sense to be defined for all schemes. As this is currently 
related to media it seems that for the moment it applies to HTTP(S) and 
RTSP(U|S).

So a general question, what is the actually possibilities to standardize 
such a solution?

Second, is the usage of the fragment identifier. It seems that you tries 
to redefine the meaning of the fragment identifier. See below.

 From RFC 2396:

  "When a URI reference is used to perform a retrieval action on the
   identified resource, the optional fragment identifier, separated from
   the URI by a crosshatch ("#") character, consists of additional
   reference information to be interpreted by the user agent after the
   retrieval action has been successfully completed.  As such, it is not
   part of a URI, but is often used in conjunction with a URI.

      fragment      = *uric

   The semantics of a fragment identifier is a property of the data
   resulting from a retrieval action, regardless of the type of URI used
   in the reference.  Therefore, the format and interpretation of
   fragment identifiers is dependent on the media type [RFC2046] of the
   retrieval result.  The character restrictions described in Section 2
   for URI also apply to the fragment in a URI-reference.  Individual
   media types may define additional restrictions or structure within
   the fragment for specifying different types of "partial views" that
   can be identified within that media type."

Two things strikes me as not being taken into account by your draft in 
relation to this quote. The first is from the first paragraph of the 
quote. The fragment identifier is interpreted by the user agent. While 
in chapter 4 of your draft you clearly specifies that the server shall 
interpret the fragment. This redefines the behavior of the fragment 
identifier.

Secondly the interpretation of the fragment identifier is in relation to 
the data resulting in the request. So in fact it seems that you need to 
define the behavior on per media basis.

I think the use of fragment to specify where in a media resource is 
appropriate in some case. For example an RTSP URL that has a fragment of 
the proposed type is useful. However as RTSP works the server shall not 
interpret the fragment as this is pointless. Only the client needs to do 
this and use that to determine the range to include in the PLAY request. 
So again the usage rules in chapter 4 seems to be wrong.

It seems that the time specification should be complemented to allow 
specifying ranges. The result of a search or the part of your friends 
wedding video that you want to point out would in most cases only be a 
part of the video. Therefore both a start and stop time seems useful. 
Then it is also a very short step to having multiple ranges allow one to 
point out multiple parts of the same resource that are interesting.

Also I think you have a few strange thing in the definition. First you 
specify an attribute "start" that I think are pointless. Either you 
totally skip the fragment or you can use "npt=0". Secondly it the 
defaulting to "npt". This clashed with the general parameters. If you 
use start then you have actually wrote "npt=start" which isn't defined. 
"now" works as long as this is based on the RTSP definitions of NPT, 
however from your BNF definitions this is evidently not the case. So my 
proposal is to skip having any default time scheme if you are going to 
have general attributes that does not belong to a time scheme.

If you would like the server to take the range into account would not 
the use of query (?) be better? However I hope that somebody that are 
more will versed in URIs can give further suggestions.

I can see the usefulness of this proposal, however there is need to 
resolve a number of issues:
- A general standardization or target certain URI schemes?
- Applicability on different media types and protocols needs to better 
understood.
- Use of fragment or other URI syntax element, like parameter or query?
- Need for having ranges?
- The actual need for general parameters?

Regards

Magnus Westerlund 

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



_______________________________________________
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Subject: Re: [AVT] I-D ACTION:draft-ietf-avt-uncomp-video-02.txt
To: Chuck Harrison <cfharr@erols.com>
Cc: avt@ietf.org, Ladan Gharai <ladan@east.isi.edu>
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Hi all,

I concur with chuck, this is looking very good indeed.

Just a few remarks

*****************************
I am not sure I always understand the "pixels" and "#words" second and third column in the tables ?

For exemple for monochrome 10 bits pixels is "4" so that would hint it means "number of pixels per Pgroup"

But then the next line is "4" for 4:1:1 while there are 8 pixels per Pgroup?

Also the #words is 6x10, but these are actually "bits" rather than "words"?

I think these tables would benefit a few sentences giving the exact definition of each column.

*****************************
The 8 bit words tables probably contains a typo for the "octet alignement" of Interlaced 4:2:0 which should be "4x8/8 = 4"
no?

*****************************

in section 4.3 it says

"For RGB format video, samples are packed in order Red-Green-Blue.  All
samples are the same bit size, which may be 8, 10, 12, or 16 bits.  If 8
bit samples are used, the pgroup is 3 octets. If 10 bit samples are
used, samples from adjacent pixels are packed with no padding, and the
pgroup is 15 octets (4 pixels).  Refer to Tables 1 thru 4.

For RGBA format video, samples are packed in order Red-Green-Blue-Alpha.
All samples are the same bit size, which may be 8, 10, 12, or 16 bits.
For pgroups refer to Tables 1 thru 4."

Unfortunately the tables 1 thru 4 do not explicitely mention any RGB or RGBA format(!)

I think it would be useful to add them, here are the tables
(not sure the notation R:G:B is appropriate, anyway  "RGB" is
what is declared in the mime type ...)


                                                 8 bit words
          Color                --------------------------------------
      Subsampling Pixels       #words octet alignment samples  pgroup
                                                               octets
     +-----------+------+---+ +------+---------------+---------------+
     |   R:G:B   |  1   | P | | 3x8  |  3x8/8 = 3    |   3   |   3   |
     +-----------+------+---+ +------+---------------+---------------+
     |  R:G:B:A  |  1   | P | | 4x8  |  4x8/8 = 4    |   4   |   4   |
     +-----------+------+---+ +------+---------------+---------------+


                                               10 bit words
          Color                --------------------------------------
      Subsampling Pixels       #words octet alignment samples  pgroup
                                                               octets
     +-----------+------+---+ +------+---------------+---------------+
     |   R:G:B   |  1   | P | | 3x10 |  4x30/8 = 15  |   12   | 15   |
     +-----------+------+---+ +------+---------------+---------------+
     |  R:G:B:A  |  1   | P | | 4x10 |    40/8 = 5   |    4   |  5   |
     +-----------+------+---+ +------+---------------+---------------+

                                            12 bit words
          Color                --------------------------------------
      Subsampling Pixels       #words octet alignment samples  pgroup
                                                               octets
     +-----------+------+---+ +------+---------------+---------------+
     |   R:G:B   |  1   | P | | 6x12 |  2x36/8 = 9   |    6   |  9   |
     +-----------+------+---+ +------+---------------+---------------+
     |  R:G:B:A  |  1   | P | | 4x12 |    48/8 = 6   |    4   |  6   |
     +-----------+------+---+ +------+---------------+---------------+

                                             16 bit words
          Color                --------------------------------------
      Subsampling Pixels       #words octet alignment samples  pgroup
                                                               octets
     +-----------+------+---+ +------+---------------+-------+-------+
     |   R:G:B   |  1   | P | | 3x16 | 3x16/8 = 6    |   3   |   6   |
     +-----------+------+---+ +------+---------------+-------+-------+
     |  R:G:B:A  |  1   | P | | 4x16 | 4x16/8 = 8    |   4   |   8   |
     +-----------+------+---+ +------+---------------+-------+-------+



*****************************

Since we have 4:4:4:4, one thing that should be added is 4:4:2:0
which is 4:2:0 (the most common raw video format
since it is the raw format of most codecs) plus
a transparency plane with one transparency word
per pixel i.e. same amount of transparency as luminance

This can be *very* useful for studio applications and is also
directly compressable using the profiles and levels of
MPEG-4 video that support the "shape" tool.

It simply needs to add 4:4:2:0 in the valid value list (section 6.1)
and in section 3 the relevant tables would be:


                                                 8 bit words
          Color                --------------------------------------
      Subsampling Pixels       #words octet alignment samples  pgroup
                                                               octets
     +-----------+------+---+ +------+---------------+---------------+
     |  4:4:2:0  |  4   | P | | 10x8 |  80/8 = 10    |   10  |   10  |
     +-----------+------+---+ +------+---------------+---------------+
     |  4:4:2:0  |  4   | I | | 8x8  |  64/8 = 8     |   8   |   8   |
     +-----------+------+---+ +------+---------------+---------------+



                                                10 bit words
          Color                --------------------------------------
      Subsampling Pixels       #words octet alignment samples  pgroup
                                                               octets
     +-----------+------+---+ +------+---------------+---------------+
     |  4:4:2:0  |  4   | P | | 10x10| 2x100/8 = 25  |   20  |   25  |
     +-----------+------+---+ +------+---------------+---------------+
     |  4:4:2:0  |  4   | I | | 8x10 |   80/8 = 10   |   8   |   10  |
     +-----------+------+---+ +------+---------------+---------------+


                                                 12 bit words
          Color                --------------------------------------
      Subsampling Pixels       #words octet alignment samples  pgroup
                                                               octets
     +-----------+------+---+ +------+---------------+---------------+
     |  4:4:2:0  |  4   | P | | 10x12|   120/8 = 15  |   10  |   15  |
     +-----------+------+---+ +------+---------------+---------------+
     |  4:4:2:0  |  4   | I | | 8x12 |   96/8 = 12   |   8   |   12  |
     +-----------+------+---+ +------+---------------+---------------+

                                                 16 bit words
          Color                --------------------------------------
      Subsampling Pixels       #words octet alignment samples  pgroup
                                                               octets
     +-----------+------+---+ +------+---------------+---------------+
     |  4:4:2:0  |  4   | P | | 10x16|   160/8 = 20  |   10  |   20  |
     +-----------+------+---+ +------+---------------+---------------+
     |  4:4:2:0  |  4   | I | | 8x16 |  128/8 = 16   |   8   |   16  |
     +-----------+------+---+ +------+---------------+---------------+




regards,


Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com


                                                                                                                                       
                                                                                                                                       
                                                   To:   avt@ietf.org                                                                  
                                                    Ladan Gharai <ladan@east.isi.edu>                                                  
                                                   cc:   (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                                    
                                                   Subject:    Re: [AVT] I-D ACTION:draft-ietf-avt-uncomp-video-02.txt                 
               Chuck Harrison                                                                                                          
               <cfharr@erols.com>                  Classification:                                                                     
                                                                                                                                       
               Sent by:                                                                                                                
               avt-admin@ietf.org                                                                                                      
                                                                                                                                       
               05/03/2003 19:19                                                                                                        
                                                                                                                                       
                                                                                                                                       




Hi, all,

A few comments on this draft, which is looking very good so far.

---
In uncomp-video-02 we have:

4.3.  Payload Data

Depending on the video format, each RTP packet can include either a
single complete scan line, a single fragment of a scan line, or one
(or
more) complete scan lines plus a fragment of a scan line. [...]


I think there are additional logical possibilities allowed by
the document as it stands: a packet may begin with the tail
fragment of a line, then continue with 0 or more full lines, and
possibly the head fragment of another line.

If the intent is to exclude these possibilities, we need normative
language to say so. If these operational modes are permitted, the
quoted text above is misleading and should be amended.

---
On page 3 of the draft, we have

[...] This payload format
requires that applications using such synchronized ancillary data MUST
deliver it in separate RTP sessions which operate concurrently with
the
video session.  The normal RTP mechanisms SHOULD be used to
synchronize
the media.

I believe the MUST is inappropriate and SHOULD is preferable.

The idea that frame-accurate timecode will be carried as a
parallel RTP session is a laudable but forward-looking conjecture.
IMHO reliable use of this payload format in the near future
will depend on carriage of vertical interval timecode (VITC)
in an unused scan line which is part of the vertical blanking
interval. (This is normal video operational practice, as
mentioned in RFC 2431.)

At the top of page 2 of the draft, we have "...the contents of
horizontal and vertical blanking SHOULD NOT be transported." This
does not bother me as SHOULD NOT language does not prevent me from
carrying VITC when I need it, until the multisession RTP sync
issues get ironed out. On the other hand the MUST language of
page 3 is burdensome.

---
<nitpick>
A minor MUST/SHOULD nit at sec. 4.3 regarding filling to a pgroup
boundary: since a receiver MUST ignore the fill data, it seems
unneccessary that the sender MUST fill with zeroes, specifically.
The sender MUST fill, and SHOULD fill with zeroes.
</nitpick>

---

The MIME and SDP parameters do not include the video frame rate.
There are many plausible scenarios in which it is important
to distinguish alternative frame rates. For example, choosing
the right VOD stream during RTSP setup would need something like
this.

---

Cheers,
  Chuck Harrison
  Far Field Associates, LLC
_______________________________________________
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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		: The MIDI Wire Protocol Packetization (MWPP)
	Author(s)	: J. Lazzaro, J. Wawrzynek
	Filename	: draft-ietf-avt-mwpp-midi-rtp-06.txt
	Pages		: 100
	Date		: 2003-3-5
	
The MIDI Wire Protocol Packetization (MWPP) is a general-purpose
RTP packetization for the MIDI command language. MWPP is suitable
for use in both interactive applications (such as the remote
operation of musical instruments) and content-delivery applications
(such as MIDI file streaming). MWPP is suitable for use over
unicast and multicast UDP, and defines tools that support the
graceful recovery from packet loss. MWPP may also be used over
reliable transport such as TCP. The SDP parameters defined for MWPP
support the customization of stream behavior (including the MIDI
rendering method) during session setup. MWPP is compatible with the
MPEG-4 generic RTP payload format, to support the MPEG 4 Audio
object types for General MIDI, DLS2, and Structured Audio.

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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 JVT Video
	Author(s)	: S. Wenger et al.
	Filename	: draft-ietf-avt-rtp-h264-01.txt
	Pages		: 27
	Date		: 2003-3-5
	
This memo describes an RTP Payload format for the ITU-T 
Recommendation H.264 video codec.  The most up-to-date draft of the 
video codec was specified in February 2003, is due for final 
approval at the committee level late March 2003, and is available 
for public review [1].  This codec was designed as a joint project 
of the Video Coding Experts Group (VCEG) of ITU-T and the Moving 
Picture Experts Group (MPEG) of ISO/IEC.  ISO/IEC International 
Standard 14496-10 will be technically identical to ITU-T 
Recommendation H.264.

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



From mailnull@www1.ietf.org  Thu Mar  6 08:01:06 2003
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Date: Thu, 06 Mar 2003 04:46:05 -0800
To: "Adam Li" <adamli@icsl.ucla.edu>
From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [AVT] File format for EVRC/SMV vocoder
Cc: "Ietf-Avt" <avt@ietf.org>, "3GPP2 TSG-C" <tsgc@3gpp2.org>
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Adam
   I have a question:  Why take a file format specification through the IETF?

Best regards, Mark
At 12:13 PM 3/5/2003 -0800, Adam Li wrote:
>Hi folks,
>
>One issue just recently been raised about the file format for EVRC and
>SMV data, which is specified the MIME registration for them. We very
>much would like to hear your ideas and comments on this issue.
>
>As you may have known, we are working the IETF specification on the RTP
>Payload Format for EVRC/SMV. It specifies how the EVRC/SMV payload is
>carried in RTP sessions, and it also registered the MIME format for the
>EVRC and SMV vocoders. The draft (draft-ietf-avt-evrc-smv) is currently
>at the final editing stage at the IETF FRC Editor's queue.
>
>There has been a file format defined in the draft-ietf-avt-evrc-smv at
>the MIME registration for EVRC/SMV. The file format specified there
>hasn't been changed much since the first version of the draft in 2000.
>
>There is a new Internet Draft, draft-garudadri-qcp-00.txt, specifies a
>different way of storage using the QCP file format, which has been used
>for QCEPL-13K and EVRC by some applications.
>
>Currently, it has been proposed to remove the file format specification
>in the draft-ietf-avt-evrc-smv, and to expand the draft-garudadri-qcp to
>be the only formal specification for the file format for EVRC/SMV. On
>the other hand, there are also suggestions to keep the file format at it
>is specified in the draft-ietf-avt-evrc-smv.
>
>We would like to hear your suggestions, and hear from you how you think
>would be the best way to proceed. Your comments are very much
>appreciated.
>
>Adam
>
>
>_______________________________________________
>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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From: Kreuter Ruediger <ruediger.kreuter@siemens.com>
To: "'Colin Perkins'" <csp@isi.edu>, avt@ietf.org
Cc: casner@acm.org
Subject: AW: [AVT] Agenda for San Francisco meeting - Clearmode
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> -----Ursprüngliche Nachricht-----
> Von: Colin Perkins [mailto:csp@isi.edu]
> Gesendet: Mittwoch, 5. März 2003 01:59
> An: avt@ietf.org
> Cc: casner@acm.org
> Betreff: [AVT] Agenda for San Francisco meeting
> 
> 
> Folks,
> 
> Enclosed is the agenda for the AVT meeting in San Francisco. 
> Please send
> any comments or corrections to the chairs. 
> 
> Those who have agenda time scheduled are requested to prepare 
> a summary of
> the issues they wish to discuss, and to send this to the 
> mailing list ahead
> of the meeting. This gives those who won't be attending the 
> opportunity to
> post their comments and concerns, and potentially even reach 
> some consensus,
> before the meeting time is consumed. It also gives other 
> participants the
> opportunity to consider the issue in more depth. Summary 
> messages SHOULD be
> sent to the mailing list by Monday 10th March.
> 
...
 	
> 
> Thursday, 20th March at 09:00-11:30
> ===================================
> 
> RTP Payload Format for a 64 kbit/s transparent call	
> (Kreuter)          15
> 	draft-kreuter-avt-rtp-clearmode-01.txt
> 	See also the TDM emulation work in the PWE3 working group
> 

Last friday I've submitted draft-kreuter-avt-rtp-clearmode-02.txt
to internet-drafts@ietf.org. This is still in the queue ...
Basically I removed Nx64 kbit/s from my draft and fixed some typos.

After discussion with A. Vainshtein, one of the author's of
draft-vainshtein-cesopsn-05.txt (PWE3), Tom Johnson, who is 
working on a similar topic, and Colin Perkins we agreed that 
it makes sense to split the Clearmode handling in two drafts, 
where my draft focuses on 1x64 kbit/s and an according MIME 
type registration, and the PWE3 draft focuses on Nx64 kbit,
T1/E1 and up for CES services.

So according draft-kreuter-avt-rtp-clearmode-xx.txt, IMHO 
the only open item now is the content type. In discussion 
is "audio" or "application".

My intention for the IETF presentation is to get an agreement
on that topic, so that this draft can be advanced for IESG
approval.

Regards, Ruediger



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



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To: Kreuter Ruediger <ruediger.kreuter@siemens.com>
CC: "'avt@ietf.org'" <avt@ietf.org>
Subject: Re: [AVT] updated "Clearmode" (64kbit/s unrestricted) Internet-draft
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Hi,

I have some comments on this draft.

1. Chapter 2: First sentence seems strange it looks like it lacks 
context. "The profile .." refers to something which I don't know what it 
is.

2. Maybe a little more text on how data is put in the payload format. 
Especially around the number of samples you are allowed to put into each 
packet.

3. End of chapter 2: "The timestamp SHALL be set according [5]"
Shouldn't this be "The timestamp SHALL be set according to [5]."

4. Chapter 3: Optional parameters. I think "maxptime" is a valid one to 
use. The existing RFC with this parameter is RFC 3267 but the updated 
SDP draft also contains a definition of it.

5. Chapter 3 is missing rules on how you apply the mime parameters to 
SDP. Please see draft-ietf-avt-rtp-mime-06.txt for such a chapter.

6. Reference list in chapter 5. Shouldn't there be nice [] around the 
numbers?

7. Time to read the ID nits. You are lacking a couple of necessary 
sections, like Copyright statement and IPR statement. See 
http://www.ietf.org/ID-nits.html

8. When you don't transport ISDN 7 kHz voice in this and actually carry 
some other type of data. Isn't of interest to mention RTP relevant 
mechanisms to protect the stream against packet loss. There should be 
significant considerations of this problem.

9. Please look at the formatting also. There are certain parts that 
would look nicer if they are done as bullet list instead of one sentence 
paragraphs.

Best Regards

Magnus

Kreuter Ruediger wrote:

>FYI, 
>I have submitted an update of the CLEARMODE Internet draft. You find it at
>http://www.ietf.org/internet-drafts/draft-kreuter-avt-rtp-clearmode-01.txt
>
>The major update, due to comments according the support of Nx64kbit/s
>and similar standardization activities in ITU's SG 11, is the introduction of
>Nx64kbit/s support. Two remarks to this issue:
>1) 1x64kbit/s is used in current VoIP products and supported by most DSP 
>    Providers. Therefore I would like to keep Nx64kbit/s with N>1 as an option.
>2) We currently do not have a VoIP product supporting Nx64kbit/s, therefore 
>    I introduced this merely as a discussion input.
>    So if anybody has a working Nx64kbit/s solution I would like to get some 
>    comments on the packatization method in that case. If nobody has a
>    solution I will delete this feature from the next version of this draft.
>
>Besides, the first draft initiated a controverse discussion, if this should be 
>"audio/CLEARMODE" or e.g. "application/CLEARMODE".
>IMHO none of the existing IANA content types really fits for CLEARMODE. 
>Since 1x64kbit/s is also used for 7kHz voice (despite G.722) I kept "audio"
>in this draft. 
>
>Regards
>   Ruediger
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt
>
>  
>

-- 

Magnus Westerlund 

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



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



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To: "'avt@ietf.org'" <avt@ietf.org>
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Date: Thu, 6 Mar 2003 09:12:34 -0600 
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Subject: [AVT] Status of SRTP
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AVT group:

What's the status of SRTP?  It expired at the end of '02.  Are there any
plans for a new draft?

DWJ

---
David W. Jarrett        Senior Technical Staff
Broadband Solutions     Texas Instruments
djarrett@ti.com         (408) 468-0830


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<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Arial">AVT group:</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Arial">What's the status of =
SRTP?&nbsp; It expired at the end of '02.&nbsp; Are there any plans for =
a new draft?</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Arial">DWJ</FONT></P>

<P ALIGN=3DLEFT><A NAME=3D"_MailAutoSig"><FONT SIZE=3D2 FACE=3D"Courier =
New">---</FONT></A></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">David W. =
Jarrett&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Senior Technical =
Staff</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Broadband =
Solutions&nbsp;&nbsp;&nbsp;&nbsp; Texas Instruments</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier =
New">djarrett@ti.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
(408) 468-0830</FONT></P>

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_______________________________________________
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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 Extended Reports (RTP XR)
	Author(s)	: T. Friedman et. al.
	Filename	: draft-ietf-avt-rtcp-report-extns-03.txt
	Pages		: 41
	Date		: 2003-3-5
	
This document defines the extended report (XR) packet type for the
RTP control protocol (RTCP).  XR packets are composed of report
blocks, and seven block types are defined here.  The purpose of the
extended reporting format is to convey information that supplements
the six statistics that are contained in the report blocks used by
RTCP's sender report (SR) and receiver report (RR) packets.  Some
applications, such as multicast inference of network characteristics
(MINC) or voice over IP (VoIP) monitoring, require other and more
detailed statistics.  In addition to the block types defined here,
additional block types may be defined in the future by adhereing to
the framework that this document provides.

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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		: RTCP Extensions for Single-Source Multicast Sessions 
			  with Unicase Feedback
	Author(s)	: J. Chesterfield, E. Schooler, J. Ott 
	Filename	: draft-ietf-avt-rtcpssm-03.txt
	Pages		: 37
	Date		: 2003-3-5
	
This document specifies a modification to the Real-time Transport 
Control Protocol (RTCP) to use unicast feedback. The proposed 
extension is useful for single source multicast sessions such as 
Source Specific Multicast (SSM) communication where the traditional 
model of many-to-many group communication is either not possible or 
not preferred. In addition, it can be applied to any group that 
might benefit from a sender controlled summarised reporting 
mechanism.

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Subject: [AVT] RE: Status of SRTP
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Status of SRTPDavid,

a new version will be coming out shortly, reflecting some discussion with
the IESG.  Thanks for pointing out that the current draft actually expired
last December.  It's been in IESG review for a while, and it has taken some
time to settle on changes to the draft.  We unintentionally let the document
lapse.

David M.

  -----Original Message-----
  From: Jarrett, David [mailto:djarrett@ti.com]
  Sent: Thursday, March 06, 2003 7:13 AM
  To: 'avt@ietf.org'
  Cc: 'mcgrew@cisco.com'
  Subject: Status of SRTP


  AVT group:

  What's the status of SRTP?  It expired at the end of '02.  Are there any
plans for a new draft?

  DWJ

  ---

  David W. Jarrett        Senior Technical Staff

  Broadband Solutions     Texas Instruments

  djarrett@ti.com         (408) 468-0830



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Status of SRTP</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 5.50.4807.2300" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D958084315-06032003><FONT face=3DArial color=3D#0000ff =

size=3D2>David,</FONT></SPAN></DIV>
<DIV><SPAN class=3D958084315-06032003><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D958084315-06032003><FONT face=3DArial color=3D#0000ff =
size=3D2>a new=20
version will be coming out shortly, reflecting some discussion with the=20
IESG.&nbsp; </FONT></SPAN><SPAN class=3D958084315-06032003><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Thanks for pointing out that the current draft =
actually=20
expired last December.&nbsp; It's been in IESG review for a while, and =
it has=20
taken some time to settle on changes to the draft.&nbsp; We =
unintentionally let=20
the document lapse.</FONT></SPAN></DIV>
<DIV><SPAN class=3D958084315-06032003><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D958084315-06032003><FONT face=3DArial color=3D#0000ff =
size=3D2>David=20
M.</FONT></SPAN></DIV>
<DIV><SPAN class=3D958084315-06032003><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Jarrett, David=20
  [mailto:djarrett@ti.com]<BR><B>Sent:</B> Thursday, March 06, 2003 7:13 =

  AM<BR><B>To:</B> 'avt@ietf.org'<BR><B>Cc:</B>=20
  'mcgrew@cisco.com'<BR><B>Subject:</B> Status of =
SRTP<BR><BR></FONT></DIV>
  <P align=3Dleft><FONT face=3DArial size=3D2>AVT group:</FONT></P>
  <P align=3Dleft><FONT face=3DArial size=3D2>What's the status of =
SRTP?&nbsp; It=20
  expired at the end of '02.&nbsp; Are there any plans for a new=20
  draft?</FONT></P>
  <P align=3Dleft><FONT face=3DArial size=3D2>DWJ</FONT></P>
  <P align=3Dleft><A name=3D_MailAutoSig><FONT face=3D"Courier New"=20
  size=3D2>---</FONT></A></P>
  <P align=3Dleft><FONT face=3D"Courier New" size=3D2>David W.=20
  Jarrett&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Senior Technical=20
  Staff</FONT></P>
  <P align=3Dleft><FONT face=3D"Courier New" size=3D2>Broadband=20
  Solutions&nbsp;&nbsp;&nbsp;&nbsp; Texas Instruments</FONT></P>
  <P align=3Dleft><FONT face=3D"Courier New"=20
  =
size=3D2>djarrett@ti.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
(408)=20
  468-0830</FONT></P>
  <P align=3Dleft></P></BLOCKQUOTE></BODY></HTML>

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_______________________________________________
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Perhaps this post is not the most appropriate; but for some time I have been
considering informing the experts here about the work done by the ATSC to
address the problem of multicast when no return path is available. See 
http://www.atsc.org/standards/a_92.pdf


Art
::{)
Art Allison
Director Advanced Engineering
NAB
1771 N St NW
Washington DC 20036
202 429 5418


-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: Thursday, March 06, 2003 10:45 AM
Cc: avt@ietf.org
Subject: [AVT] I-D ACTION:draft-ietf-avt-rtcpssm-03.txt


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		: RTCP Extensions for Single-Source Multicast
Sessions 
			  with Unicase Feedback
	Author(s)	: J. Chesterfield, E. Schooler, J. Ott 
	Filename	: draft-ietf-avt-rtcpssm-03.txt
	Pages		: 37
	Date		: 2003-3-5
	
This document specifies a modification to the Real-time Transport 
Control Protocol (RTCP) to use unicast feedback. The proposed 
extension is useful for single source multicast sessions such as 
Source Specific Multicast (SSM) communication where the traditional 
model of many-to-many group communication is either not possible or 
not preferred. In addition, it can be applied to any group that 
might benefit from a sender controlled summarised reporting 
mechanism.

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

To remove yourself from the IETF Announcement list, send a message to 
ietf-announce-request with the word unsubscribe in the body of the message.

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"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-avt-rtcpssm-03.txt".

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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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Send a message to:
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Dear Art;

There is a working group for no-return path transport, UDLR

http://www.ietf.org/html.charters/udlr-charter.html

which includes multicast. How do these approaches compare ? How
different is the ATSC work from the DVB standards ? There was I think
a  ip-dvb BOF in Atlanta, but I do not know if this will become a
working group.

                                  Regards
                                  Marshall Eubanks



On Thursday, March 6, 2003, at 11:01  AM, Allison, Art wrote:

> Perhaps this post is not the most appropriate; but for some time I have 
> been
> considering informing the experts here about the work done by the ATSC 
> to
> address the problem of multicast when no return path is available. See
> http://www.atsc.org/standards/a_92.pdf
>
>
> Art
> ::{)
> Art Allison
> Director Advanced Engineering
> NAB
> 1771 N St NW
> Washington DC 20036
> 202 429 5418
>
>
> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Sent: Thursday, March 06, 2003 10:45 AM
> Cc: avt@ietf.org
> Subject: [AVT] I-D ACTION:draft-ietf-avt-rtcpssm-03.txt
>
>
> 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		: RTCP Extensions for Single-Source Multicast
> Sessions
> 			  with Unicase Feedback
> 	Author(s)	: J. Chesterfield, E. Schooler, J. Ott
> 	Filename	: draft-ietf-avt-rtcpssm-03.txt
> 	Pages		: 37
> 	Date		: 2003-3-5
> 	
> This document specifies a modification to the Real-time Transport
> Control Protocol (RTCP) to use unicast feedback. The proposed
> extension is useful for single source multicast sessions such as
> Source Specific Multicast (SSM) communication where the traditional
> model of many-to-many group communication is either not possible or
> not preferred. In addition, it can be applied to any group that
> might benefit from a sender controlled summarised reporting
> mechanism.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avt-rtcpssm-03.txt
>
> To remove yourself from the IETF Announcement list, send a message to
> ietf-announce-request with the word unsubscribe in the body of the 
> message.
>
> Internet-Drafts are also available by anonymous FTP. Login with the 
> username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
> 	"get draft-ietf-avt-rtcpssm-03.txt".
>
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
>
> Internet-Drafts can also be obtained by e-mail.
>
> Send a message to:
> 	mailserv@ietf.org.
> In the body type:
> 	"FILE /internet-drafts/draft-ietf-avt-rtcpssm-03.txt".
> 	
> NOTE:	The mail server at ietf.org can return the document in
> 	MIME-encoded form by using the "mpack" utility.  To use this
> 	feature, insert the command "ENCODING mime" before the "FILE"
> 	command.  To decode the response(s), you will need "munpack" or
> 	a MIME-compliant mail reader.  Different MIME-compliant mail readers
> 	exhibit different behavior, especially when dealing with
> 	"multipart" MIME messages (i.e. documents which have been split
> 	up into multiple messages), so check your local documentation on
> 	how to manipulate these messages.
> 		
> 		
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>

T.M. Eubanks
Multicast Technologies, Inc.
Phone : 703-293-9601       Fax     : 703-293-9609
e-mail : tme@multicasttech.com
http://www.multicasttech.com

  Our New Multicast Workshop :
  http://www.multicasttech.com/workshop
Test your network for multicast :
http://www.multicasttech.com/mt/

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



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Dear Art;

There is a working group for no-return path transport, UDLR

http://www.ietf.org/html.charters/udlr-charter.html

which includes multicast. How do these approaches compare ? How
different is the ATSC work from the DVB standards ? There was I think
a  ip-dvb BOF in Atlanta, but I do not know if this will become a
working group.

                                  Regards
                                  Marshall Eubanks



On Thursday, March 6, 2003, at 11:01  AM, Allison, Art wrote:

> Perhaps this post is not the most appropriate; but for some time I have 
> been
> considering informing the experts here about the work done by the ATSC 
> to
> address the problem of multicast when no return path is available. See
> http://www.atsc.org/standards/a_92.pdf
>
>
> Art
> ::{)
> Art Allison
> Director Advanced Engineering
> NAB
> 1771 N St NW
> Washington DC 20036
> 202 429 5418
>
>
> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Sent: Thursday, March 06, 2003 10:45 AM
> Cc: avt@ietf.org
> Subject: [AVT] I-D ACTION:draft-ietf-avt-rtcpssm-03.txt
>
>
> 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		: RTCP Extensions for Single-Source Multicast
> Sessions
> 			  with Unicase Feedback
> 	Author(s)	: J. Chesterfield, E. Schooler, J. Ott
> 	Filename	: draft-ietf-avt-rtcpssm-03.txt
> 	Pages		: 37
> 	Date		: 2003-3-5
> 	
> This document specifies a modification to the Real-time Transport
> Control Protocol (RTCP) to use unicast feedback. The proposed
> extension is useful for single source multicast sessions such as
> Source Specific Multicast (SSM) communication where the traditional
> model of many-to-many group communication is either not possible or
> not preferred. In addition, it can be applied to any group that
> might benefit from a sender controlled summarised reporting
> mechanism.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avt-rtcpssm-03.txt
>
> To remove yourself from the IETF Announcement list, send a message to
> ietf-announce-request with the word unsubscribe in the body of the 
> message.
>
> Internet-Drafts are also available by anonymous FTP. Login with the 
> username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
> 	"get draft-ietf-avt-rtcpssm-03.txt".
>
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
>
> Internet-Drafts can also be obtained by e-mail.
>
> Send a message to:
> 	mailserv@ietf.org.
> In the body type:
> 	"FILE /internet-drafts/draft-ietf-avt-rtcpssm-03.txt".
> 	
> NOTE:	The mail server at ietf.org can return the document in
> 	MIME-encoded form by using the "mpack" utility.  To use this
> 	feature, insert the command "ENCODING mime" before the "FILE"
> 	command.  To decode the response(s), you will need "munpack" or
> 	a MIME-compliant mail reader.  Different MIME-compliant mail readers
> 	exhibit different behavior, especially when dealing with
> 	"multipart" MIME messages (i.e. documents which have been split
> 	up into multiple messages), so check your local documentation on
> 	how to manipulate these messages.
> 		
> 		
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>

T.M. Eubanks
Multicast Technologies, Inc.
Phone : 703-293-9601       Fax     : 703-293-9609
e-mail : tme@multicasttech.com
http://www.multicasttech.com

  Our New Multicast Workshop :
  http://www.multicasttech.com/workshop
Test your network for multicast :
http://www.multicasttech.com/mt/

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



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David:

 

Thanks for the info!  I'll be looking forward to seeing the new draft.  Can
you comment on the sorts of changes under discussion?  

 

DWJ

 

-----Original Message-----
From: David A. Mcgrew [mailto:mcgrew@cisco.com] 
Sent: Thursday, March 06, 2003 10:47 AM
To: Jarrett, David
Cc: avt@ietf.org
Subject: RE: Status of SRTP

 

David,

 

a new version will be coming out shortly, reflecting some discussion with
the IESG.  Thanks for pointing out that the current draft actually expired
last December.  It's been in IESG review for a while, and it has taken some
time to settle on changes to the draft.  We unintentionally let the document
lapse.

 

David M.

 

-----Original Message-----
From: Jarrett, David [mailto:djarrett@ti.com]
Sent: Thursday, March 06, 2003 7:13 AM
To: 'avt@ietf.org'
Cc: 'mcgrew@cisco.com'
Subject: Status of SRTP

AVT group:

What's the status of SRTP?  It expired at the end of '02.  Are there any
plans for a new draft?

DWJ

---

David W. Jarrett        Senior Technical Staff

Broadband Solutions     Texas Instruments

djarrett@ti.com         (408) 468-0830


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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>David:</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Thanks for the info!&nbsp; I'll be
looking forward to seeing the new draft.&nbsp; Can you comment on the sorts of
changes under discussion?&nbsp; </span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>DWJ</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'>-----Original Message-----<br>
<b><span style='font-weight:bold'>From:</span></b> David A. Mcgrew
[mailto:mcgrew@cisco.com] <br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, March 06, 2003
10:47 AM<br>
<b><span style='font-weight:bold'>To:</span></b> Jarrett, David<br>
<b><span style='font-weight:bold'>Cc:</span></b> avt@ietf.org<br>
<b><span style='font-weight:bold'>Subject:</span></b> RE: Status of SRTP</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>&nbsp;</span></font></p>

<div>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=blue face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:blue'>David,</span></font></p>

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style='font-size:12.0pt'>&nbsp;</span></font></p>

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<div>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=blue face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:blue'>a new version will be
coming out shortly, reflecting some discussion with the IESG.&nbsp; Thanks for
pointing out that the current draft actually expired last December.&nbsp; It's
been in IESG review for a while, and it has taken some time to settle on
changes to the draft.&nbsp; We unintentionally let the document lapse.</span></font></p>

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<div>

<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>&nbsp;</span></font></p>

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<p class=MsoNormal style='margin-left:.5in'><font size=2 color=blue face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:blue'>David M.</span></font></p>

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<div>

<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>&nbsp;</span></font></p>

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<blockquote style='border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4.0pt;
margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'>

<p class=MsoNormal style='margin-right:0in;margin-bottom:12.0pt;margin-left:
.5in'><font size=2 face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>-----Original
Message-----<br>
<b><span style='font-weight:bold'>From:</span></b> Jarrett, David
[mailto:djarrett@ti.com]<br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, March 06, 2003
7:13 AM<br>
<b><span style='font-weight:bold'>To:</span></b> 'avt@ietf.org'<br>
<b><span style='font-weight:bold'>Cc:</span></b> 'mcgrew@cisco.com'<br>
<b><span style='font-weight:bold'>Subject:</span></b> Status of SRTP</span></font></p>

<p style='margin-left:.5in'><font size=2 face=Arial><span style='font-size:
10.0pt;font-family:Arial'>AVT group:</span></font></p>

<p style='margin-left:.5in'><font size=2 face=Arial><span style='font-size:
10.0pt;font-family:Arial'>What's the status of SRTP?&nbsp; It expired at the
end of '02.&nbsp; Are there any plans for a new draft?</span></font></p>

<p style='margin-left:.5in'><font size=2 face=Arial><span style='font-size:
10.0pt;font-family:Arial'>DWJ</span></font></p>

<p style='margin-left:.5in'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>---</span></font></p>

<p style='margin-left:.5in'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>David W.
Jarrett&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Senior Technical Staff</span></font></p>

<p style='margin-left:.5in'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>Broadband Solutions&nbsp;&nbsp;&nbsp;&nbsp;
Texas Instruments</span></font></p>

<p style='margin-left:.5in'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>djarrett@ti.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
(408) 468-0830</span></font></p>

</blockquote>

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



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Subject: [AVT] When will draft-ietf-avt-rfc2833bis be re-published
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Dear Mr. Shulzrinne and Mr. Petrack,

I understand that your submitted draft document
"draft-ietf-avt-rfc2833bis-02 " was discussed  at the IETF meeting in
Atlanta, resulting in some changes to be made. I am interested in finding
out when do you plan in resubmitting an ubdated version to be discussed in
the next IETF meeting. 

Regards,
Dominic Walker
Nortel Networks
Tel: +44 (0)1279 403379

 

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<P><FONT SIZE=3D2 FACE=3D"Arial">Dear Mr. Shulzrinne and Mr. =
Petrack,</FONT>
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<P><FONT SIZE=3D2 FACE=3D"Arial">I understand that your submitted draft =
document &quot;draft-ietf-avt-rfc2833bis-02 &quot; was =
discussed&nbsp;</FONT> <FONT SIZE=3D2 FACE=3D"Arial">at the IETF =
meeting in Atlanta, resulting in some changes to be made. I am =
interested in finding out when do you plan in resubmitting an ubdated =
version to be discussed in the next IETF meeting. </FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Regards,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Dominic Walker</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Nortel Networks</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Tel: +44 (0)1279 403379</FONT>
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From: "David A. Mcgrew" <mcgrew@cisco.com>
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Status of SRTPDavid,
  -----Original Message-----
  From: Jarrett, David [mailto:djarrett@ti.com]
  Sent: Thursday, March 06, 2003 9:10 AM
  To: 'David A. Mcgrew'
  Cc: avt@ietf.org
  Subject: RE: Status of SRTP


  David:



  Thanks for the info!  I'll be looking forward to seeing the new draft.
Can you comment on the sorts of changes under discussion?



  I'll send out a detailed list of changes to the AVT list shortly.  The
changes are mostly to strengthen the defaults and to make more clear the
circumstances under which null authentication is acceptable.



  DAM



  DWJ



  -----Original Message-----
  From: David A. Mcgrew [mailto:mcgrew@cisco.com]
  Sent: Thursday, March 06, 2003 10:47 AM
  To: Jarrett, David
  Cc: avt@ietf.org
  Subject: RE: Status of SRTP



  David,



  a new version will be coming out shortly, reflecting some discussion with
the IESG.  Thanks for pointing out that the current draft actually expired
last December.  It's been in IESG review for a while, and it has taken some
time to settle on changes to the draft.  We unintentionally let the document
lapse.



  David M.



    -----Original Message-----
    From: Jarrett, David [mailto:djarrett@ti.com]
    Sent: Thursday, March 06, 2003 7:13 AM
    To: 'avt@ietf.org'
    Cc: 'mcgrew@cisco.com'
    Subject: Status of SRTP

    AVT group:

    What's the status of SRTP?  It expired at the end of '02.  Are there any
plans for a new draft?

    DWJ

    ---

    David W. Jarrett        Senior Technical Staff

    Broadband Solutions     Texas Instruments

    djarrett@ti.com         (408) 468-0830


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<DIV><SPAN class=3D930524617-06032003><FONT face=3DArial color=3D#0000ff =

size=3D2>David,</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Jarrett, David=20
  [mailto:djarrett@ti.com]<BR><B>Sent:</B> Thursday, March 06, 2003 9:10 =

  AM<BR><B>To:</B> 'David A. Mcgrew'<BR><B>Cc:</B>=20
  avt@ietf.org<BR><B>Subject:</B> RE: Status of =
SRTP<BR><BR></FONT></DIV>
  <DIV class=3DSection1>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">David:</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Thanks for =
the=20
  info!&nbsp; I'll be looking forward to seeing the new draft.&nbsp; Can =
you=20
  comment on the sorts of changes under discussion?&nbsp;&nbsp;<SPAN=20
  class=3D930524617-06032003><FONT=20
  color=3D#0000ff>&nbsp;</FONT></SPAN></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><SPAN=20
  class=3D930524617-06032003></SPAN></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><SPAN=20
  class=3D930524617-06032003><FONT color=3D#0000ff>I'll send out a =
detailed list of=20
  changes to the AVT list&nbsp;shortly.&nbsp;&nbsp;The changes are =
mostly to=20
  strengthen the defaults and to make more clear the circumstances under =
which=20
  null authentication is acceptable.&nbsp; =
</FONT></SPAN></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><SPAN=20
  class=3D930524617-06032003></SPAN></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><SPAN=20
  class=3D930524617-06032003><FONT=20
  color=3D#0000ff>DAM</FONT></SPAN></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">DWJ</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DTahoma =
size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">-----Original=20
  Message-----<BR><B><SPAN style=3D"FONT-WEIGHT: bold">From:</SPAN></B> =
David A.=20
  Mcgrew [mailto:mcgrew@cisco.com] <BR><B><SPAN=20
  style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Thursday, March 06, 2003 =
10:47=20
  AM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> Jarrett,=20
  David<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Cc:</SPAN></B>=20
  avt@ietf.org<BR><B><SPAN style=3D"FONT-WEIGHT: =
bold">Subject:</SPAN></B> RE:=20
  Status of SRTP</SPAN></FONT></P>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3D"Times =
New Roman"=20
  size=3D3><SPAN style=3D"FONT-SIZE: 12pt"></SPAN></FONT>&nbsp;</P>
  <DIV>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial =
color=3Dblue=20
  size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">David,</SPAN></FONT></P></DIV>
  <DIV>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3D"Times =
New Roman"=20
  size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
  <DIV>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial =
color=3Dblue=20
  size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">a new=20
  version will be coming out shortly, reflecting some discussion with =
the=20
  IESG.&nbsp; Thanks for pointing out that the current draft actually =
expired=20
  last December.&nbsp; It's been in IESG review for a while, and it has =
taken=20
  some time to settle on changes to the draft.&nbsp; We unintentionally =
let the=20
  document lapse.</SPAN></FONT></P></DIV>
  <DIV>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3D"Times =
New Roman"=20
  size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
  <DIV>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial =
color=3Dblue=20
  size=3D2><SPAN style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">David=20
  M.</SPAN></FONT></P></DIV>
  <DIV>
  <P class=3DMsoNormal style=3D"MARGIN-LEFT: 0.5in"><FONT face=3D"Times =
New Roman"=20
  size=3D3><SPAN style=3D"FONT-SIZE: =
12pt"></SPAN></FONT>&nbsp;</P></DIV>
  <BLOCKQUOTE=20
  style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: =
medium none; PADDING-LEFT: 4pt; PADDING-BOTTOM: 0in; MARGIN: 5pt 0in 5pt =
3.75pt; BORDER-LEFT: blue 1.5pt solid; PADDING-TOP: 0in; BORDER-BOTTOM: =
medium none">
    <P class=3DMsoNormal=20
    style=3D"MARGIN-BOTTOM: 12pt; MARGIN-LEFT: 0.5in; MARGIN-RIGHT: =
0in"><FONT=20
    face=3DTahoma size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">-----Original=20
    Message-----<BR><B><SPAN style=3D"FONT-WEIGHT: =
bold">From:</SPAN></B> Jarrett,=20
    David [mailto:djarrett@ti.com]<BR><B><SPAN=20
    style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Thursday, March 06, =
2003 7:13=20
    AM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B>=20
    'avt@ietf.org'<BR><B><SPAN style=3D"FONT-WEIGHT: =
bold">Cc:</SPAN></B>=20
    'mcgrew@cisco.com'<BR><B><SPAN style=3D"FONT-WEIGHT: =
bold">Subject:</SPAN></B>=20
    Status of SRTP</SPAN></FONT></P>
    <P style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">AVT =
group:</SPAN></FONT></P>
    <P style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">What's the status of =
SRTP?&nbsp;=20
    It expired at the end of '02.&nbsp; Are there any plans for a new=20
    draft?</SPAN></FONT></P>
    <P style=3D"MARGIN-LEFT: 0.5in"><FONT face=3DArial size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">DWJ</SPAN></FONT></P>
    <P style=3D"MARGIN-LEFT: 0.5in"><FONT face=3D"Courier New" =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Courier =
New'">---</SPAN></FONT></P>
    <P style=3D"MARGIN-LEFT: 0.5in"><FONT face=3D"Courier New" =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Courier New'">David W.=20
    Jarrett&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Senior Technical=20
    Staff</SPAN></FONT></P>
    <P style=3D"MARGIN-LEFT: 0.5in"><FONT face=3D"Courier New" =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Courier New'">Broadband=20
    Solutions&nbsp;&nbsp;&nbsp;&nbsp; Texas =
Instruments</SPAN></FONT></P>
    <P style=3D"MARGIN-LEFT: 0.5in"><FONT face=3D"Courier New" =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Courier =
New'">djarrett@ti.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
    (408) =
468-0830</SPAN></FONT></P></BLOCKQUOTE></DIV></BLOCKQUOTE></BODY></HTML>

------=_NextPart_000_1935_01C2E3CA.070285E0--

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



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Subject: [AVT] Re: When will draft-ietf-avt-rfc2833bis be re-published
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Unfortunately, I didn't quite get around to issuing a revised version 
before the deadline. I hope to do this shortly after the IETF. There 
have been no major changes, just minor edits. I'm also still looking for 
people that can disentangle the in-band-signaling-tone mess (CAS/MF), as 
I'm pretty sure that the current text is neither correct nor complete, 
but I don't have the right resources to fix them.

Dominic Walker wrote:
> Dear Mr. Shulzrinne and Mr. Petrack,
> 
> I understand that your submitted draft document 
> "draft-ietf-avt-rfc2833bis-02 " was discussed  at the IETF meeting in 
> Atlanta, resulting in some changes to be made. I am interested in 
> finding out when do you plan in resubmitting an ubdated version to be 
> discussed in the next IETF meeting.
> 
> Regards,
> Dominic Walker
> Nortel Networks
> Tel: +44 (0)1279 403379
> 
>  
> 

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On Thu, 6 Mar 2003, Jarrett, David wrote:

> AVT group:
>
> What's the status of SRTP?  It expired at the end of '02.  Are there any
> plans for a new draft?

Documents under IESG review do not expire, even if the expiration date
on the title page passes.

                                                        -- Steve

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From: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>
To: Adam Li <adamli@icsl.ucla.edu>
Cc: Ietf-Avt <avt@ietf.org>
Subject: RE: [AVT] File format for EVRC/SMV vocoder
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Adam and others,

I thought the reason for defining also the file format in the IETF
was to make it in line with the transport format, and also because
you would get it almost "for free" when you define the transport
format. The proposed new approach seems to take away those motives,
and then I do not understand why we should do it in the IETF.

Further, I wonder what would be the rationales for using this new
QCP format instead of the file format defined in the EVRC/SMV draft?

If we are to standardize a file format in this body, I can not see
why we should not stick to the format that is already finalized and
has been taken through our standards process. 

/L-E


> -----Original Message-----
> From: Mark Baugher [mailto:mbaugher@cisco.com]
> 
> Adam
>    I have a question:  Why take a file format specification 
> through the IETF?
> 
> Best regards, Mark
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Hi.

I have a number of comments on this draft. Most are structure and 
clarification request comments.

1. The short copyright statement is missing from the first page. See 
ID-Nits http://www.ietf.org/ID-nits.html.

2. The two first pages length is not of the appropriate length.

3. The footer and header is very crowed as the running text uses the 
lines directly after and before. To make this more readable and separate 
page header and footer from text please leave on empty line between them 
and the running text on all pages.

4. I would recommend that you move the changes chapter to last in the 
draft and do not include it in the table of contents. That way you can 
include it with a note to the RFC editor to remove this section and keep 
it being used until the draft is approved.

5. All figures and table are missing number and all references are 
either xxxx or xxx for figures respectively tables.

6. Chapter 6.1: The extension bit in the RTP header, shouldn't that also 
be used according to RFC 1889 and profile definitions instead of only 
the profile?

7. Chapter 6.1: Payload Type. I think that one can use a better language 
in this explanation. I especially object to the use of the word 
expected. In accordance with today's understanding  the text could read: 
"The assignment of a payload type needs to be performed either through 
the profile used or in a dynamic way.".

8. Chapter 6.2 I think one can clarify the language in the first 
paragraph. I would also like to rename it simple payload as this format 
are after all only a RTP payload.
Old text:
    The RTP payload of a Simple Packet according to this specification
    consists of one NALU as depicted in Figure xxxx.  The type of the
    NALU MUST be specified in [1], i.e., the NALU MUST NOT be an
    aggregation packet or a fragmentation unit.  A NAL unit stream
    composed by decapsulating Simple Packets in RTP sequence number
    order MUST conform to the NAL unit decoding order.

New text proposal:
The simple payload defined here MUST contain one and only one NALU of 
the types defined in [1]. This means that neither an aggregation payload 
or a fragmentation payload can be used. A NAL unit stream composed by 
decapsulating Simple Packets in RTP sequence number order MUST conform 
to the NAL unit decoding order. The simple payload is shown in figure xxxx.

9. Figure in 6.2: For clarification would it not be better to to show a 
RTP header before it:

   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
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |                          RTP Header                           |
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |                                                               |
  |                       Single NALU                             |
  |                                                               |
  |                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |                               :...OPTIONAL RTP padding        |
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    
    Figure xxxx. RTP payload format for Simple Packet.
 

10. Chapter 6.3  I have some concerns about the figure structure in this 
and the following sub chapters. Combined with the text the complete 
picture is not clear. Either you rename the text that says NALU payload 
to something clearer or you change the headings of the following sub 
chapters to include NALU payload instead of packet in their titles.

11. Chapter 6.3 The use of the F bit for aggregated units. Is it really 
wise to set this bit as bad as soon as any of the NALUs part of the 
aggregate is bad. This is useless information because you still needs to 
go into the packet an look at each NALU to determine which is actually 
useful. A cheap implementation that just throws away NALUs which is 
damaged can't use this flag anyway. It could be useful it the flag is 
instead defined as: "The F bit MUST only be set if all NALUs part of the 
aggregate has their F bit set". That way the cheap implementation can 
throw the whole packet as soon as it looks at the first bit.

12. Chapter 6.3.1: The DON text paragraph. First it is a very heavy 
section of text please look if you could break it into more paragraphs. 
A empty line before the numbered list would also be good.  Secondly 
there is a strange thing in the text that determine if D1 is before D2 
or not. " or if D1 > D2 and D1 . D2 >= 32768, then NAL unit ..." the "D1 
. D2" should probably be "D1 - D2" or?

13. To clarify how a complete STAP RTP payload looks a example of this 
should be provided.

   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
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |                          RTP Header                           |
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |STAP NAL HDR   | DON                           | NALU 1 Size   |
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  | NALU 1 Size   | NALU 1 HDR    | NALU DATA                     |
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               +
  :                                                               :
  +               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |               | NALU 2 SIZE                   | NALU 2 HDR    |
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |                       NALU Data                               |
  |                                                               |
  |                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |                               :...OPTIONAL RTP padding        |
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


This would be a nice figure in the end of  chapter 6.3.1 with a simple 
text declaring it to be STAP RTP packet with two NALU.

14. Chapter 6.3.2: Second paragraph. There is missing text about the use 
of modular arithmetic. The text says:
"DON of the following NALU is equal to DONB + DOND and MUST NOT be 
larger than 65535." Shouldn't this text say that the DON of the NALU is 
(DONB+DOND) mod 65535.

15. Chapter 6.3.2: Second paragraph at the end. I think the text 
declaring how the timestamp shall be calculated is written to 
informative. The text should be more normative and clearly state how it 
shall be calculated.

16. At the end of the chapter 6.3.2 I think a complete example should be 
shown containing all fields from RTP header to the end. You can never 
have to many examples as long as they are correct. 

17. Chapter 6.4: I think this chapter is missing some text on the 
considerations to make in regards to using FU payloads. What is the 
advantage over IP fragmentation? What is the drawbacks? I am also 
missing the packetization rules for this format. Even if I later found 
them in the next chapter. I think that the rules directly applicable to 
FU should be part of this chapter. Otherwise it is hard to understand 
that this works. The rules I am talking about is: "Fragments of the same 
NALU MUST be sent in consecutive order with no other RTP packets  sent 
between the first and last fragment" and "Fragments SHALL be reassembled 
in RTP sequence number order."

18. Chapter 7.1: Is the rules for how simple packets shall relate to 
aggregated packets necessary? Isn't this already covered by bullet 
number 2?

19. Chapter 71. The bullet that start with "An Aggregation Packet MUST 
succeed a Fragmentation Unit in transmission order if the NAL units in 
the Simple Packet precede the NAL unit conveyed in the Fragmentation 
Unit in NAL unit  decoding order. " Halfway through this sentence there 
is a simple packet, shouldn't that be fragment packet.

20. Chapter 8. These depacketization rules seems to not to take possible 
packet losses into account. For example:
"If the receiver buffer contains at least N VCL NAL units, NAL units are 
removed from the receiver buffer and passed to the decoder in the order 
specified below until the buffer contains N-1 VCL NAL units." The quoted 
sentence clearly does not take packet loss resulting in lost NAL units. 

21. Chapter 9.1."Profile-interoperability". These parameters are they 
often set in either direction. If that is the case I would see that 
declaring a default value and then spilt out each capability into its 
own parameter. Also the name is a bit long.

22. Chapter 9.1: "num-reorder-vcl-NAL-units". First why not rename it to 
"interleaving-depth" because this is what this parameter is about. 
Secondly the language about the default is strange. The sentence "If the 
parameter is not present, num-reorder-VCL-NAL-units equal to 0 MUST be 
implied." it seems very unsure even though a MUST is used. I would 
formulate it more like "If the parameter is not present the the value of 
num-reorder-vcl-NAL-units MUST be set to 0".

23. Chapter 9.1: "aggregation-mode" I would rename this parameter to 
what it really represents namely the use of unrestricted mode. 
Aggregation is still possible in restricted mode and therefore the 
present parameter name is not good. My proposal is "unrestricted". If 
value is 1 the unrestricted mode MAY be used. If not present the value 
MUST be set to 0. A value of 0 MUST only use restricted mode functionality.

24. Chapter 9.1 Required parameter. The RTP timestamp rate parameter is 
missing. I think it is a required parameter unless it is possible to 
hard code it to a single value.

25. Chapter 9.2: The second bullet about encoding name. It is missing 
text that the RTP timestamp rate shall also be part of the "a=rtpmap" line

26. Chapter 10: One standard thing that usually is part of the security 
consideration is if the codec shows any non-linear behavior. If one 
knows that switching a single bit in a NALU will result in that the 
processing requirements becomes the double or more this should be 
expressed.

27. Chapter 14. The correct IETF IPR text is not present. See ID nits: 
 From RFC 2026 sec 10.4 (A) and 10.4(B) and possibly 10.4(D)

Best Regards

Magnus Westerlund 

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



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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		: An RTP Payload Format for Erasure-Resilient 
			  Transmission of Progressive Multimedia Streams
	Author(s)	: G. Liebl et. al.
	Filename	: draft-ietf-avt-uxp-05.txt
	Pages		: 25
	Date		: 2003-3-6
	
This document specifies an efficient way to ensure erasure-
resilient transmission of progressively encoded multimedia
sources via RTP using Reed-Solomon (RS) codes together with
interleaving. The level of erasure protection can be explicitly
adapted to the importance of the respective parts in the source
stream, thus allowing a graceful degradation of application
quality with increasing packet loss rate on the network. Hence,
this type of unequal erasure protection (UXP) schemes is intended
to cope with the rapidly varying channel conditions on wireless
access links to the Internet backbone. Nevertheless, backward
compatibility to currently standardized non-progressive
multimedia codecs is ensured, since equal erasure protection
(EXP) represents a subset of generic UXP. By applying
interleaving and RS codes a  payload format is defined, which can
be easily integrated into the existing framework for RTP.

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A couple of minor updates, mostly fixing draft version numbers. Once again,
please send any comments and corrections to the chairs.

Colin



-------------------------------------------------------------------------------
Audio/Video Transport Working Group

CHAIRS: Steve Casner <casner@acm.org>
        Colin Perkins <csp@csperkins.org>


Wednesday, 19th March at 13:00-15:00
====================================

Introduction and status update				(Casner/Perkins)   15
	draft-ietf-avt-rtp-new-12.txt
	draft-ietf-avt-profile-new-13.txt
	draft-ietf-avt-uxp-05.txt
	draft-ietf-avt-ulp-07.txt          (Under revision)
	draft-ietf-avt-tcrtp-07.txt
	draft-ietf-avt-rtp-retransmission-06.txt
	draft-ietf-avt-rtcp-feedback-05.txt
	draft-ietf-avt-rtcp-feedback-sim-01.txt
	draft-ietf-avt-crtp-enhance-07.txt (IETF last call)
	draft-ietf-avt-rtcp-bw-05.txt      (RFC Editor)
	draft-ietf-avt-rtp-mime-06.txt     (RFC Editor)
	draft-ietf-avt-dsr-05.txt          (RFC Editor)
	draft-ietf-avt-evrc-smv-03.txt     (RFC Editor)
		See also draft-garudadri-qcp-00.txt
	RFC 3497

The Secure Real-time Transport Protocol                                    15
	draft-ietf-avt-srtp-05.txt

Framing RTP over Connection-Oriented Transport 		(Lazzaro)          15
	draft-lazzaro-avt-rtp-framing-contrans-00.txt

RTP Extended Reports (RTP XR) 				(Friedman/Clark)   15
	draft-ietf-avt-rtcp-report-extns-03.txt

RTCP Extensions for SSM with unicast feedback		(Chesterfield)     15
	draft-ietf-avt-rtcpssm-03.txt

RTP Payload Format for JVT Video 			(Wenger)	   15
 	draft-ietf-avt-rtp-h264-01.txt

RTP Payload Format for uncompressed video 		(Gharai)	   15
	draft-ietf-avt-uncomp-video-02.txt
	

Thursday, 20th March at 09:00-11:30
===================================

RTP Payload Format for a 64 kbit/s transparent call	(Kreuter)          15
	draft-kreuter-avt-rtp-clearmode-02.txt
	See also the TDM emulation work in the PWE3 working group

The MIDI Wire Protocol Packetization (MWPP) 		(Lazzaro)          15
	draft-ietf-avt-mwpp-midi-rtp-06.txt
	draft-lazzaro-avt-mwpp-coding-guidelines-02.txt

RTP Payload Format for ATRAC-X				(Romaine)	   15
	draft-hatanaka-avt-rtp-atracx-01.txt

RTP Payload Format for iLBC Speech 			(Duric)            15
	draft-ietf-avt-rtp-ilbc-01.txt
	draft-ietf-avt-ilbc-codec-01.txt

RTP Payload Format for the Speex Codec 			(Herlein)          15
	draft-herlein-speex-rtp-profile-00.txt

RTP Payload Format for RGL Codec			(Ramalho)          15
	draft-ramalho-rgl-desc-01.txt 
	draft-ramalho-rgl-rtpformat-01.txt

RTP Payload Format for ETSI ES 202 050 DSR		(Xie)	           15
	draft-xie-avt-dsr-es202050-00.txt

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Subject: Re: [AVT] Question on RTP sequence number in RFC2833 and bis-02
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Hi, I am still hoping that I can get some clarification from the WG on 
this question.  Apologies if this was discussed before, but I could not 
find anything by searching on the archive.

Thanks
Eric

Eric Cheung wrote:
> Hi Henning, Scott
> 
> Want to clarify a fine point.
> 
> In RFC2833:
> 
>    If there has been no new event
>    in the last interval, the event SHOULD be retransmitted three times
>    or until the next event is recognized.
> 
> Q1: For a device conforming to RFC2833, should the RTP sequence number 
> in the 3 packets be incremented, or should they be the same?
> 
> 
> In RFC2833bis-02, this seems to be clarified:
> 
>    If an event has ended and
>    there has been no new event in the last interval, the event SHOULD be
>    retransmitted three times at the interval used by the source for
>    updates.
> 
>    In all cases, the RTP sequence number MUST be incremented by one in
>    each RTP packet.
> 
> Q2: Can you confirm that if a device conforms to bis-02, the sequence 
> number in the 3 packets will be monotonically increasing?
> 
> Q3: In writing this, I realize that I am not sure whether there should 
> be 4 packets (1 original transmission + 3 retransmission) or 3 packets. 
>  Can you clarify this?
> 
> 
> Sorry for the nit-picking, but when testing it seems many devices do 
> many different things when it comes to 2833.
> 
> Thanks
> Eric


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Date: Fri, 07 Mar 2003 12:03:02 -0500
From: Colin Perkins <csp@isi.edu>
Subject: [AVT] WG last call: RTP Profile for RTCP-based Feedback (RTP/AVPF)
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This is to announce a working group last call on the Extended RTP Profile
for RTCP-based Feedback (RTP/AVPF) (draft-ietf-avt-rtcp-feedback-05.txt),
and simulation results (draft-burmeister-avt-rtcp-feedback-sim-01.txt).

Please send any final comments on this work to the <avt@ietf.org> mailing
list by 19th March 2003. If no substantive technical comments are received
by this time, we propose to submit the profile to the IESG for publication
as a Proposed Standard RFC, and the simulation results for publication as
an Informational RFC.

Thanks,
Steve and Colin
_______________________________________________
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Date: Fri, 07 Mar 2003 11:12:00 -0600
From: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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> RTP Payload Format for ETSI ES 202 050 DSR              (Xie)              15
>         draft-xie-avt-dsr-es202050-00.txt
> 

Hi, all,

This draft is a follow-up to draft-ietf-avt-dsr-05.txt (which has passed
IESG and currently in the RFC Editor's queue for an RFC #). The
difference is that this new draft specifies a 2nd front-end encoding
scheme published by ETSI in its ES202050 (the previous DSR draft defines
RTP payload format for ES201108).

I'd appreciate your comments on this new draft.

-Qiaobing
_______________________________________________
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From: Chuck Harrison <cfharr@erols.com>
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Philippe, all,

philippe.gentric@philips.com wrote:
> 
> Hi all,
> 
> I concur with chuck, this is looking very good indeed.
> 
[...]
> 
> Unfortunately the tables 1 thru 4 do not explicitely mention any RGB or RGBA format(!)
> 
> I think it would be useful to add them, here are the tables
> (not sure the notation R:G:B is appropriate, anyway  "RGB" is
> what is declared in the mime type ...) [...]

I had assumed that the conjunction of color-mode = RGB and 
sub-sampling = 4:4:4 would describe this. Thus the tables 1 thru
4 would not require additional entries distinguishing one color-
mode from another.

As you say, the tables could use a legend/explanation. I think
this could be clarified there.

While we're on color specs, how about this, to encourage *complete*
color specifications:

----
6.2.  Parameter Registration

New values of the "colorimetry" parameter MAY be registered with the
IANA provided they reference an RFC or other permanent and readily
available specification (the Specification Required policy of RFC 2434
[2434]). <New text:> New registrations SHOULD define all parameters
of a color image encoding as described in [22028].
----

Note: ISO 22028-1 is still in ballot with TC42 for another few days
but I am hopeful that it will be adopted by the time this internet-
draft reaches IESG.

If anyone is interested in the contents of 22028-1 please contact
me off list.

Cheers,
  Chuck Harrison
  Far Field Associates, LLC
_______________________________________________
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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		: Internet Low Bit Rate Codec
	Author(s)	: S. Andersen et al.
	Filename	: draft-ietf-avt-ilbc-codec-01.txt
	Pages		: 175
	Date		: 2003-3-7
	
This document specifies a speech codec suitable for robust voice 
communication over IP. The codec is developed by Global IP Sound 
(GIPS). It is designed for narrow band speech and results in a payload 
bit rate of 13.33 kbit/s for 30 ms frames and 15.20 kbit/s for 20 ms 
frames. The codec enables graceful speech quality degradation in the 
case of lost frames, which occurs in connection with lost or delayed IP 
packets.

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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 iLBC Speech
	Author(s)	: A. Duric, S. Andersen
	Filename	: draft-ietf-avt-rtp-ilbc-01.txt
	Pages		: 10
	Date		: 2003-3-7
	
This document describes the RTP payload format for the internet Low 
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(GIPS). Also, within the document there are included necessary 
details for the use of iLBC with MIME and SDP.

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

	Title		: RTP Profile for Audio and Video Conferences with 
                          Minimal Control
	Author(s)	: H. Schulzrinne, S. Casner
	Filename	: draft-ietf-avt-profile-new-13.txt,.ps
	Pages		: 45
	Date		: 2003-3-7
	
This document describes a profile called 'RTP/AVP' for the use of the
real-time transport protocol (RTP), version 2, and the associated
control protocol, RTCP, within audio and video multiparticipant
conferences with minimal control.  It provides interpretations of
generic fields within the RTP specification suitable for audio and
video conferences.  In particular, this document defines a set of
default mappings from payload type numbers to encodings.
This document also describes how audio and video data may be carried
within RTP.  It defines a set of standard encodings and their names
when used within RTP.  The descriptions provide pointers to reference
implementations and the detailed standards.  This document is meant
as an aid for implementors of audio, video and other real-time
multimedia applications.
This memorandum obsoletes RFC 1890.  It is mostly backwards-
compatible except for functions removed because two interoperable
implementations were not found.  The additions to RFC 1890 codify
existing practice in the use of payload formats under this profile
and include new payload formats defined since RFC 1890 was published.

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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: A Transport Protocol for Real-Time Applications
	Author(s)	: H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson
	Filename	: draft-ietf-avt-rtp-new-12.txt,.ps
	Pages		: 107
	Date		: 2003-3-7
	
This memorandum describes RTP, the real-time transport protocol.  RTP
provides end-to-end network transport functions suitable for
applications transmitting real-time data, such as audio, video or
simulation data, over multicast or unicast network services.  RTP
does not address resource reservation and does not guarantee
quality-of-service for real-time services.  The data transport is
augmented by a control protocol (RTCP) to allow monitoring of the
data delivery in a manner scalable to large multicast networks, and
to provide minimal control and identification functionality.  RTP and
RTCP are designed to be independent of the underlying transport and
network layers.  The protocol supports the use of RTP-level
translators and mixers.
Most of the text in this memorandum is identical to RFC 1889 which it
obsoletes.  There are no changes in the packet formats on the wire,
only changes to the rules and algorithms governing how the protocol
is used.  The biggest change is an enhancement to the scalable timer
algorithm for calculating when to send RTCP packets in order to
minimize transmission in excess of the intended rate when many
participants join a session simultaneously.

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From mailnull@www1.ietf.org  Fri Mar  7 17:13:49 2003
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To: Colin Perkins <csp@csperkins.org>, avt@ietf.org
Subject: Re: [AVT] I-D ACTION:draft-ietf-avt-uncomp-video-02.txt 
In-Reply-To: Your message of "Wed, 05 Mar 2003 20:47:37 EST."
             <200303060147.h261lbdk005611@purple.nge.isi.edu> 
Date: Fri, 07 Mar 2003 17:10:52 -0500
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Hello Chuck, please see some additional comments in line:

thanx,
L.


 > Hi,
 > 
 > Thanks for the comments. A couple of replies inline.
 > 
 > --> Chuck Harrison writes:
 > >
 > >A few comments on this draft, which is looking very good so far.
 > >
 > >---
 > >In uncomp-video-02 we have:
 > >
 > >4.3.  Payload Data
 > >
 > >Depending on the video format, each RTP packet can include either a
 > >single complete scan line, a single fragment of a scan line, or one
 > >(or
 > >more) complete scan lines plus a fragment of a scan line. [...]
 > >
 > >
 > >I think there are additional logical possibilities allowed by
 > >the document as it stands: a packet may begin with the tail
 > >fragment of a line, then continue with 0 or more full lines, and
 > >possibly the head fragment of another line.
 > >
 > >If the intent is to exclude these possibilities, we need normative
 > >language to say so. If these operational modes are permitted, the
 > >quoted text above is misleading and should be amended.
 > 
 > Those modes are permitted, so I guess we need to clarify the text. Can you
 > suggest some wording?

 Yes - the draft does not mean to list all possible combinations. 
 Basically anything goes, as long the information in the payload 
 header (offset and scan lines#) can be used to place the data in its
 frame.  Perhaps the wording can be changed to:

 Depending on the video format, each RTP packet can include multiple
 scan lines and fragments of scan lines. 

 > 
 > >---
 > >On page 3 of the draft, we have
 > >
 > >[...] This payload format
 > >requires that applications using such synchronized ancillary data MUST
 > >deliver it in separate RTP sessions which operate concurrently with
 > >the
 > >video session.  The normal RTP mechanisms SHOULD be used to
 > >synchronize
 > >the media.
 > >
 > >I believe the MUST is inappropriate and SHOULD is preferable.
 > >
 > >The idea that frame-accurate timecode will be carried as a
 > >parallel RTP session is a laudable but forward-looking conjecture.
 > >IMHO reliable use of this payload format in the near future
 > >will depend on carriage of vertical interval timecode (VITC)
 > >in an unused scan line which is part of the vertical blanking
 > >interval. (This is normal video operational practice, as
 > >mentioned in RFC 2431.)
 > >
 > >At the top of page 2 of the draft, we have "...the contents of
 > >horizontal and vertical blanking SHOULD NOT be transported." This
 > >does not bother me as SHOULD NOT language does not prevent me from
 > >carrying VITC when I need it, until the multisession RTP sync
 > >issues get ironed out. On the other hand the MUST language of
 > >page 3 is burdensome.
 > 
 > I'm still not convinced that an additional timecode is necessary: it should
 > be derivable from the RTP timestamp and RTCP SR packets. Accordingly if you
 > wish to synchronize the video to another source, you should use the normal
 > RTCP method, rather than sending an additional timecode stream (indeed, if
 > you use RTSP, I _think_ there is already a mapping function between the RTP
 > timestamp and SMPTE timecode).

   I guess the big questions in this regard is can the timecode be 
   derived from  RTP timestamp and RTCP SR packets without any loss of
   information?  and how will this impact the interval for sending
   RTCP SR packets?

 > 
 > >---
 > ><nitpick>
 > >A minor MUST/SHOULD nit at sec. 4.3 regarding filling to a pgroup
 > >boundary: since a receiver MUST ignore the fill data, it seems
 > >unneccessary that the sender MUST fill with zeroes, specifically.
 > >The sender MUST fill, and SHOULD fill with zeroes.
 > ></nitpick>
 > 
 > Okay. Is there a reason why zero fill doesn't work?
 > 
 > >---
 > >
 > >The MIME and SDP parametersdo not include the video frame rate.
 > >There are many plausible scenarios in which it is important
 > >to distinguish alternative frame rates. For example, choosing
 > >the right VOD stream during RTSP setup would need something like
 > >this.
 > 
 > You can use the SDP "a=framerate:" attribute, defined in RFC 2327.
 > 
 > Cheers,
 > Colin

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Hello Philippe - thank you for your comments. Please see reponses inline.

Ladan

 > Hi all,
 > 
 > I concur with chuck, this is looking very good indeed.
 > 
 > Just a few remarks
 > 
 > *****************************
 > I am not sure I always understand the "pixels" and "#words" second and third column in the tables ?

   Pixels refer to the smallest number of pixels a related of subsamples 
   represent, so for example for 4:4:4 Cb-Y-Cr cover 1 pixel whereas for 
   4:2:2 Cb0-Y0-Cr0-Y1 covers 2 pixels.
    
   #words, on the other hand expresses the number of words used to
   represent those pixels: 
   so for 4:4:4 requires 3 words Cb-Y-Cr and
          4:2:2 needs    4 words Cb0-Y0-Cr0-Y1

 > For exemple for monochrome 10 bits pixels is "4" so that would hint it means "number of pixels per Pgroup"

   It seems to me now that "number of pixels per Pgroup" would be a more
   intuitive way to look at pgroup - however pgroup is actually
   the number of bytes need to represent the least number of
   pixels, while being both byte aligned and observing ALF. 

   thats why, for the 10bit and 12bit tables, the two last columns
   #samples and pgroup, dont agree, whereas they agree for the 8bit table.
  
 > But then the next line is "4" for 4:1:1 while there are 8 pixels per Pgroup?

   so what that means is that the smallest group of related samples for a
   4:1:1 subsampling is 4 pixels p0 p1 p4 p3  represented as 
   Cb0-Y0-Y1-Cr0-Y2-Y3, this is 6 samples, which with 10bit words, align
   after 12 samples, or 15 octets - which is representing 8 pixels.
  
 > Also the #words is 6x10, but these are actually "bits" rather than "words"?

    true - its 6 words of 10 bits each.
  
 > I think these tables would benefit a few sentences giving the exact definition of each column.

   I couldn't agree more. The tables have acquired a life of their own. I hope my
   explanations have helped to clarify somewhat.
 > 
 > *****************************
 > The 8 bit words tables probably contains a typo for the "octet alignement" of Interlaced 4:2:0 which should be "4
x8/8 = 4"
 > no?

   yes - thank you :)
 > 
 > *****************************
 > 
 > in section 4.3 it says
 > 
 > "For RGB format video, samples are packed in order Red-Green-Blue.  All
 > samples are the same bit size, which may be 8, 10, 12, or 16 bits.  If 8
 > bit samples are used, the pgroup is 3 octets. If 10 bit samples are
 > used, samples from adjacent pixels are packed with no padding, and the
 > pgroup is 15 octets (4 pixels).  Refer to Tables 1 thru 4.
 > 
 > For RGBA format video, samples are packed in order Red-Green-Blue-Alpha.
 > All samples are the same bit size, which may be 8, 10, 12, or 16 bits.
 > For pgroups refer to Tables 1 thru 4."
 > 
 > Unfortunately the tables 1 thru 4 do not explicitely mention any RGB or RGBA format(!)

   True. I guess I was assuming that it is obvious that RGB and RGBA 
   would act the same as 4:4:4 and 4:4:4:4 subsampling.

   I wonder if it would be best to expand the tables, with the rows you
   have supplied, or add a reference in the text to the effect that 
   RGB and RGBA  pgroups are  like 4:4:4 and 4:4:4:4 pgroups? 

 > 
 > I think it would be useful to add them, here are the tables
 > (not sure the notation R:G:B is appropriate, anyway  "RGB" is
 > what is declared in the mime type ...)
 > 
 > 
 > 8 bit words
 > Color                --------------------------------------
 > Subsampling Pixels       #words octet alignment samples  pgroup
 > octets
 > +-----------+------+---+ +------+---------------+---------------+
 > |   R:G:B   |  1   | P | | 3x8  |  3x8/8 = 3    |   3   |   3   |
 > +-----------+------+---+ +------+---------------+---------------+
 > |  R:G:B:A  |  1   | P | | 4x8  |  4x8/8 = 4    |   4   |   4   |
 > +-----------+------+---+ +------+---------------+---------------+
 > 
 > 
 > 10 bit words
 > Color                --------------------------------------
 > Subsampling Pixels       #words octet alignment samples  pgroup
 > octets
 > +-----------+------+---+ +------+---------------+---------------+
 > |   R:G:B   |  1   | P | | 3x10 |  4x30/8 = 15  |   12   | 15   |
 > +-----------+------+---+ +------+---------------+---------------+
 > |  R:G:B:A  |  1   | P | | 4x10 |    40/8 = 5   |    4   |  5   |
 > +-----------+------+---+ +------+---------------+---------------+
 > 
 > 12 bit words
 > Color                --------------------------------------
 > Subsampling Pixels       #words octet alignment samples  pgroup
 > octets
 > +-----------+------+---+ +------+---------------+---------------+
 > |   R:G:B   |  1   | P | | 6x12 |  2x36/8 = 9   |    6   |  9   |
 > +-----------+------+---+ +------+---------------+---------------+
 > |  R:G:B:A  |  1   | P | | 4x12 |    48/8 = 6   |    4   |  6   |
 > +-----------+------+---+ +------+---------------+---------------+
 > 
 > 16 bit words
 > Color                --------------------------------------
 > Subsampling Pixels       #words octet alignment samples  pgroup
 > octets
 > +-----------+------+---+ +------+---------------+-------+-------+
 > |   R:G:B   |  1   | P | | 3x16 | 3x16/8 = 6    |   3   |   6   |
 > +-----------+------+---+ +------+---------------+-------+-------+
 > |  R:G:B:A  |  1   | P | | 4x16 | 4x16/8 = 8    |   4   |   8   |
 > +-----------+------+---+ +------+---------------+-------+-------+
 > 
 > 
 > 
 > *****************************

 Yes. I shall add in values for 4:4:2:0.  Thank you.

 > 
 > Since we have 4:4:4:4, one thing that should be added is 4:4:2:0
 > which is 4:2:0 (the most common raw video format
 > since it is the raw format of most codecs) plus
 > a transparency plane with one transparency word
 > per pixel i.e. same amount of transparency as luminance
 > 
 > This can be *very* useful for studio applications and is also
 > directly compressable using the profiles and levels of
 > MPEG-4 video that support the "shape" tool.
 > 
 > It simply needs to add 4:4:2:0 in the valid value list (section 6.1)
 > and in section 3 the relevant tables would be:
 > 
 > 
 > 8 bit words
 > Color                --------------------------------------
 > Subsampling Pixels       #words octet alignment samples  pgroup
 > octets
 > +-----------+------+---+ +------+---------------+---------------+
 > |  4:4:2:0  |  4   | P | | 10x8 |  80/8 = 10    |   10  |   10  |
 > +-----------+------+---+ +------+---------------+---------------+
 > |  4:4:2:0  |  4   | I | | 8x8  |  64/8 = 8     |   8   |   8   |
 > +-----------+------+---+ +------+---------------+---------------+
 > 
 > 
 > 
 > 10 bit words
 > Color                --------------------------------------
 > Subsampling Pixels       #words octet alignment samples  pgroup
 > octets
 > +-----------+------+---+ +------+---------------+---------------+
 > |  4:4:2:0  |  4   | P | | 10x10| 2x100/8 = 25  |   20  |   25  |
 > +-----------+------+---+ +------+---------------+---------------+
 > |  4:4:2:0  |  4   | I | | 8x10 |   80/8 = 10   |   8   |   10  |
 > +-----------+------+---+ +------+---------------+---------------+
 > 
 > 
 > 12 bit words
 > Color                --------------------------------------
 > Subsampling Pixels       #words octet alignment samples  pgroup
 > octets
 > +-----------+------+---+ +------+---------------+---------------+
 > |  4:4:2:0  |  4   | P | | 10x12|   120/8 = 15  |   10  |   15  |
 > +-----------+------+---+ +------+---------------+---------------+
 > |  4:4:2:0  |  4   | I | | 8x12 |   96/8 = 12   |   8   |   12  |
 > +-----------+------+---+ +------+---------------+---------------+
 > 
 > 16 bit words
 > Color                --------------------------------------
 > Subsampling Pixels       #words octet alignment samples  pgroup
 > octets
 > +-----------+------+---+ +------+---------------+---------------+
 > |  4:4:2:0  |  4   | P | | 10x16|   160/8 = 20  |   10  |   20  |
 > +-----------+------+---+ +------+---------------+---------------+
 > |  4:4:2:0  |  4   | I | | 8x16 |  128/8 = 16   |   8   |   16  |
 > +-----------+------+---+ +------+---------------+---------------+
 > 
 > 
 > 
 > 
 > regards,
 > 
 > 
 > Philippe Gentric
 > Software Architect
 > Philips MP4Net
 > philippe.gentric@philips.com
 > http://www.platform4.philips.com
 > 
 > 
 > 
 > 
 > To:   avt@ietf.org                                                                  
 > Ladan Gharai <ladan@east.isi.edu>                                                  
 > cc:   (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                                    
 > Subject:    Re: [AVT] I-D ACTION:draft-ietf-avt-uncomp-video-02.txt                 
 > Chuck Harrison                                                                                                   
       
 > <cfharr@erols.com>                  Classification:                                                              
       
 > 
 > Sent by:                                                                                                         
       
 > avt-admin@ietf.org                                                                                               
       
 > 
 > 05/03/2003 19:19                                                                                                 
       
 > 
 > 
 > 
 > 
 > 
 > 
 > Hi, all,
 > 
 > A few comments on this draft, which is looking very good so far.
 > 
 > ---
 > In uncomp-video-02 we have:
 > 
 > 4.3.  Payload Data
 > 
 > Depending on the video format, each RTP packet can include either a
 > single complete scan line, a single fragment of a scan line, or one
 > (or
 > more) complete scan lines plus a fragment of a scan line. [...]
 > 
 > 
 > I think there are additional logical possibilities allowed by
 > the document as it stands: a packet may begin with the tail
 > fragment of a line, then continue with 0 or more full lines, and
 > possibly the head fragment of another line.
 > 
 > If the intent is to exclude these possibilities, we need normative
 > language to say so. If these operational modes are permitted, the
 > quoted text above is misleading and should be amended.
 > 
 > ---
 > On page 3 of the draft, we have
 > 
 > [...] This payload format
 > requires that applications using such synchronized ancillary data MUST
 > deliver it in separate RTP sessions which operate concurrently with
 > the
 > video session.  The normal RTP mechanisms SHOULD be used to
 > synchronize
 > the media.
 > 
 > I believe the MUST is inappropriate and SHOULD is preferable.
 > 
 > The idea that frame-accurate timecode will be carried as a
 > parallel RTP session is a laudable but forward-looking conjecture.
 > IMHO reliable use of this payload format in the near future
 > will depend on carriage of vertical interval timecode (VITC)
 > in an unused scan line which is part of the vertical blanking
 > interval. (This is normal video operational practice, as
 > mentioned in RFC 2431.)
 > 
 > At the top of page 2 of the draft, we have "...the contents of
 > horizontal and vertical blanking SHOULD NOT be transported." This
 > does not bother me as SHOULD NOT language does not prevent me from
 > carrying VITC when I need it, until the multisession RTP sync
 > issues get ironed out. On the other hand the MUST language of
 > page 3 is burdensome.
 > 
 > ---
 > <nitpick>
 > A minor MUST/SHOULD nit at sec. 4.3 regarding filling to a pgroup
 > boundary: since a receiver MUST ignore the fill data, it seems
 > unneccessary that the sender MUST fill with zeroes, specifically.
 > The sender MUST fill, and SHOULD fill with zeroes.
 > </nitpick>
 > 
 > ---
 > 
 > The MIME and SDP parameters do not include the video frame rate.
 > There are many plausible scenarios in which it is important
 > to distinguish alternative frame rates. For example, choosing
 > the right VOD stream during RTSP setup would need something like
 > this.
 > 
 > ---
 > 
 > Cheers,
 > Chuck Harrison
 > Far Field Associates, LLC
 > _______________________________________________
 > 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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Subject: Re: [AVT] I-D ACTION:draft-ietf-avt-uncomp-video-02.txt
Date: Fri, 07 Mar 2003 17:13:21 -0500
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Chuck:

I think what would be interesting and what would really
add value to the draft, is having a solution for carrying 
timecodes that is accepted and understood by both worlds.

If RTP and RTCP SR packtes are to be used, the draft should 
be explicit about it and discuss the conversion process. 
Likewise, if VITC is to transported on the video signals,
what happens in instances of discrepancy with the RTP timestamps?

However, as you wisely point out, a good start would be to
get in touch with the real users.

Ladan

 > Hi, Colin
 > 
 > Colin Perkins wrote:
 > > 
 > > 
 > > I'm still not convinced that an additional timecode is necessary:
 > > it should be derivable from the RTP timestamp and RTCP SR packets.
 > 
 > Please understand I am not saying that systems will not work
 > without VITC. In fact, even in an editorial environment where
 > VITC may be carrying user bits and there are jump cuts in code
 > I am sure that something could be lashed together with enough
 > c hackers on the job. I look forward to seeing these things
 > work the way they should, as totally flexible RTP-synchronized
 > metadata streams (are you monitoring the MXF work at
 > http://www.g-fors.com/ , http://www.ist-nuggets.tv/mxf/ ?)
 > 
 > In the mean time we need to connect with actual users, of which
 > I am not one. Let's see what people are or aren't doing with
 > RFC2431. Let's find a few people who are interested in applying
 > draft-avt-uncomp-video to a real-world application.
 > 
 > My intuition is that anyone who says "don't worry about VITC,
 > we can always simulate it" has never been seriously involved
 > in video production. However I do not speak authoritatively.
 > 
 > Meanwhile, just what is the major threat to humanity that
 > comes from allowing a few poor slobs to put VITC on our video
 > signals if it makes us feel warm and fuzzy? Is it the extra
 > 0.5% in bandwidth?
 > 
 > Cheers,
 > Chuck



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



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In-Reply-To: Your message of "Fri, 07 Mar 2003 12:18:09 PST."
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 > Philippe, all,
 > 
 > philippe.gentric@philips.com wrote:
 > > 
 > > Hi all,
 > > 
 > > I concur with chuck, this is looking very good indeed.
 > > 
 > [...]
 > > 
 > > Unfortunately the tables 1 thru 4 do not explicitely mention any RGB or RGBA 
format(!)
 > > 
 > > I think it would be useful to add them, here are the tables
 > > (not sure the notation R:G:B is appropriate, anyway  "RGB" is
 > > what is declared in the mime type ...) [...]
 > 
 > I had assumed that the conjunction of color-mode = RGB and 
 > sub-sampling = 4:4:4 would describe this. Thus the tables 1 thru
 > 4 would not require additional entries distinguishing one color-
 > mode from another.

   Well, the problem is that section 4.3 refers back to the tables,
   when discussing RGB and RGBA. A solution would be to eliminate
   the reference and instead add some text that clarifies how RGB and
   RGBA are packetized.
 
 > As you say, the tables could use a legend/explanation. I think
 > this could be clarified there.

   true.

 > 
 > While we're on color specs, how about this, to encourage *complete*
 > color specifications:
 > 
 > ----
 > 6.2.  Parameter Registration
 > 
 > New values of the "colorimetry" parameter MAY be registered with the
 > IANA provided they reference an RFC or other permanent and readily
 > available specification (the Specification Required policy of RFC 2434
 > [2434]). <New text:> New registrations SHOULD define all parameters
 > of a color image encoding as described in [22028].
 > ----
 > 
 > Note: ISO 22028-1 is still in ballot with TC42 for another few days
 > but I am hopeful that it will be adopted by the time this internet-
 > draft reaches IESG.
 > 
 > If anyone is interested in the contents of 22028-1 please contact
 > me off list.
 > 
 > Cheers,
 > Chuck Harrison
 > Far Field Associates, LLC

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



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Subject: [AVT] RFC 3497 on RTP Payload Format for Society of Motion Picture and Television Engineers (SMPTE) 292M Video
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3497

        Title:      RTP Payload Format for Society of Motion Picture
                    and Television Engineers (SMPTE) 292M Video
        Author(s):  L. Gharai, C. Perkins, G. Goncher, A. Mankin
        Status:     Standards Track
        Date:       March 2003
        Mailbox:    ladan@isi.edu, csp@isi.edu, mankin@psg.com,
                    Gary.Goncher@tek.com
        Pages:      12
        Characters: 26094
        Updates/Obsoletes/SeeAlso:  None

        I-D Tag:    draft-ietf-avt-smpte292-video-08.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3497.txt


This memo specifies an RTP payload format for encapsulating
uncompressed High Definition Television (HDTV) as defined by the
Society of Motion Picture and Television Engineers (SMPTE) standard,
SMPTE 292M.  SMPTE is the main standardizing body in the motion
imaging industry and the SMPTE 292M standard defines a bit-serial
digital interface for local area HDTV transport.

This document is a product of the Audio/Video Transport Working Group
of the IETF.

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
standardization state and status of this protocol.  Distribution
of this memo is unlimited.

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RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

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_______________________________________________
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Hi everyone,

	Here is the combined status report for the MWPP normative
(draft-ietf-avt-mwpp-midi-rtp-06.txt) and informative
(draft-lazzaro-avt-mwpp-coding-guidelines-02.txt) I-D's, also at:

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

	The normative document went through one revision (-06.txt),
fixing up many small details (see change log for the list). The
informative document was completed (-01.txt) and then touched up to
conform to the normative -06.txt (-02.txt).

	I'm assuming there is no reason for the informative document
to be a WG item, let me know if there is a wish for the document to be
relabeled as a WG item and I will retitle and resubmit.

The Road to Last Call ...
-------------------------

	Since Atlanta, many hours has been spent in e-dialogue with
MMA (MIDI Manufacturers Association) folks. The MMA has taken a more
active interest in MWPP recently, Jim Wright is the transport-expert
MMA contact. We have an informal list of open issues of potential
interest to the MMA (mixed in with a few from other sources sent in
recently), and I'm doing design work point-by-point off this list.

	Once the design is complete, and consensus occurs on the
result, we may be only an ID-nit-check away from Last Call. Most of
these items are better discussed after I release an I-D with a
proposed solution, since they are much more about MIDI than RTP
issues.

	However, below I bring up two points that would benefit from
pre-design discussion (on list, and at the meeting if there is
interest). Both are timing-related, and concern MWPP on the LAN (or
well-behaved WAN) in a stage or studio context. Both describe cases
where the natural MWPP interpretation of the default AVP audio timing
semantics do not completely address the needs of the application
domain.


[1] Sending-time proxies
------------------------

	One issue concerns "operating at the underlying latency of the
network, on a network known to have low jitter". This is of particular
interest with musicians who play a (soundless) keyboard controller
that generates MWPP, which is received by a separate box that makes
noise. As network latency is in the human feedback loop, low latency
is essential.

	In these environments, it may be desirable to set up a timing
mode where a sender takes on responsibility to transmit, at a precise
time interval relative to the MIDI source timing, RTP MWPP packets
that (usually) encode a single MIDI command. The receiver, aware that
the timing mode is in use (via an SDP parameter), uses this "time of
flight" timing together with the timestamp of the packet, as inputs
into whatever algorithm it chooses to use for playback. One potential
playback algorithm assumes zero network jitter, ignores the timestamp,
and executes the command on receipt. Other algorithms incorporate RTP
timestamps into the algorithm.
 	
	Aside from the new SDP fmtp parameter to code that the sender
is communicating timing information in its sending dynamics, a few
other MWPP additions are helpful for this mode:

 -- Defining the timestamp of the last MIDI command in the
    MIDI command section as the proxy for the sending time. This
    definition is particularly useful when combined with MIDI
    command section syntax (in the current I-D) that permits
    the last MIDI command in this section to be the "null" command.

 -- Defining the ptime and maxptime SDP values as originally intended
    in SDP -- the length of time encoded by the payload. A second
    set of MWPP fmtp parameters should be introduced to take over
    current MWPP use of ptime/maxptime (the delta of the RTP
    timestamps of two successive packets).


[2] Multi-format sync
---------------------

	A second LAN timing issue concerns multi-format sync.
Consider this block diagram (fixed-font recommended):



   -------------                                           --------
  | Digital     |  Digital Audio (ADAT, AES/EBU, S/PDIF)  | Digital| 
  | Audio       |----------->----------------------->-----| Mixer  |
  | Workstation |                                         | With   |-->out
   -------------                                          | Word   |
          |                                               | Clock  |
          |                                                --------
          | RTP/MWPP                                          |
  	  |                                                   |
          |                                                   ^
   -------------                                              |
   |           |  Digital Audio (ADAT, AES/EBU, S/PDIF)       |
   | SoftSynth |----------->--------------------------->-------
   |           |
   -------------


	The Digital Audio Workstation (DAW) generates two pre-recorded
streams: digital audio, sent out over one of the standard digital I/O
wire/fiber formats (ADAT, AES/EBU, S/PDIF, TDM, etc), and RTP/MWPP
over a LAN. The RTP/MWPP stream goes to a separate box running a
synth, that also has standard digital I/O format out. A digital mixer,
with word-clock to get frame-accurate audio, does the final mix (sent
to studio monitors, a mastering rig, etc).

	The DAW is playing back pre-recorded data here (overdubs are a
separate issue, but I believe solving the playback-only issue also
solves overdubs). DAWs normally let you time-shift all of the tracks
(MIDI and audio), to compensate for delays on the way to the final out.

	But, how does one compensate for a softsynth with an RTP
receiver, whose receiver-buffer latency may be variable? I can
imagine two approaches:

  [A] Assuming the softsynth and the DAW have NTP clocks that are kept
  in near-frame-accurate sync, you want SDP semantics that request a
  time differential between the RTCP NTP of the sender and the synced
  NTP clock of the receiver.

  [B] You want the ability to specify a fixed receiver latency, perhaps
  with options that break apart packet buffering latency from post-RTP
  synth latency (or maybe not).

[A] is potentially "calibration free", while [B] merely gives a studio
engineer a shot at manually calibrating the rig.

So, the question for the WG is, how do you feel about [A] and [B] as
design paths to pursue? [A], [B], both, neither? As always, an option
is to declare the problem domain "out of scope of RTP." But it would
be good to include this "anti-recommendation" in the memo.

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------

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



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        Adam Li <adamli@icsl.ucla.edu>
From: Harinath Garudadri <hgarudad@qualcomm.com>
Subject: RE: [AVT] File format for EVRC/SMV vocoder
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At 12:58 PM 3/7/2003 +0100, Lars-Erik Jonsson (EAB) wrote:
>Adam and others,
>
>I thought the reason for defining also the file format in the IETF
>was to make it in line with the transport format, and also because
>you would get it almost "for free" when you define the transport
>format.

In RFC2658, the payload format for PureVoice, the fileformat for storing 
data was not specified. We wanted rectify this and hence the 
draft-garudadri-qcp-00.txt.

>The proposed new approach seems to take away those motives,
>and then I do not understand why we should do it in the IETF.
>
>Further, I wonder what would be the rationales for using this new
>QCP format instead of the file format defined in the EVRC/SMV draft?

It turns out that there have been multiple, independent implementations of 
the qcp format for quite some. Our understanding is that the qcp format is 
quite mature and many applications use this format to exchange PureVoice 
and EVRC data. It is also widely used in many CDMA handsets. 
draft-garudadri-qcp-00.txt did not propose storage format for SMV, but 
there have been suggestions to include SMV in the next revision.


>If we are to standardize a file format in this body, I can not see
>why we should not stick to the format that is already finalized and
>has been taken through our standards process.

I think it would be useful to understand the extent to which the format 
defined in EVRC/SMV draft has been implemented, tested and being used. We 
proposed MIME type audio/qcp with qcelp and evrc as parameters, but we were 
advised that a better technical solution was to reuse audio/evrc 
andaudio/qcelp. Thanks,

hari.


>/L-E
>
>
> > -----Original Message-----
> > From: Mark Baugher [mailto:mbaugher@cisco.com]
> >
> > Adam
> >    I have a question:  Why take a file format specification
> > through the IETF?
> >
> > Best regards, Mark
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt

Dr. Hari Garudadri
phone : 858-651-6383
mobile : 858-229-2801
pager  : 858-653-0963 or hgarudadri@pager.qualcomm.com
--=====================_9754085==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
<font size=3>At 12:58 PM 3/7/2003 +0100, Lars-Erik Jonsson (EAB)
wrote:<br>
<blockquote type=cite class=cite cite>Adam and others,<br><br>
I thought the reason for defining also the file format in the IETF<br>
was to make it in line with the transport format, and also because<br>
you would get it almost &quot;for free&quot; when you define the
transport<br>
format. </font></blockquote><br>
In RFC2658, the payload format for PureVoice, the fileformat for storing
data was not specified. We wanted rectify this and hence the
<font size=3>draft-garudadri-qcp-00.txt. <br><br>
<blockquote type=cite class=cite cite>The proposed new approach seems to
take away those motives,<br>
and then I do not understand why we should do it in the IETF.<br><br>
Further, I wonder what would be the rationales for using this new<br>
QCP format instead of the file format defined in the EVRC/SMV
draft?</font></blockquote><br>
It turns out that there have been multiple, independent implementations
of the qcp format for quite some. Our understanding is that the qcp
format is quite mature and many applications use this format to exchange
PureVoice and EVRC data. It is also widely used in many CDMA handsets.
<font size=3>draft-garudadri-qcp-00.txt did not propose storage format
for SMV, but there have been suggestions to include SMV in the next
revision.<br><br>
<br>
<blockquote type=cite class=cite cite>If we are to standardize a file
format in this body, I can not see<br>
why we should not stick to the format that is already finalized and<br>
has been taken through our standards process. </font></blockquote><br>
I think it would be useful to understand the extent to which the format
defined in EVRC/SMV draft has been implemented, tested and being used. We
proposed MIME type audio/qcp with qcelp and evrc as parameters, but we
were advised that a better technical solution was to reuse audio/evrc
andaudio/qcelp. Thanks,<br><br>
hari.<br><br>
<br>
<blockquote type=cite class=cite cite><font size=3>/L-E<br><br>
<br>
&gt; -----Original Message-----<br>
&gt; From: Mark Baugher
[<a href="mailto:mbaugher@cisco.com" eudora="autourl">mailto:mbaugher@cisco.com</a>]<br>
&gt; <br>
&gt; Adam<br>
&gt;&nbsp;&nbsp;&nbsp; I have a question:&nbsp; Why take a file format
specification <br>
&gt; through the IETF?<br>
&gt; <br>
&gt; Best regards, Mark<br>
_______________________________________________<br>
Audio/Video Transport Working Group<br>
avt@ietf.org<br>
<a href="https://www1.ietf.org/mailman/listinfo/avt" eudora="autourl">https://www1.ietf.org/mailman/listinfo/avt</a></blockquote>
<x-sigsep><p></x-sigsep>
Dr. Hari Garudadri<br>
phone : 858-651-6383 <br>
mobile : 858-229-2801<br>
pager&nbsp; : 858-653-0963 or
hgarudadri@pager.qualcomm.com</font></html>

--=====================_9754085==_.ALT--

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



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From: Harinath Garudadri <hgarudad@qualcomm.com>
Subject: Re: [AVT] File format for EVRC/SMV vocoder
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At 04:46 AM 3/6/2003 -0800, Mark Baugher wrote:
>Adam
>   I have a question:  Why take a file format specification through the IETF?

Hi Mark,
did you mean AVT above? We were not sure if draft-garudadri-qcp-00.txt 
should go in to AVT or as an independent submission. I do not think this 
has been resolved yet.

If you meant IETF, our motivation was if a multimedia object (message?) was 
delivered over some mechanism other than RTP/UDP, (e.g. http, ftp, etc.) 
the receiver should be able to render the multimedia content appropriately. 
One example is if you create a voice attachment in Eudora email client, it 
is stored as a qcp file described in draft-garudadri-qcp-00.txt. If you 
send this email to someone who uses a different email client, that person 
could play the attachment in the QuickTime player, as QuickTime supports 
the above format.

Thanks,
hari.


>Best regards, Mark
>At 12:13 PM 3/5/2003 -0800, Adam Li wrote:
>>Hi folks,
>>
>>One issue just recently been raised about the file format for EVRC and
>>SMV data, which is specified the MIME registration for them. We very
>>much would like to hear your ideas and comments on this issue.
>>
>>As you may have known, we are working the IETF specification on the RTP
>>Payload Format for EVRC/SMV. It specifies how the EVRC/SMV payload is
>>carried in RTP sessions, and it also registered the MIME format for the
>>EVRC and SMV vocoders. The draft (draft-ietf-avt-evrc-smv) is currently
>>at the final editing stage at the IETF FRC Editor's queue.
>>
>>There has been a file format defined in the draft-ietf-avt-evrc-smv at
>>the MIME registration for EVRC/SMV. The file format specified there
>>hasn't been changed much since the first version of the draft in 2000.
>>
>>There is a new Internet Draft, draft-garudadri-qcp-00.txt, specifies a
>>different way of storage using the QCP file format, which has been used
>>for QCEPL-13K and EVRC by some applications.
>>
>>Currently, it has been proposed to remove the file format specification
>>in the draft-ietf-avt-evrc-smv, and to expand the draft-garudadri-qcp to
>>be the only formal specification for the file format for EVRC/SMV. On
>>the other hand, there are also suggestions to keep the file format at it
>>is specified in the draft-ietf-avt-evrc-smv.
>>
>>We would like to hear your suggestions, and hear from you how you think
>>would be the best way to proceed. Your comments are very much
>>appreciated.
>>
>>Adam
>>
>>
>>_______________________________________________
>>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

Dr. Hari Garudadri
phone : 858-651-6383
mobile : 858-229-2801
pager  : 858-653-0963 or hgarudadri@pager.qualcomm.com
--=====================_1210941==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
<font size=3>At 04:46 AM 3/6/2003 -0800, Mark Baugher wrote:<br>
<blockquote type=cite class=cite cite>Adam<br>
&nbsp; I have a question:&nbsp; Why take a file format specification
through the IETF?</font></blockquote><br>
Hi Mark,<br>
did you mean AVT above? We were not sure if
<font size=3>draft-garudadri-qcp-00.txt should go in to AVT or as an
independent submission. I do not think this has been resolved
yet.<br><br>
</font>If you meant IETF, our motivation was if a multimedia object
(message?) was delivered over some mechanism other than RTP/UDP, (e.g.
http, ftp, etc.) the receiver should be able to render the multimedia
content appropriately. One example is if you create a voice attachment in
Eudora email client, it is stored as a qcp file described in
<font size=3>draft-garudadri-qcp-00.txt. If you send this email to
someone who uses a different email client, that person could play the
attachment in the QuickTime player, as QuickTime supports the above
format.<br><br>
</font>Thanks,<br>
hari.<br><br>
<br>
<blockquote type=cite class=cite cite><font size=3>Best regards,
Mark<br>
At 12:13 PM 3/5/2003 -0800, Adam Li wrote:<br>
<blockquote type=cite class=cite cite>Hi folks,<br><br>
One issue just recently been raised about the file format for EVRC
and<br>
SMV data, which is specified the MIME registration for them. We 
very<br>
much would like to hear your ideas and comments on this issue.<br><br>
As you may have known, we are working the IETF specification on the
RTP<br>
Payload Format for EVRC/SMV. It specifies how the EVRC/SMV payload
is<br>
carried in RTP sessions, and it also registered the MIME format for
the<br>
EVRC and SMV vocoders. The draft (draft-ietf-avt-evrc-smv) is
currently<br>
at the final editing stage at the IETF FRC Editor's queue.<br><br>
There has been a file format defined in the draft-ietf-avt-evrc-smv
at<br>
the MIME registration for EVRC/SMV. The file format specified there<br>
hasn't been changed much since the first version of the draft in
2000.<br><br>
There is a new Internet Draft, draft-garudadri-qcp-00.txt, specifies
a<br>
different way of storage using the QCP file format, which has been
used<br>
for QCEPL-13K and EVRC by some applications.<br><br>
Currently, it has been proposed to remove the file format
specification<br>
in the draft-ietf-avt-evrc-smv, and to expand the draft-garudadri-qcp
to<br>
be the only formal specification for the file format for EVRC/SMV.
On<br>
the other hand, there are also suggestions to keep the file format at
it<br>
is specified in the draft-ietf-avt-evrc-smv.<br><br>
We would like to hear your suggestions, and hear from you how you
think<br>
would be the best way to proceed. Your comments are very much<br>
appreciated.<br><br>
Adam<br><br>
<br>
_______________________________________________<br>
Audio/Video Transport Working Group<br>
avt@ietf.org<br>
<a href="https://www1.ietf.org/mailman/listinfo/avt" eudora="autourl">https://www1.ietf.org/mailman/listinfo/avt</a></blockquote><br>
_______________________________________________<br>
Audio/Video Transport Working Group<br>
avt@ietf.org<br>
<a href="https://www1.ietf.org/mailman/listinfo/avt" eudora="autourl">https://www1.ietf.org/mailman/listinfo/avt</a><br>
</blockquote>
<x-sigsep><p></x-sigsep>
Dr. Hari Garudadri<br>
phone : 858-651-6383 <br>
mobile : 858-229-2801<br>
pager&nbsp; : 858-653-0963 or
hgarudadri@pager.qualcomm.com</font></html>

--=====================_1210941==_.ALT--

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



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Hari,

At 09:57 AM 3/7/2003 -0800, Harinath Garudadri wrote:
>At 04:46 AM 3/6/2003 -0800, Mark Baugher wrote:
>>Adam
>>   I have a question:  Why take a file format specification through the IETF?
>
>Hi Mark,
>did you mean AVT above?

No, I meant the IETF.  I asked the question because I don't think of the 
IETF as having a lot of file-format expertise so it's not the obvious place 
to develop file-format technology.

>We were not sure if draft-garudadri-qcp-00.txt should go in to AVT or as 
>an independent submission. I do not think this has been resolved yet.
>
>If you meant IETF, our motivation was if a multimedia object (message?) 
>was delivered over some mechanism other than RTP/UDP, (e.g. http, ftp, 
>etc.) the receiver should be able to render the multimedia content 
>appropriately. One example is if you create a voice attachment in Eudora 
>email client, it is stored as a qcp file described in 
>draft-garudadri-qcp-00.txt. If you send this email to someone who uses a 
>different email client, that person could play the attachment in the 
>QuickTime player, as QuickTime supports the above format.

Is it related to MP4, then?

Best Regards, Mark


>Thanks,
>hari.
>
>
>>Best regards, Mark
>>At 12:13 PM 3/5/2003 -0800, Adam Li wrote:
>>>Hi folks,
>>>
>>>One issue just recently been raised about the file format for EVRC and
>>>SMV data, which is specified the MIME registration for them. We very
>>>much would like to hear your ideas and comments on this issue.
>>>
>>>As you may have known, we are working the IETF specification on the RTP
>>>Payload Format for EVRC/SMV. It specifies how the EVRC/SMV payload is
>>>carried in RTP sessions, and it also registered the MIME format for the
>>>EVRC and SMV vocoders. The draft (draft-ietf-avt-evrc-smv) is currently
>>>at the final editing stage at the IETF FRC Editor's queue.
>>>
>>>There has been a file format defined in the draft-ietf-avt-evrc-smv at
>>>the MIME registration for EVRC/SMV. The file format specified there
>>>hasn't been changed much since the first version of the draft in 2000.
>>>
>>>There is a new Internet Draft, draft-garudadri-qcp-00.txt, specifies a
>>>different way of storage using the QCP file format, which has been used
>>>for QCEPL-13K and EVRC by some applications.
>>>
>>>Currently, it has been proposed to remove the file format specification
>>>in the draft-ietf-avt-evrc-smv, and to expand the draft-garudadri-qcp to
>>>be the only formal specification for the file format for EVRC/SMV. On
>>>the other hand, there are also suggestions to keep the file format at it
>>>is specified in the draft-ietf-avt-evrc-smv.
>>>
>>>We would like to hear your suggestions, and hear from you how you think
>>>would be the best way to proceed. Your comments are very much
>>>appreciated.
>>>
>>>Adam
>>>
>>>
>>>_______________________________________________
>>>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
>
>Dr. Hari Garudadri
>phone : 858-651-6383
>mobile : 858-229-2801
>pager  : 858-653-0963 or hgarudadri@pager.qualcomm.com

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



From mailnull@www1.ietf.org  Mon Mar 10 22:13:00 2003
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	(envelope-from casner@acm.org)
Date: Mon, 10 Mar 2003 19:07:36 -0800 (PST)
From: Stephen Casner <casner@acm.org>
To: AVT WG <avt@ietf.org>
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Subject: [AVT] Changes in response to IESG review of RTP
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My main topic for discussion in the "Introduction and document status"
item at the top of the AVT agenda is the set of changes to the RTP
spec and RTP/AVP profile resulting from the exchange of email on the
set of issues I sent out as separate messages in January.  The results
are in

http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-12.txt
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-12.ps
http://www.ietf.org/internet-drafts/draft-ietf-avt-profile-new-13.txt
http://www.ietf.org/internet-drafts/draft-ietf-avt-profile-new-13.ps

Actually, I hope that everyone finds the changes I have made to be
a satisfactory exposition of the results of the email exchange.  If
not, please speak up in email or at the meeting.

I have written up a detailed accounting of these changes to facilitate
the IESG's review and (I hope) acceptance of them.  I sent the
following message to the IESG.  You are welcome to look at the package
of files I mention at the end.

                                                        -- Steve

---------- Forwarded message ----------
Date: Mon, 10 Mar 2003 18:41:17 -0800 (PST)
From: Stephen Casner <casner@acm.org>
To: iesg@ietf.org
Cc: Allison Mankin <mankin@psg.com>
Subject: Changes in response to IESG review of RTP

Dear IESG members:

You may have observed that the following updated drafts have been
posted:

    draft-ietf-avt-rtp-new-12
    draft-ietf-avt-profile-new-13

These drafts update -11 and -12, respectively, which have already been
"tentatively approved" by the IESG for publication as Draft Standards.
I realize that the IESG normally does not allow drafts to be updated
after they have been reviewed and approved because you do not want to
have to review the revised drafts.  However, in this case I received
permission from our responsible AD, Allison, to submit revised drafts
because the set of changes that I believe should be made is large
enough that trying to pass them to the RFC Editor as "RFC Editor
notes" is not appropriate.  There are three reasons for this:

  - There were several changes that the IESG requested.

  - The drafts have been under review for more than a year (this is
    not a complaint -- some of that time is mine) and during that
    period there have been several questions from implementers that
    pointed out places where clarification of the text was needed.
    I'd like to have the document be the best it can as it goes to
    Draft Standard.

  - There are changes in several places where I have tried to
    carefully scrub down these documents according to ID-nits and
    rfc2223bis.  There are two examples that I think will let you see
    why I wanted to handle these revisions through new I-D's:

      - I have changed the order of a few end-sections.  By doing that
        in the source, all the refereces are correctly updated.

      - I have fixed all the places where there were not two spaces
        after the end of a sentence.

Marking all of these in RFC-Editor-note format would not be pleasant
or efficient for me, the IESG, or the RFC Editor.  It would be
unreasonable, I think, to have a long list of RFC-Editor-notes in the
message announcing approval to the ietf list and the RFC Editor.

To minimize the effort required for you to review these changes, I am
providing a package of files to allow you to easily satisfy yourselves
that the changes are appropriate:

  - For the differences of substance, I am providing a description and
    motivation in RFC-Editor-note format.  This should be all that the
    IESG as a whole needs to consider.

  - To let you verify that I have not tried to sneak in any changes of
    substance other than those I describe, I have separated the
    changes into a series of "layers" (in the Photoshop sense) so that
    it is easy to see the differences between each successive pair of
    layers using diff or wdiff.  I have provided the diff or wdiff
    output as well, but you may also run those programs for yourself.
    I assume it would be sufficient for just one IESG member to
    perform this verification.  I know there is at least one wdiff
    aficionado on the IESG.

This package of files is available in a tarball at

    ftp://ftp.packetdesign.com/outgoing/casner/rtp-diffs.tgz

Since this tarball contains multiple copies of each draft, it is
1.1MB.  Please see the files README-rtp.txt and README-profile.txt for
a listing of the layers and the changes they contain.  I would be
happy to answer any questions you may have.

                                                        -- Steve

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



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Folks,
here are the issues re the H.264/JVT packetization:

1. Status of JVT -- should be out for the final ballot by the time of the 
meeting.
2. Fragmentation -- we decided on adding application layer fragmentation of 
NALUs to support some exotic scenarios.  I plan to describe the pros and 
cons of the fragmentation, and hope for more giudance from the meeting re 
these feature.  We had discussed this briefly in Atlanta already, but it 
has turned out that I forgot some applications where fragmentation is useful.
3. MIME/SDP sections: are they ok?
4. Interaction with MPEG-4 simple packetization (since H.264 is also 
ISO/IEC 14496-10).  Miska and I are currently in Pattaya at the JVT/MPEG 
meeting, as are Jan Van-der-Meer, Dave Singer, Young-Kown Lim (and other 
knowledgeable people).  We will have some discussions about the interaction 
of the two drafts, and report probably to the mailing list, but certainly 
orally at the meeting, about the outcome of these discussions.
5. I currently believe that the problem mentioned in section 12 of the 
draft doesn't exist, but we will discuss this here in Pattaya as well and 
take action as necessary.
6. Security section seems to need more work, but this is not reflected in 
the draft yet.  We plan to present a summary of the problems and proposed 
solution if there is time left.

Stephan

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



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        Adam Li <adamli@icsl.ucla.edu>
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Subject: RE: [AVT] File format for EVRC/SMV vocoder
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At 12:58 PM +0100 3/7/03, Lars-Erik Jonsson (EAB) wrote:

>  If we are to standardize a file format in this body, I can not see
>  why we should not stick to the format that is already finalized and
>  has been taken through our standards process.

Given that there are a number of deployed implementations which have 
been using the qcp format to exchange qcelp-13k and evrc, what is the 
advantage of having the audio/evrc media type specify a different 
format?
-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly-selected tag: ---------------
Despite an abundance of devoted followers, the production of great
leaders been discontinued.
_______________________________________________
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Folks,

first of all I need to thank Magnus for his careful review.  Most of his 
points will be part of a -02 submission of the draft.  However, there are a 
few points that need more discussion, and one where I would like to get an 
opinion from the group:

Magnus:
>11. Chapter 6.3 The use of the F bit for aggregated units. Is it really 
>wise to set this bit as bad as soon as any of the NALUs part of the 
>aggregate is bad. This is useless information because you still needs to 
>go into the packet an look at each NALU to determine which is actually 
>useful. A cheap implementation that just throws away NALUs which is 
>damaged can't use this flag anyway. It could be useful it the flag is 
>instead defined as: "The F bit MUST only be set if all NALUs part of the 
>aggregate has their F bit set". That way the cheap implementation can 
>throw the whole packet as soon as it looks at the first bit.

To make it short: how do you set an error bit for an aggregation packet 
based on the error bits of the carried sub-packets.  Two options:

You logically AND the sub-packet error bits: this is what is in the draft
You logically OR the sub-packet error bits: this is what Magnus proposed

It's IMHO a minor issue, there are good arguments for going both ways, and 
I would be fine either way.
Opinions?

Thanks,
Stephan

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



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From: "Alan Duric" <alan.duric@globalipsound.com>
To: "'Colin Perkins'" <csp@isi.edu>, <avt@ietf.org>, <casner@acm.org>
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Subject: [AVT] Agenda item RTP Payload Format for iLBC Speech
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Hi all,

here is some info about the agenda item:

RTP Payload Format for iLBC Speech 		(Duric)            15
min
	draft-ietf-avt-rtp-ilbc-01.txt
	draft-ietf-avt-ilbc-codec-01.txt

and the issues that will be addressed/discussed at the meeting on
Thursday, March the 20th:

- details about the changes made from 00 versions (introduction of 20 ms
frame size mode, improvements on PLC, ...)
- implications of 20 ms frame size mode on the RTP Payload Format draft
- feedback from SIPit 12, where beside SIP interop tests were conducted
interop tests of the running iLBC implementations
- next steps 
- time slot for Q&A 


Drafts are available under the following links: 

http://www.ietf.org/internet-drafts/draft-ietf-avt-ilbc-codec-01.txt
- iLBC codec spec 

http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-ilbc-01.txt
- RTP payload profile for iLBC

Best regards,
Alan


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From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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Hi,

Here is my comments on the latest version of the AVPF draft. I am glad 
that we are coming very near a finished specification for AVPF. However 
I still have some things that we might need to discuss a little bit, see 
point 14-17. I hope this will not delay the progressing to much. I would 
very much hope that we can see a updated version just weeks after meeting.

Best Regards

Magnus


1. Before Abstract: The short copyright is missing.

2. Section 1.1: Early RTCP packet definition. It might be good in this 
definition to explain what [1] is by including the word "RTP/RTCP" 
before the reference.

3. The documents page lengths seem to be highly variable. It is not 
pretty when the footer jumps up and down the different pages.

4. Section 2, RTCP Report interval: The second sentence in the 
explanation says: " In Regular RTCP mode, all rules from [1] apply." 
This is not true as the minimal 5 second rules is removed in this case. 
Also the following sentence makes it clear that the five second rules is 
only removed for immediate and early RTCP modes.

5. Isn't usually that the page header has the document title centered on 
the line?

6. Section 3.2 Third paragraph: A sentence beginning with "Hence" is 
missing space before it.

7. Section 3.2 Fourth paragraph: The first sentence is a bit unclear. 
The use of following lacks a clear context. I understand that one may 
send a earlier then what the normal rules would allow. However this 
sentence may also be interpreted that scheduling earlier is done by 
following the previous normal rule.

8. Section 3.2 Fourth paragraph. In the middle of the section there is a 
number of spaces missing and a double ".". See sentence: "The receiver 
waits for a (short)random dithering intervalto checkwhether it sees a 
corresponding FB message from any other receiver reporting the same event.."

9. Section 3.2 Fourth paragraph, last sentence: "The permission to send 
Early feedback is derived from the time Early feedback was sent." I 
think this is a bit unclear. I think that one must better clarify that 
it depend on if early feedback has been sent or not in the last 
interval. A proposal could be :" The permission to send Early feedback 
is dependent if an Early feedback has been sent since the last regular 
RTCP packet."

10. Section 3.4 bullet d: I think the word "deterministic" should be 
added to the first sentence to clarify the impact Tmin has.

11. Section 3.4 bullet g: I think one could include text to clarify what 
the value will be if in a multicast session.

12. Section 3.4 Bullet m: The following sentence should be reformulated 
to become more clear and precise: "If T_rr_interval!= 0 then Regular 
RTCP packets will not be scheduled T_rr after the last Regular RTCP 
transmission (at tp+T_rr) but at least T_rr_interval after the last 
Regular RTCP transmission, i.e. later than or at tp+T_rr_interval." 
Mostly I think that "but" is the wrong word here.

13. Section 3.4 Bullet o: Prior to this section there is a stray 
occurrence of "Regular".

14. Before section 3.5: I think that there should be a section 
describing Immediate Feedback mode also. This mode is a bit under 
explained. One get some general impression of the mode in section 3.3. I 
think that this section could clarify that in immediate more one can 
send FB messages anytime. Also how one connects immediate packets to 
regular reports.  However one MUST track the number of reports that are 
sent to determine when one passes the FB threshold.

15. Section 3.5.3 bullet 2C: I think this paragraph is not specified 
enough on what to do. In all other cases one always determines what to 
do next. Here one simple says that it will not be sent. I think there 
are two ways forward:
A) As currently specify that one does not send and then included which 
variables that needs to be updated so that a new scheduling calculation 
is performed. Otherwise the RTCP sending simple stops here.
B) Specify that one shall in this case schedule the next regular packet 
for t_rr_last+T_rr_current_interval. As this is a random value, it meats 
the criteria for a value possible to schedule.

16. Section 4: The SDP attribute: I have problems with the current 
definitions of the values. I don't understand why the payload specific 
FB messages must be signaled in either ack or nack. What if you do not 
want to use generic NACK or ACK and only a payload format specific type 
like RPSI. Wouldn't it be better to create this as a flat structure. 
Having a attribute definition like the following one:

rtcp-fb-syntax = "a=rtcp-fb:" rtcp-fb-pt SP rtcp-fb-val CRLF
   
   rtcp-fb-pt         = "*"   ; wildcard: applies to all formats
                      / fmt   ; as defined in SDP spec
   
   rtcp-fb-val        = "ack"
                      / "nack"
                      / "app" [SP byte-string]
                      / "rpsi"
                      / "sli"
                      / "pli"
                      / "trr-int" SP 1*DIGIT
                      / rtcp-fb-id [SP byte-string]
   
   rtcp-fb-id         = 1*(alpha-numeric | "-" | "_")
   


I can accept the choice to only have one parameter per attribute line. 
The gain is of course that extension types are easier to write and parse.

17. Section 5. bullet for AVPF receivers. Similar to the note in AVP 
enties that AVP receivers may timeout AVPF sources that run with 
T_rr_interval larger than 5 seconds, a problem for AVPF exist.  A AVPF 
receiver setting T_rr_interval smaller than 5 seconds may actually time 
out AVP entities. So unless only AVPF senders are present T_rr_interval 
SHOULD NOT be set smaller than 5 seconds.


18. Section 6.3.3.2 Native RPSI bit string: Shouldn't it be good to put 
some references here to codes that has RPSI and have RTP payload formats?

19. Section 8, Second paragraph: "Group members of the
   associated RTP session (possibly pretending to represent a large
   number of entities) may disturb the operation of RTCP by sending
   large numbers of RTCP packets thereby reducing the RTCP bandwidth
   available for Regular RTCP reporting as well as for Early FB
   messages."

This section covers one case. When a attacker sends with different SSRC 
every time to make the number of participants large. There is another 
case that effect RTCP reporting interval that might also be mentioned. 
May however not be as potent. This is to send very large RTCP packets. 
This increase the avg_rtcp_size variable and also results in lower 
reporting interval.

20. Section 9: The registration of Application Layer Feedback claims 
that it will use value "1" instead of "15" which the earlier tables 
indicate.


-- 

Magnus Westerlund 

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



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To: "'Randall Gellens'" <randy@qualcomm.com>, Adam Li <adamli@icsl.ucla.edu>
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Subject: RE: [AVT] File format for EVRC/SMV vocoder
Date: Tue, 11 Mar 2003 13:52:36 +0100
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> >  If we are to standardize a file format in this body, I can not see
> >  why we should not stick to the format that is already finalized and
> >  has been taken through our standards process.
> 
> Given that there are a number of deployed implementations which have 
> been using the qcp format to exchange qcelp-13k and evrc, what is the 
> advantage of having the audio/evrc media type specify a different 
> format?

OK, but why has not this issue been brought up earlier? The EVRC/SMV
format work in AVT has been going on for a long time, and therefore one
can question whether it is correct (from a procedural point of view) to
suddenly not use what has been developed in this WG, but use something
defined elsewhere. IETF standardization is usually an engineering
process, not rubber-stamping of already deployed technical solutions.

Note that I am not aware of all the details in this matter, but what
has been communicated on the list does not provide enough material for
motivating what has been proposed. "We have already deployed this 
alternative format while you were trying to agree on a standard, and
now we think you should use our format instead since it is more
widely deployed" is a rather strange way of arguing in standards
development, I think.

Once again, note that I am not addressing the technical aspects, and
the above analysis is of course a little bit extreme. However, I hope
it clarified my point. Personally, I have no strong opinions in favor
of either of the proposals, my concern is more about encouraging active
participation in our standards work, instead of encouraging people to
work in parallel with our standards efforts. It is a process concern.

Cheers,
/L-E
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Subject: [AVT] H.264 payload spec: a few remarks
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stephen, all,

a few remarks on H264 RTP payload:

********************

      General remarks:

   The title(!) and the text keeps on using "JVT" while the file
   name, the abtract and introduction use the term H.264. (and
   introduction explains why H.264 is the correct name!) I think a
   global find-and-replace(JVT->H.264) is needed!

   The term "NALU" is defined but after that "NAL unit" is also
   used, consistency could help.

There are quite a few reference problems i.e.
      * figures and table numbers
      * "above" used for things that are below and vice versa
      * wrong internal section references

   Some acronyms are not explained: SEI, IDC (also "Idc"?), DPA,
   DPB, DPC (even FEC etc), (could be more I did not check), also I
   am always fascinated by recursivity in acronyms: "NRI: NAL
   Reference IDC" (this one is superb!-)


      Detailed remarks:


      Section 1.1

   I suspect the sentence "Neither VCL nor NAL are claimed to be
   media or network independent" should be removed because it is not
   logically correlated to the rest of the paragraph.

   About the paragraph:

   "Also, there are NAL units that are affecting many pictures and
   are, hence, inherently time-less. For this reason, the handling
   of the RTP timestamp requires some special considerations for
   those NALUs for which the sampling or presentation time is not
   defined, or, at transmission time, unknown."

   Describing these NALU as being "time-less" and of having
   "presentation time [] not defined" could cause some mis-
   understanding, another formulation could be:

   "Some NALU transport information that affect the decoding process
   over a period of time corresponding to many video frames; For
   these units, the handling of the RTP timestamp requires some
   special considerations especially in cases where this duration is
   unknown at transmission time." BTW the same remark applies in section 6.1 for "If the NALU has
   no own timing properties". You should also elaborate i.e. start
   to present _how_ the "special considerations" are implemented in
   the spec ...

   About the sentence:

   "The combination of the higher level parameters is called a
   Parameter Set".

   It would be useful to add:

   "which is a data structure akin to what is described as "decoder
   configuration" in MPEG-4 specifications."

   since this concept is quite specific to MPEG-4 codecs and H.264
   (e.g. most speech or "older" codecs do not have this etc).

   About the sentence:

   "Although, conceptually, the Parameter Set updates are not
   designed to be sent in the synchronous packet stream, this memo
   contains means to convey them in the RTP packet stream."
   It is strange (actually potentially contradictory: if they are not
   "designed" to be sent in packets you would need to document how
   to map them into NALU but then why the draft does not do that?),
   perhaps you should replace "designed" with "intended"? in any
   case it could benefit a little explanation of the "why".

   The sentence:

   "The JVT specification declares a value of 1 as a syntax
   violation"

   I like this old joke ... ah syntax violation ... is it a
   disgusting crime or something one might indulge oneself in?-)
   but at first it _really_ reads as "it is forbiden to set this bit to
   1" so your reader will start scratching his (her) head... what
   about:

   "The F bit when set to 1 signals that the NALU or header contains
   a syntax error, in the context of this specification it may have
   been set by a network device upon detection that a transmission
   error has affected the corresponding data. A prudent reaction of
   decoders that are incapable of handling bit errors is to discard
   such packets. "


   In the "4. scope" section one finds:

   "Likely, the first applications of a Standard Track RFC resulting
   from this draft will be in the conversational multimedia field,
   video telephone or video conference"

   This is all very good but something is missing there. I think the
   problem is not what will be the "first" application but rather
   what is the _intention_ and _suitability_. This section does not
   say anything about the suitability of the specification for say
   broadcast or video on demand, which is at minimum extremely
   surprising since in the introduction the reader was entertained
   about "a very broad application range" and how good H264 is to
   transport satellite quality at 1.5 Mb/s!


   In section 6.3.1

   I could not find _how_ DON is computed except "The DON of the
   first NALU in transmission order MAY be set to any value", if you
   dont mandate anything it should be explicit for example "It is
   not required that DON are monotonously incremented"

   in the figure "NALU payload format for STAP." (same for the next
   2 figures) what are the first 8 bits ?

   Sometimes one finds "NALU type octet" some other times "|F|NRI|
   type   |"; consistency would help.


   By "D1 . D2 >= 32768" you probably mean "D1 - D2 >= 32768" ?

   It says: "NAL units associated with different primary coded
   pictures according to subclause 7.4.1.2 of [1] MUST NOT have the
   same value of DON." Is it possible to have an accidental
   collision? Then what could be the consequence?

   In section 6.3.2

   "DON of the following NALU is equal to DONB + DOND and MUST NOT
   be larger than 65535" You mean that DON=DONB+DOND MUST NOT roll
   over inside a packet, right? It would help to add if DONB roll
   over can occur at any value (also see above "not monotonously
   incremented" rule?) for the next packet...

   In Section 6.4. "Fragmentation Units " I could not find the
   indication that there cannot be more than one FU per RTP packet?
   I think it is the case? if it is  it should be more explicit.
   (and also, why then call it a "unit" rather than a "packet" ?)

   "R: 1 bit The Reserved bit MUST be 0." sheer curiosity: what is
   the purpose of a reserved bit with a _forever_ fixed value ?

   "A FU payload can have any number of octets and can be empty" why
   (what for?) can it be empty?


   In Section 7.1 about the sentence:

   "Some receivers MAY NOT support a transmission order that does
   not conform to the NAL unit decoding order. "

   By what mean the sender can find out about that?


   In Section 10. Security Considerations:

   What about crashing a decoder by sending a lethal Parameter Set ?

   References:

   for [14] I found it online at: http://www.diee.unica.it/pv2000/proceedings/papers/30.pdf

   The link http://www.pv2002.org is broken(?) but I found that paper at:
   http://www-video.eecs.berkeley.edu/papers/sangk7/pv2002.doc

************************************************************



regards,


Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com

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



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Subject: Re: [AVT] H.264 payload spec: a question
To: Stephan Wenger <stewe@cs.tu-berlin.de>
Cc: avt@ietf.org
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Magnus,

I see your point but such a rule could make the F bit almost never used
i.e. if all the packets are corrupted you are in bad trouble to start with (!)
and otherwise the F bit will "almost always" be zero, No?

regards,


Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com


                                                                                                                                       
                                                                                                                                       
                                                   To:   avt@ietf.org                                                                  
                                                   cc:   (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                                    
                                                   Subject:    [AVT] H.264 payload spec: a question                                    
                                                                                                                                       
               Stephan Wenger                      Classification:                                                                     
               <stewe@cs.tu-berlin.de>                                                                                                 
                                                                                                                                       
               Sent by:                                                                                                                
               avt-admin@ietf.org                                                                                                      
                                                                                                                                       
               11/03/2003 06:38                                                                                                        
                                                                                                                                       
                                                                                                                                       




Folks,

first of all I need to thank Magnus for his careful review.  Most of his
points will be part of a -02 submission of the draft.  However, there are a
few points that need more discussion, and one where I would like to get an
opinion from the group:

Magnus:
>11. Chapter 6.3 The use of the F bit for aggregated units. Is it really
>wise to set this bit as bad as soon as any of the NALUs part of the
>aggregate is bad. This is useless information because you still needs to
>go into the packet an look at each NALU to determine which is actually
>useful. A cheap implementation that just throws away NALUs which is
>damaged can't use this flag anyway. It could be useful it the flag is
>instead defined as: "The F bit MUST only be set if all NALUs part of the
>aggregate has their F bit set". That way the cheap implementation can
>throw the whole packet as soon as it looks at the first bit.

To make it short: how do you set an error bit for an aggregation packet
based on the error bits of the carried sub-packets.  Two options:

You logically AND the sub-packet error bits: this is what is in the draft
You logically OR the sub-packet error bits: this is what Magnus proposed

It's IMHO a minor issue, there are good arguments for going both ways, and
I would be fine either way.
Opinions?

Thanks,
Stephan

_______________________________________________
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Hi Phillipe and Stephan,

philippe.gentric@philips.com wrote:

>Magnus,
>
>I see your point but such a rule could make the F bit almost never used
>i.e. if all the packets are corrupted you are in bad trouble to start with (!)
>and otherwise the F bit will "almost always" be zero, No?
>
>  
>
Yes, however my point is that this field is useless unless it provides 
any advantage over looking at each NALU. The F bit in STAP and MTAPs are 
a redundant field. The information is available for in each NALU header. 
So with the current definition of the F in STAP, MTAP what advantage 
does it give? To my understanding the only knowledge it gives is: At 
least one NALU is damaged. However you will still need to look at which 
they are. So I don't see any advantage of reading the F bit in the STAP, 
MTAP header.

However if you redefine it as I propose it is a little bit less 
redundant, now the F bit gives the following information: All NALUs are 
damaged, unless I can repair them I can throw away them. This means that 
simple implementations (not trying any repair) can throw away the whole 
packet with less processing. Of course it is correct that the bit will 
be set much less often.

Cheers

Magnus Westerlund 

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



_______________________________________________
Audio/Video Transport Working Group
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From mailnull@www1.ietf.org  Tue Mar 11 08:59:34 2003
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Subject: Re: [AVT] H.264 payload spec: a question
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Cc: avt@ietf.org, Stephan Wenger <stewe@cs.tu-berlin.de>
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Hi Magnus, stephan,

I agree this bit is not extremely useful

... and it is a better idea then to set it if all are discardable...

regards,

Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com


                                                                                                                                              
                                                                                                                                              
                                                          To:   avt@ietf.org                                                                  
                                                          cc:   Stephan Wenger <stewe@cs.tu-berlin.de>                                        
                                                           (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                                         
                                                          Subject:    Re: [AVT] H.264 payload spec: a question                                
               Magnus Westerlund                                                                                                              
               <magnus.westerlund@era.ericsson            Classification:                                                                     
               .se>                                                                                                                           
                                                                                                                                              
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               avt-admin@ietf.org                                                                                                             
                                                                                                                                              
               11/03/2003 14:43                                                                                                               
                                                                                                                                              
                                                                                                                                              




Hi Phillipe and Stephan,

philippe.gentric@philips.com wrote:

>Magnus,
>
>I see your point but such a rule could make the F bit almost never used
>i.e. if all the packets are corrupted you are in bad trouble to start with (!)
>and otherwise the F bit will "almost always" be zero, No?
>
>
>
Yes, however my point is that this field is useless unless it provides
any advantage over looking at each NALU. The F bit in STAP and MTAPs are
a redundant field. The information is available for in each NALU header.
So with the current definition of the F in STAP, MTAP what advantage
does it give? To my understanding the only knowledge it gives is: At
least one NALU is damaged. However you will still need to look at which
they are. So I don't see any advantage of reading the F bit in the STAP,
MTAP header.

However if you redefine it as I propose it is a little bit less
redundant, now the F bit gives the following information: All NALUs are
damaged, unless I can repair them I can throw away them. This means that
simple implementations (not trying any repair) can throw away the whole
packet with less processing. Of course it is correct that the bit will
be set much less often.

Cheers

Magnus Westerlund

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



_______________________________________________
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,

I have some comments on the latest version of the draft. It starts to look good, and I hope for quick addressing of the problems. 


1. The title: I don't like the title, I think it is a bit miss leading. This is not a only RTP extended reports. Some of the things only are related to RTCP functionality. So maybe "RTP/RTCP Extended Reports" would be a better title.

2. The page length of the first page is different from the others. Maybe not that important as the RFC-Editor will change this page to be 1 line longer than all other pages in the draft. 

3. Section 1: Should be page break to a new page for readability.

4. The page header is not the normal one. See draft-rfc-editor-rfc2223bis-04.txt section 3.3. In drafts that results in the following example from above draft:

Internet-Draft        Instructions to RFC Authors            10 Feb 2003 

5. Section 1, bullet list with report blocks: All bullets with second lines should be indented to look nicer and be more readable. 

6. Section 2, Figure: The "PT=XP=207" shouldn't that be PT=XR=207?

7. Section 4.1, listing for format figure: The bullet for "SSRC of Source" is missing in the list. 

8. Section 4.3, The RTP Timestamp i, bullet after format figure: I still think that this format should be reduced. I also think there is a problem of using the RTP timestamp clock to stamp arrival time. First it is unnecessary as there is no connection between used RTP TS in the packet and the arrival sampling. Secondly, this section does not describe how you handle changes of RTP TS rates due to change of PT in the incoming data. It is also possible that different sources uses different RTP TS rates due to usage of different PTs. All these problems go away if one defines a single clock rate to use in this format. One can also then consider that the number of bits per report can be reduced significantly. 

9. Section 4.4 first paragraph. Why is not the TTL fields defined for the IPv6 Hop Limit field which is identical to TTL?

10. Section 4.4, The statistics summary format: I have earlier had the misconception that this format would actually be a bit smarter and having variable fields present. However I know understand that even if only a single part of the report is used the whole 40 bytes must be sent. I would be more efficient to remove the parts of the format that are not being reported on. I don't see the little complexity of such an operation as having much significance. If for example only a single part, e.g. TTL, of the block is of interest the overhead could be reduced with up to 24 bytes. 24 bytes may be as much as 20 % increase in packet size, i.e. the reporting interval could be reduced with the same amount. 

11. Section 4.4, The definition of standard deviation fields. All fields seems to be defined as they are integer fields why standard deviation calculations normal generate a real value. I think one needs to mention what type of operation is used to produce the integer value, rounding, ceiling, flooring?

12. Section 4.5 First sentence: Please include NTP in this sentence so that it reads: "This block extends RTCP's timestamp reporting so that non-senders may also send NTP timestamps.". This makes it clears directly what type of timestamp is used. 

13. Section 4.5 Last sentence of NTP timestamp definition: If a sender has no notion of time it shall not send this format at all. It is very unnecessary for other entities to waste bits on something no one can use. So I would suggest that the last sentence is changes to: "A report sender that has no notion of wallclock or elapsed time SHALL NOT send this report block.".

14. Section 4.6, block length definition: I think one should clarify that the length is used to derive the number of sources that are reported on in the reporting block. The formula seems to be as simple as taking "block length / 3". 

15. Section 4.6, Format definition: For completeness please include the SSRC_i in the fields explained. 

16. Section 4.6, DLRR definition: I think that senders of this block SHALL only send reports on sources sending the NTP Timestamp report. It is useless to spend bits on source that has not asked for this information. It is a total waste as the SSRC being reported has nothing to calculate the RTT unless it has sent a NTP Timestamp reporting block. I would also make a an addition to this paragraph that a reporter may stop reporting on a source after sending three times on the same LRR. A proposed text: 

"A Sender of DLRR reports MAY stop sending them for any given source when it has reported three times on the same LRR for that source. It MUST resume the reporting when it receives a new LRR in the form of a Receiver Timestamp Report block from that source."

17. Section 4.7.4: The signal level calculations are not clear as Log10 has been separated into "log 10". 

18. Section 4.7.7: How does one use the jitter buffer sizes when one has a sample based codec and have no defined frame size, or for that matter if the frame size is variable? 

19. Section 5: The first sentence, replace RTP with RTCP. 

20. Section 5.1: I don't think one can order IANA to give one a certain number. So I think changing SHALL to request is better. However this the chairs may answer better. 

21. Section 5.2: I think this format should register 255 as reserved for future extending of the number space. 

22. Section 5.2, last paragraph: I don't see much reason for the recommendation that 0 and 255 should not be used. There are several ways of checking integrity of the different reports. The simplest is to check the XR report block length against the different report blocks length field. This is simply a continuation of what should be performed at RTCP packet types level. I would in fact recommend that a new section containing decoding considerations is written to explain these method. 

23. Section 6: Second and third paragraph: Can we actually write something about the AS's right to keep network topology and characteristics confidential? 

24. Section A.1: Why is the extended sequence number initiated to the middle of the space? Normally one would start with an extended sequence number of 0. Shouldn't one try to be consistent with extended highest sequence number received in the receiver reporting. 

25. A SDP section. I still think that a SDP section should be included in this document. However if the authors of XR format draft are not willing to include it in their document, I volunteer myself to write and drive a separate SDP draft towards and RFC. 


Best Regards

Magnus Westerlund 

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



_______________________________________________
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From mailnull@www1.ietf.org  Tue Mar 11 10:01:32 2003
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From: Kreuter Ruediger <ruediger.kreuter@siemens.com>
To: "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>
Cc: "'avt@ietf.org'" <avt@ietf.org>
Subject: AW: [AVT] updated "Clearmode" (64kbit/s unrestricted) Internet-dr
	aft
Date: Tue, 11 Mar 2003 16:01:43 +0100
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Hi Magnus,

thanks for your indepth review and the according comments.
They are still true for version 02, so the updates will 
come in version 03 after the upcoming IETF Meeting.

See more detailed answers below.

Regards,
Rüdiger Kreuter
Siemens AG
ICN CP Systems Engineering
D-81359 München
Tel    +49 89 722 62553
Fax    +49 89 722 130 62553
ruediger.kreuter@siemens.com

> -----Ursprüngliche Nachricht-----
> Von: Magnus Westerlund [mailto:magnus.westerlund@era.ericsson.se]
> Gesendet: Donnerstag, 6. März 2003 16:28
> 
...
> 
> 1. Chapter 2: First sentence seems strange it looks like it lacks 
> context. "The profile .." refers to something which I don't 
> know what it 
> is.

This will be changed to "This document .." in version 03.


> 
> 2. Maybe a little more text on how data is put in the payload format. 
> Especially around the number of samples you are allowed to 
> put into each 
> packet.

Well, this chapter was copied and adapted from the PCMA/PCMU
section of draft-ietf-avt-profile-new-12.txt, because the
CLEARMODE handling is similar (simple).

I will add some of the generic payload handling descriptions of
draft-ietf-avt-profile-new-13.txt, which are relevant for
CLEARMODE too.

As PCMA/PCMU the number of samples just depends on the supported 
packet sizes, since CLEARMODE is sample oriented.


> 
> 3. End of chapter 2: "The timestamp SHALL be set according [5]"
> Shouldn't this be "The timestamp SHALL be set according to [5]."

O.k. this will be changed in version 03


> 
> 4. Chapter 3: Optional parameters. I think "maxptime" is a 
> valid one to 
> use. The existing RFC with this parameter is RFC 3267 but the updated 
> SDP draft also contains a definition of it.

Accepted, this will be added in version 03:
*******************************************
Optional parameters: ptime, maxptime
"ptime" gives the length of time in milliseconds represented 
by the media in a packet [7].
"maxptime" represents the maximum amount of media which can be 
encapsulated in each packet, expressed as time in milliseconds [10].
*******************************************

... where [10] is a new reference to:
M. Handley, V. Jacobson and C. Perkins, "SDP: Session Description 
Protocol", revision of 2327, work in progress.


> 
> 5. Chapter 3 is missing rules on how you apply the mime parameters to 
> SDP. Please see draft-ietf-avt-rtp-mime-06.txt for such a chapter.

A new chapter will be added in version 03:
******************************************
4. Mapping to SDP parameters

Parameters are mapped to SDP [7] in a standard way.
o  The MIME type (audio) goes in SDP "m=" as the media name.
o  The MIME subtype (CLEARMODE) goes in SDP "a=rtpmap"
   as the encoding name.
o  The optional parameters "ptime" and "maxptime" go in the 
   SDP "a=ptime" and "a=maxptime" attributes, respectively.

An example mapping is as follows:

        audio/CLEARMODE; ptime=10

        m=audio 12345 RTP/AVP 97
        a=rtpmap:97 CLEARMODE/8000
        a=ptime:10

Note that the payload format (encoding) names defined in the 
RTP Profile 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.
******************************************


> 
> 6. Reference list in chapter 5. Shouldn't there be nice [] around the 
> numbers?

They will be added in version 03 :-)


> 
> 7. Time to read the ID nits. You are lacking a couple of necessary 
> sections, like Copyright statement and IPR statement. See 
> http://www.ietf.org/ID-nits.html

Copyright and IPR statement will be added in version 03.
I didn't find other missing chapters.


> 
> 8. When you don't transport ISDN 7 kHz voice in this and 
> actually carry 
> some other type of data. Isn't of interest to mention RTP relevant 
> mechanisms to protect the stream against packet loss. There should be 
> significant considerations of this problem.

Since the application that uses Clearmode is unknown, it is 
hard to require an adequate method for packet loss concealment. 
So IMHO this is application dependent and therefore must be 
dealt with by methods outside the scope of this document. 

_______________________________________________
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From: Greg Herlein <gherlein@herlein.com>
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Regarding:

RTP Payload Format for the Speex Codec
(Herlein)          15

scheduled for the San Francisco meeting on Thursday, 20th March
at 09:00-11:30.

This will essentially be an introduction of Speex and the draft
profile to the WG.  I'll cover:

 - a short introduction to Speex
 - a summary of what an RTP profile must provide to Speex for use
in an RTP stream
 - a very short discussion of the things that Speex does in-band
that are not required to be handled in the RTP profile
 - a brief introduction to the draft profile itself
 - a summary of issues we already know are outstanding

Please review the draft profile at:

http://www.ietf.org/internet-drafts/draft-herlein-speex-rtp-profile-00.txt

The complete manual regarding the Speex codec is available at:

http://www.speex.org/docs.html

Thank you,

Greg Herlein









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Subject: [AVT] Agenda item RTP Payload Format for RGL Codec
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Greetings,

Below is a summary of the issues I will be discussing
at the Thursday, 20th March at 09:00-11:30 AVT time
slot concerning the following agenda item:

RTP Payload Format for RGL Codec (Ramalho) 15 min
draft-ramalho-rgl-desc-01.txt
draft-ramalho-rgl-rtpformat-01.txt

The text versions of these drafts can be found at:

http://www.ietf.org/internet-drafts/draft-ramalho-rgl-desc-01.txt
http://www.ietf.org/internet-drafts/draft-ramalho-rgl-rtpformat-01.txt

and html versions (easier to view) may be found at:

http://www.winlab.rutgers.edu/~ramalho/rgl_codec_01.html
http://www.winlab.rutgers.edu/~ramalho/rgl_rtp_01.html

1) RGL Codec draft (~5 min):

*The new RGL codec draft describes a revision to the previous
  version of the RGL codec. The reason the RGL codec was
  revised was to accommodate a simple RTP payload format for it.

* To create a Table of Contents (TOC) required for some of the
  packetization formats proposed in the RTP payload draft, the
  RGL codec was revised to create seven "reserved first RGL bytes"
  that are never produced as the first byte of a RGL frame.

* The revision was made backwardly compatible; that is, an RGL
  frame produced by the revised encoder will be successfully
  decoded by the previous version decoder.

2) The RGL RTP payload format draft (~10 min):

This draft proposes several RTP payload formats for RGL encoded
frames. When more than one RGL frame is placed in a RTP payload,
a TOC is created at the beginning of the RTP payload. This
draft proposes a few specialized TOCs and one generic one
to determine the boundaries and necessary parameters of the
RGL frame(s) contained in the RTP payload.

3) Next Steps and Q&A

Regards,

Michael Ramalho

Michael A. Ramalho, Ph.D.
Office email: mramalho@cisco.com
Personal email: mar42@cornell.edu
Office: +1.941.708.4650
Cell: +1.941.544.2844

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



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--=====================_1367241299==_.ALT
Content-Type: text/plain; charset="us-ascii"; format=flowed

Gunnar and Steve,
While RFC 2793 defines an RTP encapsulation for the text media, it does not 
clearly state that the SDP protocol name associated with the MIME text/t140 
is RTP/AVP. Can one assume that?

Of the five protocol names (SDP proto parameter values) listed by the IANA 
at http://www.iana.org/assignments/sdp-parameters, RTP/AVP does seem to be 
the most suitable. The other ones are rtp (historic), vat (historic), udptl 
and tcp. One may, of course, define a new protocol name/profile that 
includes RTP encapsulations of text, but I would advise against doing that. 
TTY devices often interleave voice and text, and it would be simpler to 
keep text/t140 under the RTP/AVP umbrella.

It would be nice if RFC 2793 has SDP examples just as RFC 2198, RFC 2733, 
RFC 2833 etc. do. But there are none.  Here is a possible example. Please 
let us know if this is what you had in mind when you registered the MIME 
type text/t140.

If an IP endpoint/gateway dedicates a UDP port to RFC 2793 text relay, the 
media line for the text relay stream is as follows:
m=text <port #> RTP/AVP <payload type>
a=rtpmap:<payload type> t140

This is consistent with the text/t140 MIME definition in RFC 2793. It maps 
an RTP payload type to the MIME text/t140.   For example,

m=text 49232 RTP/AVP 100
a=rtpmap:100 t140

indicates that the IP endpoint or gateway is willing to receive an RFC 
2793-based RTP packet stream with a payload type of 100 on port  49232.

Thanks for your guidance in thismatter.

Rajesh

--=====================_1367241299==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
Gunnar and Steve,<br>
While RFC 2793 defines an RTP encapsulation for the text media, it does
not clearly state that the SDP protocol name associated with the MIME
text/t140 is RTP/AVP. Can one assume that?<br>
<br>
Of the five protocol names (SDP <i>proto </i>parameter values) listed by
the IANA at
<a href="http://www.iana.org/assignments/sdp-parameters" eudora="autourl">http://www.iana.org/assignments/sdp-parameters</a>,
RTP/AVP does seem to be the most suitable. The other ones are rtp
(historic), vat (historic), udptl and tcp. One may, of course, define a
new protocol name/profile that includes RTP encapsulations of text, but I
would advise against doing that. TTY devices often interleave voice and
text, and it would be simpler to keep text/t140 under the RTP/AVP
umbrella.<br>
<br>
It would be nice if RFC 2793 has SDP examples just as RFC 2198, RFC 2733,
RFC 2833 etc. do. But there are none.&nbsp; Here is a possible example.
Please let us know if this is what you had in mind when you registered
the MIME type text/t140.<br>
<br>
<font face="Courier New, Courier" size=3>If an IP endpoint/gateway
dedicates a UDP port to RFC 2793 text relay, the media line for the text
relay stream is as follows:<br>
m=text &lt;port #&gt; RTP/AVP &lt;payload type&gt;<br>
a=rtpmap:&lt;payload type&gt; t140 <br>
<br>
This is consistent with the text/t140 MIME definition in RFC 2793. It
maps an RTP payload type to the MIME text/t140.&nbsp;&nbsp; For example,
<br>
&nbsp;<br>
m=text 49232 RTP/AVP 100<br>
a=rtpmap:100 t140 <br>
<br>
indicates that the IP endpoint or gateway is willing to receive an RFC
2793-based RTP packet stream with a payload type of 100 on port&nbsp;
49232.<br>
<br>
</font>Thanks for your guidance in thismatter.<br>
<br>
Rajesh <font face="Courier New, Courier" size=3>&nbsp;&nbsp; <br>
</font></html>

--=====================_1367241299==_.ALT--

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



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To: Rajesh Kumar <rkumar@cisco.com>
cc: avt@ietf.org, gunnar.hellstrom@omnitor.se, casner@acm.org,
        mario Garakani <mgarakan@cisco.com>, paulej@packetizer.com,
        bfoster@cisco.com, rmahy@cisco.com
Subject: Re: [AVT] Need clarification on protocol type for RFC 2793 text media 
In-Reply-To: Your message of "Tue, 11 Mar 2003 10:04:39 PST."
             <4.3.2.7.2.20030311095004.0160ffd8@mira-sjc5-8.cisco.com> 
Date: Tue, 11 Mar 2003 13:17:36 -0500
From: Colin Perkins <csp@isi.edu>
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Rajesh,

--> Rajesh Kumar writes:
>
>Gunnar and Steve,
>While RFC 2793 defines an RTP encapsulation for the text media, it does not 
>clearly state that the SDP protocol name associated with the MIME text/t140 
>is RTP/AVP. Can one assume that?

An RTP payload format can be used under multiple RTP profiles.  To date,
there is only one profile [RFC 1890] and that is denoted by RTP/AVP. You
almost certainly want to use this.

In the future, we have two more profiles in development:
	RTP/SAVP	draft-ietf-avt-srtp-05.txt
	RTP/AVPF	draft-ietf-avt-rtcp-feedback-05.txt
If you use either of those profiles, you would use the appropriate name to
identify them.

>Of the five protocol names (SDP proto parameter values) listed by the IANA 
>at http://www.iana.org/assignments/sdp-parameters, RTP/AVP does seem to be 
>the most suitable. The other ones are rtp (historic), vat (historic), udptl 
>and tcp. One may, of course, define a new protocol name/profile that 
>includes RTP encapsulations of text, but I would advise against doing that. 
>TTY devices often interleave voice and text, and it would be simpler to 
>keep text/t140 under the RTP/AVP umbrella.
>
>It would be nice if RFC 2793 has SDP examples just as RFC 2198, RFC 2733, 
>RFC 2833 etc. do. But there are none.  Here is a possible example. Please 
>let us know if this is what you had in mind when you registered the MIME 
>type text/t140.
>
>If an IP endpoint/gateway dedicates a UDP port to RFC 2793 text relay, the 
>media line for the text relay stream is as follows:
>m=text <port #> RTP/AVP <payload type>
>a=rtpmap:<payload type> t140
>
>This is consistent with the text/t140 MIME definition in RFC 2793. It maps 
>an RTP payload type to the MIME text/t140.   For example,
>
>m=text 49232 RTP/AVP 100
>a=rtpmap:100 t140
>
>indicates that the IP endpoint or gateway is willing to receive an RFC 
>2793-based RTP packet stream with a payload type of 100 on port  49232.

This is the standard mapping, provided you are operating under the RTP
profile defined in RFC 1890.

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



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Date: Tue, 11 Mar 2003 10:26:42 -0800 (PST)
From: Stephen Casner <casner@acm.org>
To: Rajesh Kumar <rkumar@cisco.com>
cc: avt@ietf.org, <gunnar.hellstrom@omnitor.se>,
        mario Garakani <mgarakan@cisco.com>, <paulej@packetizer.com>,
        <bfoster@cisco.com>, <rmahy@cisco.com>
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Subject: [AVT] Re: Need clarification on protocol type for RFC 2793 text media
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Rajesh,

> While RFC 2793 defines an RTP encapsulation for the text media, it does not
> clearly state that the SDP protocol name associated with the MIME text/t140
> is RTP/AVP. Can one assume that?

It is intentional that the payload format does not specify the RTP
profile under which it may be used because it may be used under
different profiles for different applications if needed.

At this moment, RTP/AVP is the only profile that has been published as
an RFC, but two more (RTP/SRTP and RTP/AVPF) are very close.

> Of the five protocol names (SDP proto parameter values) listed by the IANA
> at http://www.iana.org/assignments/sdp-parameters, RTP/AVP does seem to be
> the most suitable. The other ones are rtp (historic), vat (historic), udptl
> and tcp. One may, of course, define a new protocol name/profile that
> includes RTP encapsulations of text, but I would advise against doing that.
> TTY devices often interleave voice and text, and it would be simpler to
> keep text/t140 under the RTP/AVP umbrella.

Yes, it is appropriate to use text/t140 under rtp/avp.

> It would be nice if RFC 2793 has SDP examples just as RFC 2198, RFC 2733,
> RFC 2833 etc. do. But there are none.  Here is a possible example. Please
> let us know if this is what you had in mind when you registered the MIME
> type text/t140.

Yes, having SDP examples is handy, although not required (since using
SDP is not required).

> If an IP endpoint/gateway dedicates a UDP port to RFC 2793 text relay, the
> media line for the text relay stream is as follows:
> m=text <port #> RTP/AVP <payload type>
> a=rtpmap:<payload type> t140
>
> This is consistent with the text/t140 MIME definition in RFC 2793. It maps
> an RTP payload type to the MIME text/t140.   For example,
>
> m=text 49232 RTP/AVP 100
> a=rtpmap:100 t140
>
> indicates that the IP endpoint or gateway is willing to receive an RFC
> 2793-based RTP packet stream with a payload type of 100 on port  49232.

The mapping you have shown is valid.  It follows the established rules
in draft-ietf-mmusic-sdp-new-12.txt and draft-ietf-avt-rtp-mime-06.txt.

                                                        -- Steve

_______________________________________________
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To: philippe.gentric@philips.com
From: Stephan Wenger <stewe@cs.tu-berlin.de>
Subject: Re: [AVT] H.264 payload spec: a few remarks
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Folks,

At 02:09 PM 3/11/2003 +0100, philippe.gentric@philips.com wrote:
>[...]
>    The title(!) and the text keeps on using "JVT" while the file name, 
> the abtract and introduction use the term H.264. [...].  I think a
>    global find-and-replace(JVT->H.264) is needed!

or a change back to JVT, if MPEG endorses this draft as a way to transport 
this new video bit stream.  The inconsistency will be fixed, but IMHO it 
was not worth to spend the time before the I-D deadline.

>    The term "NALU" is defined but after that "NAL unit" is also used, 
> consistency could help.

Agreed, will fix

>There are quite a few reference problems i.e.
>       * figures and table numbers
>       * "above" used for things that are below and vice versa
>       * wrong internal section references

Agreed, will fix

>Some acronyms are not explained: SEI, IDC (also "Idc"?), DPA, DPB, DPC 
>(even FEC etc), (could be more I did not check), also I am always 
>fascinated by recursivity in acronyms: "NRI: NAL Reference IDC" (this one 
>is superb!-)

All these are either taken verbatim from the definitions section or from 
the syntax/semantic description of the H.264 spec.  Clearly, the payload 
format will not make much sense to anyone who doesn't have the payload spec 
itself in his/her hands, but isn't this true for all payload 
formats?  Nevertheless, we may add some non-normative, explanatory text 
(don't know yet how much -- is a minor issue).

With respect to the rest of this Email: A few points can be answered 
directly, without much discussions between the authors.  On others I need 
to consult first.  I removed the latter from the EMail to save space and 
reading time, and replaced them by [...].


[...]
>       Section 1.1
>[...]   Describing these NALU as being "time-less" and of having
>    "presentation time [] not defined" could cause some mis-
>    understanding, another formulation could be:
>
>    "Some NALU transport information that affect the decoding process
>    over a period of time corresponding to many video frames; For
>    these units, the handling of the RTP timestamp requires some
>    special considerations especially in cases where this duration is
>    unknown at transmission time." BTW the same remark applies in section 
> 6.1 for "If the NALU has
>    no own timing properties". You should also elaborate i.e. start
>    to present _how_ the "special considerations" are implemented in
>    the spec ...

Here in Pattaya we decided a few hours ago that, for the purpose of the 
video buffer management (compliance tests and such, has nothing to do with 
RTP), all NAL units have a picture associated with and hence carry time.  I 
heard that the MPEG systems folk use the same type of 
definition.  Following this course would be the easy way out.
However, there are NALU types which -- I continue to believe -- still 
inherently timeless.  The two parameter set NALU types, for example, are, 
once they are transmitted and unless they are overwritten, valid for the 
lifetime of the session.  We will try to add clarifying remarks to this 
section.

>    "The combination of the higher level parameters is called a Parameter 
> Set".
>    "which is a data structure akin to what is described as "decoder 
> configuration" in MPEG-4 specifications."

Not sure whether they are really "akin" (I tend to believe they are not, 
but I will talk to people here and check whether such a comparison is 
valid.  I also tend to believe that the MPEG-4 specs are not that well 
known that this remark would benefit a large user community, while it could 
guide others on the wrong path.  Finally, I would like to avoid normatively 
referencing other MPEG documents but 14496 Part 10 (as I try to avoid 
mentioning ITU Recs such as H.245, which could be used to carry Parameter 
Sets as well and has a similar concept).

>[...]   "Although, conceptually, the Parameter Set updates are not 
>designed to be sent in the synchronous packet stream, this memo contains 
>means to convey them in the RTP packet stream."  It is strange (actually 
>potentially contradictory: if they are not "designed" to be sent in 
>packets you would need to document how to map them into NALU but then why 
>the draft does not do that?) [...]

Agreed to fix.  Don't know how, yet, but this language is strange, indeed.

>[...]
>"The F bit when set to 1 signals that the NALU or header contains a syntax 
>error, in the context of this specification it may have been set by a 
>network device upon detection that a transmission error has affected the 
>corresponding data. A prudent reaction of decoders that are incapable of 
>handling bit errors is to discard such packets. "

This language, while technically correct, is politically incorrect (at 
least for the JVT spec itself).  I think what we have is good enough, but 
we will check whether informative language along the lines of your 
suggestion are possible.

[...]

>    In section 6.3.1 [...]

Miska and I are working on the complete DON and FU sections (6.3 and 6.4) 
here in Pattaya.  I agree that both sections need some serious work.

>    In Section 7.1 about the sentence:
>"Some receivers MAY NOT support a transmission order that does not conform 
>to the NAL unit decoding order. "
>By what mean the sender can find out about that?

unspecified control protocol means, or implicit (application dependent).  I 
think this is mentioned, but we will double-check.

>    In Section 10. Security Considerations:
>    What about crashing a decoder by sending a lethal Parameter Set ?

Or sending a lethal other NAL unit, e.g. an SEI that sets the video buffer 
size to 1 byte, or, or, or.  There are many ways to kill a video 
codec.  But I agree that there is work needed in the security section re 
asymmetric operation characteristics and user data.  I hope to make some 
progress on that in San Francisco

>    References:
>    for [14] I found it online at: 
> http://www.diee.unica.it/pv2000/proceedings/papers/30.pdf
>    The link http://www.pv2002.org is broken(?) but I found that paper at:
>    http://www-video.eecs.berkeley.edu/papers/sangk7/pv2002.doc

Thanks, will be added.


I want to thank you (and Magnus) again for your careful 
proof-reading.  This helps a lot.

Stephan


_______________________________________________
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Subject: Re: [AVT] H.264 payload spec: a question
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Folks,

At 02:52 PM 3/11/2003 +0100, philippe.gentric@philips.com wrote:
>I agree this bit is not extremely useful
>... and it is a better idea then to set it if all are discardable...

Ok, logical OR.  We'll change it in the next revision of the draft.

Thanks,
Stephan


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We would like to progress this draft to last call at, or shortly after, this IETF meeting.  Discussion items include:-

- Title (RTP XR or RTCP Reporting Extensions?)

- Have any IANA issues been resolved satistfactorily

- Should SDP be included in this draft to permit negotiation or should that be in a separate more general draft

- Support for sample based CODECs in VoIP metrics section

We do plan to discuss and hopefully resolve some issues on the mailing list prior to the meeting.

Alan Clark, Timur Friedman

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Hi Magnus, all,

You wrote as a summary:
 > I can see the usefulness of this proposal, however there is need to
 > resolve a number of issues:
 > - A general standardization or target certain URI schemes?
 > - Applicability on different media types and protocols needs to better
 > understood.
 > - Use of fragment or other URI syntax element, like parameter or query?
 > - Need for having ranges?
 > - The actual need for general parameters?

Great to hear you liked the idea. Many thanks for your elaborate answer 
and well thought through comments, which I am very happy to address.


1) General mechanism questions, applicability to different media types 
and protocols:

> First is the possibility to specify this as a general mechanism. It 
> seems that you want to have this to apply to all URI schemes. Is that 
> really possible? 

The question that you are addressing goes beyond whether our mechanism 
applies to all URI schemes. In 
http://www.w3.org/DesignIssues/Fragment.html Tim Berners-Lee states as 
an axiom that "The fragment ID spec for a new MIME type should be part 
of the MIME type registration process". This implies that the meaning of 
a fragment specification depends on the MIME type of the addressed 
resource. Therefore, the interpretation of a fragment has to be 
specified for each MIME type, and then probably again for each MIME type 
on each different scheme. Our I-D can only propose a generic way to 
specify time-offsets for time-continuous data and we are proposing it 
for http and rtsp where they make sense.

As an example take QuickTime files. Apple registered the MIME type 
video/quicktime but the documentation does not have a specification of 
how to interpret fragment parts when the files are used in URIs. Our I-D 
would not automatically enable timed fragment offsets to be used with 
QuickTime files - it needs an active move from Apple to change their 
MIME type specification. The same holds true for RealMedia and 
WindowsMedia and any other time-continuous MIME type. In fact, I believe 
that no MIME type registration except for text/html includes a fragment 
interpretation section.

 > How do you know that the use of @ will not collide with
> any usage?

The "@" sign is a reserved character in the URI Generic Syntax 
(RFC2396). It is allowed as part of a fragment, but its use is reserved 
for special cases. No special case of the use of "@" is know to us or 
published with IETF and therefore we asume there will be no collisions.

> For example the rtsp scheme does not define how to interpret 
> the fragment. So this is up to the server responding to a request. So 
> this can potentially already be used by some implementations.

As this has never been suggested before, there won't be any existing 
implementations that use it. Streaming servers should however find it 
easy, to support this mechanism as they only have to copy the 
specification after the "@" sign into the appropriate rtsp timestamp field.

> So a general question, what is the actually possibilities to standardize 
> such a solution?

It needs some minor adjustments in the URI standard. As that standard is 
currently being changed/adapted, this could be the right time to get it 
adjusted. Then our mechanism can be provided as a Generic Syntax 
proposal and MIME type specifications for time-continuous data can elect 
to support it. That's my take.


2) Questions on "fragment" definition, ranges:

> Second, is the usage of the fragment identifier. It seems that you tries 
> to redefine the meaning of the fragment identifier. See below.
> 
>  From RFC 2396:
> 
>  "When a URI reference is used to perform a retrieval action on the
>   identified resource, the optional fragment identifier, separated from
>   the URI by a crosshatch ("#") character, consists of additional
>   reference information to be interpreted by the user agent after the
>   retrieval action has been successfully completed.  As such, it is not
>   part of a URI, but is often used in conjunction with a URI.
> 
>      fragment      = *uric
> 
>   The semantics of a fragment identifier is a property of the data
>   resulting from a retrieval action, regardless of the type of URI used
>   in the reference.  Therefore, the format and interpretation of
>   fragment identifiers is dependent on the media type [RFC2046] of the
>   retrieval result.  The character restrictions described in Section 2
>   for URI also apply to the fragment in a URI-reference.  Individual
>   media types may define additional restrictions or structure within
>   the fragment for specifying different types of "partial views" that
>   can be identified within that media type."
> 
> Two things strikes me as not being taken into account by your draft in 
> relation to this quote. The first is from the first paragraph of the 
> quote. The fragment identifier is interpreted by the user agent. While 
> in chapter 4 of your draft you clearly specifies that the server shall 
> interpret the fragment. This redefines the behavior of the fragment 
> identifier.

We are aware that with the current URI specification, fragments only get 
interpreted on the user agent side. One way to overcome this problem is 
to use the Query part of the URI ("?") instead, but this creates other 
problems, iin the least overloading the meaning of "querying". 
Inherently, a temporal offset is a specification of a "fragment" in its 
core meaning and therefore we would prefer to use the "fragment" part. 
It makes everything much simpler. If there is no burning need to 
interpret the fragment identifier at the user agent, a change of the 
standard may be adequate. As the URI standard is currently reworked, we 
may get a chance to just do that (see URIBOF at next IETF meeting).

> Secondly the interpretation of the fragment identifier is in relation to 
> the data resulting in the request. So in fact it seems that you need to 
> define the behavior on per media basis.

Agreed.

> I think the use of fragment to specify where in a media resource is 
> appropriate in some case. For example an RTSP URL that has a fragment of 
> the proposed type is useful. However as RTSP works the server shall not 
> interpret the fragment as this is pointless. Only the client needs to do 
> this and use that to determine the range to include in the PLAY request. 
> So again the usage rules in chapter 4 seems to be wrong.

Thanks. We'll take that into account when working over it.

> It seems that the time specification should be complemented to allow 
> specifying ranges. The result of a search or the part of your friends 
> wedding video that you want to point out would in most cases only be a 
> part of the video. Therefore both a start and stop time seems useful. 
> Then it is also a very short step to having multiple ranges allow one to 
> point out multiple parts of the same resource that are interesting.

As we've had several requests for that feature, we should probably work 
that as an option into the next version of the document.

> Also I think you have a few strange thing in the definition. First you 
> specify an attribute "start" that I think are pointless. Either you 
> totally skip the fragment or you can use "npt=0". 

Possibly. But sometimes you have a file that does not start at 0 but 
rather with a time offset (because e.g. it is mirroring a timecode from 
an analog band from where it got digitized). In this case, "start" makes 
sense and does not default to "npt=0".

 > Secondly it the defaulting to "npt".
 > This clashed with the general parameters. If you
> use start then you have actually wrote "npt=start" which isn't defined. 

That's a matter of parsing sequence and probably adjustment of the BNF 
that we wrote in the document. The use of "npt" as default came from our 
experience with using the mechanism and the short form added much usability.

> "now" works as long as this is based on the RTSP definitions of NPT, 
> however from your BNF definitions this is evidently not the case. So my 
> proposal is to skip having any default time scheme if you are going to 
> have general attributes that does not belong to a time scheme.

I guess, a change of the BNF definition is more appropriate. Also, 
"start" and "now" seem to be useful for streaming and thus should be 
covered in URIs, too.


Best Regards,

Silvia.
_______________________________________________
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Hi Silvia,

First I have a question on how you are going to go forward in IETF. Are 
you planning to take this work into any WG or do a individual submission?

Then more comments below:

Silvia.Pfeiffer@csiro.au wrote:

> 1) General mechanism questions, applicability to different media types 
> and protocols:
>
>> First is the possibility to specify this as a general mechanism. It 
>> seems that you want to have this to apply to all URI schemes. Is that 
>> really possible? 
>
>
> The question that you are addressing goes beyond whether our mechanism 
> applies to all URI schemes. In 
> http://www.w3.org/DesignIssues/Fragment.html Tim Berners-Lee states as 
> an axiom that "The fragment ID spec for a new MIME type should be part 
> of the MIME type registration process". This implies that the meaning 
> of a fragment specification depends on the MIME type of the addressed 
> resource. Therefore, the interpretation of a fragment has to be 
> specified for each MIME type, and then probably again for each MIME 
> type on each different scheme. Our I-D can only propose a generic way 
> to specify time-offsets for time-continuous data and we are proposing 
> it for http and rtsp where they make sense.
>
> As an example take QuickTime files. Apple registered the MIME type 
> video/quicktime but the documentation does not have a specification of 
> how to interpret fragment parts when the files are used in URIs. Our 
> I-D would not automatically enable timed fragment offsets to be used 
> with QuickTime files - it needs an active move from Apple to change 
> their MIME type specification. The same holds true for RealMedia and 
> WindowsMedia and any other time-continuous MIME type. In fact, I 
> believe that no MIME type registration except for text/html includes a 
> fragment interpretation section.


Thanks, for the explanation, it greatly clarifies this. However to my 
understanding the usage of MIME types is actually depending on the URI 
scheme. One simple example is to compare HTTP and RTSP. In HTTP you: 1) 
download the object specified by the URI. 2) Look at the MIME type of 
the downloaded object and sends it to its parser together with the 
fragment. 3) The parser takes the fragment into account when displaying 
the content.

However for RTSP URI with a fragment one possible interpretation is to 
setup the presentation indicated by the URI. This setup results in that 
a media description is retrieved. This description has of course a MIME 
type. Is it to this description the the fragment applies? Or is it to a 
the presentation in general which does not have a MIME type, or is it to 
the media streams which can contain one or more different encodings 
perhaps identified by MIME types.

My point with this example is that the schemes seems to be taken into 
account. For RTSP it seems in fact that it is time to start defining 
some fragment identifier behavior. 

>
> > How do you know that the use of @ will not collide with
>
>> any usage?
>
>
> The "@" sign is a reserved character in the URI Generic Syntax 
> (RFC2396). It is allowed as part of a fragment, but its use is 
> reserved for special cases. No special case of the use of "@" is know 
> to us or published with IETF and therefore we assume there will be no 
> collisions.

Ok.

>
>> For example the rtsp scheme does not define how to interpret the 
>> fragment. So this is up to the server responding to a request. So 
>> this can potentially already be used by some implementations.
>
>
> As this has never been suggested before, there won't be any existing 
> implementations that use it. Streaming servers should however find it 
> easy, to support this mechanism as they only have to copy the 
> specification after the "@" sign into the appropriate rtsp timestamp 
> field.


I don't think we need to consider servers in this case, rather the 
clients as fragments should have been treated in accordance with the URI 
specification saying that fragment is a user agent problem.

>
>> So a general question, what is the actually possibilities to 
>> standardize such a solution?
>
>
> It needs some minor adjustments in the URI standard. As that standard 
> is currently being changed/adapted, this could be the right time to 
> get it adjusted. Then our mechanism can be provided as a Generic 
> Syntax proposal and MIME type specifications for time-continuous data 
> can elect to support it. That's my take.


I am not fully aware of what is happening with this. However I have seen 
a draft for making URI generic syntax to standard. Is that what you have 
in mind? To my knowledge the changes performed on a specification when 
going from draft to full standard are only clarifications. Technical 
changes are very limited if allowed at all.

I think that any such extension as you are proposing will be a proposed 
standard. 

>
>
> 2) Questions on "fragment" definition, ranges:
>
>> Second, is the usage of the fragment identifier. It seems that you 
>> tries to redefine the meaning of the fragment identifier. See below.
>>
>>  From RFC 2396:
>>
>>  "When a URI reference is used to perform a retrieval action on the
>>   identified resource, the optional fragment identifier, separated from
>>   the URI by a crosshatch ("#") character, consists of additional
>>   reference information to be interpreted by the user agent after the
>>   retrieval action has been successfully completed.  As such, it is not
>>   part of a URI, but is often used in conjunction with a URI.
>>
>>      fragment      = *uric
>>
>>   The semantics of a fragment identifier is a property of the data
>>   resulting from a retrieval action, regardless of the type of URI used
>>   in the reference.  Therefore, the format and interpretation of
>>   fragment identifiers is dependent on the media type [RFC2046] of the
>>   retrieval result.  The character restrictions described in Section 2
>>   for URI also apply to the fragment in a URI-reference.  Individual
>>   media types may define additional restrictions or structure within
>>   the fragment for specifying different types of "partial views" that
>>   can be identified within that media type."
>>
>> Two things strikes me as not being taken into account by your draft 
>> in relation to this quote. The first is from the first paragraph of 
>> the quote. The fragment identifier is interpreted by the user agent. 
>> While in chapter 4 of your draft you clearly specifies that the 
>> server shall interpret the fragment. This redefines the behavior of 
>> the fragment identifier.
>
>
> We are aware that with the current URI specification, fragments only 
> get interpreted on the user agent side. One way to overcome this 
> problem is to use the Query part of the URI ("?") instead, but this 
> creates other problems, in the least overloading the meaning of 
> "querying". Inherently, a temporal offset is a specification of a 
> "fragment" in its core meaning and therefore we would prefer to use 
> the "fragment" part. It makes everything much simpler. If there is no 
> burning need to interpret the fragment identifier at the user agent, a 
> change of the standard may be adequate. As the URI standard is 
> currently reworked, we may get a chance to just do that (see URIBOF at 
> next IETF meeting).


For me it seems very bad step to redefine this. I think one should look 
at both use cases. There will still be plenty of reasons to use fragment 
on the client side. And as there is a large number of HTTP clients out 
there using the current definition it seems a bad move to redefine it.  
However one should also look into how to address this when one wants the 
server to perform the action. I can clearly understand the use case of 
having a server reduce large media files to only contain what one is 
interested of. 

I looked through the URIBOF agenda: 
http://www.ietf.org/ietf/03mar/uribof.txt . I could however not see that 
this issue would be part of the BOF. Are you planning to take up your 
proposal there?

Best Regards

Magnus Westerlund 

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



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



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Magnus

Thanks for the comments.  Timur will respond on the non-VoIP issues:

>17. Section 4.7.4: The signal level calculations are not clear as Log10 has been separated into "log 10". 

We will correct to read "Log10"

>18. Section 4.7.7: How does one use the jitter buffer sizes when one has a sample based codec and have no defined frame size, or for that matter if the frame size is variable? 

The goal of these metrics is to indicate the mean and max delays due to the jitter buffer and also provide a means of determining if the cause of excessive discards is an incorrectly configured jitter buffer.  This can be achieved by specifying the jitter buffer characteristics in terms of milliseconds rather than packets or frames.  If we took this approach then would it be sufficient to have the mean/max jitter buffer delay range from 0 to "255 or greater" mS?

Regards

Alan Clark
Telchemy

_______________________________________________
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Hi Alan,

Alan Clark wrote:

>>18. Section 4.7.7: How does one use the jitter buffer sizes when one has a sample based codec and have no defined frame size, or for that matter if the frame size is variable? 
>>    
>>
>
>The goal of these metrics is to indicate the mean and max delays due to the jitter buffer and also provide a means of determining if the cause of excessive discards is an incorrectly configured jitter buffer.  This can be achieved by specifying the jitter buffer characteristics in terms of milliseconds rather than packets or frames.  If we took this approach then would it be sufficient to have the mean/max jitter buffer delay range from 0 to "255 or greater" mS?
>  
>

I would say no. Why not define it as 5 or 10 ms. This would give more 
substantial range.

Cheers

Magnus Westerlund 

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



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



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From: Kreuter Ruediger <ruediger.kreuter@siemens.com>
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Subject: [AVT] Summary for the Clearmode draft presentation
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Greetings,

regarding:
 
Thursday, 20th March at 09:00-11:30
===================================
RTP Payload Format for a 64 kbit/s transparent call
	(Kreuter)          15
	draft-kreuter-avt-rtp-clearmode-02.txt


This presentation will 
- give a motivation for Clearmode
- discuss the MIME Media Type for Clearmode (which is the primary intention of this draft)
- introduce a new parameter: "maxptime" that will show up in version 03 of this draft
- introduce a new SDP parameter chapter that will show up in version 03 of this draft

The current version 02 of this draft can be found at
http://www.ietf.org/internet-drafts/draft-kreuter-avt-rtp-clearmode-02.txt

For preparing the meeting you may also want to check
http://www.ietf.org/internet-drafts/draft-vainshtein-cesopsn-05.txt
which deals with TDM Circuit Emulation Service over Packet Switched Network

Best regards
Rüdiger Kreuter

_______________________________________________
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From mailnull@www1.ietf.org  Wed Mar 12 10:18:21 2003
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Proposed updated wording for Section 4.7.7

4.7.7 Jitter Buffer Parameters

   jitter buffer nominal delay in 5mS increments (JB nominal): 8 bits
      This is the current nominal delay through the jitter buffer which
      corresponds to the additional delay for packets that arrive on-time.  
      This parameter MUST be provided for both fixed and adaptive jitter
      buffer implementations.

   jitter buffer maximum delay in 5mS increments (JB Maximum): 8 bits
      This is the current maximum jitter buffer delay corresponding
      to the earliest......

   jitter buffer absolute maximum delay in 5mS increments (JB abs max): 8 bits
      This is the absolute maximum delay that an adaptive jitter buffer
      can reach under..........

Regards

Alan Clark
Telchemy  


---------- Original Message ----------------------------------
From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Date:  Wed, 12 Mar 2003 14:52:11 +0100

>Hi Alan,
>
>Alan Clark wrote:
>
>>>18. Section 4.7.7: How does one use the jitter buffer sizes when one has a sample based codec and have no defined frame size, or for that matter if the frame size is variable? 
>>>    
>>>
>>
>>The goal of these metrics is to indicate the mean and max delays due to the jitter buffer and also provide a means of determining if the cause of excessive discards is an incorrectly configured jitter buffer.  This can be achieved by specifying the jitter buffer characteristics in terms of milliseconds rather than packets or frames.  If we took this approach then would it be sufficient to have the mean/max jitter buffer delay range from 0 to "255 or greater" mS?
>>  
>>
>
>I would say no. Why not define it as 5 or 10 ms. This would give more 
>substantial range.
>
>Cheers
>
>Magnus Westerlund 
>
>Multimedia Technologies, Ericsson Research ERA/TVA/A
>----------------------------------------------------------------------
>Ericsson AB                | Phone +46 8 4048287
>Torshamsgatan 23           | Fax   +46 8 7575550
>S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
>
>
>
>
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Wed Mar 12 16:15:43 2003
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From: =?us-ascii?Q?Gunnar_Hellstrom?= <gunnar.hellstrom@omnitor.se>
To: "Stephen Casner" <casner@acm.org>, "Rajesh Kumar" <rkumar@cisco.com>
Cc: <avt@ietf.org>, "mario Garakani" <mgarakan@cisco.com>,
        <paulej@packetizer.com>, <bfoster@cisco.com>, <rmahy@cisco.com>
Subject: SV: [AVT] Re: Need clarification on protocol type for RFC 2793 text media
Date: Wed, 12 Mar 2003 22:06:46 +0100
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Rajesh,
Stephen and Colin have already given you good answers.

I can only add that it has become a habit to specify RFC2793 under the
RTP/AVP model.

It is used in

1.the examples in ITU-T H.248.2

2.In RFC 3551

http://www.ietf.org/rfc/rfc3351.txt?number=3351

3. In a transcoding draft
http://www.ietf.org/internet-drafts/draft-camarillo-sip-deaf-01.txt

Please notice that it would be a good habit to specify some redundancy. That
could imply that a suitable m= specification ends up like this:

m=text 49232 RTP/AVP 100 102
a=rtpmap:102 red
a=fmtp: 102 100/100
a=rtpmap:100 t140

Best regards
Gunnar
-------------------------------------------
Gunnar Hellstrom
Omnitor AB
Renathvagen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------

> -----Ursprungligt meddelande-----
> Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Stephen Casner
> Skickat: den 11 mars 2003 19:27
> Till: Rajesh Kumar
> Kopia: avt@ietf.org; gunnar.hellstrom@omnitor.se; mario Garakani;
> paulej@packetizer.com; bfoster@cisco.com; rmahy@cisco.com
> Amne: [AVT] Re: Need clarification on protocol type for RFC 2793 text
> media
>
>
> Rajesh,
>
> > While RFC 2793 defines an RTP encapsulation for the text media,
> it does not
> > clearly state that the SDP protocol name associated with the
> MIME text/t140
> > is RTP/AVP. Can one assume that?
>
> It is intentional that the payload format does not specify the RTP
> profile under which it may be used because it may be used under
> different profiles for different applications if needed.
>
> At this moment, RTP/AVP is the only profile that has been published as
> an RFC, but two more (RTP/SRTP and RTP/AVPF) are very close.
>
> > Of the five protocol names (SDP proto parameter values) listed
> by the IANA
> > at http://www.iana.org/assignments/sdp-parameters, RTP/AVP does
> seem to be
> > the most suitable. The other ones are rtp (historic), vat
> (historic), udptl
> > and tcp. One may, of course, define a new protocol name/profile that
> > includes RTP encapsulations of text, but I would advise against
> doing that.
> > TTY devices often interleave voice and text, and it would be simpler to
> > keep text/t140 under the RTP/AVP umbrella.
>
> Yes, it is appropriate to use text/t140 under rtp/avp.
>
> > It would be nice if RFC 2793 has SDP examples just as RFC 2198,
> RFC 2733,
> > RFC 2833 etc. do. But there are none.  Here is a possible
> example. Please
> > let us know if this is what you had in mind when you registered the MIME
> > type text/t140.
>
> Yes, having SDP examples is handy, although not required (since using
> SDP is not required).
>
> > If an IP endpoint/gateway dedicates a UDP port to RFC 2793 text
> relay, the
> > media line for the text relay stream is as follows:
> > m=text <port #> RTP/AVP <payload type>
> > a=rtpmap:<payload type> t140
> >
> > This is consistent with the text/t140 MIME definition in RFC
> 2793. It maps
> > an RTP payload type to the MIME text/t140.   For example,
> >
> > m=text 49232 RTP/AVP 100
> > a=rtpmap:100 t140
> >
> > indicates that the IP endpoint or gateway is willing to receive an RFC
> > 2793-based RTP packet stream with a payload type of 100 on port  49232.
>
> The mapping you have shown is valid.  It follows the established rules
> in draft-ietf-mmusic-sdp-new-12.txt and draft-ietf-avt-rtp-mime-06.txt.
>
>                                                         -- Steve
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
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From: Ladan Gharai <ladan@isi.edu>
Subject: [AVT] Summary for draft-ietf-avt-uncomp-video-02.txt
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The following is a list of changes made to draft
draft-ietf-avt-uncomp-video-02.txt. 

1. the standard RTP header has been extended to 32bits.
2. 4:2:0 interlaced and progressive  packetization
3. addition of  4:1:1 color coding
4. separate time stamps for interlaced fields (to accommodate 
    reversing 3:2 pulldown) 
5. definition of required and optional SDP parameters
6. text on congestion control in the security considerations.


In addition to the changes, we would appreciate feedback on the
remaining open issues:  
 - should planar video be added to the draft or does it unduly 
   complicate the draft?
 - is the SDP parameter for colorimetry sufficient?
 - should time codes have independent representation? or are the
   RTP timestamps and RTCP SR timestamps sufficient?
 

thank you,
Ladan

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Date: Thu, 13 Mar 2003 16:38:25 +0700
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From: Stephan Wenger <stewe@cs.tu-berlin.de>
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Subject: [AVT] Updated issues to discuss -- draft-ietf-avt-rtp-h264-01.txt
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Folks,
here is an updated list of the issues re the H.264/JVT packetization.  It 
reflects mostly the discussions here in Pattaya, where MPEG and JVT 
currently meets:

Unchanged items:
1. Status of JVT -- should be out for the final ballot by the time of the 
meeting.
2. Fragmentation -- we decided on adding application layer fragmentation of 
NALUs to support some exotic scenarios.  I plan to describe the pros and 
cons of the fragmentation, and hope for more guidance from the meeting re 
these feature.  We had discussed this briefly in Atlanta already, but it 
has turned out that I forgot some applications where fragmentation is useful.
3. MIME/SDP sections: are they ok?
4. Interaction with MPEG-4 simple packetization (since H.264 is also 
ISO/IEC 14496-10).  Miska and I are currently in Pattaya at the JVT/MPEG 
meeting, as are Jan Van-der-Meer, Dave Singer, Young-Kown Lim (and other 
knowledgeable people).  We will have some discussions about the interaction 
of the two drafts, and report probably to the mailing list, but certainly 
orally at the meeting, about the outcome of these discussions.
5. I currently believe that the problem mentioned in section 12 of the 
draft doesn't exist, but we will discuss this here in Pattaya as well and 
take action as necessary.
6. Security section seems to need more work, but this is not reflected in 
the draft yet.  We plan to present a summary of the problems and proposed 
solution if there is time left.

New items:
7. Interoperability between MPEG-4 simple and H.264 packetization -- as it 
has turned out, we already have one common operation point.  Now that AUs 
are syntactically defined in MPEG-4, we can add something similar to a STAP 
that adds another common operation point.  Our goal is to make this draft a 
binary-compatible superset of the MP4-Simple draft when carrying H.264, but 
it is unclear whether this can be achieved.
8. MIME/SDP codepoints to support out-of-level picture sizes -- the video 
conferencing industry needs to support extreme picture sizes (e.g. 1500 x 
1000 pels) at low frame rates, for example for document cameras.  However, 
they don't want to signal a profile/level combination that allows for this 
picture size (memory requirements), because that also implies the ability 
to decode high speed bit streams at high frame rates (computational 
requirements).  Such codepoints are in principle agreed in the ITU-T H.245 
protocol, and we are asked to add similar codepoints in the MIME/SDP.
9. I will be prepared to explain DON (why it is there, and it works), but 
we admit that there is an urgent need for clearer language for both.

Regards,
Stephan

_______________________________________________
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Subject: Re: [AVT] Summary for draft-ietf-avt-uncomp-video-02.txt
To: ladan@isi.edu
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ladan,

>In addition to the changes, we would appreciate feedback on the
>remaining open issues:
> - should planar video be added to the draft or does it unduly
>   complicate the draft?

Since I was the original proposer of this I feel it is my duty to kill it:-)

the purpose would have been to optimize the transport in terms of memory management
and data locality when processing the format (at both sender and receiver)
After analysis, this type of optimization is dependent on the application
and probably brings only marginal performance gain.

regards,


Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com

_______________________________________________
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From mailnull@www1.ietf.org  Thu Mar 13 05:21:55 2003
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From: "Even, Roni" <roni.even@polycom.co.il>
To: "'Stephan Wenger'" <stewe@cs.tu-berlin.de>, avt@ietf.org
Subject: RE: [AVT] Updated issues to discuss -- draft-ietf-avt-rtp-h264-01
	.txt
Date: Thu, 13 Mar 2003 12:22:11 +0200
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Hi,
I would like to point out that there are problems with supporting parameters
sets in multipoint video conferencing scenarios. When using out of band
signalling for the parameter sets if a video switch need to occur then there
is a need for a new parameter set for the new video stream and it has to be
available to the receiver before the actual stream.
The current work at the ITU suggests to use in band parameter sets and to
send the parameter sets before the full picture as a response to a fast
update request. (available in the RTCP feedback draft)
Regards
Roni Even

> -----Original Message-----
> From: Stephan Wenger [mailto:stewe@cs.tu-berlin.de]
> Sent: Thursday, March 13, 2003 11:38 AM
> To: avt@ietf.org
> Subject: [AVT] Updated issues to discuss -- 
> draft-ietf-avt-rtp-h264-01.txt
> 
> 
> Folks,
> here is an updated list of the issues re the H.264/JVT 
> packetization.  It 
> reflects mostly the discussions here in Pattaya, where MPEG and JVT 
> currently meets:
> 
> Unchanged items:
> 1. Status of JVT -- should be out for the final ballot by the 
> time of the 
> meeting.
> 2. Fragmentation -- we decided on adding application layer 
> fragmentation of 
> NALUs to support some exotic scenarios.  I plan to describe 
> the pros and 
> cons of the fragmentation, and hope for more guidance from 
> the meeting re 
> these feature.  We had discussed this briefly in Atlanta 
> already, but it 
> has turned out that I forgot some applications where 
> fragmentation is useful.
> 3. MIME/SDP sections: are they ok?
> 4. Interaction with MPEG-4 simple packetization (since H.264 is also 
> ISO/IEC 14496-10).  Miska and I are currently in Pattaya at 
> the JVT/MPEG 
> meeting, as are Jan Van-der-Meer, Dave Singer, Young-Kown Lim 
> (and other 
> knowledgeable people).  We will have some discussions about 
> the interaction 
> of the two drafts, and report probably to the mailing list, 
> but certainly 
> orally at the meeting, about the outcome of these discussions.
> 5. I currently believe that the problem mentioned in section 
> 12 of the 
> draft doesn't exist, but we will discuss this here in Pattaya 
> as well and 
> take action as necessary.
> 6. Security section seems to need more work, but this is not 
> reflected in 
> the draft yet.  We plan to present a summary of the problems 
> and proposed 
> solution if there is time left.
> 
> New items:
> 7. Interoperability between MPEG-4 simple and H.264 
> packetization -- as it 
> has turned out, we already have one common operation point.  
> Now that AUs 
> are syntactically defined in MPEG-4, we can add something 
> similar to a STAP 
> that adds another common operation point.  Our goal is to 
> make this draft a 
> binary-compatible superset of the MP4-Simple draft when 
> carrying H.264, but 
> it is unclear whether this can be achieved.
> 8. MIME/SDP codepoints to support out-of-level picture sizes 
> -- the video 
> conferencing industry needs to support extreme picture sizes 
> (e.g. 1500 x 
> 1000 pels) at low frame rates, for example for document 
> cameras.  However, 
> they don't want to signal a profile/level combination that 
> allows for this 
> picture size (memory requirements), because that also implies 
> the ability 
> to decode high speed bit streams at high frame rates (computational 
> requirements).  Such codepoints are in principle agreed in 
> the ITU-T H.245 
> protocol, and we are asked to add similar codepoints in the MIME/SDP.
> 9. I will be prepared to explain DON (why it is there, and it 
> works), but 
> we admit that there is an urgent need for clearer language for both.
> 
> Regards,
> Stephan
> 
> _______________________________________________
> 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 mailnull@www1.ietf.org  Thu Mar 13 06:52:32 2003
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Date: Thu, 13 Mar 2003 18:49:11 +0700
To: "Even, Roni" <roni.even@polycom.co.il>
From: Stephan Wenger <stewe@cs.tu-berlin.de>
Subject: RE: [AVT] Updated issues to discuss --
  draft-ietf-avt-rtp-h264-01 .txt
Cc: "'Stephan Wenger'" <stewe@cs.tu-berlin.de>, avt@ietf.org
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Hi Roni,

At 12:22 PM 3/13/2003 +0200, Even, Roni wrote:
>[...]
>I would like to point out that there are problems with supporting 
>parameters sets in multipoint video conferencing scenarios.
>[...]

First, please let me point out that Roni's multipoint is not the multipoint 
most of the AVT'ers would think of.  Roni speaks, I believe, primarily 
about a multipoint scenario where a central Multipoint Control Unit (MCU) 
is employed.  For those who don't know, an MCU is something similar to, but 
not quite the same as, an RTP Mixer.  It also allows for stream switching 
in the same RTP session, in addition to spatial mixing (actually, there are 
product that do switching only).

Second, I agree that there are issues with parameter sets in these 
scenarios, and I know that people in the ITU are working on those.  I am 
one of them.  If the outcome of the ITU discussions is that they will use 
in-band parameter sets over IP links, then there is indeed a need to send 
parameter sets closely aligned to the IDR pictures.  Hence, technically, 
Roni and me are in a perfect technical agreement.

However, the question is whether the RTP payload specification is the right 
place to express the mentioned need.  There are many totally valid 
examples, where RTP-based video, even when using in-band parameter sets, 
will NOT require the parameter sets being sent with the IDR picture.  And I 
think I could also come up with at least one example where it would 
actually be harmful to do so.

Also, I haven't seen similar statements in RFC2429, where the analog 
language would be that you have to send the full picture header with all 
it's extensions with every I picture (or, at least, after every Full Intra 
Request or something).

We may add an informal comment on the close relationship between parameter 
sets and IDR pictures in H.264, but more seems to be 
inappropriate.  However, I would be more than happy to learn what draft/RFC 
would be the right place (in the ITU world, IMHO the right place would be 
the system overview spec, e.g. H.320 or H.323).

Regards,
Stephan


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



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From: "Even, Roni" <roni.even@polycom.co.il>
To: "'Stephan Wenger'" <stewe@cs.tu-berlin.de>,
        "Even, Roni"
	 <roni.even@polycom.co.il>
Cc: avt@ietf.org
Subject: RE: [AVT] Updated issues to discuss -- draft-ietf-avt-rtp-h264-01
	 .txt
Date: Thu, 13 Mar 2003 13:56:45 +0200
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Stephan,
In the RTP world how will parameter sets work when a mixer is involved. It
is the same problem.
Roni

> -----Original Message-----
> From: Stephan Wenger [mailto:stewe@cs.tu-berlin.de]
> Sent: Thursday, March 13, 2003 1:49 PM
> To: Even, Roni
> Cc: 'Stephan Wenger'; avt@ietf.org
> Subject: RE: [AVT] Updated issues to discuss -- 
> draft-ietf-avt-rtp-h264-01 .txt
> 
> 
> Hi Roni,
> 
> At 12:22 PM 3/13/2003 +0200, Even, Roni wrote:
> >[...]
> >I would like to point out that there are problems with supporting 
> >parameters sets in multipoint video conferencing scenarios.
> >[...]
> 
> First, please let me point out that Roni's multipoint is not 
> the multipoint 
> most of the AVT'ers would think of.  Roni speaks, I believe, 
> primarily 
> about a multipoint scenario where a central Multipoint 
> Control Unit (MCU) 
> is employed.  For those who don't know, an MCU is something 
> similar to, but 
> not quite the same as, an RTP Mixer.  It also allows for 
> stream switching 
> in the same RTP session, in addition to spatial mixing 
> (actually, there are 
> product that do switching only).
> 
> Second, I agree that there are issues with parameter sets in these 
> scenarios, and I know that people in the ITU are working on 
> those.  I am 
> one of them.  If the outcome of the ITU discussions is that 
> they will use 
> in-band parameter sets over IP links, then there is indeed a 
> need to send 
> parameter sets closely aligned to the IDR pictures.  Hence, 
> technically, 
> Roni and me are in a perfect technical agreement.
> 
> However, the question is whether the RTP payload 
> specification is the right 
> place to express the mentioned need.  There are many totally valid 
> examples, where RTP-based video, even when using in-band 
> parameter sets, 
> will NOT require the parameter sets being sent with the IDR 
> picture.  And I 
> think I could also come up with at least one example where it would 
> actually be harmful to do so.
> 
> Also, I haven't seen similar statements in RFC2429, where the analog 
> language would be that you have to send the full picture 
> header with all 
> it's extensions with every I picture (or, at least, after 
> every Full Intra 
> Request or something).
> 
> We may add an informal comment on the close relationship 
> between parameter 
> sets and IDR pictures in H.264, but more seems to be 
> inappropriate.  However, I would be more than happy to learn 
> what draft/RFC 
> would be the right place (in the ITU world, IMHO the right 
> place would be 
> the system overview spec, e.g. H.320 or H.323).
> 
> Regards,
> Stephan
> 
> 
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Even, Roni" <roni.even@polycom.co.il>
To: "'Stephan Wenger'" <stewe@cs.tu-berlin.de>,
        "Even, Roni"
	 <roni.even@polycom.co.il>
Cc: "'avt@ietf.org'" <avt@ietf.org>
Subject: RE: [AVT] Updated issues to discuss -- draft-ietf-avt-rtp-h264-01
	 .txt
Date: Thu, 13 Mar 2003 14:07:33 +0200
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Stephan,
I do not see how H.264 work through RTP level mixers and transcoders without
sending the parameter sets in band
Roni

Stephan,
In the RTP world how will parameter sets work when a mixer is involved. It
is the same problem.
Roni

> -----Original Message-----
> From: Stephan Wenger [mailto:stewe@cs.tu-berlin.de]
> Sent: Thursday, March 13, 2003 1:49 PM
> To: Even, Roni
> Cc: 'Stephan Wenger'; avt@ietf.org
> Subject: RE: [AVT] Updated issues to discuss -- 
> draft-ietf-avt-rtp-h264-01 .txt
> 
> 
> Hi Roni,
> 
> At 12:22 PM 3/13/2003 +0200, Even, Roni wrote:
> >[...]
> >I would like to point out that there are problems with supporting 
> >parameters sets in multipoint video conferencing scenarios.
> >[...]
> 
> First, please let me point out that Roni's multipoint is not 
> the multipoint 
> most of the AVT'ers would think of.  Roni speaks, I believe, 
> primarily 
> about a multipoint scenario where a central Multipoint 
> Control Unit (MCU) 
> is employed.  For those who don't know, an MCU is something 
> similar to, but 
> not quite the same as, an RTP Mixer.  It also allows for 
> stream switching 
> in the same RTP session, in addition to spatial mixing 
> (actually, there are 
> product that do switching only).
> 
> Second, I agree that there are issues with parameter sets in these 
> scenarios, and I know that people in the ITU are working on 
> those.  I am 
> one of them.  If the outcome of the ITU discussions is that 
> they will use 
> in-band parameter sets over IP links, then there is indeed a 
> need to send 
> parameter sets closely aligned to the IDR pictures.  Hence, 
> technically, 
> Roni and me are in a perfect technical agreement.
> 
> However, the question is whether the RTP payload 
> specification is the right 
> place to express the mentioned need.  There are many totally valid 
> examples, where RTP-based video, even when using in-band 
> parameter sets, 
> will NOT require the parameter sets being sent with the IDR 
> picture.  And I 
> think I could also come up with at least one example where it would 
> actually be harmful to do so.
> 
> Also, I haven't seen similar statements in RFC2429, where the analog 
> language would be that you have to send the full picture 
> header with all 
> it's extensions with every I picture (or, at least, after 
> every Full Intra 
> Request or something).
> 
> We may add an informal comment on the close relationship 
> between parameter 
> sets and IDR pictures in H.264, but more seems to be 
> inappropriate.  However, I would be more than happy to learn 
> what draft/RFC 
> would be the right place (in the ITU world, IMHO the right 
> place would be 
> the system overview spec, e.g. H.320 or H.323).
> 
> Regards,
> Stephan
> 
> 
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] draft-ietf-avt-rtp-h264-01.txt in band PS
To: "Even, Roni" <roni.even@polycom.co.il>
Cc: avt@ietf.org, "'Stephan Wenger'" <stewe@cs.tu-berlin.de>
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Roni,

Thank you for bringing this up.

Stephan,

After reflexion the sentence in the draft:

"    Although, conceptually, the Parameter Set updates are not designed
    to be sent in the synchronous packet stream, this memo contains
    means to convey them in the RTP packet stream.   "

looks very dangerous to me.

let me explain.

*************************************************************************

H.264 has been designed with the capability to  change the configuration
dynamically and possibly very often (at every frame!) I wish it was clear for
everyone that this directly impacts the concept of  "Random Access Point",
which is very different depending on applications:

I assume you want to use the "fast update request" to send the parameter set ID(s)
of the missing PS when a client "jumps" into a stream so as to cause the sender to insert
the relevant (missing) PS in the stream.

I conclude that in your application it is acceptable to have a "black screen" while
the decoder waits for the full list of useful PS(s)....

BTW I am concerned by the possibility  that -since each single frame of the stream
may have a different PS  and that H264 supports 8k different PS for a stream-
you may spend the first minutes sending requests for missing PS(s)
and in the meanwhile display either black or a bad slide show!

but Ok, this is your application right ?

Now thinking about that made me push the reasoning one more step:
It looks to me that H264 has pushed the flexibility so far that using
"fast update request" to get PS(s) can rapidly become
more like resorting to retransmission in case of "vital" packet losses
... a bad idea usually :-(

I.e. the problem of random access generalizes to an error resilience issue,
and I have to insist that:

1) in band PS is *bad* for error resilience

2) and that this is true for *all applications* using non-reliable transport e.g. RTP

***************

The obvious *safe* alternative is to  include all the required PS(s) in the initial session setup.

***************

Anyway ...let me translate this to the VOD media distribution case.

A) a first concern is the conformance of a decoder with regard to the support of PS in band.

We need to scope it in the RTP payload  because
implementing a VOD system that supports PS inband is
*much* more difficult  and *not desirable* as follows:

B) with video on demand you would get in trouble when a media player
performs a jump and discovers it cannot decode the video
it is receiving because it misses some PS(s) that are embedded in the stream
"before" the jump !

there are 3 solutions there

* jump on "complete" PS(s) updates only (unefficient and unpractical)

* implement a dedicated trick in the streaming server so that it scans
the stream and looks up all the required PS(s) (ouch!) to include them in
some kind of "primer" in the first RTP packets (and risk to loose them!)

* do not use in band PS, put all the required PS(s) in the SDP instead!

So you understand the choice is simple: for VOD we dont want in band PS !

***********

Conclusion: I can only marginaly understand you may want to have in band PS
but there MUST be a way to make conformant systems that do not need to implement it,

therefore that sentence should be:

"This memo contains means to convey Parameter Set updates in the RTP packet stream;
but since, conceptually, these are not designed to be sent in the synchronous packet stream,
supporting in band Parameter Set transmission is OPTIONAL".

regards,


Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com


                                                                                                                                       
                                                                                                                                       
                                                   To:   "'Stephan Wenger'" <stewe@cs.tu-berlin.de>                                    
                                                    avt@ietf.org                                                                       
                                                   cc:   (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                                    
                                                   Subject:    RE: [AVT] Updated issues to discuss -- draft-ietf-avt-rtp-h264-01       
               "Even, Roni"                         .txt                                                                               
               <roni.even@polycom.co.il                                                                                                
               >                                   Classification:                                                                     
                                                                                                                                       
               Sent by:                                                                                                                
               avt-admin@ietf.org                                                                                                      
                                                                                                                                       
               13/03/2003 11:22                                                                                                        
                                                                                                                                       
                                                                                                                                       




Hi,
I would like to point out that there are problems with supporting parameters
sets in multipoint video conferencing scenarios. When using out of band
signalling for the parameter sets if a video switch need to occur then there
is a need for a new parameter set for the new video stream and it has to be
available to the receiver before the actual stream.
The current work at the ITU suggests to use in band parameter sets and to
send the parameter sets before the full picture as a response to a fast
update request. (available in the RTCP feedback draft)
Regards
Roni Even

> -----Original Message-----
> From: Stephan Wenger [mailto:stewe@cs.tu-berlin.de]
> Sent: Thursday, March 13, 2003 11:38 AM
> To: avt@ietf.org
> Subject: [AVT] Updated issues to discuss --
> draft-ietf-avt-rtp-h264-01.txt
>
>
> Folks,
> here is an updated list of the issues re the H.264/JVT
> packetization.  It
> reflects mostly the discussions here in Pattaya, where MPEG and JVT
> currently meets:
>
> Unchanged items:
> 1. Status of JVT -- should be out for the final ballot by the
> time of the
> meeting.
> 2. Fragmentation -- we decided on adding application layer
> fragmentation of
> NALUs to support some exotic scenarios.  I plan to describe
> the pros and
> cons of the fragmentation, and hope for more guidance from
> the meeting re
> these feature.  We had discussed this briefly in Atlanta
> already, but it
> has turned out that I forgot some applications where
> fragmentation is useful.
> 3. MIME/SDP sections: are they ok?
> 4. Interaction with MPEG-4 simple packetization (since H.264 is also
> ISO/IEC 14496-10).  Miska and I are currently in Pattaya at
> the JVT/MPEG
> meeting, as are Jan Van-der-Meer, Dave Singer, Young-Kown Lim
> (and other
> knowledgeable people).  We will have some discussions about
> the interaction
> of the two drafts, and report probably to the mailing list,
> but certainly
> orally at the meeting, about the outcome of these discussions.
> 5. I currently believe that the problem mentioned in section
> 12 of the
> draft doesn't exist, but we will discuss this here in Pattaya
> as well and
> take action as necessary.
> 6. Security section seems to need more work, but this is not
> reflected in
> the draft yet.  We plan to present a summary of the problems
> and proposed
> solution if there is time left.
>
> New items:
> 7. Interoperability between MPEG-4 simple and H.264
> packetization -- as it
> has turned out, we already have one common operation point.
> Now that AUs
> are syntactically defined in MPEG-4, we can add something
> similar to a STAP
> that adds another common operation point.  Our goal is to
> make this draft a
> binary-compatible superset of the MP4-Simple draft when
> carrying H.264, but
> it is unclear whether this can be achieved.
> 8. MIME/SDP codepoints to support out-of-level picture sizes
> -- the video
> conferencing industry needs to support extreme picture sizes
> (e.g. 1500 x
> 1000 pels) at low frame rates, for example for document
> cameras.  However,
> they don't want to signal a profile/level combination that
> allows for this
> picture size (memory requirements), because that also implies
> the ability
> to decode high speed bit streams at high frame rates (computational
> requirements).  Such codepoints are in principle agreed in
> the ITU-T H.245
> protocol, and we are asked to add similar codepoints in the MIME/SDP.
> 9. I will be prepared to explain DON (why it is there, and it
> works), but
> we admit that there is an urgent need for clearer language for both.
>
> Regards,
> Stephan
>
> _______________________________________________
> 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




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



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From: "Even, Roni" <roni.even@polycom.co.il>
To: "'philippe.gentric@philips.com'" <philippe.gentric@philips.com>,
        "Even, Roni" <roni.even@polycom.co.il>
Cc: avt@ietf.org, "'Stephan Wenger'" <stewe@cs.tu-berlin.de>
Subject: RE: [AVT] draft-ietf-avt-rtp-h264-01.txt in band PS
Date: Thu, 13 Mar 2003 17:15:47 +0200
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Hi Phillippe,

About your VOD example. I do not see the difference between the PS and the
full picture. If you skip frames you may need to look for some full frame if
you the new frame is not referencing a full frame that is in the decoder
buffers. In the case of Parameter Sets it is even simpler since the right
way will be to send all the parameter sets you intend to use in the
beginning of the connection between the media player client and the server.

As for error resilience I am OK with your comment.

The number of parameter sets is not the only issue but also if you have for
example a multicast group and one end point is transmitting it may have used
3 parameter sets, than a different end point is transmitting using the same
parameter sets but their order in the set is different. There is a need to
send the new parameter sets.

As for the paragraph I would like to have it as

"This memo contains means to convey Parameter Set updates in the RTP packet
stream;
supporting of in band Parameter Set transmission is application dependent.
When using in band Parameter Set, the Parameter sets should be sent as part
of the response  to an RTCP picture loss indication ".

Regards
Roni Even
 


> -----Original Message-----
> From: philippe.gentric@philips.com 
> [mailto:philippe.gentric@philips.com]
> Sent: Thursday, March 13, 2003 4:31 PM
> To: Even, Roni
> Cc: avt@ietf.org; 'Stephan Wenger'
> Subject: RE: [AVT] draft-ietf-avt-rtp-h264-01.txt in band PS
> 
> 
> Roni,
> 
> Thank you for bringing this up.
> 
> Stephan,
> 
> After reflexion the sentence in the draft:
> 
> "    Although, conceptually, the Parameter Set updates are 
> not designed
>     to be sent in the synchronous packet stream, this memo contains
>     means to convey them in the RTP packet stream.   "
> 
> looks very dangerous to me.
> 
> let me explain.
> 
> **************************************************************
> ***********
> 
> H.264 has been designed with the capability to  change the 
> configuration
> dynamically and possibly very often (at every frame!) I wish 
> it was clear for
> everyone that this directly impacts the concept of  "Random 
> Access Point",
> which is very different depending on applications:
> 
> I assume you want to use the "fast update request" to send 
> the parameter set ID(s)
> of the missing PS when a client "jumps" into a stream so as 
> to cause the sender to insert
> the relevant (missing) PS in the stream.
> 
> I conclude that in your application it is acceptable to have 
> a "black screen" while
> the decoder waits for the full list of useful PS(s)....
> 
> BTW I am concerned by the possibility  that -since each 
> single frame of the stream
> may have a different PS  and that H264 supports 8k different 
> PS for a stream-
> you may spend the first minutes sending requests for missing PS(s)
> and in the meanwhile display either black or a bad slide show!
> 
> but Ok, this is your application right ?
> 
> Now thinking about that made me push the reasoning one more step:
> It looks to me that H264 has pushed the flexibility so far that using
> "fast update request" to get PS(s) can rapidly become
> more like resorting to retransmission in case of "vital" packet losses
> ... a bad idea usually :-(
> 
> I.e. the problem of random access generalizes to an error 
> resilience issue,
> and I have to insist that:
> 
> 1) in band PS is *bad* for error resilience
> 
> 2) and that this is true for *all applications* using 
> non-reliable transport e.g. RTP
> 
> ***************
> 
> The obvious *safe* alternative is to  include all the 
> required PS(s) in the initial session setup.
> 
> ***************
> 
> Anyway ...let me translate this to the VOD media distribution case.
> 
> A) a first concern is the conformance of a decoder with 
> regard to the support of PS in band.
> 
> We need to scope it in the RTP payload  because
> implementing a VOD system that supports PS inband is
> *much* more difficult  and *not desirable* as follows:
> 
> B) with video on demand you would get in trouble when a media player
> performs a jump and discovers it cannot decode the video
> it is receiving because it misses some PS(s) that are 
> embedded in the stream
> "before" the jump !
> 
> there are 3 solutions there
> 
> * jump on "complete" PS(s) updates only (unefficient and unpractical)
> 
> * implement a dedicated trick in the streaming server so that it scans
> the stream and looks up all the required PS(s) (ouch!) to 
> include them in
> some kind of "primer" in the first RTP packets (and risk to 
> loose them!)
> 
> * do not use in band PS, put all the required PS(s) in the 
> SDP instead!
> 
> So you understand the choice is simple: for VOD we dont want 
> in band PS !
> 
> ***********
> 
> Conclusion: I can only marginaly understand you may want to 
> have in band PS
> but there MUST be a way to make conformant systems that do 
> not need to implement it,
> 
> therefore that sentence should be:
> 
> "This memo contains means to convey Parameter Set updates in 
> the RTP packet stream;
> but since, conceptually, these are not designed to be sent in 
> the synchronous packet stream,
> supporting in band Parameter Set transmission is OPTIONAL".
> 
> regards,
> 
> 
> Philippe Gentric
> Software Architect
> Philips MP4Net
> philippe.gentric@philips.com
> http://www.platform4.philips.com
> 
> 
>                                                               
>                                                               
>            
>                                                               
>                                                               
>            
>                                                    To:   
> "'Stephan Wenger'" <stewe@cs.tu-berlin.de>                    
>                 
>                                                     
> avt@ietf.org                                                  
>                      
>                                                    cc:   
> (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                    
>                 
>                                                    Subject:   
>  RE: [AVT] Updated issues to discuss -- 
> draft-ietf-avt-rtp-h264-01       
>                "Even, Roni"                         .txt      
>                                                               
>            
>                <roni.even@polycom.co.il                       
>                                                               
>            
>                >                                   
> Classification:                                               
>                       
>                                                               
>                                                               
>            
>                Sent by:                                       
>                                                               
>            
>                avt-admin@ietf.org                             
>                                                               
>            
>                                                               
>                                                               
>            
>                13/03/2003 11:22                               
>                                                               
>            
>                                                               
>                                                               
>            
>                                                               
>                                                               
>            
> 
> 
> 
> 
> Hi,
> I would like to point out that there are problems with 
> supporting parameters
> sets in multipoint video conferencing scenarios. When using 
> out of band
> signalling for the parameter sets if a video switch need to 
> occur then there
> is a need for a new parameter set for the new video stream 
> and it has to be
> available to the receiver before the actual stream.
> The current work at the ITU suggests to use in band parameter 
> sets and to
> send the parameter sets before the full picture as a response 
> to a fast
> update request. (available in the RTCP feedback draft)
> Regards
> Roni Even
> 
> > -----Original Message-----
> > From: Stephan Wenger [mailto:stewe@cs.tu-berlin.de]
> > Sent: Thursday, March 13, 2003 11:38 AM
> > To: avt@ietf.org
> > Subject: [AVT] Updated issues to discuss --
> > draft-ietf-avt-rtp-h264-01.txt
> >
> >
> > Folks,
> > here is an updated list of the issues re the H.264/JVT
> > packetization.  It
> > reflects mostly the discussions here in Pattaya, where MPEG and JVT
> > currently meets:
> >
> > Unchanged items:
> > 1. Status of JVT -- should be out for the final ballot by the
> > time of the
> > meeting.
> > 2. Fragmentation -- we decided on adding application layer
> > fragmentation of
> > NALUs to support some exotic scenarios.  I plan to describe
> > the pros and
> > cons of the fragmentation, and hope for more guidance from
> > the meeting re
> > these feature.  We had discussed this briefly in Atlanta
> > already, but it
> > has turned out that I forgot some applications where
> > fragmentation is useful.
> > 3. MIME/SDP sections: are they ok?
> > 4. Interaction with MPEG-4 simple packetization (since H.264 is also
> > ISO/IEC 14496-10).  Miska and I are currently in Pattaya at
> > the JVT/MPEG
> > meeting, as are Jan Van-der-Meer, Dave Singer, Young-Kown Lim
> > (and other
> > knowledgeable people).  We will have some discussions about
> > the interaction
> > of the two drafts, and report probably to the mailing list,
> > but certainly
> > orally at the meeting, about the outcome of these discussions.
> > 5. I currently believe that the problem mentioned in section
> > 12 of the
> > draft doesn't exist, but we will discuss this here in Pattaya
> > as well and
> > take action as necessary.
> > 6. Security section seems to need more work, but this is not
> > reflected in
> > the draft yet.  We plan to present a summary of the problems
> > and proposed
> > solution if there is time left.
> >
> > New items:
> > 7. Interoperability between MPEG-4 simple and H.264
> > packetization -- as it
> > has turned out, we already have one common operation point.
> > Now that AUs
> > are syntactically defined in MPEG-4, we can add something
> > similar to a STAP
> > that adds another common operation point.  Our goal is to
> > make this draft a
> > binary-compatible superset of the MP4-Simple draft when
> > carrying H.264, but
> > it is unclear whether this can be achieved.
> > 8. MIME/SDP codepoints to support out-of-level picture sizes
> > -- the video
> > conferencing industry needs to support extreme picture sizes
> > (e.g. 1500 x
> > 1000 pels) at low frame rates, for example for document
> > cameras.  However,
> > they don't want to signal a profile/level combination that
> > allows for this
> > picture size (memory requirements), because that also implies
> > the ability
> > to decode high speed bit streams at high frame rates (computational
> > requirements).  Such codepoints are in principle agreed in
> > the ITU-T H.245
> > protocol, and we are asked to add similar codepoints in the 
> MIME/SDP.
> > 9. I will be prepared to explain DON (why it is there, and it
> > works), but
> > we admit that there is an urgent need for clearer language for both.
> >
> > Regards,
> > Stephan
> >
> > _______________________________________________
> > 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
> 
> 
> 
> 
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] draft-ietf-avt-rtp-h264-01.txt in band PS
To: "Even, Roni" <roni.even@polycom.co.il>
Cc: avt@ietf.org, "'Stephan Wenger'" <stewe@cs.tu-berlin.de>
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Roni, you write:

> In the case of Parameter Sets it is even simpler since the right
> way will be to send all the parameter sets you intend to use in the
> beginning of the connection between the media player client and the server.

Ah but that is *exactly* the point !

I know I need to convey all the required parameter sets to my decoder *before* starting to decode

My point is that since I have 2 ways to do it:
* out of band and reliable (typically SDP  on  TCP)
or
* in band and not reliable

I prefer to do it out of band  and *reliable* of course!

All the more so since anyway I have already a session to setup
with several round trips and a SDP to exchange etc ...

How can it be better to do it in band ?

regards


Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com


                                                                                                                                       
                                                                                                                                       
                                                   To:   Philippe Gentric/MP4-SUR/CE/PHILIPS@EMEA1                                     
                                                    "Even, Roni" <roni.even@polycom.co.il>                                             
                                                   cc:   avt@ietf.org                                                                  
                                                    "'Stephan Wenger'" <stewe@cs.tu-berlin.de>                                         
               "Even, Roni"                        Subject:    RE: [AVT] draft-ietf-avt-rtp-h264-01.txt in band PS                     
               <roni.even@polycom.co.il                                                                                                
               >                                   Classification:                                                                     
                                                                                                                                       
               13/03/2003 16:15                                                                                                        
                                                                                                                                       
                                                                                                                                       




Hi Phillippe,

About your VOD example. I do not see the difference between the PS and the
full picture. If you skip frames you may need to look for some full frame if
you the new frame is not referencing a full frame that is in the decoder
buffers. In the case of Parameter Sets it is even simpler since the right
way will be to send all the parameter sets you intend to use in the
beginning of the connection between the media player client and the server.

As for error resilience I am OK with your comment.

The number of parameter sets is not the only issue but also if you have for
example a multicast group and one end point is transmitting it may have used
3 parameter sets, than a different end point is transmitting using the same
parameter sets but their order in the set is different. There is a need to
send the new parameter sets.

As for the paragraph I would like to have it as

"This memo contains means to convey Parameter Set updates in the RTP packet
stream;
supporting of in band Parameter Set transmission is application dependent.
When using in band Parameter Set, the Parameter sets should be sent as part
of the response  to an RTCP picture loss indication ".

Regards
Roni Even



> -----Original Message-----
> From: philippe.gentric@philips.com
> [mailto:philippe.gentric@philips.com]
> Sent: Thursday, March 13, 2003 4:31 PM
> To: Even, Roni
> Cc: avt@ietf.org; 'Stephan Wenger'
> Subject: RE: [AVT] draft-ietf-avt-rtp-h264-01.txt in band PS
>
>
> Roni,
>
> Thank you for bringing this up.
>
> Stephan,
>
> After reflexion the sentence in the draft:
>
> "    Although, conceptually, the Parameter Set updates are
> not designed
>     to be sent in the synchronous packet stream, this memo contains
>     means to convey them in the RTP packet stream.   "
>
> looks very dangerous to me.
>
> let me explain.
>
> **************************************************************
> ***********
>
> H.264 has been designed with the capability to  change the
> configuration
> dynamically and possibly very often (at every frame!) I wish
> it was clear for
> everyone that this directly impacts the concept of  "Random
> Access Point",
> which is very different depending on applications:
>
> I assume you want to use the "fast update request" to send
> the parameter set ID(s)
> of the missing PS when a client "jumps" into a stream so as
> to cause the sender to insert
> the relevant (missing) PS in the stream.
>
> I conclude that in your application it is acceptable to have
> a "black screen" while
> the decoder waits for the full list of useful PS(s)....
>
> BTW I am concerned by the possibility  that -since each
> single frame of the stream
> may have a different PS  and that H264 supports 8k different
> PS for a stream-
> you may spend the first minutes sending requests for missing PS(s)
> and in the meanwhile display either black or a bad slide show!
>
> but Ok, this is your application right ?
>
> Now thinking about that made me push the reasoning one more step:
> It looks to me that H264 has pushed the flexibility so far that using
> "fast update request" to get PS(s) can rapidly become
> more like resorting to retransmission in case of "vital" packet losses
> ... a bad idea usually :-(
>
> I.e. the problem of random access generalizes to an error
> resilience issue,
> and I have to insist that:
>
> 1) in band PS is *bad* for error resilience
>
> 2) and that this is true for *all applications* using
> non-reliable transport e.g. RTP
>
> ***************
>
> The obvious *safe* alternative is to  include all the
> required PS(s) in the initial session setup.
>
> ***************
>
> Anyway ...let me translate this to the VOD media distribution case.
>
> A) a first concern is the conformance of a decoder with
> regard to the support of PS in band.
>
> We need to scope it in the RTP payload  because
> implementing a VOD system that supports PS inband is
> *much* more difficult  and *not desirable* as follows:
>
> B) with video on demand you would get in trouble when a media player
> performs a jump and discovers it cannot decode the video
> it is receiving because it misses some PS(s) that are
> embedded in the stream
> "before" the jump !
>
> there are 3 solutions there
>
> * jump on "complete" PS(s) updates only (unefficient and unpractical)
>
> * implement a dedicated trick in the streaming server so that it scans
> the stream and looks up all the required PS(s) (ouch!) to
> include them in
> some kind of "primer" in the first RTP packets (and risk to
> loose them!)
>
> * do not use in band PS, put all the required PS(s) in the
> SDP instead!
>
> So you understand the choice is simple: for VOD we dont want
> in band PS !
>
> ***********
>
> Conclusion: I can only marginaly understand you may want to
> have in band PS
> but there MUST be a way to make conformant systems that do
> not need to implement it,
>
> therefore that sentence should be:
>
> "This memo contains means to convey Parameter Set updates in
> the RTP packet stream;
> but since, conceptually, these are not designed to be sent in
> the synchronous packet stream,
> supporting in band Parameter Set transmission is OPTIONAL".
>
> regards,
>
>
> Philippe Gentric
> Software Architect
> Philips MP4Net
> philippe.gentric@philips.com
> http://www.platform4.philips.com
>
>
>
>
>
>
>
>
>                                                    To:
> "'Stephan Wenger'" <stewe@cs.tu-berlin.de>
>
>
> avt@ietf.org
>
>                                                    cc:
> (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)
>
>                                                    Subject:
>  RE: [AVT] Updated issues to discuss --
> draft-ietf-avt-rtp-h264-01
>                "Even, Roni"                         .txt
>
>
>                <roni.even@polycom.co.il
>
>
>                >
> Classification:
>
>
>
>
>                Sent by:
>
>
>                avt-admin@ietf.org
>
>
>
>
>
>                13/03/2003 11:22
>
>
>
>
>
>
>
>
>
>
>
>
> Hi,
> I would like to point out that there are problems with
> supporting parameters
> sets in multipoint video conferencing scenarios. When using
> out of band
> signalling for the parameter sets if a video switch need to
> occur then there
> is a need for a new parameter set for the new video stream
> and it has to be
> available to the receiver before the actual stream.
> The current work at the ITU suggests to use in band parameter
> sets and to
> send the parameter sets before the full picture as a response
> to a fast
> update request. (available in the RTCP feedback draft)
> Regards
> Roni Even
>
> > -----Original Message-----
> > From: Stephan Wenger [mailto:stewe@cs.tu-berlin.de]
> > Sent: Thursday, March 13, 2003 11:38 AM
> > To: avt@ietf.org
> > Subject: [AVT] Updated issues to discuss --
> > draft-ietf-avt-rtp-h264-01.txt
> >
> >
> > Folks,
> > here is an updated list of the issues re the H.264/JVT
> > packetization.  It
> > reflects mostly the discussions here in Pattaya, where MPEG and JVT
> > currently meets:
> >
> > Unchanged items:
> > 1. Status of JVT -- should be out for the final ballot by the
> > time of the
> > meeting.
> > 2. Fragmentation -- we decided on adding application layer
> > fragmentation of
> > NALUs to support some exotic scenarios.  I plan to describe
> > the pros and
> > cons of the fragmentation, and hope for more guidance from
> > the meeting re
> > these feature.  We had discussed this briefly in Atlanta
> > already, but it
> > has turned out that I forgot some applications where
> > fragmentation is useful.
> > 3. MIME/SDP sections: are they ok?
> > 4. Interaction with MPEG-4 simple packetization (since H.264 is also
> > ISO/IEC 14496-10).  Miska and I are currently in Pattaya at
> > the JVT/MPEG
> > meeting, as are Jan Van-der-Meer, Dave Singer, Young-Kown Lim
> > (and other
> > knowledgeable people).  We will have some discussions about
> > the interaction
> > of the two drafts, and report probably to the mailing list,
> > but certainly
> > orally at the meeting, about the outcome of these discussions.
> > 5. I currently believe that the problem mentioned in section
> > 12 of the
> > draft doesn't exist, but we will discuss this here in Pattaya
> > as well and
> > take action as necessary.
> > 6. Security section seems to need more work, but this is not
> > reflected in
> > the draft yet.  We plan to present a summary of the problems
> > and proposed
> > solution if there is time left.
> >
> > New items:
> > 7. Interoperability between MPEG-4 simple and H.264
> > packetization -- as it
> > has turned out, we already have one common operation point.
> > Now that AUs
> > are syntactically defined in MPEG-4, we can add something
> > similar to a STAP
> > that adds another common operation point.  Our goal is to
> > make this draft a
> > binary-compatible superset of the MP4-Simple draft when
> > carrying H.264, but
> > it is unclear whether this can be achieved.
> > 8. MIME/SDP codepoints to support out-of-level picture sizes
> > -- the video
> > conferencing industry needs to support extreme picture sizes
> > (e.g. 1500 x
> > 1000 pels) at low frame rates, for example for document
> > cameras.  However,
> > they don't want to signal a profile/level combination that
> > allows for this
> > picture size (memory requirements), because that also implies
> > the ability
> > to decode high speed bit streams at high frame rates (computational
> > requirements).  Such codepoints are in principle agreed in
> > the ITU-T H.245
> > protocol, and we are asked to add similar codepoints in the
> MIME/SDP.
> > 9. I will be prepared to explain DON (why it is there, and it
> > works), but
> > we admit that there is an urgent need for clearer language for both.
> >
> > Regards,
> > Stephan
> >
> > _______________________________________________
> > 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
>
>
>
>




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



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Subject: RE: [AVT] draft-ietf-avt-rtp-h264-01.txt in band PS
To: "Even, Roni" <roni.even@polycom.co.il>
Cc: avt@ietf.org, "'Stephan Wenger'" <stewe@cs.tu-berlin.de>
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Roni,

you write:

"This memo contains means to convey Parameter Set updates in the RTP packet
stream;
supporting of in band Parameter Set transmission is application dependent.
When using in band Parameter Set, the Parameter sets should be sent as part
of the response  to an RTCP picture loss indication ".

Hum.

You imply that the server should send all the "required" Parameters Sets,
it would be better if it was explicit.

anyway this does not solve my problem which is that I dont want
to be forced to implement it in my VOD servers and clients
and I prefer some clear normative text for it!


regards,


Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com


                                                                                                                                       
                                                                                                                                       
                                                   To:   Philippe Gentric/MP4-SUR/CE/PHILIPS@EMEA1                                     
                                                    "Even, Roni" <roni.even@polycom.co.il>                                             
                                                   cc:   avt@ietf.org                                                                  
                                                    "'Stephan Wenger'" <stewe@cs.tu-berlin.de>                                         
               "Even, Roni"                        Subject:    RE: [AVT] draft-ietf-avt-rtp-h264-01.txt in band PS                     
               <roni.even@polycom.co.il                                                                                                
               >                                   Classification:                                                                     
                                                                                                                                       
               13/03/2003 16:15                                                                                                        
                                                                                                                                       
                                                                                                                                       




Hi Phillippe,

About your VOD example. I do not see the difference between the PS and the
full picture. If you skip frames you may need to look for some full frame if
you the new frame is not referencing a full frame that is in the decoder
buffers. In the case of Parameter Sets it is even simpler since the right
way will be to send all the parameter sets you intend to use in the
beginning of the connection between the media player client and the server.

As for error resilience I am OK with your comment.

The number of parameter sets is not the only issue but also if you have for
example a multicast group and one end point is transmitting it may have used
3 parameter sets, than a different end point is transmitting using the same
parameter sets but their order in the set is different. There is a need to
send the new parameter sets.

As for the paragraph I would like to have it as

"This memo contains means to convey Parameter Set updates in the RTP packet
stream;
supporting of in band Parameter Set transmission is application dependent.
When using in band Parameter Set, the Parameter sets should be sent as part
of the response  to an RTCP picture loss indication ".

Regards
Roni Even



> -----Original Message-----
> From: philippe.gentric@philips.com
> [mailto:philippe.gentric@philips.com]
> Sent: Thursday, March 13, 2003 4:31 PM
> To: Even, Roni
> Cc: avt@ietf.org; 'Stephan Wenger'
> Subject: RE: [AVT] draft-ietf-avt-rtp-h264-01.txt in band PS
>
>
> Roni,
>
> Thank you for bringing this up.
>
> Stephan,
>
> After reflexion the sentence in the draft:
>
> "    Although, conceptually, the Parameter Set updates are
> not designed
>     to be sent in the synchronous packet stream, this memo contains
>     means to convey them in the RTP packet stream.   "
>
> looks very dangerous to me.
>
> let me explain.
>
> **************************************************************
> ***********
>
> H.264 has been designed with the capability to  change the
> configuration
> dynamically and possibly very often (at every frame!) I wish
> it was clear for
> everyone that this directly impacts the concept of  "Random
> Access Point",
> which is very different depending on applications:
>
> I assume you want to use the "fast update request" to send
> the parameter set ID(s)
> of the missing PS when a client "jumps" into a stream so as
> to cause the sender to insert
> the relevant (missing) PS in the stream.
>
> I conclude that in your application it is acceptable to have
> a "black screen" while
> the decoder waits for the full list of useful PS(s)....
>
> BTW I am concerned by the possibility  that -since each
> single frame of the stream
> may have a different PS  and that H264 supports 8k different
> PS for a stream-
> you may spend the first minutes sending requests for missing PS(s)
> and in the meanwhile display either black or a bad slide show!
>
> but Ok, this is your application right ?
>
> Now thinking about that made me push the reasoning one more step:
> It looks to me that H264 has pushed the flexibility so far that using
> "fast update request" to get PS(s) can rapidly become
> more like resorting to retransmission in case of "vital" packet losses
> ... a bad idea usually :-(
>
> I.e. the problem of random access generalizes to an error
> resilience issue,
> and I have to insist that:
>
> 1) in band PS is *bad* for error resilience
>
> 2) and that this is true for *all applications* using
> non-reliable transport e.g. RTP
>
> ***************
>
> The obvious *safe* alternative is to  include all the
> required PS(s) in the initial session setup.
>
> ***************
>
> Anyway ...let me translate this to the VOD media distribution case.
>
> A) a first concern is the conformance of a decoder with
> regard to the support of PS in band.
>
> We need to scope it in the RTP payload  because
> implementing a VOD system that supports PS inband is
> *much* more difficult  and *not desirable* as follows:
>
> B) with video on demand you would get in trouble when a media player
> performs a jump and discovers it cannot decode the video
> it is receiving because it misses some PS(s) that are
> embedded in the stream
> "before" the jump !
>
> there are 3 solutions there
>
> * jump on "complete" PS(s) updates only (unefficient and unpractical)
>
> * implement a dedicated trick in the streaming server so that it scans
> the stream and looks up all the required PS(s) (ouch!) to
> include them in
> some kind of "primer" in the first RTP packets (and risk to
> loose them!)
>
> * do not use in band PS, put all the required PS(s) in the
> SDP instead!
>
> So you understand the choice is simple: for VOD we dont want
> in band PS !
>
> ***********
>
> Conclusion: I can only marginaly understand you may want to
> have in band PS
> but there MUST be a way to make conformant systems that do
> not need to implement it,
>
> therefore that sentence should be:
>
> "This memo contains means to convey Parameter Set updates in
> the RTP packet stream;
> but since, conceptually, these are not designed to be sent in
> the synchronous packet stream,
> supporting in band Parameter Set transmission is OPTIONAL".
>
> regards,
>
>
> Philippe Gentric
> Software Architect
> Philips MP4Net
> philippe.gentric@philips.com
> http://www.platform4.philips.com
>
>
>
>
>
>
>
>
>                                                    To:
> "'Stephan Wenger'" <stewe@cs.tu-berlin.de>
>
>
> avt@ietf.org
>
>                                                    cc:
> (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)
>
>                                                    Subject:
>  RE: [AVT] Updated issues to discuss --
> draft-ietf-avt-rtp-h264-01
>                "Even, Roni"                         .txt
>
>
>                <roni.even@polycom.co.il
>
>
>                >
> Classification:
>
>
>
>
>                Sent by:
>
>
>                avt-admin@ietf.org
>
>
>
>
>
>                13/03/2003 11:22
>
>
>
>
>
>
>
>
>
>
>
>
> Hi,
> I would like to point out that there are problems with
> supporting parameters
> sets in multipoint video conferencing scenarios. When using
> out of band
> signalling for the parameter sets if a video switch need to
> occur then there
> is a need for a new parameter set for the new video stream
> and it has to be
> available to the receiver before the actual stream.
> The current work at the ITU suggests to use in band parameter
> sets and to
> send the parameter sets before the full picture as a response
> to a fast
> update request. (available in the RTCP feedback draft)
> Regards
> Roni Even
>
> > -----Original Message-----
> > From: Stephan Wenger [mailto:stewe@cs.tu-berlin.de]
> > Sent: Thursday, March 13, 2003 11:38 AM
> > To: avt@ietf.org
> > Subject: [AVT] Updated issues to discuss --
> > draft-ietf-avt-rtp-h264-01.txt
> >
> >
> > Folks,
> > here is an updated list of the issues re the H.264/JVT
> > packetization.  It
> > reflects mostly the discussions here in Pattaya, where MPEG and JVT
> > currently meets:
> >
> > Unchanged items:
> > 1. Status of JVT -- should be out for the final ballot by the
> > time of the
> > meeting.
> > 2. Fragmentation -- we decided on adding application layer
> > fragmentation of
> > NALUs to support some exotic scenarios.  I plan to describe
> > the pros and
> > cons of the fragmentation, and hope for more guidance from
> > the meeting re
> > these feature.  We had discussed this briefly in Atlanta
> > already, but it
> > has turned out that I forgot some applications where
> > fragmentation is useful.
> > 3. MIME/SDP sections: are they ok?
> > 4. Interaction with MPEG-4 simple packetization (since H.264 is also
> > ISO/IEC 14496-10).  Miska and I are currently in Pattaya at
> > the JVT/MPEG
> > meeting, as are Jan Van-der-Meer, Dave Singer, Young-Kown Lim
> > (and other
> > knowledgeable people).  We will have some discussions about
> > the interaction
> > of the two drafts, and report probably to the mailing list,
> > but certainly
> > orally at the meeting, about the outcome of these discussions.
> > 5. I currently believe that the problem mentioned in section
> > 12 of the
> > draft doesn't exist, but we will discuss this here in Pattaya
> > as well and
> > take action as necessary.
> > 6. Security section seems to need more work, but this is not
> > reflected in
> > the draft yet.  We plan to present a summary of the problems
> > and proposed
> > solution if there is time left.
> >
> > New items:
> > 7. Interoperability between MPEG-4 simple and H.264
> > packetization -- as it
> > has turned out, we already have one common operation point.
> > Now that AUs
> > are syntactically defined in MPEG-4, we can add something
> > similar to a STAP
> > that adds another common operation point.  Our goal is to
> > make this draft a
> > binary-compatible superset of the MP4-Simple draft when
> > carrying H.264, but
> > it is unclear whether this can be achieved.
> > 8. MIME/SDP codepoints to support out-of-level picture sizes
> > -- the video
> > conferencing industry needs to support extreme picture sizes
> > (e.g. 1500 x
> > 1000 pels) at low frame rates, for example for document
> > cameras.  However,
> > they don't want to signal a profile/level combination that
> > allows for this
> > picture size (memory requirements), because that also implies
> > the ability
> > to decode high speed bit streams at high frame rates (computational
> > requirements).  Such codepoints are in principle agreed in
> > the ITU-T H.245
> > protocol, and we are asked to add similar codepoints in the
> MIME/SDP.
> > 9. I will be prepared to explain DON (why it is there, and it
> > works), but
> > we admit that there is an urgent need for clearer language for both.
> >
> > Regards,
> > Stephan
> >
> > _______________________________________________
> > 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
>
>
>
>




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



From mailnull@www1.ietf.org  Thu Mar 13 10:48:02 2003
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From: "Even, Roni" <roni.even@polycom.co.il>
To: "'philippe.gentric@philips.com'" <philippe.gentric@philips.com>,
        "Even, Roni" <roni.even@polycom.co.il>
Cc: avt@ietf.org, "'Stephan Wenger'" <stewe@cs.tu-berlin.de>
Subject: RE: [AVT] draft-ietf-avt-rtp-h264-01.txt in band PS
Date: Thu, 13 Mar 2003 17:48:30 +0200
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Phillipe,
In your case it may be better to use reliable protocol to send the parameter
set. The problem is with different applications that may switch the RTP
source during the call. Sending all the parameter sets in the beginning may
be difficult since each source may want to use the same parameter set
numbers with different values so some mechanism to allocate the numbers is
needed. The other option is to use out of band signalling with each switch
of source. If you use out of band in this case it will be long and you will
need to synchronize so that the sender will not send before the receiver
arrives. Using in band non reliable has lower latency and losing the PS is
the same as losing the full picture that you need from the new source so you
need to protect both if error resilience is a concern

Roni

> -----Original Message-----
> From: philippe.gentric@philips.com 
> [mailto:philippe.gentric@philips.com]
> Sent: Thursday, March 13, 2003 5:25 PM
> To: Even, Roni
> Cc: avt@ietf.org; 'Stephan Wenger'
> Subject: RE: [AVT] draft-ietf-avt-rtp-h264-01.txt in band PS
> 
> 
> 
> 
> Roni, you write:
> 
> > In the case of Parameter Sets it is even simpler since the right
> > way will be to send all the parameter sets you intend to use in the
> > beginning of the connection between the media player client 
> and the server.
> 
> Ah but that is *exactly* the point !
> 
> I know I need to convey all the required parameter sets to my 
> decoder *before* starting to decode
> 
> My point is that since I have 2 ways to do it:
> * out of band and reliable (typically SDP  on  TCP)
> or
> * in band and not reliable
> 
> I prefer to do it out of band  and *reliable* of course!
> 
> All the more so since anyway I have already a session to setup
> with several round trips and a SDP to exchange etc ...
> 
> How can it be better to do it in band ?
> 
> regards
> 
> 
> Philippe Gentric
> Software Architect
> Philips MP4Net
> philippe.gentric@philips.com
> http://www.platform4.philips.com
> 
> 
>                                                               
>                                                               
>            
>                                                               
>                                                               
>            
>                                                    To:   
> Philippe Gentric/MP4-SUR/CE/PHILIPS@EMEA1                     
>                 
>                                                     "Even, 
> Roni" <roni.even@polycom.co.il>                               
>               
>                                                    cc:   
> avt@ietf.org                                                  
>                 
>                                                     "'Stephan 
> Wenger'" <stewe@cs.tu-berlin.de>                              
>            
>                "Even, Roni"                        Subject:   
>  RE: [AVT] draft-ietf-avt-rtp-h264-01.txt in band PS          
>            
>                <roni.even@polycom.co.il                       
>                                                               
>            
>                >                                   
> Classification:                                               
>                       
>                                                               
>                                                               
>            
>                13/03/2003 16:15                               
>                                                               
>            
>                                                               
>                                                               
>            
>                                                               
>                                                               
>            
> 
> 
> 
> 
> Hi Phillippe,
> 
> About your VOD example. I do not see the difference between 
> the PS and the
> full picture. If you skip frames you may need to look for 
> some full frame if
> you the new frame is not referencing a full frame that is in 
> the decoder
> buffers. In the case of Parameter Sets it is even simpler 
> since the right
> way will be to send all the parameter sets you intend to use in the
> beginning of the connection between the media player client 
> and the server.
> 
> As for error resilience I am OK with your comment.
> 
> The number of parameter sets is not the only issue but also 
> if you have for
> example a multicast group and one end point is transmitting 
> it may have used
> 3 parameter sets, than a different end point is transmitting 
> using the same
> parameter sets but their order in the set is different. There 
> is a need to
> send the new parameter sets.
> 
> As for the paragraph I would like to have it as
> 
> "This memo contains means to convey Parameter Set updates in 
> the RTP packet
> stream;
> supporting of in band Parameter Set transmission is 
> application dependent.
> When using in band Parameter Set, the Parameter sets should 
> be sent as part
> of the response  to an RTCP picture loss indication ".
> 
> Regards
> Roni Even
> 
> 
> 
> > -----Original Message-----
> > From: philippe.gentric@philips.com
> > [mailto:philippe.gentric@philips.com]
> > Sent: Thursday, March 13, 2003 4:31 PM
> > To: Even, Roni
> > Cc: avt@ietf.org; 'Stephan Wenger'
> > Subject: RE: [AVT] draft-ietf-avt-rtp-h264-01.txt in band PS
> >
> >
> > Roni,
> >
> > Thank you for bringing this up.
> >
> > Stephan,
> >
> > After reflexion the sentence in the draft:
> >
> > "    Although, conceptually, the Parameter Set updates are
> > not designed
> >     to be sent in the synchronous packet stream, this memo contains
> >     means to convey them in the RTP packet stream.   "
> >
> > looks very dangerous to me.
> >
> > let me explain.
> >
> > **************************************************************
> > ***********
> >
> > H.264 has been designed with the capability to  change the
> > configuration
> > dynamically and possibly very often (at every frame!) I wish
> > it was clear for
> > everyone that this directly impacts the concept of  "Random
> > Access Point",
> > which is very different depending on applications:
> >
> > I assume you want to use the "fast update request" to send
> > the parameter set ID(s)
> > of the missing PS when a client "jumps" into a stream so as
> > to cause the sender to insert
> > the relevant (missing) PS in the stream.
> >
> > I conclude that in your application it is acceptable to have
> > a "black screen" while
> > the decoder waits for the full list of useful PS(s)....
> >
> > BTW I am concerned by the possibility  that -since each
> > single frame of the stream
> > may have a different PS  and that H264 supports 8k different
> > PS for a stream-
> > you may spend the first minutes sending requests for missing PS(s)
> > and in the meanwhile display either black or a bad slide show!
> >
> > but Ok, this is your application right ?
> >
> > Now thinking about that made me push the reasoning one more step:
> > It looks to me that H264 has pushed the flexibility so far 
> that using
> > "fast update request" to get PS(s) can rapidly become
> > more like resorting to retransmission in case of "vital" 
> packet losses
> > ... a bad idea usually :-(
> >
> > I.e. the problem of random access generalizes to an error
> > resilience issue,
> > and I have to insist that:
> >
> > 1) in band PS is *bad* for error resilience
> >
> > 2) and that this is true for *all applications* using
> > non-reliable transport e.g. RTP
> >
> > ***************
> >
> > The obvious *safe* alternative is to  include all the
> > required PS(s) in the initial session setup.
> >
> > ***************
> >
> > Anyway ...let me translate this to the VOD media distribution case.
> >
> > A) a first concern is the conformance of a decoder with
> > regard to the support of PS in band.
> >
> > We need to scope it in the RTP payload  because
> > implementing a VOD system that supports PS inband is
> > *much* more difficult  and *not desirable* as follows:
> >
> > B) with video on demand you would get in trouble when a media player
> > performs a jump and discovers it cannot decode the video
> > it is receiving because it misses some PS(s) that are
> > embedded in the stream
> > "before" the jump !
> >
> > there are 3 solutions there
> >
> > * jump on "complete" PS(s) updates only (unefficient and 
> unpractical)
> >
> > * implement a dedicated trick in the streaming server so 
> that it scans
> > the stream and looks up all the required PS(s) (ouch!) to
> > include them in
> > some kind of "primer" in the first RTP packets (and risk to
> > loose them!)
> >
> > * do not use in band PS, put all the required PS(s) in the
> > SDP instead!
> >
> > So you understand the choice is simple: for VOD we dont want
> > in band PS !
> >
> > ***********
> >
> > Conclusion: I can only marginaly understand you may want to
> > have in band PS
> > but there MUST be a way to make conformant systems that do
> > not need to implement it,
> >
> > therefore that sentence should be:
> >
> > "This memo contains means to convey Parameter Set updates in
> > the RTP packet stream;
> > but since, conceptually, these are not designed to be sent in
> > the synchronous packet stream,
> > supporting in band Parameter Set transmission is OPTIONAL".
> >
> > regards,
> >
> >
> > Philippe Gentric
> > Software Architect
> > Philips MP4Net
> > philippe.gentric@philips.com
> > http://www.platform4.philips.com
> >
> >
> >
> >
> >
> >
> >
> >
> >                                                    To:
> > "'Stephan Wenger'" <stewe@cs.tu-berlin.de>
> >
> >
> > avt@ietf.org
> >
> >                                                    cc:
> > (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)
> >
> >                                                    Subject:
> >  RE: [AVT] Updated issues to discuss --
> > draft-ietf-avt-rtp-h264-01
> >                "Even, Roni"                         .txt
> >
> >
> >                <roni.even@polycom.co.il
> >
> >
> >                >
> > Classification:
> >
> >
> >
> >
> >                Sent by:
> >
> >
> >                avt-admin@ietf.org
> >
> >
> >
> >
> >
> >                13/03/2003 11:22
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > Hi,
> > I would like to point out that there are problems with
> > supporting parameters
> > sets in multipoint video conferencing scenarios. When using
> > out of band
> > signalling for the parameter sets if a video switch need to
> > occur then there
> > is a need for a new parameter set for the new video stream
> > and it has to be
> > available to the receiver before the actual stream.
> > The current work at the ITU suggests to use in band parameter
> > sets and to
> > send the parameter sets before the full picture as a response
> > to a fast
> > update request. (available in the RTCP feedback draft)
> > Regards
> > Roni Even
> >
> > > -----Original Message-----
> > > From: Stephan Wenger [mailto:stewe@cs.tu-berlin.de]
> > > Sent: Thursday, March 13, 2003 11:38 AM
> > > To: avt@ietf.org
> > > Subject: [AVT] Updated issues to discuss --
> > > draft-ietf-avt-rtp-h264-01.txt
> > >
> > >
> > > Folks,
> > > here is an updated list of the issues re the H.264/JVT
> > > packetization.  It
> > > reflects mostly the discussions here in Pattaya, where 
> MPEG and JVT
> > > currently meets:
> > >
> > > Unchanged items:
> > > 1. Status of JVT -- should be out for the final ballot by the
> > > time of the
> > > meeting.
> > > 2. Fragmentation -- we decided on adding application layer
> > > fragmentation of
> > > NALUs to support some exotic scenarios.  I plan to describe
> > > the pros and
> > > cons of the fragmentation, and hope for more guidance from
> > > the meeting re
> > > these feature.  We had discussed this briefly in Atlanta
> > > already, but it
> > > has turned out that I forgot some applications where
> > > fragmentation is useful.
> > > 3. MIME/SDP sections: are they ok?
> > > 4. Interaction with MPEG-4 simple packetization (since 
> H.264 is also
> > > ISO/IEC 14496-10).  Miska and I are currently in Pattaya at
> > > the JVT/MPEG
> > > meeting, as are Jan Van-der-Meer, Dave Singer, Young-Kown Lim
> > > (and other
> > > knowledgeable people).  We will have some discussions about
> > > the interaction
> > > of the two drafts, and report probably to the mailing list,
> > > but certainly
> > > orally at the meeting, about the outcome of these discussions.
> > > 5. I currently believe that the problem mentioned in section
> > > 12 of the
> > > draft doesn't exist, but we will discuss this here in Pattaya
> > > as well and
> > > take action as necessary.
> > > 6. Security section seems to need more work, but this is not
> > > reflected in
> > > the draft yet.  We plan to present a summary of the problems
> > > and proposed
> > > solution if there is time left.
> > >
> > > New items:
> > > 7. Interoperability between MPEG-4 simple and H.264
> > > packetization -- as it
> > > has turned out, we already have one common operation point.
> > > Now that AUs
> > > are syntactically defined in MPEG-4, we can add something
> > > similar to a STAP
> > > that adds another common operation point.  Our goal is to
> > > make this draft a
> > > binary-compatible superset of the MP4-Simple draft when
> > > carrying H.264, but
> > > it is unclear whether this can be achieved.
> > > 8. MIME/SDP codepoints to support out-of-level picture sizes
> > > -- the video
> > > conferencing industry needs to support extreme picture sizes
> > > (e.g. 1500 x
> > > 1000 pels) at low frame rates, for example for document
> > > cameras.  However,
> > > they don't want to signal a profile/level combination that
> > > allows for this
> > > picture size (memory requirements), because that also implies
> > > the ability
> > > to decode high speed bit streams at high frame rates 
> (computational
> > > requirements).  Such codepoints are in principle agreed in
> > > the ITU-T H.245
> > > protocol, and we are asked to add similar codepoints in the
> > MIME/SDP.
> > > 9. I will be prepared to explain DON (why it is there, and it
> > > works), but
> > > we admit that there is an urgent need for clearer 
> language for both.
> > >
> > > Regards,
> > > Stephan
> > >
> > > _______________________________________________
> > > 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
> >
> >
> >
> >
> 
> 
> 
> 
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Colin Perkins <csp@isi.edu>
Subject: [AVT] Updated AVT agenda
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Audio/Video Transport Working Group

CHAIRS: Steve Casner <casner@acm.org>
Colin Perkins <csp@csperkins.org>


Wednesday, 19th March at 13:00-15:00
====================================

Introduction and status update				(Casner/Perkins)   15
draft-ietf-avt-rtp-new-12.txt      (Approved!)
draft-ietf-avt-profile-new-13.txt  (Approved!)
draft-ietf-avt-uxp-05.txt
draft-ietf-avt-ulp-07.txt          (Under revision)
draft-ietf-avt-tcrtp-07.txt
draft-ietf-avt-rtp-retransmission-06.txt
draft-ietf-avt-rtcp-feedback-05.txt
draft-burmeister-avt-rtcp-feedback-sim-01.txt
draft-ietf-avt-mpeg4-simple-07.txt (with IESG)
draft-ietf-avt-crtp-enhance-07.txt (IETF last call)
draft-ietf-avt-rtcp-bw-05.txt      (RFC Editor)
draft-ietf-avt-rtp-mime-06.txt     (RFC Editor)
draft-ietf-avt-dsr-05.txt          (RFC Editor)
draft-ietf-avt-evrc-smv-03.txt     (RFC Editor)
	See also draft-garudadri-qcp-00.txt
RFC 3497

The Secure Real-time Transport Protocol                                    15
draft-ietf-avt-srtp-05.txt

Framing RTP over Connection-Oriented Transport 		(Lazzaro)          15
	draft-lazzaro-avt-rtp-framing-contrans-00.txt

RTP Extended Reports (RTP XR) 				(Friedman/Clark)   15
	draft-ietf-avt-rtcp-report-extns-03.txt

RTCP Extensions for SSM with unicast feedback		(Chesterfield)     15
	draft-ietf-avt-rtcpssm-03.txt

RTP Payload Format for JVT Video 			(Wenger)	   15
 	draft-ietf-avt-rtp-h264-01.txt

RTP Payload Format for uncompressed video 		(Gharai)	   15
	draft-ietf-avt-uncomp-video-02.txt
	

Thursday, 20th March at 09:00-11:30
===================================

RTP Payload Format for a 64 kbit/s transparent call	(Kreuter)          15
	draft-kreuter-avt-rtp-clearmode-02.txt
	See also the TDM emulation work in the PWE3 working group

The MIDI Wire Protocol Packetization (MWPP) 		(Lazzaro)          15
	draft-ietf-avt-mwpp-midi-rtp-06.txt
	draft-lazzaro-avt-mwpp-coding-guidelines-02.txt

RTP Payload Format for ATRAC-X				(Romaine)	   15
	draft-hatanaka-avt-rtp-atracx-01.txt

RTP Payload Format for iLBC Speech 			(Duric)            15
	draft-ietf-avt-rtp-ilbc-01.txt
	draft-ietf-avt-ilbc-codec-01.txt

RTP Payload Format for the Speex Codec 			(Herlein)          15
	draft-herlein-speex-rtp-profile-00.txt

RTP Payload Format for RGL Codec			(Ramalho)          15
	draft-ramalho-rgl-desc-01.txt 
	draft-ramalho-rgl-rtpformat-01.txt

RTP Payload Format for ETSI ES 202 050 DSR		(Xie)	           15
	draft-xie-avt-dsr-es202050-00.txt

File format for EVRC/SMV vs QCP                                            15

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



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From: "David A. Mcgrew" <mcgrew@cisco.com>
To: "Jose Rey" <rey@panasonic.de>, "Avt@Ietf. Org" <avt@ietf.org>
Cc: <mbaugher@cisco.com>, <elisabetta.carrara@era.ericsson.se>,
        <mats.naslund@era.ericsson.se>, <karl.norrman@era.ericsson.se>,
        <oran@cisco.com>
Subject: RE: [AVT] RE: Security for draft-ietf-avt-rtp-retransmission
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Jose,

> -----Original Message-----
> From: Jose Rey [mailto:rey@panasonic.de]
> Sent: Monday, February 17, 2003 7:41 AM
> To: David A. Mcgrew; Avt@Ietf. Org
> Cc: mbaugher@cisco.com; elisabetta.carrara@era.ericsson.se;
> mats.naslund@era.ericsson.se; karl.norrman@era.ericsson.se;
> oran@cisco.com
> Subject: RE: [AVT] RE: Security for draft-ietf-avt-rtp-retransmission
>
>
> David,
>
> thanks for your comments, please see inline.
>
> > -----Original Message-----
> > From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of David
> > A. Mcgrew
> > Sent: Thursday, February 06, 2003 4:53 PM
> > To: Jose Rey; Avt@Ietf. Org
> > Cc: mbaugher@cisco.com; elisabetta.carrara@era.ericsson.se;
> > mats.naslund@era.ericsson.se; karl.norrman@era.ericsson.se;
> > oran@cisco.com
> > Subject: [AVT] RE: Security for draft-ietf-avt-rtp-retransmission
> >
> >
> > José,
> >
> > > -----Original Message-----
> > > From: Jose Rey [mailto:rey@panasonic.de]
> > > Sent: Thursday, January 23, 2003 8:35 AM
> > > To: Avt@Ietf. Org
> > > Cc: mbaugher@cisco.com; elisabetta.carrara@era.ericsson.se;
> > > mcgrew@cisco.com; mats.naslund@era.ericsson.se;
> > > karl.norrman@era.ericsson.se; oran@cisco.com
> > > Subject: Security for draft-ietf-avt-rtp-retransmission
> > >
> > >
> > > Hi all,
> > >
> > > we are doing the final edits to the
> > draft-ietf-avt-rtp-retransmission.
> > > We have considered all your comments until the date, but there's
> > > something for which we would like to get some feedback on: security.
> > >
> > > A couple of questions:
> > >
> > > 1.-As discussed on the list:
> > > ...
> > > > -----Original Message-----
> > > > From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On
> > Behalf Of Colin
> > > > Perkins
> > > > Sent: Friday, January 10, 2003 4:39 AM
> > > > To: David.Leon@nokia.com
> > > > Cc: avt@ietf.org
> > > > Subject: Re: [AVT] Comments on
> > > > draft-ietf-avt-rtp-retransmission-04.txt
> > > >
> > > ...snip...
> > >
> > > > >>  - In section 12, how does this draft affect SRTP? Is it
> > > > possible to use
> > > > >>    the two formats together?
> > > > >
> > > > >We'll try to draft something on that. My understanding
> > is that the
> > > > >retransmission stream is seen as any other stream by SRTP
> > > > and there should
> > > > >thus be no problem.
> > > >
> > > > I was not clear if retransmitting packets caused problems for
> > > > SRTP; maybe
> > > > one of the SRTP authors can comment on any potential
> > security issue?
> > > >
> > >
> > > we don't see any problems (except for SRTCP as below in 2)
> > to use SRTP
> > > with RTX since the RTP retransmission packets (with a two byte
> > > retransmission payload header) are seen by SRTP as 'normal'
> > RTP packets.
> > > Is our assumption correct?
> >
> > I think so.  Certainly if the retransmission packets are
> > treated as a separate
> > session or separate stream with its own set of SRTP keys,
> > then there's no
> > problem.
>
> Fine.
>
> >  The only potential issue that I see is that the
> > SRTP protection for
> > the retransmission data will need to get keys (and other
> > parameters) somehow.
> > Is the plan to signal the retransmission session/stream at
> > the same time as the
> > original stream, but then only use it if it is needed?  In
> > that case, I'd expect
> > that all the SRTP info for both streams/sessions would be
> > provided by the
> > signaling, which seems like it would work well.
> >
>
> This is also my understanding.
>
> > >
> > > 2.-The retransmission payload format needs of the AVPF
> > profile to enable
> > > more frequent feedback and to request packets. This profile defines
> > > (besides the new timing rules) some general-purpose messages such as
> > > ACKs and NACKs and a new RTCP packet format: the early
> > feedback packet,
> > > which just contains 1 SR or RR and 1 SDES with just the CNAME. Now,
> > > there should be no problem to encrypt/authenticate early packets,
> > > according to the SRTP draft:
> > >
> > > "
> > >    According to [RFC1889] there is a "recommended" packet format for
> > >    compound packets. SRTCP MUST be given packets according to that
> > >    recommendation in the sense that the first part MUST be a sender
> > >    report or a receiver report. However, the encryption
> > prefix (Section
> > >    6.1 of [RFC1889]), a random 32-bit quantity intended to
> > deter known
> > >    plaintext attacks, MUST NOT be used (see below).
> > > "
> > >
> > >
> > > Also there should be no problem to authenticate/encrypt
> > ACKs and NACKs
> > > with SRTCP since they are sent in compound RTCP packets
> > starting with an
> > > RR or SR packet. Let us know if you disagree with this point.
> > >
> > >
> > > 3.- if a new profile needs be defined, something like "SAVPF", where
> > > shall this profile "SAVPF" be specified and registered?
> > wouldn't it be
> > > reasonable to do this in the SRTP draft which already registers the
> > > "SAVP" profile given that the feedback draft would
> > (presumably) have a
> > > similar schedule to the SRTP draft? In this way the
> > question 2. could
> > > also be anwered there.
> >
> > The SRTP draft is in IESG review, so at this point it
> > wouldn't be appropriate to
> > drag it back.
>
> Yes, I understand.
>
> > But if the new profile is uncomplicated, it
> > shouldn't be hard to
> > throw together a new draft that references AVPF and SRTP to
> > make SAVPF.  Could
> > you point me in the right direction as to what differences
> > between AVP and AVPF
> > would be important w.r.t. SRTP?
> >
>
> AVPF introduces new feedback messages ACKs/NACKs and application layer
> messages which, as concluded above, don't represent any problem for
> SRTP. This is because the RTCP packets used by AVPF-compliant entities
> also use the "recommended" RTCP packet format as in RTP, thus not being
> a problem (see point 2 above).
>
> The other major change is that the AVPF profile abolishes the 5-second
> rule; this shouldn't be a problem for SRTP either (?)
> > >
> > > 4.- the security section in the retransmission draft
> > already points out
> > > the risks of using the same keys across sessions, e.g.
> > two.-time pads
> > > and points to the SRTP for solutions. That much is done.
> > Are we missing
> > > something else?
> >
> > The draft says that "Applications utilising encryption SHOULD
> > encrypt both the
> > original    and the retransmission stream."   IIUC, it might
> > be better to state
> > that "If cryptography is used to provide security services on
> > the original
> > stream, then the same services, with equivalent cryptographic
> > strength, SHOULD
> > be provided on the retransmission stream."  Also, I'm
> > wondering why you would
> > want a SHOULD and not a MUST here!
> >
>
> Right, thanks for the rewording. I like a MUST instead better.
>
> > Correct me if I'm wrong, but here's my mental picture of how
> > the combined
> > srtp/retransmission system would work:
> >
> > sender side
> > -----------
> >
> > original rtp source -+------------------> srtp encrypter
> > (key1) ---> ...
> >                      |
> >                      +-> rexmit buffer -> srtp encrypter
> > (key2) ---> ...
> >
> > Another approach would be to move the retransmission buffer
> > to the other side of
> > the srtp encrypter, and get rid of key2.  This would be
> > workable, since the
> > original sequence number is provided in the retransmitted
> > packet.  However, this
> > scheme would not provide message authentication on the
> > retransmission packet
> > flow.  Another demerit of this scheme is that it would
> > interact badly with the
> > SRTP anti-replay window; the recommended size for this window
> > is 128, and this
> > scheme might require an even larger one.  So I like the first
> > method (the way I
> > think that you're doing it) better!
> >
> > At any rate, it might be good to describe how you want the
> > retransmission
> > mechanism to work with SRTP or other crypto mechanisms, to
> > avoid confusion down
> > the road.
>
>
> The 1st possibility as depicted by you is the one we have in mind. Your
> other "option" could be given as an example of what not to do and why.
> This would  rule out possible questions.
>
>
>
> >
> > >
> > > 5.-It is also pointed out that the Timestamp of the retransmission
> > > packets is the same as in the original packets. Since this
> > is random it
> > > should be no problem for security. Is this assumption correct?
> >
> > I think that the non-randomness of the initial timestamp of
> > the retransmission
> > stream is not an issue.  The unpredictability of the initial
> > RTP timestamp makes
> > a DoS attack more difficult for an attacker who is not able
> > to observe the RTP
> > flow, but who knows the destination address, destination port
> > number, and SSRC
> > of that flow.  The same argument could be made for the
> > avt-rtp-retranmission
> > mechanism.  If the attacker can't see the original flow, she
> > can't predict the
> > initial timestamp of the retransmission flow.  If she can see
> > the original flow,
> > then there is no protection provided by having random
> > timestamp on either flow.
> > So there is no disadvantage to the avt-rtp-retransmission
> > mechanism's re-use of
> > the initial timestamp.
> >
> OK.
>
> Now, I got a question:
>
> as discussed in previous emails (see the email from Elisabetta), in the
> case of SSRC-multiplexing the same master key might be used for both
> streams since the SSRC is unique among the streams.
>
> In the original packets the SN  is sent in the header, thus accessible.
> However in the retransmission packets, the retransmission payload header
> (the two bytes containing the original SN, OSN) is sent encrypted and
> its location is known: right after the RTP header before the payload.
> Since both original and retransmission packets have the same TS, an
> attacker could easily identify an original packet and its retransmission
> by comparing the TS. Once this is done he could easily see how the
> cleartext SN is encrypted. Now, I am not a security expert but,
> intuitively, this is not secure. Is it not a security threat? Does SRTP
> solve this?

In crypto jargon, the SN in the retransmission packet is known-plaintext.  The
ciphers used in SRTP, like most modern ciphers, provide good protection against
known-plaintext attacks, even where there is really great of known plaintext.
So the point that you bring up isn't really a security problem.

David

>
> thanks,
>
> José
>
>
>

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



From mailnull@www1.ietf.org  Thu Mar 13 14:38:15 2003
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From: Eve Schooler <schooler@research.att.com>
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Fiandra,

Finally, here are pointers to the paper that extends the 
ideas proposed in the I-D, and that I promised below.
  
The paper analyzes the bandwidth savings of summarisation,
and presents a couple additional techniques (biasing and
hierarchical summarisation) that 
- combine well with unicast feedback and generic summarisation, 
- provide further rtcp scalability, and 
- offer additional application contexts for rtcp.


http://www.research.att.com/~schooler/papers/rtcp-summary-paper.techreport.ps

http://www.research.att.com/~schooler/papers/rtcp-summary-paper.techreport.pdf
  
A shorter version will appear in the 2nd IEEE International 
Symposium on Network Computing and Applications, NCA-03, 
April 16-18, 2003, Cambridge, MA.
 
 http://www.research.att.com/~schooler/papers/rtcp-summary-paper.ps
 http://www.research.att.com/~schooler/papers/rtcp-summary-paper.pdf

E.

> -----Original Message-----
> From: Eve Schooler [mailto:schooler@research.att.com]
> Sent: martedì 4 febbraio 2003 2.02
> To: Fiandra Riccardo
> Cc: Eve Schooler; Julian Chesterfield; Joerg Ott; avt@ietf.org
> Subject: Re: [AVT] RTCP unicast feedback for measurement applications?
> 
> 
> Fiandra Riccardo <riccardo.fiandra@fastweb.it> wrote:
> 
> >I was interested in the "RTCP Extensions for Single-Source Multicast
> >Sessions with Unicast Feedback" draft.
> >Actually, I was wondering if it is also targeted to have a very large and
> >distribuited network measurement framework.
> >Where multicast receivers (thousands) use RTCP to send report to a
> >measurement collector about packet loss, jitter, or other.
> >I see really many advantages on that for e2e measurements, especially in a
> >scenario where thousands of receivers are always connected to the network
> >and cannot send multicast (common scenario for an operator, even without
> >SSM...).
> >
> >My questions are:
> >-Was this a scenario of application when writing the draft?
> 
> Absolutely, though the primary focus of the draft was to solve
> the problem posed by the removal/restriction of the backchannel
> caused by SSM and other uni-directional or asymmetric topologies.
> Solving the feedback problem for SSM gives us the blueprint for
> other unicast-based feedback architectures, including ones for
> network measurement in general.
> 
> >-In this case, could I use the RTCP channel multicasted to the receivers to
> >control them?
> 
> Yes you could.  This is certainly what we envision in the RTCP
> context; use the multicast channel in the source-to-receiver
> direction, and use the unicast backchannels in the
> receiver-to-source direction.
> 
> We also propose to use the multicast channel either:
> - to reflect receiver feedback (received on the unicast backchannels)
>   back out on the multicast channel to reach all receivers, or
> - to redistribute to the receivers feedback summaries
>   (receiver feedback collected at the source and aggregated into
>   a single mathematical distribution) back out on the multicast channel.
> 
> >-Do you see any extension possible also for ptp rtp connections (forward
> >rtcp reports to another device?)
> 
> In section 10 of draft-ietf-avt-rtcpssm-02.txt, we use
> an extension to SDP to indicate the device to which the
> feedback gets reported.  This is done following the proposal
> for SDP source filters documented in
> draft-ietf-mmusic-sdp-srcfilter-00.txt.
> 
> We also just completed a paper that expands upon the ideas
> in the I-D and that should provide further application
> scenarios.  I'll post it here shortly.
> 
> Eve
> 
> -=-=-=-=-
> 
> Eve M. Schooler                 Voice:  1-650-330-7913
> AT&T Labs - Research            FAX:    1-650-463-7037
> 75 Willow Road                  E-mail: schooler@research.att.com
> Menlo Park, CA, USA 94025       http://www.research.att.com/~schooler
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] Protocol Action: RTP: A Transport Protocol for Real-Time
 Applications to Draft Standard
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The IESG has approved the following Internet-Drafts as Draft
Standards:


  o RTP: A Transport Protocol for Real-Time Applications
    <draft-ietf-avt-rtp-new-12.txt>, obsoleting RFC1889, currently a
    Proposed Standard.

  o RTP Profile for Audio and Video Conferences with Minimal Control
    <draft-ietf-avt-profile-new-13.txt> obsoleting RFC1890, currently
    a Proposed Standard.


These documents are the products of the Audio/Video Transport Working
Group. The IESG contact persons are Allison Mankin and Scott Bradner.
   
Technical Summary
   
    The first document defines RTP, consisting of two closely-linked 
    parts:

    o the real-time transport protocol (RTP), to carry data that has
            real-time properties.

    o the RTP control protocol (RTCP), to monitor the quality of
            service and to convey information about the participants in 
  		an on-going session.
       
    A well-exercised capability for DES encryption of RTP and RTCP
    packets is part of the document, along with extensive
    considerations on use of the feature. A new profile of RTP
    called Secure RTP (SRTP) is under way that will provide encryption
    with AES since DES is not considered strong, and provide much
    better security support. However it was deemed that RTP and
    RTCP should be advanced to Draft Standard with their currently
    deployed security feature rather than waiting for the future
    standardization of SRTP (and its eligibility for Draft), because
    the practical use of DES-encrypted media with RTP has been
    extensive.

    This document obsoletes RFC 1889.
    
    The second document defines aspects of RTP that complete a 
    profile for the use within audio and video conferences with 
    minimal session control. In particular, no support for the 
    negotiation of parameters or membership control is provided. The 
    profile is useful in sessions where no negotiation or membership 
    control are used (e.g., using the static payload types and the 
    membership indications provided by RTCP), but this profile is also 
    useful in conjunction with a higher-level control protocol, and it 
    is invoked by uses of media managed by Session Description Protocol
    (RFC2327) as well in frequent practice. 
     
    This document obsoletes RFC 1890.

Working Group Summary
   
       The AVT Working Group conducted careful review of the two
       documents and their clarification. There was a strong effort 
       to ensure the implementation test plan and report had extensive 
       review as well. The differences from the Proposed Standard to 
       the Draft Standards are minor, are well-documented in sections 
       of the two documents, and do not result in any on-the-wire 
       changes to the protocols.

       There was strong working group support for the advancement
       to Draft Standard.
           
Protocol Quality
   
       RTP and RTCP have been in use on the Internet for many years.
       As the implementation report shows, there are two independent 
       open source implementations that are very full, as well as many 
       commercial ones. RTP and RTCP and their payloads are used in 
       numerous applications, in conjunction with other established 
       protocols, such as Session Initiation Protocol in telephony 
       applications. The documents were reviewed for the IESG by 
       Allison Mankin.
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi Magnus,

> First I have a question on how you are going to go forward in IETF. Are 
> you planning to take this work into any WG or do a individual submission?

I am most happy to take it into a WG but am uncertain which one that 
would be.

> Thanks, for the explanation, it greatly clarifies this. However to my 
> understanding the usage of MIME types is actually depending on the URI 
> scheme. One simple example is to compare HTTP and RTSP. In HTTP you: 1) 
> download the object specified by the URI. 2) Look at the MIME type of 
> the downloaded object and sends it to its parser together with the 
> fragment. 3) The parser takes the fragment into account when displaying 
> the content.

Not necessarily. That's just the way current browsers work because of 
the recommendations for the use of the fragment. I know for a fact that 
the Apache Web server can receive, parse and handle fragment offsets.

> However for RTSP URI with a fragment one possible interpretation is to 
> setup the presentation indicated by the URI. This setup results in that 
> a media description is retrieved. This description has of course a MIME 
> type. Is it to this description the the fragment applies? Or is it to a 
> the presentation in general which does not have a MIME type, or is it to 
> the media streams which can contain one or more different encodings 
> perhaps identified by MIME types.

That should be discussed, but my take is that it applies to each one of 
the media streams that are part of the general presentation.

> My point with this example is that the schemes seems to be taken into 
> account. For RTSP it seems in fact that it is time to start defining 
> some fragment identifier behavior.

I agree, we definately need to take the schemes into account when 
describing the expected behaviour of servers and clients when 
encountering these temporal offsets in URIs.

> I don't think we need to consider servers in this case, rather the 
> clients as fragments should have been treated in accordance with the URI 
> specification saying that fragment is a user agent problem.

It is important to consider servers, too. I does not make sense to 
burden the Internet or the user with a huge amount of video data when 
all that the user requires is a clip 20 minutes into the video. I am 
aware that the current use of fragments does not conform to this idea 
and therefore suggested to discuss the possibilities of changing that 
usage specification.

>>> So a general question, what is the actually possibilities to 
>>> standardize such a solution?
>>
>> It needs some minor adjustments in the URI standard. As that standard 
>> is currently being changed/adapted, this could be the right time to 
>> get it adjusted. Then our mechanism can be provided as a Generic 
>> Syntax proposal and MIME type specifications for time-continuous data 
>> can elect to support it. That's my take.
> 
> I am not fully aware of what is happening with this. However I have seen 
> a draft for making URI generic syntax to standard. Is that what you have 
> in mind? To my knowledge the changes performed on a specification when 
> going from draft to full standard are only clarifications. Technical 
> changes are very limited if allowed at all.

As far as I understand it, there are more changes than just 
clarifications happening to RFC2396. But if you are right, we are 
probably forced to go with the query "?" solution rather than fragments, 
which makes it a lot less elegant. Why should we be using a different 
scheme for addressing offsets in audio/video than in html? Also, the use 
of query is already overloaded and therefore a sensible use of fragments 
for media would make sense.

> I think that any such extension as you are proposing will be a proposed 
> standard.

I'd be more than happy to have it become that and through an IETF WG.

> For me it seems very bad step to redefine this. I think one should look 
> at both use cases. There will still be plenty of reasons to use fragment 
> on the client side.

Could you please tell me some? I tend to disagree. I believe that as 
time-continuous media are usually high data rates, fragment addressing 
of such data only makes sense if we can avoid having to serve out the 
complete file before we can address the offset. By the way, even with 
HTML I have often been upset about having to wait (especially on a modem 
line) for the complete document to download when all I wanted was to 
read a specific paragraph in it.

> And as there is a large number of HTTP clients out 
> there using the current definition it seems a bad move to redefine it.

No. That would  not affect them. It is still possible to say that if the 
user (agent) wishes to interpret the fragment offset locally, he's free 
to strip off the fragment offset from the URI (as happening today) and 
not sending that to the server. However, if a server receives a fragment 
offset request, he requires to send data only from that offset onwards.

> However one should also look into how to address this when one wants the 
> server to perform the action. I can clearly understand the use case of 
> having a server reduce large media files to only contain what one is 
> interested of.
> I looked through the URIBOF agenda: 
> http://www.ietf.org/ietf/03mar/uribof.txt . I could however not see that 
> this issue would be part of the BOF. Are you planning to take up your 
> proposal there?

I only found out about the URIBOF agenda after having submitted the I-D. 
I have however sent the I-D to the relevant mailing lists and hope to 
get them to pick it up and discuss the issues. I cannot attend the 
meeting personally, unfortunately.

Best Regards,

Silvia.

---
Silvia Pfeiffer
Research Scientist
CSIRO Australia
_______________________________________________
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From: Colin Perkins <csp@isi.edu>
Subject: [AVT] Slides for San Francisco meeting
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As slides for the San Francisco AVT meeting are submitted, I'm making
copies available at http://www.east.isi.edu/~csp/avt56/ 

They will eventually migrate to http://www.ietf.org/proceedings/03mar/

Colin
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Subject: [AVT] Comments on draft-ietf-avt-rtp-retransmission-06.txt
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Hi,

Here is a few comments on the RTP retransmission draft I think the draft 
is in very good state and I propose that it is moved forward with due 
swiftness. I at least has use of this format and would strongly 
encourage authors and chairs that this draft is progressed and becomes 
an Proposed standard RFC before september. My use case would be to 
include this in the 3GPP Packet-based streaming service (PSS) Release 6.

1. Section 4, last paragraph: "The M, CC and CSRC bit of the original 
RTP packet MUST be copied "as is" into the RTP header of the 
retransmission packet." I think one should change the start of the 
sentence as the use of bit is wrong. I would propose "The marker bit 
(M), the CSRC count (CC) field and CSRC list of the original packet MUST 
..."

2. Section 8.1, Third paragraph: "If multiple payload types are used for 
the original streams, then multiple "apt" parameters MUST be included to 
map each original stream payload type to a different retransmission 
payload type." I think this is a bit unclearly formulated as it can give 
the wrong impression that multiple apt can be given for a single PT. I 
would suggest that one changes this to: "Each RTP retransmission payload 
type MUST be mapped to one and only one original payload type. If there 
is multiple original payload types, each of them needs to be mapped to 
its own retransmission payload type."

3. Section 8.1, Fourth paragraph: "An OPTIONAL payload-format-specific 
parameter, "rtx-time," indicates the maximum time a server will try to 
retransmit a packet." I think this is wrongly formulated. To my 
understanding the rtx-time correspond to the time that the sender will 
keep old packets in their retransmission buffer. This is also formulated 
on some other place in the draft. The above text also gives the 
impression that a server will perform retransmission automatically. I 
would suggest that one reformulates this to something like: "An OPTIONAL 
payload format specific parameter "rtx-time", indicates the time in 
milliseconds a sender will keep an original packet available for 
retransmission since its first transmission." The text needs to be 
updated also in all MIME registrations.

4. Section 9.1 First paragraph: I would like to add the following 
sentence to the end of that paragraph. "The SSRC value included in SETUP 
response's Transport header SHOULD be that of the original stream." This 
makes it clear on how to use the SSRC field when doing SSRC-multiplexed 
retransmission.

5 Section 12. With respect to the latest discussion on the aspects of 
using SRTP and what's written in this section I would recommend that 
some text is added to this section. I would propose to add to the end of 
the paragraph that starts "However, since ..". My proposal for text 
would be:
"Still as it is possible to infer a lot of the RTP fields, including the 
OSN of the retransmission payload format, it is RECOMMENDED that any 
encryption has good properties against known plain-text attacks." I 
would like to have some comment on this from our SRTP experts.

6. Section 14.1 Update reference 3 to the new RTP draft. Could one also 
include [ ] around the reference numbers. I think this is clearer and 
follows the well used IEEE way of doing reference lists.

7. Please also look over how the page breaks falls out after editing. 
There are definitely cases when page breaks should not be allowed in the 
middle. One such example is the author list where Rolf Hakenberg's 
information is split over two pages. 


Best Regards

Magnus Westerlund 

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



_______________________________________________
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Subject: [AVT] Issues for the file format for EVRC/SMV vocoder
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Hi folks,

The topic of the file format for EVRC/SMV vocoders hopefully will be
discussed in this meeting at San Francisco. Below is a list of the
issues that might be related to this topic. They are listed here as
potential issues for you to consider before the discussion at the
meeting.

(1) Technically differences. Is there much difference in efficiency and
performance between the format in draft-ietf-avt-evrc-smv and
draft-garudadri-qcp? Are the differences simply on the file syntax?

(2) Completeness. For EVRC codec data, formats defined in both document
can handle it. For SMV codec data, the format defined in
draft-ietf-avt-evrc-smv is currently the only file format that handles
SMV data. draft-garudadri-qcp does not handle SMV at this time. 

(3) Maturity of the format definition. draft-ietf-avt-evrc-smv has its
first version submitted to AVT on November 2000. It has been actively
worked on all these years, and is co-authored by people from seven
companies and universities. It is currently on the RFC editor's queue.
draft-garudadri-qcp is submitted in February 2003. Will there be
concerns about the maturity of the drafts, particularly the handling of
SMV codec hasn't been written in the later draft yet?

(4) The recognition. Since EVRC and SMV codec are designed in 3GPP2 for
their CDMA networks, formats recognized by them are likely to be the
most widely used format for those codecs. Which of the formats, as
defined in draft-ietf-avt-evrc-smv and draft-garudadri-qcp, has been
considered by 3GPP2 for the format for storing EVRC and SMV data?

(5) Document organization. Even though this is a rather minor point, but
would there be enough reasons to trade-off for the additional complexity
for having the MIME registration of EVRC/SMV refering to a separate and
yet to be complete draft?

Draft-ietf-avt-evrc-smv is currently on the RFC editor's queue. If we
want to consider deleting the file format that is in the draft for
almost two years in the last minute before it becomes an RFC and using
another yet to be complete new draft instead, we should be providing
ourselves with the clear justification for doing so. 

I will not be there for the discussion unfortunately since I have not
made the plan to attend in advance. However, I hope the list of
potential issues above might be useful for us to discuss and consider.

Wish we have a fruitful meeting in San Francisco. 

Adam


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



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To: "Adam Li" <adamli@icsl.ucla.edu>, "'Ietf-Avt'" <avt@ietf.org>
From: Randall Gellens <randy@qualcomm.com>
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Subject: [AVT] Re: Issues for the file format for EVRC/SMV vocoder
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At 9:18 PM -0800 3/15/03, Adam Li wrote:

>  The topic of the file format for EVRC/SMV vocoders hopefully will be
>  discussed in this meeting at San Francisco.

We may be able to resolve this before the meeting, in email.  If not, 
then I think it may be discussed at the AVT meeting.

>  Technically differences.

While the formats do have some differences, the main difference is 
that the qcp format is currently used on different hardware and 
software to interoperably store codec frames.  It doesn't appear to 
me that there is any difference in efficiency.


>  For EVRC codec data, formats defined in both document
>  can handle it. For SMV codec data, the format defined in
>  draft-ietf-avt-evrc-smv is currently the only file format that handles
>  SMV data. draft-garudadri-qcp does not handle SMV at this time.

SMV is being added to the qcp format; it's just a trivial matter of 
adding a new identifier.

>   Maturity of the format definition.

The format defined in draft-ietf-avt-evrc-smv was specified primarily 
for completeness of the document.  At the time we were working on 
this draft, I and the other authors were unaware of the qcp file 
format.  (I'm very sorry for this.)  It turns out that the qcp format 
has been widely used on multiple platforms for some time, including 
Eudora email and Quicktime media player (both on both Mac and 
Windows), and millions of CDMA handsets  I'm not aware of any current 
interoperable independent implementations of the format specified in 
draft-ietf-avt-evrc-smv.

>  Which of the formats, as
>  defined in draft-ietf-avt-evrc-smv and draft-garudadri-qcp, has been
>  considered by 3GPP2 for the format for storing EVRC and SMV data?

The issue of which format to specify in 3GPP2 is currently being 
discussed.  Most of the discussions to date have been on how to 
include evrc, smv, and qcelp-13k in an iso-based mp4 file format. 
The issue of specifying an independent file format for vocoders in 
3GPP2 is new.

>  would there be enough reasons to trade-off for the additional complexity
>  for having the MIME registration of EVRC/SMV refering to a separate and
>  yet to be complete draft?

This is very minor.  However, if the qcp format were something new, 
and not currently deployed, I'd be the first to say that we should 
stick with the format in draft-ietf-avt-evrc-smv.  It's only because 
of the existing interoperable use of the qcp format that I agree with 
making it the file format for the registrations.

>  Draft-ietf-avt-evrc-smv is currently on the RFC editor's queue. If we
>  want to consider deleting the file format that is in the draft for
>  almost two years in the last minute before it becomes an RFC and using
>  another yet to be complete new draft instead, we should be providing
>  ourselves with the clear justification for doing so.

I completely agree with you.  I have zero desire to delay 
draft-ietf-avt-evrc-smv.  In fact, my initial suggestion, as written 
in the qcp draft, was to take the easier path of not changing 
draft-ietf-avt-evrc-smv at all, and instead to have the qcp draft 
register a new MIME type (audio/qcp) which can be used to store evrc, 
qcelp (and SMV).  I knew this was a kludge (since the qcp format is 
already in use) but wanted to do what was quickest and easiest.

However, I was clearly told that this was not acceptable -- that we 
should do what is technically correct: allow draft-ietf-avt-evrc-smv 
to progress but without the file format, and change the qcp draft to 
modify the MIME registrations.  I agree that this is the correct way 
to proceed.


-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
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There is so much to be said in favor of modern journalism.
By giving us the opinions of the uneducated, it keeps us in touch
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_______________________________________________
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Subject: RE: [AVT] File format for EVRC/SMV vocoder
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At 1:52 PM +0100 3/11/03, Lars-Erik Jonsson (EAB) wrote:

>  OK, but why has not this issue been brought up earlier?

Because of an unfortunate oversight.  Neither myself nor the other 
authors of the evrc/smv draft were aware of the qcp format.  I very 
much regret this.  At the time we were working on the evrc/smv draft, 
we specified a file format to make the draft complete.  Had we been 
aware of an existing format that was currently being used to 
interoperably exchange vocoder frames, I'm sure it would have been 
considered instead of inventing something new.

>   The EVRC/SMV
>  format work in AVT has been going on for a long time, and therefore one
>  can question whether it is correct (from a procedural point of view) to
>  suddenly not use what has been developed in this WG, but use something
>  defined elsewhere. IETF standardization is usually an engineering
>  process, not rubber-stamping of already deployed technical solutions.

When I learned of the qcp format, my suggestion for how to proceed 
was to register a new mime type (audio/qcp) so as to not risk 
delaying the evrc draft.  (As one of the authors of that draft, I 
have no desire to delay it at all.)  This is what the current qcp 
draft says.  However, I was told to do the technically correct thing 
instead.

>  Note that I am not aware of all the details in this matter, but what
>  has been communicated on the list does not provide enough material for
>  motivating what has been proposed. "We have already deployed this
>  alternative format while you were trying to agree on a standard, and
>  now we think you should use our format instead since it is more
>  widely deployed" is a rather strange way of arguing in standards
>  development, I think.

The QCP format was in deployment when we started working on the 
evrc/smv draft, but neither I not the other authors were aware of 
this.  It's not the case that the qcp format was deployed afterwards 
or in parallel, in an attempt to circumvent process.

The question now is how best to proceed.  As I said, my initial 
suggestion was to take the easy path of not deleting the file format 
from the evrc/smv draft, and instead to use a kludge of registering a 
new mime type for the qcp file format.

-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly-selected tag: ---------------
Laughter is the closest distance between two people.
                                     --Victor Borge
_______________________________________________
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Subject: RE: [AVT] Comments on draft-ietf-avt-rtp-retransmission-06.txt
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Hi Magnus,

Thanks a lot for your detailed review.
We will make the changes you recommend to the draft.

David.

> -----Original Message-----
> From: ext Magnus Westerlund [mailto:magnus.westerlund@era.ericsson.se]
> Sent: Sunday, March 16, 2003 6:42 AM
> To: avt@ietf.org
> Subject: [AVT] Comments on draft-ietf-avt-rtp-retransmission-06.txt
> 
> 
> Hi,
> 
> Here is a few comments on the RTP retransmission draft I 
> think the draft 
> is in very good state and I propose that it is moved forward with due 
> swiftness. I at least has use of this format and would strongly 
> encourage authors and chairs that this draft is progressed 
> and becomes 
> an Proposed standard RFC before september. My use case would be to 
> include this in the 3GPP Packet-based streaming service (PSS) 
> Release 6.
> 
> 1. Section 4, last paragraph: "The M, CC and CSRC bit of the original 
> RTP packet MUST be copied "as is" into the RTP header of the 
> retransmission packet." I think one should change the start of the 
> sentence as the use of bit is wrong. I would propose "The marker bit 
> (M), the CSRC count (CC) field and CSRC list of the original 
> packet MUST 
> ..."
> 
> 2. Section 8.1, Third paragraph: "If multiple payload types 
> are used for 
> the original streams, then multiple "apt" parameters MUST be 
> included to 
> map each original stream payload type to a different retransmission 
> payload type." I think this is a bit unclearly formulated as 
> it can give 
> the wrong impression that multiple apt can be given for a 
> single PT. I 
> would suggest that one changes this to: "Each RTP 
> retransmission payload 
> type MUST be mapped to one and only one original payload 
> type. If there 
> is multiple original payload types, each of them needs to be 
> mapped to 
> its own retransmission payload type."
> 
> 3. Section 8.1, Fourth paragraph: "An OPTIONAL 
> payload-format-specific 
> parameter, "rtx-time," indicates the maximum time a server 
> will try to 
> retransmit a packet." I think this is wrongly formulated. To my 
> understanding the rtx-time correspond to the time that the 
> sender will 
> keep old packets in their retransmission buffer. This is also 
> formulated 
> on some other place in the draft. The above text also gives the 
> impression that a server will perform retransmission automatically. I 
> would suggest that one reformulates this to something like: 
> "An OPTIONAL 
> payload format specific parameter "rtx-time", indicates the time in 
> milliseconds a sender will keep an original packet available for 
> retransmission since its first transmission." The text needs to be 
> updated also in all MIME registrations.
> 
> 4. Section 9.1 First paragraph: I would like to add the following 
> sentence to the end of that paragraph. "The SSRC value 
> included in SETUP 
> response's Transport header SHOULD be that of the original 
> stream." This 
> makes it clear on how to use the SSRC field when doing 
> SSRC-multiplexed 
> retransmission.
> 
> 5 Section 12. With respect to the latest discussion on the aspects of 
> using SRTP and what's written in this section I would recommend that 
> some text is added to this section. I would propose to add to 
> the end of 
> the paragraph that starts "However, since ..". My proposal for text 
> would be:
> "Still as it is possible to infer a lot of the RTP fields, 
> including the 
> OSN of the retransmission payload format, it is RECOMMENDED that any 
> encryption has good properties against known plain-text attacks." I 
> would like to have some comment on this from our SRTP experts.
> 
> 6. Section 14.1 Update reference 3 to the new RTP draft. 
> Could one also 
> include [ ] around the reference numbers. I think this is clearer and 
> follows the well used IEEE way of doing reference lists.
> 
> 7. Please also look over how the page breaks falls out after editing. 
> There are definitely cases when page breaks should not be 
> allowed in the 
> middle. One such example is the author list where Rolf Hakenberg's 
> information is split over two pages. 
> 
> 
> Best Regards
> 
> Magnus Westerlund 
> 
> Multimedia Technologies, Ericsson Research ERA/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
> 
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 
_______________________________________________
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From: =?iso-8859-1?Q?Jos=E9_Rey?= <rey@panasonic.de>
Subject: AW: [AVT] Comments on draft-ietf-avt-rtp-retransmission-06.txt
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Hi Magnus,

thanks for your comments. I clipped off the comments on editing and wording,
which will be taken care of. I only address the following:


> 4. Section 9.1 First paragraph: I would like to add the following
> sentence to the end of that paragraph. "The SSRC value included in SETUP
> response's Transport header SHOULD be that of the original stream." This
> makes it clear on how to use the SSRC field when doing SSRC-multiplexed
> retransmission.
>
OK.

> 5 Section 12. With respect to the latest discussion on the aspects of
> using SRTP and what's written in this section I would recommend that
> some text is added to this section. I would propose to add to the end of
> the paragraph that starts "However, since ..". My proposal for text
> would be:
> "Still as it is possible to infer a lot of the RTP fields, including the
> OSN of the retransmission payload format, it is RECOMMENDED that any
> encryption has good properties against known plain-text attacks." I
> would like to have some comment on this from our SRTP experts.
>
Yeah, a comment on this is needed . Your proposed text above names a lot of
"RTP fields", I am not sure what you mean. The only field that is
potentially send both as clear tex and encrypted is the (O)SN field. I would
limit the text to that.


cheers,

José

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



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José Rey wrote:

>>5 Section 12. With respect to the latest discussion on the aspects of
>>using SRTP and what's written in this section I would recommend that
>>some text is added to this section. I would propose to add to the end of
>>the paragraph that starts "However, since ..". My proposal for text
>>would be:
>>"Still as it is possible to infer a lot of the RTP fields, including the
>>OSN of the retransmission payload format, it is RECOMMENDED that any
>>encryption has good properties against known plain-text attacks." I
>>would like to have some comment on this from our SRTP experts.
>>
>Yeah, a comment on this is needed . Your proposed text above names a lot of
>"RTP fields", I am not sure what you mean. The only field that is
>potentially send both as clear tex and encrypted is the (O)SN field. I would
>limit the text to that.
>

Actually I was a little bit to restricted in my description. First this 
does not only apply to SRTP, but also like the DES encryption of the 
complete RTP packet. Which means that all of RTP header can be known if 
you know the entry point, for example through RTSP signaling. Then some 
payload formats has rather lot of known fields. So what I mean with RTP 
fields are both RTP header fields, and RTP payload.

I hope that this clarifies and I understand that improvements of the 
language seems to be requirement.

Cheers

Magnus Westerlund 

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



_______________________________________________
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From: Silvia.Pfeiffer@csiro.au
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To: William.C.Miller@abc.com
Cc: Conrad.Parker@csiro.au, s22-list@eng.smpte.org, avt@ietf.org
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Subject: [AVT] Re: Comments, draft-pfeiffer-temporal-fragments-00
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Dear Mr. Miller,

I am delighted to get feedback from you and the SMPTE community on our 
timed URI fragment specification proposal. I would like to thank you 
very much for your well elaborated comments which we are more than happy 
to include into our proposal.

We will be developing another version of the document over the next few 
weeks. If I may, I would like to forward you the next version, just to 
make sure we haven't overlooked anything.

Also, in case there are any relevant changes in the new version of SMPTE 
12M, I would be very happy if you could make me aware of them.

Thank you very much indeed.

Best Regards,

Silvia Pfeiffer.


Miller, William C wrote:
> Dear Ms. Pfeiffer and Mr. Parker,
> 
> Your draft was recently passed on to me by a SMPTE colleague.  I'd like 
> to offer a couple of comments.  I am not familiar with all of the 
> practices of the IETF community, so forgive me if I commit a solecism or 
> two along the way.
> 
> I'm chairman of SMPTE S22, the TV Systems Technology Committee, which 
> has responsibility for SMPTE time code.  I'm writing as an individual 
> commenter.
> 
> 1.  SMPTE has about 400 Standards, Recommended Practices and Engineering 
> Guidelines, of which SMPTE time code is just one.  The reference you 
> should cite is SMPTE 12M-1999, Time and Control Code.  Be aware that we 
> are in the process of considering revisions to this document.
> 
> 2.  The current version of SMPTE 12M defines time code for 24, 25 and 30 
> frames per second (fps).  Video production standards for all of these 
> exist and are in use.  A drop-frame mode is defined for 30 fps (actually 
> 30/1.001 fps) to cover 525-line systems (NTSC time-compensated mode).  
> However, there are now 50,  60 and 60/1.001 fps video production 
> standards, and practices have evolved to deal with these within the 
> structure of SMPTE 12M.  These practices will be added to 12M as it is 
> revised.  A possibility also exists that there will be 24/1.001 fps video.
> 
> To be safe you should define syntax to handle all 8 of these rates (all 
> are allowable under MPEG-2, for instance).  Recapping them in order:
> 
> 24/1.001 "24-drop" (this algorithm is not defined)
> 24
> 25
> 30/1.001 "30-drop"
> 30
> 50
> 60/1.001 "60-drop" (same algorithm as 30-drop, but 4 frames dropped 
> rather than 2)
> 60
> 
> Your draft gives 30 fps (non-drop) as the default; in practice this is 
> the least likely frame rate to be encountered.  I believe it is 
> preferable to require explicit coding of the frame rate, e. g. 
> "smpte-30-time".  "smpte-time" should be allowed only for those cases 
> where the user does not know the frame rate; this would serve as a wild 
> card, with best-effort address resolution (the error would be no more 
> than one second, typically much less).
> 
> 3.  Although not generally understood as such, SMPTE time code is a 
> label.  Its lowest granularity is the frame for progressive-scan and 
> film material and the field for interlaced material.  Fractional frames 
> (fields) are meaningless for video, and in the case of the /1.001 rates 
> are ambiguous for audio.  I urge you to delete "smpte-sf".
> 
> 4.  The first frame of video of a program is normally 01:00:00:00, not 
> 00:00:00:00.  This practice arose from the requirements of program 
> production and analog videotape recording technology, and it has 
> subsequently become a uniform, almost ironclad practice worldwide.  
> Users should be made aware of this in an informative note.
> 
> I hope these comments are of help in refining your draft.  I am 
> attaching a copy of SMPTE 12M for your use.  Please do not redistribute 
> it except to those involved in the development of this RFC.  Thanks.
> 
> If I can be of further help please let me know.
> 
> Best regards,
> Bill Miller
> 
> 
> 
> William C. Miller
> ABC-TV
> 212-456-4448
> 212-456-2284 fax
> william.c.miller@abc.com
> 

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



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        "Randall Gellens" <randy@qualcomm.com>
From: Harinath Garudadri <hgarudad@qualcomm.com>
Subject: RE: [AVT] File format for EVRC/SMV vocoder
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At 08:04 AM 3/17/2003 +0900, Jae-Yong Lee wrote:
>Hi all,
>
>To my best knowledge, QT is supporting QCELP "codec"
>(in QT file format), not the .qcp file format.

I was able to create a qcp file in my Eudora client and send to someone who 
uses a different email client. When he launched Quicktime for this file, he 
was able to play the file as is and also convert the file to .mov format. 
Here is the .mov version of the file for those who have Quicktime.


>Jay
>
> >
> > Hi, Randy,
> >
> > Are there any intellectual property issues with the qcp format?
> >
> > Also, were the interoperable implementations of the qcp format really
> > done independently?  My understanding is that Eudora is a Qualcomm
> > product and that Apple acted merely as an integrator and distributer
> > of the PureVoice technology from Qualcomm.  If these implementations
> > are not independent it may be misleading for you to talk about
> > interoperability.
> >
> > -Pete
> >
> >
> > Randall Gellens writes:
> >  > At 1:52 PM +0100 3/11/03, Lars-Erik Jonsson (EAB) wrote:
> >  >
> >  > >  OK, but why has not this issue been brought up earlier?
> >  >
> >  > Because of an unfortunate oversight.  Neither myself nor the other
> >  > authors of the evrc/smv draft were aware of the qcp
> > format.  I very
> >  > much regret this.  At the time we were working on the
> > evrc/smv draft,
> >  > we specified a file format to make the draft complete.
> > Had we been
> >  > aware of an existing format that was currently being used to
> >  > interoperably exchange vocoder frames, I'm sure it would have been
> >  > considered instead of inventing something new.
> >  >
> >  > >   The EVRC/SMV
> >  > >  format work in AVT has been going on for a long time,
> > and therefore one
> >  > >  can question whether it is correct (from a procedural
> > point of view) to
> >  > >  suddenly not use what has been developed in this WG,
> > but use something
> >  > >  defined elsewhere. IETF standardization is usually an
> > engineering
> >  > >  process, not rubber-stamping of already deployed
> > technical solutions.
> >  >
> >  > When I learned of the qcp format, my suggestion for how to proceed
> >  > was to register a new mime type (audio/qcp) so as to not risk
> >  > delaying the evrc draft.  (As one of the authors of that draft, I
> >  > have no desire to delay it at all.)  This is what the current qcp
> >  > draft says.  However, I was told to do the technically
> > correct thing
> >  > instead.
> >  >
> >  > >  Note that I am not aware of all the details in this
> > matter, but what
> >  > >  has been communicated on the list does not provide
> > enough material for
> >  > >  motivating what has been proposed. "We have already
> > deployed this
> >  > >  alternative format while you were trying to agree on a
> > standard, and
> >  > >  now we think you should use our format instead since it is more
> >  > >  widely deployed" is a rather strange way of arguing in standards
> >  > >  development, I think.
> >  >
> >  > The QCP format was in deployment when we started working on the
> >  > evrc/smv draft, but neither I not the other authors were aware of
> >  > this.  It's not the case that the qcp format was deployed
> > afterwards
> >  > or in parallel, in an attempt to circumvent process.
> >  >
> >  > The question now is how best to proceed.  As I said, my initial
> >  > suggestion was to take the easy path of not deleting the
> > file format
> >  > from the evrc/smv draft, and instead to use a kludge of
> > registering a
> >  > new mime type for the qcp file format.
> >
> >

Hari Garudadri (from Vancouver, B.C.)
mobile : 858-229-2801

--=====================_172821323==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
<font size=3>At 08:04 AM 3/17/2003 +0900, Jae-Yong Lee wrote:<br>
<blockquote type=cite class=cite cite>Hi all,<br><br>
To my best knowledge, QT is supporting QCELP &quot;codec&quot; <br>
(in QT file format), not the .qcp file format.</font></blockquote><br>
I was able to create a qcp file in my Eudora client and send to someone
who uses a different email client. When he launched Quicktime for this
file, he was able to play the file as is and also convert the file to
.mov format. Here is the .mov version of the file for those who have
Quicktime.<br><br>
<br>
<blockquote type=cite class=cite cite><font size=3>Jay<br><br>
&gt; <br>
&gt; Hi, Randy,<br>
&gt; <br>
&gt; Are there any intellectual property issues with the qcp 
format?<br>
&gt; <br>
&gt; Also, were the interoperable implementations of the qcp format
really<br>
&gt; done independently?&nbsp; My understanding is that Eudora is a
Qualcomm<br>
&gt; product and that Apple acted merely as an integrator and
distributer<br>
&gt; of the PureVoice technology from Qualcomm.&nbsp; If these
implementations<br>
&gt; are not independent it may be misleading for you to talk about<br>
&gt; interoperability.<br>
&gt; <br>
&gt; -Pete<br>
&gt; <br>
&gt; <br>
&gt; Randall Gellens writes:<br>
&gt;&nbsp; &gt; At 1:52 PM +0100 3/11/03, Lars-Erik Jonsson (EAB)
wrote:<br>
&gt;&nbsp; &gt; <br>
&gt;&nbsp; &gt; &gt;&nbsp; OK, but why has not this issue been brought up
earlier?<br>
&gt;&nbsp; &gt; <br>
&gt;&nbsp; &gt; Because of an unfortunate oversight.&nbsp; Neither myself
nor the other <br>
&gt;&nbsp; &gt; authors of the evrc/smv draft were aware of the qcp 
<br>
&gt; format.&nbsp; I very <br>
&gt;&nbsp; &gt; much regret this.&nbsp; At the time we were working on
the <br>
&gt; evrc/smv draft, <br>
&gt;&nbsp; &gt; we specified a file format to make the draft
complete.&nbsp; <br>
&gt; Had we been <br>
&gt;&nbsp; &gt; aware of an existing format that was currently being used
to <br>
&gt;&nbsp; &gt; interoperably exchange vocoder frames, I'm sure it would
have been <br>
&gt;&nbsp; &gt; considered instead of inventing something new.<br>
&gt;&nbsp; &gt; <br>
&gt;&nbsp; &gt; &gt;&nbsp;&nbsp; The EVRC/SMV<br>
&gt;&nbsp; &gt; &gt;&nbsp; format work in AVT has been going on for a
long time, <br>
&gt; and therefore one<br>
&gt;&nbsp; &gt; &gt;&nbsp; can question whether it is correct (from a
procedural <br>
&gt; point of view) to<br>
&gt;&nbsp; &gt; &gt;&nbsp; suddenly not use what has been developed in
this WG, <br>
&gt; but use something<br>
&gt;&nbsp; &gt; &gt;&nbsp; defined elsewhere. IETF standardization is
usually an <br>
&gt; engineering<br>
&gt;&nbsp; &gt; &gt;&nbsp; process, not rubber-stamping of already
deployed <br>
&gt; technical solutions.<br>
&gt;&nbsp; &gt; <br>
&gt;&nbsp; &gt; When I learned of the qcp format, my suggestion for how
to proceed <br>
&gt;&nbsp; &gt; was to register a new mime type (audio/qcp) so as to not
risk <br>
&gt;&nbsp; &gt; delaying the evrc draft.&nbsp; (As one of the authors of
that draft, I <br>
&gt;&nbsp; &gt; have no desire to delay it at all.)&nbsp; This is what
the current qcp <br>
&gt;&nbsp; &gt; draft says.&nbsp; However, I was told to do the
technically <br>
&gt; correct thing <br>
&gt;&nbsp; &gt; instead.<br>
&gt;&nbsp; &gt; <br>
&gt;&nbsp; &gt; &gt;&nbsp; Note that I am not aware of all the details in
this <br>
&gt; matter, but what<br>
&gt;&nbsp; &gt; &gt;&nbsp; has been communicated on the list does not
provide <br>
&gt; enough material for<br>
&gt;&nbsp; &gt; &gt;&nbsp; motivating what has been proposed. &quot;We
have already <br>
&gt; deployed this<br>
&gt;&nbsp; &gt; &gt;&nbsp; alternative format while you were trying to
agree on a <br>
&gt; standard, and<br>
&gt;&nbsp; &gt; &gt;&nbsp; now we think you should use our format instead
since it is more<br>
&gt;&nbsp; &gt; &gt;&nbsp; widely deployed&quot; is a rather strange way
of arguing in standards<br>
&gt;&nbsp; &gt; &gt;&nbsp; development, I think.<br>
&gt;&nbsp; &gt; <br>
&gt;&nbsp; &gt; The QCP format was in deployment when we started working
on the <br>
&gt;&nbsp; &gt; evrc/smv draft, but neither I not the other authors were
aware of <br>
&gt;&nbsp; &gt; this.&nbsp; It's not the case that the qcp format was
deployed <br>
&gt; afterwards <br>
&gt;&nbsp; &gt; or in parallel, in an attempt to circumvent 
process.<br>
&gt;&nbsp; &gt; <br>
&gt;&nbsp; &gt; The question now is how best to proceed.&nbsp; As I said,
my initial <br>
&gt;&nbsp; &gt; suggestion was to take the easy path of not deleting the
<br>
&gt; file format <br>
&gt;&nbsp; &gt; from the evrc/smv draft, and instead to use a kludge of
<br>
&gt; registering a <br>
&gt;&nbsp; &gt; new mime type for the qcp file format.<br>
&gt; <br>
&gt; </blockquote>
<x-sigsep><p></x-sigsep>
Hari Garudadri (from Vancouver, B.C.)<br>
mobile : 858-229-2801<br>
</font></html>

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--=====================_172821323==_--

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



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From: "Jae-Yong Lee" <jlee@ieee.org>
To: "Adam Li" <adamli@icsl.ucla.edu>, "'Ietf-Avt'" <avt@ietf.org>,
        "Randall Gellens" <randy@qualcomm.com>
Cc: <casner@acm.org>, "'Scott Bradner'" <sob@harvard.edu>,
        "'Colin Perkins'" <csp@csperkins.org>,
        "'Allison Mankin'" <mankin@psg.com>, <randy@qualcomm.com>,
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References: <001301c2eb7b$b0cc77d0$6c7ba8c0@divxnetworks.com> <a06000b0bba9a6726a547@[66.114.232.124]>
Subject: Re: [AVT] Re: Issues for the file format for EVRC/SMV vocoder
Date: Mon, 17 Mar 2003 10:14:58 +0900
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Hello all,

My comments are inlined.

> At 9:18 PM -0800 3/15/03, Adam Li wrote:
> 
> >  The topic of the file format for EVRC/SMV vocoders hopefully will be
> >  discussed in this meeting at San Francisco.
> 
> We may be able to resolve this before the meeting, in email.  If not, 
> then I think it may be discussed at the AVT meeting.
> 
> >  Technically differences.
> 
> While the formats do have some differences, the main difference is 
> that the qcp format is currently used on different hardware and 
> software to interoperably store codec frames.  It doesn't appear to 
> me that there is any difference in efficiency.
> 

The difference that I see is that the storage format described in 
draft-ietf-avt-evrc-smv is simple, straightforward and easy to generate 
from the RTP payloads and vice versa while draft-garudadri-qcp is rather 
a much more complicated scheme. This would become one of the most 
important factor if we take mobile devices into account.
What would be the technical advantage of draft-garudadri-qcp for storage
of EVRC over the current draft-ietf-avt-evrc-smv? 

> 
> >  For EVRC codec data, formats defined in both document
> >  can handle it. For SMV codec data, the format defined in
> >  draft-ietf-avt-evrc-smv is currently the only file format that handles
> >  SMV data. draft-garudadri-qcp does not handle SMV at this time.
> 
> SMV is being added to the qcp format; it's just a trivial matter of 
> adding a new identifier.
> 

Would it be necessary if we have draft-ietf-avt-evrc-smv?
Would it not be possible to extend RFC 2658 to include its storage format
and make it similar to draft-ietf-avt-evrc-smv?

> >   Maturity of the format definition.
> 
> The format defined in draft-ietf-avt-evrc-smv was specified primarily 
> for completeness of the document.  At the time we were working on 
> this draft, I and the other authors were unaware of the qcp file 
> format.  (I'm very sorry for this.)  It turns out that the qcp format 
> has been widely used on multiple platforms for some time, including 
> Eudora email and Quicktime media player (both on both Mac and 
> Windows), and millions of CDMA handsets  I'm not aware of any current 
> interoperable independent implementations of the format specified in 
> draft-ietf-avt-evrc-smv.
> 

To the best of my knowledge, Quicktime does not support .qcp file format
although it does support QCELP codec in QT file format.
I have some comments below (under the next point) regarding interoperable 
independent implementations of the format in draft-ietf-avt-evrc-smv.

> >  Which of the formats, as
> >  defined in draft-ietf-avt-evrc-smv and draft-garudadri-qcp, has been
> >  considered by 3GPP2 for the format for storing EVRC and SMV data?
> 
> The issue of which format to specify in 3GPP2 is currently being 
> discussed.  Most of the discussions to date have been on how to 
> include evrc, smv, and qcelp-13k in an iso-based mp4 file format. 
> The issue of specifying an independent file format for vocoders in 
> 3GPP2 is new.
> 

The 3GPP2's goal with multimedia file format is to have a single common 
file format for various applications as stated in its Stage-1 (requirements) 
document. People in 3GPP2 have agreed to have it based on the ISO base 
media file format which was already adopted for many other international 
standards, e.g., 3GPP, MPEG, JPEG.
Also the FFMS (file format for multimedia services) baseline text is refering
to draft-ietf-avt-evrc-smv for storing EVRC and SMV data in the media track.
Many (defintely more than two) handset vendors and solution providers
have already implemented this and tested interoperability and operators are
preparing services based on that.
If wanted, the media data would be easily taken and converted into the format 
specified in draft-ietf-avt-evrc-smv simply by adding the magic numbers.
I believe this sort of simple conversion would be really useful if speech-only
formats are also considered for some applications or services.

> >  would there be enough reasons to trade-off for the additional complexity
> >  for having the MIME registration of EVRC/SMV refering to a separate and
> >  yet to be complete draft?
> 
> This is very minor.  However, if the qcp format were something new, 
> and not currently deployed, I'd be the first to say that we should 
> stick with the format in draft-ietf-avt-evrc-smv.  It's only because 
> of the existing interoperable use of the qcp format that I agree with 
> making it the file format for the registrations.
> 
> >  Draft-ietf-avt-evrc-smv is currently on the RFC editor's queue. If we
> >  want to consider deleting the file format that is in the draft for
> >  almost two years in the last minute before it becomes an RFC and using
> >  another yet to be complete new draft instead, we should be providing
> >  ourselves with the clear justification for doing so.
> 
> I completely agree with you.  I have zero desire to delay 
> draft-ietf-avt-evrc-smv.  In fact, my initial suggestion, as written 
> in the qcp draft, was to take the easier path of not changing 
> draft-ietf-avt-evrc-smv at all, and instead to have the qcp draft 
> register a new MIME type (audio/qcp) which can be used to store evrc, 
> qcelp (and SMV).  I knew this was a kludge (since the qcp format is 
> already in use) but wanted to do what was quickest and easiest.
> 
> However, I was clearly told that this was not acceptable -- that we 
> should do what is technically correct: allow draft-ietf-avt-evrc-smv 
> to progress but without the file format, and change the qcp draft to 
> modify the MIME registrations.  I agree that this is the correct way 
> to proceed.
> 

I'm not sure what is the technically correct way.
What is the technical benefit of draft-garudadri-qcp?

Above all, it would be very much undesirable to make a significant 
change at the very last minute without sufficient justification. 

> 
> -- 
> Randall Gellens

Regards,
Jay


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



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Date: Sun, 16 Mar 2003 15:52:03 -0600
From: Pete McCann <mccap@lucent.com>
To: Randall Gellens <randy@qualcomm.com>
Cc: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>,
        Adam Li <adamli@icsl.ucla.edu>, Ietf-Avt <avt@ietf.org>,
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Subject: RE: [AVT] File format for EVRC/SMV vocoder
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Hi, Randy,

Are there any intellectual property issues with the qcp format?

Also, were the interoperable implementations of the qcp format really
done independently?  My understanding is that Eudora is a Qualcomm
product and that Apple acted merely as an integrator and distributer
of the PureVoice technology from Qualcomm.  If these implementations
are not independent it may be misleading for you to talk about
interoperability.

-Pete


Randall Gellens writes:
 > At 1:52 PM +0100 3/11/03, Lars-Erik Jonsson (EAB) wrote:
 > 
 > >  OK, but why has not this issue been brought up earlier?
 > 
 > Because of an unfortunate oversight.  Neither myself nor the other 
 > authors of the evrc/smv draft were aware of the qcp format.  I very 
 > much regret this.  At the time we were working on the evrc/smv draft, 
 > we specified a file format to make the draft complete.  Had we been 
 > aware of an existing format that was currently being used to 
 > interoperably exchange vocoder frames, I'm sure it would have been 
 > considered instead of inventing something new.
 > 
 > >   The EVRC/SMV
 > >  format work in AVT has been going on for a long time, and therefore one
 > >  can question whether it is correct (from a procedural point of view) to
 > >  suddenly not use what has been developed in this WG, but use something
 > >  defined elsewhere. IETF standardization is usually an engineering
 > >  process, not rubber-stamping of already deployed technical solutions.
 > 
 > When I learned of the qcp format, my suggestion for how to proceed 
 > was to register a new mime type (audio/qcp) so as to not risk 
 > delaying the evrc draft.  (As one of the authors of that draft, I 
 > have no desire to delay it at all.)  This is what the current qcp 
 > draft says.  However, I was told to do the technically correct thing 
 > instead.
 > 
 > >  Note that I am not aware of all the details in this matter, but what
 > >  has been communicated on the list does not provide enough material for
 > >  motivating what has been proposed. "We have already deployed this
 > >  alternative format while you were trying to agree on a standard, and
 > >  now we think you should use our format instead since it is more
 > >  widely deployed" is a rather strange way of arguing in standards
 > >  development, I think.
 > 
 > The QCP format was in deployment when we started working on the 
 > evrc/smv draft, but neither I not the other authors were aware of 
 > this.  It's not the case that the qcp format was deployed afterwards 
 > or in parallel, in an attempt to circumvent process.
 > 
 > The question now is how best to proceed.  As I said, my initial 
 > suggestion was to take the easy path of not deleting the file format 
 > from the evrc/smv draft, and instead to use a kludge of registering a 
 > new mime type for the qcp file format.

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



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Subject: RE: [AVT] File format for EVRC/SMV vocoder
Date: Mon, 17 Mar 2003 08:04:22 +0900
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From: "Jae-Yong Lee" <jlee@nextreaming.com>
To: "Pete McCann" <mccap@lucent.com>, "Randall Gellens" <randy@qualcomm.com>
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Hi all,

To my best knowledge, QT is supporting QCELP "codec" 
(in QT file format), not the .qcp file format.

Jay

> 
> Hi, Randy,
> 
> Are there any intellectual property issues with the qcp format?
> 
> Also, were the interoperable implementations of the qcp format really
> done independently?  My understanding is that Eudora is a Qualcomm
> product and that Apple acted merely as an integrator and distributer
> of the PureVoice technology from Qualcomm.  If these implementations
> are not independent it may be misleading for you to talk about
> interoperability.
> 
> -Pete
> 
> 
> Randall Gellens writes:
>  > At 1:52 PM +0100 3/11/03, Lars-Erik Jonsson (EAB) wrote:
>  > 
>  > >  OK, but why has not this issue been brought up earlier?
>  > 
>  > Because of an unfortunate oversight.  Neither myself nor the other 
>  > authors of the evrc/smv draft were aware of the qcp 
> format.  I very 
>  > much regret this.  At the time we were working on the 
> evrc/smv draft, 
>  > we specified a file format to make the draft complete.  
> Had we been 
>  > aware of an existing format that was currently being used to 
>  > interoperably exchange vocoder frames, I'm sure it would have been 
>  > considered instead of inventing something new.
>  > 
>  > >   The EVRC/SMV
>  > >  format work in AVT has been going on for a long time, 
> and therefore one
>  > >  can question whether it is correct (from a procedural 
> point of view) to
>  > >  suddenly not use what has been developed in this WG, 
> but use something
>  > >  defined elsewhere. IETF standardization is usually an 
> engineering
>  > >  process, not rubber-stamping of already deployed 
> technical solutions.
>  > 
>  > When I learned of the qcp format, my suggestion for how to proceed 
>  > was to register a new mime type (audio/qcp) so as to not risk 
>  > delaying the evrc draft.  (As one of the authors of that draft, I 
>  > have no desire to delay it at all.)  This is what the current qcp 
>  > draft says.  However, I was told to do the technically 
> correct thing 
>  > instead.
>  > 
>  > >  Note that I am not aware of all the details in this 
> matter, but what
>  > >  has been communicated on the list does not provide 
> enough material for
>  > >  motivating what has been proposed. "We have already 
> deployed this
>  > >  alternative format while you were trying to agree on a 
> standard, and
>  > >  now we think you should use our format instead since it is more
>  > >  widely deployed" is a rather strange way of arguing in standards
>  > >  development, I think.
>  > 
>  > The QCP format was in deployment when we started working on the 
>  > evrc/smv draft, but neither I not the other authors were aware of 
>  > this.  It's not the case that the qcp format was deployed 
> afterwards 
>  > or in parallel, in an attempt to circumvent process.
>  > 
>  > The question now is how best to proceed.  As I said, my initial 
>  > suggestion was to take the easy path of not deleting the 
> file format 
>  > from the evrc/smv draft, and instead to use a kludge of 
> registering a 
>  > new mime type for the qcp file format.
> 
> 
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] File format for EVRC/SMV vocoder
Date: Mon, 17 Mar 2003 10:55:15 +0900
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Thread-Topic: [AVT] File format for EVRC/SMV vocoder
Thread-Index: AcLsGxYo0W/iFtcuSZeilHaHP2LTUQACOY/Q
From: "Jae-Yong Lee" <jlee@nextreaming.com>
To: "Harinath Garudadri" <hgarudad@qualcomm.com>,
        "Pete McCann" <mccap@lucent.com>,
        "Randall Gellens" <randy@qualcomm.com>
Cc: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>,
        "Adam Li" <adamli@icsl.ucla.edu>, "Ietf-Avt" <avt@ietf.org>,
        <casner@acm.org>, "Scott Bradner" <sob@harvard.edu>,
        "Colin Perkins" <csp@csperkins.org>, "Allison Mankin" <mankin@psg.com>,
        <mdturner@lucent.com>, <smathai@lucent.com>, <lioy@qualcomm.com>,
        <zeng@packetvideo.com>, <sherwood@packetvideo.com>,
        <villa@icsl.ucla.edu>, <yllee@samsung.com>, <jeonghoon@samsung.com>,
        <tom.hiller@lucent.com>, <David.Leon@nokia.com>, <nleung@qualcomm.com>,
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Hello Hari and all,
=20
Well there is a sort of plug-in for Quicktime for that.
But I'm not sure if we can count this in since, with plug-in, virtually =
everything is possible.
Maybe someone from Apple can clarify hopefully.
=20
Jay
=20

-----Original Message-----
From: Harinath Garudadri [mailto:hgarudad@qualcomm.com]
Sent: Monday, March 17, 2003 9:12 AM
To: Jae-Yong Lee; Pete McCann; Randall Gellens
Cc: Lars-Erik Jonsson (EAB); Adam Li; Ietf-Avt; casner@acm.org; Scott =
Bradner; Colin Perkins; Allison Mankin; mdturner@lucent.com; =
smathai@lucent.com; lioy@qualcomm.com; zeng@packetvideo.com; =
sherwood@packetvideo.com; villa@icsl.ucla.edu; yllee@samsung.com; =
jeonghoon@samsung.com; tom.hiller@lucent.com; David.Leon@nokia.com; =
nleung@qualcomm.com; dgal@lucent.com; ajayrajkumar@lucent.com; =
magnus.westerlund@era-t.ericsson.se; vbharga@cisco.com; =
craig.greer@nokia.com; magda@qualcomm.com; ned.freed@mrochek.com; =
hgarudad@qualcomm.com; csp@isi.edu; sakazawa@kddilabs.jp
Subject: RE: [AVT] File format for EVRC/SMV vocoder


At 08:04 AM 3/17/2003 +0900, Jae-Yong Lee wrote:


Hi all,

To my best knowledge, QT is supporting QCELP "codec"=20
(in QT file format), not the .qcp file format.


I was able to create a qcp file in my Eudora client and send to someone =
who uses a different email client. When he launched Quicktime for this =
file, he was able to play the file as is and also convert the file to =
.mov format. Here is the .mov version of the file for those who have =
Quicktime.




Jay

>=20
> Hi, Randy,
>=20
> Are there any intellectual property issues with the qcp format?
>=20
> Also, were the interoperable implementations of the qcp format really
> done independently?  My understanding is that Eudora is a Qualcomm
> product and that Apple acted merely as an integrator and distributer
> of the PureVoice technology from Qualcomm.  If these implementations
> are not independent it may be misleading for you to talk about
> interoperability.
>=20
> -Pete
>=20
>=20
> Randall Gellens writes:
>  > At 1:52 PM +0100 3/11/03, Lars-Erik Jonsson (EAB) wrote:
>  >=20
>  > >  OK, but why has not this issue been brought up earlier?
>  >=20
>  > Because of an unfortunate oversight.  Neither myself nor the other=20
>  > authors of the evrc/smv draft were aware of the qcp=20
> format.  I very=20
>  > much regret this.  At the time we were working on the=20
> evrc/smv draft,=20
>  > we specified a file format to make the draft complete. =20
> Had we been=20
>  > aware of an existing format that was currently being used to=20
>  > interoperably exchange vocoder frames, I'm sure it would have been=20
>  > considered instead of inventing something new.
>  >=20
>  > >   The EVRC/SMV
>  > >  format work in AVT has been going on for a long time,=20
> and therefore one
>  > >  can question whether it is correct (from a procedural=20
> point of view) to
>  > >  suddenly not use what has been developed in this WG,=20
> but use something
>  > >  defined elsewhere. IETF standardization is usually an=20
> engineering
>  > >  process, not rubber-stamping of already deployed=20
> technical solutions.
>  >=20
>  > When I learned of the qcp format, my suggestion for how to proceed=20
>  > was to register a new mime type (audio/qcp) so as to not risk=20
>  > delaying the evrc draft.  (As one of the authors of that draft, I=20
>  > have no desire to delay it at all.)  This is what the current qcp=20
>  > draft says.  However, I was told to do the technically=20
> correct thing=20
>  > instead.
>  >=20
>  > >  Note that I am not aware of all the details in this=20
> matter, but what
>  > >  has been communicated on the list does not provide=20
> enough material for
>  > >  motivating what has been proposed. "We have already=20
> deployed this
>  > >  alternative format while you were trying to agree on a=20
> standard, and
>  > >  now we think you should use our format instead since it is more
>  > >  widely deployed" is a rather strange way of arguing in standards
>  > >  development, I think.
>  >=20
>  > The QCP format was in deployment when we started working on the=20
>  > evrc/smv draft, but neither I not the other authors were aware of=20
>  > this.  It's not the case that the qcp format was deployed=20
> afterwards=20
>  > or in parallel, in an attempt to circumvent process.
>  >=20
>  > The question now is how best to proceed.  As I said, my initial=20
>  > suggestion was to take the easy path of not deleting the=20
> file format=20
>  > from the evrc/smv draft, and instead to use a kludge of=20
> registering a=20
>  > new mime type for the qcp file format.
>=20
>=20

Hari Garudadri (from Vancouver, B.C.)
mobile : 858-229-2801



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">


<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D297422401-17032003>Hello=20
Hari and all,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D297422401-17032003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D297422401-17032003>Well=20
there is&nbsp;a sort of plug-in for Quicktime for =
that.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D297422401-17032003>But=20
I'm not sure if we can count this in since, with plug-in, virtually =
everything=20
is possible.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D297422401-17032003>Maybe=20
someone from Apple can clarify hopefully.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D297422401-17032003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D297422401-17032003>Jay</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D297422401-17032003></SPAN></FONT>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DArial=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Harinath Garudadri =

  [mailto:hgarudad@qualcomm.com]<BR><B>Sent:</B> Monday, March 17, 2003 =
9:12=20
  AM<BR><B>To:</B> Jae-Yong Lee; Pete McCann; Randall =
Gellens<BR><B>Cc:</B>=20
  Lars-Erik Jonsson (EAB); Adam Li; Ietf-Avt; casner@acm.org; Scott =
Bradner;=20
  Colin Perkins; Allison Mankin; mdturner@lucent.com; =
smathai@lucent.com;=20
  lioy@qualcomm.com; zeng@packetvideo.com; sherwood@packetvideo.com;=20
  villa@icsl.ucla.edu; yllee@samsung.com; jeonghoon@samsung.com;=20
  tom.hiller@lucent.com; David.Leon@nokia.com; nleung@qualcomm.com;=20
  dgal@lucent.com; ajayrajkumar@lucent.com; =
magnus.westerlund@era-t.ericsson.se;=20
  vbharga@cisco.com; craig.greer@nokia.com; magda@qualcomm.com;=20
  ned.freed@mrochek.com; hgarudad@qualcomm.com; csp@isi.edu;=20
  sakazawa@kddilabs.jp<BR><B>Subject:</B> RE: [AVT] File format for =
EVRC/SMV=20
  vocoder<BR><BR></FONT></DIV><FONT face=3DArial size=3D2>At 08:04 AM =
3/17/2003=20
  +0900, Jae-Yong Lee wrote:<BR></FONT>
  <BLOCKQUOTE class=3Dcite cite=3D"" type=3D"cite"><FONT face=3DArial =
size=3D2>Hi=20
    all,<BR><BR>To my best knowledge, QT is supporting QCELP "codec" =
<BR>(in QT=20
    file format), not the .qcp file format.</FONT></BLOCKQUOTE><BR><FONT =

  face=3DArial size=3D2>I was able to create a qcp file in my Eudora =
client and send=20
  to someone who uses a different email client. When he launched =
Quicktime for=20
  this file, he was able to play the file as is and also convert the =
file to=20
  .mov format. Here is the .mov version of the file for those who have=20
  Quicktime.<BR><BR><BR></FONT>
  <BLOCKQUOTE class=3Dcite cite=3D"" type=3D"cite"><FONT face=3DArial=20
    size=3D2>Jay<BR><BR>&gt; <BR>&gt; Hi, Randy,<BR>&gt; <BR>&gt; Are =
there any=20
    intellectual property issues with the qcp format?<BR>&gt; <BR>&gt; =
Also,=20
    were the interoperable implementations of the qcp format =
really<BR>&gt; done=20
    independently?&nbsp; My understanding is that Eudora is a =
Qualcomm<BR>&gt;=20
    product and that Apple acted merely as an integrator and =
distributer<BR>&gt;=20
    of the PureVoice technology from Qualcomm.&nbsp; If these=20
    implementations<BR>&gt; are not independent it may be misleading for =
you to=20
    talk about<BR>&gt; interoperability.<BR>&gt; <BR>&gt; -Pete<BR>&gt; =
<BR>&gt;=20
    <BR>&gt; Randall Gellens writes:<BR>&gt;&nbsp; &gt; At 1:52 PM +0100 =

    3/11/03, Lars-Erik Jonsson (EAB) wrote:<BR>&gt;&nbsp; &gt; =
<BR>&gt;&nbsp;=20
    &gt; &gt;&nbsp; OK, but why has not this issue been brought up=20
    earlier?<BR>&gt;&nbsp; &gt; <BR>&gt;&nbsp; &gt; Because of an =
unfortunate=20
    oversight.&nbsp; Neither myself nor the other <BR>&gt;&nbsp; &gt; =
authors of=20
    the evrc/smv draft were aware of the qcp <BR>&gt; format.&nbsp; I =
very=20
    <BR>&gt;&nbsp; &gt; much regret this.&nbsp; At the time we were =
working on=20
    the <BR>&gt; evrc/smv draft, <BR>&gt;&nbsp; &gt; we specified a file =
format=20
    to make the draft complete.&nbsp; <BR>&gt; Had we been =
<BR>&gt;&nbsp; &gt;=20
    aware of an existing format that was currently being used to =
<BR>&gt;&nbsp;=20
    &gt; interoperably exchange vocoder frames, I'm sure it would have =
been=20
    <BR>&gt;&nbsp; &gt; considered instead of inventing something=20
    new.<BR>&gt;&nbsp; &gt; <BR>&gt;&nbsp; &gt; &gt;&nbsp;&nbsp; The=20
    EVRC/SMV<BR>&gt;&nbsp; &gt; &gt;&nbsp; format work in AVT has been =
going on=20
    for a long time, <BR>&gt; and therefore one<BR>&gt;&nbsp; &gt; =
&gt;&nbsp;=20
    can question whether it is correct (from a procedural <BR>&gt; point =
of=20
    view) to<BR>&gt;&nbsp; &gt; &gt;&nbsp; suddenly not use what has =
been=20
    developed in this WG, <BR>&gt; but use something<BR>&gt;&nbsp; &gt;=20
    &gt;&nbsp; defined elsewhere. IETF standardization is usually an =
<BR>&gt;=20
    engineering<BR>&gt;&nbsp; &gt; &gt;&nbsp; process, not =
rubber-stamping of=20
    already deployed <BR>&gt; technical solutions.<BR>&gt;&nbsp; &gt;=20
    <BR>&gt;&nbsp; &gt; When I learned of the qcp format, my suggestion =
for how=20
    to proceed <BR>&gt;&nbsp; &gt; was to register a new mime type =
(audio/qcp)=20
    so as to not risk <BR>&gt;&nbsp; &gt; delaying the evrc draft.&nbsp; =
(As one=20
    of the authors of that draft, I <BR>&gt;&nbsp; &gt; have no desire =
to delay=20
    it at all.)&nbsp; This is what the current qcp <BR>&gt;&nbsp; &gt; =
draft=20
    says.&nbsp; However, I was told to do the technically <BR>&gt; =
correct thing=20
    <BR>&gt;&nbsp; &gt; instead.<BR>&gt;&nbsp; &gt; <BR>&gt;&nbsp; &gt;=20
    &gt;&nbsp; Note that I am not aware of all the details in this =
<BR>&gt;=20
    matter, but what<BR>&gt;&nbsp; &gt; &gt;&nbsp; has been communicated =
on the=20
    list does not provide <BR>&gt; enough material for<BR>&gt;&nbsp; =
&gt;=20
    &gt;&nbsp; motivating what has been proposed. "We have already =
<BR>&gt;=20
    deployed this<BR>&gt;&nbsp; &gt; &gt;&nbsp; alternative format while =
you=20
    were trying to agree on a <BR>&gt; standard, and<BR>&gt;&nbsp; &gt;=20
    &gt;&nbsp; now we think you should use our format instead since it =
is=20
    more<BR>&gt;&nbsp; &gt; &gt;&nbsp; widely deployed" is a rather =
strange way=20
    of arguing in standards<BR>&gt;&nbsp; &gt; &gt;&nbsp; development, I =

    think.<BR>&gt;&nbsp; &gt; <BR>&gt;&nbsp; &gt; The QCP format was in=20
    deployment when we started working on the <BR>&gt;&nbsp; &gt; =
evrc/smv=20
    draft, but neither I not the other authors were aware of =
<BR>&gt;&nbsp; &gt;=20
    this.&nbsp; It's not the case that the qcp format was deployed =
<BR>&gt;=20
    afterwards <BR>&gt;&nbsp; &gt; or in parallel, in an attempt to =
circumvent=20
    process.<BR>&gt;&nbsp; &gt; <BR>&gt;&nbsp; &gt; The question now is =
how best=20
    to proceed.&nbsp; As I said, my initial <BR>&gt;&nbsp; &gt; =
suggestion was=20
    to take the easy path of not deleting the <BR>&gt; file format=20
    <BR>&gt;&nbsp; &gt; from the evrc/smv draft, and instead to use a =
kludge of=20
    <BR>&gt; registering a <BR>&gt;&nbsp; &gt; new mime type for the qcp =
file=20
    format.<BR>&gt; <BR>&gt; </FONT></BLOCKQUOTE><X-SIGSEP>
  <P></X-SIGSEP><FONT face=3DArial size=3D2>Hari Garudadri (from =
Vancouver,=20
  B.C.)<BR>mobile : =
858-229-2801<BR></FONT></P></BLOCKQUOTE></BODY></HTML>

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



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	(envelope-from casner@acm.org)
Date: Sun, 16 Mar 2003 21:11:11 -0800 (PST)
From: Stephen Casner <casner@acm.org>
To: Randall Gellens <randy@qualcomm.com>
cc: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>,
        Adam Li <adamli@icsl.ucla.edu>, Ietf-Avt <avt@ietf.org>,
        "'Scott Bradner'" <sob@harvard.edu>,
        "'Colin Perkins'" <csp@csperkins.org>,
        "'Allison Mankin'" <mankin@psg.com>, <mccap@lucent.com>,
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        <sakazawa@kddilabs.jp>
Subject: RE: [AVT] File format for EVRC/SMV vocoder
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On Sun, 16 Mar 2003, Randall Gellens wrote:

> When I learned of the qcp format, my suggestion for how to proceed
> was to register a new mime type (audio/qcp) so as to not risk
> delaying the evrc draft.  (As one of the authors of that draft, I
> have no desire to delay it at all.)  This is what the current qcp
> draft says.  However, I was told to do the technically correct thing
> instead.

My guidance to you on this was in two parts:

1.  I don't think it makes sense he have audio/qcp as a type that does
    not say what encoding is inside the qcp format file.  That led to
    my suggestion to use the existing audio/qcelp and audio/evrc
    types.  When I wrote that, I was forgetting that there already was
    a file format associated with audio/evrc.

2.  It sounded to me as if there was substantial following for qcp
    file format as the container for EVRC, and that the format in the
    evrc-smv draft was created only because the authors were unaware
    of the qcp format.

If the second point accurately reflected the situation, then it would
not be technically correct to proceed with publishing the file format
in the evrc-smv draft, and it would be fortunate that we discovered
this before publication.

However, we are now hearing significant objection to the idea of
deleting the file format in the evrc-smv draft on a variety of
points.  There can be two formats, but all of you, on both sides of
the question, are going to suffer pain in the future if there are two
formats rather than one.  You know its true.

So, the process here is to honestly assess the demand for the two
formats, without ego as to who wrote what.  If there are IPR claims
for one or the other, then that factors into the decision.  If it is a
question of market penetration, and competitors have chosen sides,
then we may be forced into the position of specifying both (or
neither).

My first point above can be satisfied while still defining both
formats by associating the file format with audio/qcelp (where, as I
understand it, there is no conflict) and audio/evrc-qcp.

                                                        -- Steve

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



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References: <F286138BFB1A724ABDD9F0A27978ED1C1AA429@lapis.nextreaming.com>
Date: Mon, 17 Mar 2003 11:00:27 -0800
To: Jae-Yong Lee <jlee@nextreaming.com>,
        Harinath Garudadri <hgarudad@qualcomm.com>,
        Pete McCann <mccap@lucent.com>, Randall Gellens <randy@qualcomm.com>
From: Dave Singer <singer@apple.com>
Subject: RE: [AVT] File format for EVRC/SMV vocoder
Cc: "Lars-Erik Jonsson (EAB)" <Lars-Erik.Jonsson@epl.ericsson.se>,
        Adam Li <adamli@icsl.ucla.edu>, Ietf-Avt <avt@ietf.org>,
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At 10:55 +0900 3/17/03, Jae-Yong Lee wrote:
>Hello Hari and all,
>
>Well there is a sort of plug-in for Quicktime for that.
>But I'm not sure if we can count this in since, with plug-in, 
>virtually everything is possible.
>Maybe someone from Apple can clarify hopefully.

This FAQ seems to say that we ship a QCP importer.  I couldn't 
remember, and can't find a QCP file.

<http://www.cdmatech.com/solutions/products/purevoice_quicktime.html>

>
>Jay
>
>
>-----Original Message-----
>From: Harinath Garudadri [mailto:hgarudad@qualcomm.com]
>Sent: Monday, March 17, 2003 9:12 AM
>To: Jae-Yong Lee; Pete McCann; Randall Gellens
>Cc: Lars-Erik Jonsson (EAB); Adam Li; Ietf-Avt; casner@acm.org; 
>Scott Bradner; Colin Perkins; Allison Mankin; mdturner@lucent.com; 
>smathai@lucent.com; lioy@qualcomm.com; zeng@packetvideo.com; 
>sherwood@packetvideo.com; villa@icsl.ucla.edu; yllee@samsung.com; 
>jeonghoon@samsung.com; tom.hiller@lucent.com; David.Leon@nokia.com; 
>nleung@qualcomm.com; dgal@lucent.com; ajayrajkumar@lucent.com; 
>magnus.westerlund@era-t.ericsson.se; vbharga@cisco.com; 
>craig.greer@nokia.com; magda@qualcomm.com; ned.freed@mrochek.com; 
>hgarudad@qualcomm.com; csp@isi.edu; sakazawa@kddilabs.jp
>Subject: RE: [AVT] File format for EVRC/SMV vocoder
>
>At 08:04 AM 3/17/2003 +0900, Jae-Yong Lee wrote:
>
>>Hi all,
>>
>>To my best knowledge, QT is supporting QCELP "codec"
>>(in QT file format), not the .qcp file format.
>>
>
>I was able to create a qcp file in my Eudora client and send to 
>someone who uses a different email client. When he launched 
>Quicktime for this file, he was able to play the file as is and also 
>convert the file to .mov format. Here is the .mov version of the 
>file for those who have Quicktime.
>
>>Jay
>>
>  >
>  > Hi, Randy,
>  >
>  > Are there any intellectual property issues with the qcp format?
>  >
>  > Also, were the interoperable implementations of the qcp format really
>  > done independently?  My understanding is that Eudora is a Qualcomm
>>  product and that Apple acted merely as an integrator and distributer
>>  of the PureVoice technology from Qualcomm.  If these implementations
>>  are not independent it may be misleading for you to talk about
>>  interoperability.
>>
>>  -Pete
>>
>>
>>  Randall Gellens writes:
>>   > At 1:52 PM +0100 3/11/03, Lars-Erik Jonsson (EAB) wrote:
>>   >
>>   > >  OK, but why has not this issue been brought up earlier?
>>   >
>>   > Because of an unfortunate oversight.  Neither myself nor the other
>>   > authors of the evrc/smv draft were aware of the qcp
>>  format.  I very
>>   > much regret this.  At the time we were working on the
>>  evrc/smv draft,
>>   > we specified a file format to make the draft complete. 
>>  Had we been
>>   > aware of an existing format that was currently being used to
>>   > interoperably exchange vocoder frames, I'm sure it would have been
>>   > considered instead of inventing something new.
>>   >
>>   > >   The EVRC/SMV
>>   > >  format work in AVT has been going on for a long time,
>>  and therefore one
>>   > >  can question whether it is correct (from a procedural
>>  point of view) to
>>   > >  suddenly not use what has been developed in this WG,
>>  but use something
>>   > >  defined elsewhere. IETF standardization is usually an
>>  engineering
>  >  > >  process, not rubber-stamping of already deployed
>>  technical solutions.
>>   >
>>   > When I learned of the qcp format, my suggestion for how to proceed
>>   > was to register a new mime type (audio/qcp) so as to not risk
>>   > delaying the evrc draft.  (As one of the authors of that draft, I
>>   > have no desire to delay it at all.)  This is what the current qcp
>>   > draft says.  However, I was told to do the technically
>>  correct thing
>>   > instead.
>>   >
>>   > >  Note that I am not aware of all the details in this
>>  matter, but what
>>   > >  has been communicated on the list does not provide
>>  enough material for
>>   > >  motivating what has been proposed. "We have already
>>  deployed this
>>   > >  alternative format while you were trying to agree on a
>>  standard, and
>>   > >  now we think you should use our format instead since it is more
>>   > >  widely deployed" is a rather strange way of arguing in standards
>>   > >  development, I think.
>>   >
>>   > The QCP format was in deployment when we started working on the
>>   > evrc/smv draft, but neither I not the other authors were aware of
>>   > this.  It's not the case that the qcp format was deployed
>>  afterwards
>>   > or in parallel, in an attempt to circumvent process.
>>   >
>>   > The question now is how best to proceed.  As I said, my initial
>>   > suggestion was to take the easy path of not deleting the
>>  file format
>>   > from the evrc/smv draft, and instead to use a kludge of
>>  registering a
>>   > new mime type for the qcp file format.
>>
>>
>
>Hari Garudadri (from Vancouver, B.C.)
>mobile : 858-229-2801


-- 
David Singer
Apple Computer/QuickTime
--============_-1164194718==_ma============
Content-Type: text/html; charset="us-ascii"

<!doctype html public "-//W3C//DTD W3 HTML//EN">
<html><head><style type="text/css"><!--
blockquote, dl, ul, ol, li { padding-top: 0 ; padding-bottom: 0 }
 --></style><title>RE: [AVT] File format for EVRC/SMV
vocoder</title></head><body>
<div>At 10:55 +0900 3/17/03, Jae-Yong Lee wrote:</div>
<blockquote type="cite" cite><font face="Arial" size="-1"
color="#0000FF">Hello Hari and all,</font></blockquote>
<blockquote type="cite" cite>&nbsp;</blockquote>
<blockquote type="cite" cite><font face="Arial" size="-1"
color="#0000FF">Well there is&nbsp;a sort of plug-in for Quicktime for
that.</font></blockquote>
<blockquote type="cite" cite><font face="Arial" size="-1"
color="#0000FF">But I'm not sure if we can count this in since, with
plug-in, virtually everything is possible.</font></blockquote>
<blockquote type="cite" cite><font face="Arial" size="-1"
color="#0000FF">Maybe someone from Apple can clarify
hopefully.</font></blockquote>
<div><br></div>
<div>This FAQ seems to say that we ship a QCP importer.&nbsp; I
couldn't remember, and can't find a QCP file.</div>
<div><br></div>
<div
>&lt;http://www.cdmatech.com/solutions/products/purevoice_quicktime.h<span
></span>tml&gt;</div>
<div><br></div>
<blockquote type="cite" cite>&nbsp;</blockquote>
<blockquote type="cite" cite><font face="Arial" size="-1"
color="#0000FF">Jay</font></blockquote>
<blockquote type="cite" cite>&nbsp;<br>
<blockquote><font face="Arial" size="-1">-----Original
Message-----<br>
<b>From:</b> Harinath Garudadri [mailto:hgarudad@qualcomm.com]<br>
<b>Sent:</b> Monday, March 17, 2003 9:12 AM<br>
<b>To:</b> Jae-Yong Lee; Pete McCann; Randall Gellens<br>
<b>Cc:</b> Lars-Erik Jonsson (EAB); Adam Li; Ietf-Avt; casner@acm.org;
Scott Bradner; Colin Perkins; Allison Mankin; mdturner@lucent.com;
smathai@lucent.com; lioy@qualcomm.com; zeng@packetvideo.com;
sherwood@packetvideo.com; villa@icsl.ucla.edu; yllee@samsung.com;
jeonghoon@samsung.com; tom.hiller@lucent.com; David.Leon@nokia.com;
nleung@qualcomm.com; dgal@lucent.com; ajayrajkumar@lucent.com;
magnus.westerlund@era-t.ericsson.se; vbharga@cisco.com;
craig.greer@nokia.com; magda@qualcomm.com; ned.freed@mrochek.com;
hgarudad@qualcomm.com; csp@isi.edu; sakazawa@kddilabs.jp<br>
<b>Subject:</b> RE: [AVT] File format for EVRC/SMV vocoder</font><br>
<font face="Arial" size="-1"></font></blockquote>
<blockquote><font face="Arial" size="-1">At 08:04 AM 3/17/2003 +0900,
Jae-Yong Lee wrote:</font><br>
<blockquote type="cite" cite><font face="Arial" size="-1">Hi all,<br>
<br>
To my best knowledge, QT is supporting QCELP &quot;codec&quot;<br>
(in QT file format), not the .qcp file format.</font><br>
</blockquote>
</blockquote>
<blockquote><br>
<font face="Arial" size="-1">I was able to create a qcp file in my
Eudora client and send to someone who uses a different email client.
When he launched Quicktime for this file, he was able to play the file
as is and also convert the file to .mov format. Here is the .mov
version of the file for those who have Quicktime.<br>
</font>
<blockquote type="cite" cite><font face="Arial"
size="-1">Jay</font><br>
<font face="Arial" size="-1"></font></blockquote>
</blockquote>
<blockquote>
<blockquote><font face="Arial" size="-1">&gt;</font></blockquote>
<blockquote><font face="Arial" size="-1">&gt; Hi,
Randy,</font></blockquote>
<blockquote><font face="Arial" size="-1">&gt;</font></blockquote>
<blockquote><font face="Arial" size="-1">&gt; Are there any
intellectual property issues with the qcp format?</font></blockquote>
<blockquote><font face="Arial" size="-1">&gt;</font></blockquote>
<blockquote><font face="Arial" size="-1">&gt; Also, were the
interoperable implementations of the qcp format
really</font></blockquote>
<blockquote><font face="Arial" size="-1">&gt; done independently?&nbsp;
My understanding is that Eudora is a Qualcomm<br>
&gt; product and that Apple acted merely as an integrator and
distributer<br>
&gt; of the PureVoice technology from Qualcomm.&nbsp; If these
implementations<br>
&gt; are not independent it may be misleading for you to talk
about<br>
&gt; interoperability.<br>
&gt;<br>
&gt; -Pete<br>
&gt;<br>
&gt;<br>
&gt; Randall Gellens writes:<br>
&gt;&nbsp; &gt; At 1:52 PM +0100 3/11/03, Lars-Erik Jonsson (EAB)
wrote:<br>
&gt;&nbsp; &gt;<br>
&gt;&nbsp; &gt; &gt;&nbsp; OK, but why has not this issue been brought
up earlier?<br>
&gt;&nbsp; &gt;<br>
&gt;&nbsp; &gt; Because of an unfortunate oversight.&nbsp; Neither
myself nor the other<br>
&gt;&nbsp; &gt; authors of the evrc/smv draft were aware of the
qcp<br>
&gt; format.&nbsp; I very<br>
&gt;&nbsp; &gt; much regret this.&nbsp; At the time we were working on
the<br>
&gt; evrc/smv draft,<br>
&gt;&nbsp; &gt; we specified a file format to make the draft
complete.&nbsp;<br>
&gt; Had we been<br>
&gt;&nbsp; &gt; aware of an existing format that was currently being
used to<br>
&gt;&nbsp; &gt; interoperably exchange vocoder frames, I'm sure it
would have been<br>
&gt;&nbsp; &gt; considered instead of inventing something new.<br>
&gt;&nbsp; &gt;<br>
&gt;&nbsp; &gt; &gt;&nbsp;&nbsp; The EVRC/SMV<br>
&gt;&nbsp; &gt; &gt;&nbsp; format work in AVT has been going on for a
long time,<br>
&gt; and therefore one<br>
&gt;&nbsp; &gt; &gt;&nbsp; can question whether it is correct (from a
procedural<br>
&gt; point of view) to<br>
&gt;&nbsp; &gt; &gt;&nbsp; suddenly not use what has been developed in
this WG,<br>
&gt; but use something<br>
&gt;&nbsp; &gt; &gt;&nbsp; defined elsewhere. IETF standardization is
usually an<br>
&gt; engineering</font></blockquote>
<blockquote><font face="Arial" size="-1">&gt;&nbsp; &gt; &gt;&nbsp;
process, not rubber-stamping of already deployed<br>
&gt; technical solutions.<br>
&gt;&nbsp; &gt;<br>
&gt;&nbsp; &gt; When I learned of the qcp format, my suggestion for
how to proceed<br>
&gt;&nbsp; &gt; was to register a new mime type (audio/qcp) so as to
not risk<br>
&gt;&nbsp; &gt; delaying the evrc draft.&nbsp; (As one of the authors
of that draft, I<br>
&gt;&nbsp; &gt; have no desire to delay it at all.)&nbsp; This is what
the current qcp<br>
&gt;&nbsp; &gt; draft says.&nbsp; However, I was told to do the
technically<br>
&gt; correct thing<br>
&gt;&nbsp; &gt; instead.<br>
&gt;&nbsp; &gt;<br>
&gt;&nbsp; &gt; &gt;&nbsp; Note that I am not aware of all the details
in this<br>
&gt; matter, but what<br>
&gt;&nbsp; &gt; &gt;&nbsp; has been communicated on the list does not
provide<br>
&gt; enough material for<br>
&gt;&nbsp; &gt; &gt;&nbsp; motivating what has been proposed. &quot;We
have already<br>
&gt; deployed this<br>
&gt;&nbsp; &gt; &gt;&nbsp; alternative format while you were trying to
agree on a<br>
&gt; standard, and<br>
&gt;&nbsp; &gt; &gt;&nbsp; now we think you should use our format
instead since it is more<br>
&gt;&nbsp; &gt; &gt;&nbsp; widely deployed&quot; is a rather strange
way of arguing in standards<br>
&gt;&nbsp; &gt; &gt;&nbsp; development, I think.<br>
&gt;&nbsp; &gt;<br>
&gt;&nbsp; &gt; The QCP format was in deployment when we started
working on the<br>
&gt;&nbsp; &gt; evrc/smv draft, but neither I not the other authors
were aware of<br>
&gt;&nbsp; &gt; this.&nbsp; It's not the case that the qcp format was
deployed<br>
&gt; afterwards<br>
&gt;&nbsp; &gt; or in parallel, in an attempt to circumvent
process.<br>
&gt;&nbsp; &gt;<br>
&gt;&nbsp; &gt; The question now is how best to proceed.&nbsp; As I
said, my initial<br>
&gt;&nbsp; &gt; suggestion was to take the easy path of not deleting
the<br>
&gt; file format<br>
&gt;&nbsp; &gt; from the evrc/smv draft, and instead to use a kludge
of<br>
&gt; registering a<br>
&gt;&nbsp; &gt; new mime type for the qcp file format.<br>
&gt;<br>
&gt;</font><br>
</blockquote>
</blockquote>
<blockquote><font face="Arial" size="-1">Hari Garudadri (from
Vancouver, B.C.)<br>
mobile : 858-229-2801</font></blockquote>
</blockquote>
<div><br></div>
<div><br></div>
<x-sigsep><pre>-- 
</pre></x-sigsep>
<div>David Singer<br>
Apple Computer/QuickTime</div>
</body>
</html>
--============_-1164194718==_ma============--
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Thomas Zeng <zeng@PacketVideo.COM>
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Subject: [AVT] Server RTCP RR demultiplexing when one port is used to receive RR
 s from lots of clients
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Hi Steve:

As promised, I went back to  double checke RFC1889.

It is said in section 6.1:

	"SR or RR: The first RTCP packet in the compound packet must always
be a report packet to facilitate header validation as described in Appendix
A.2. This is true even if no data has been sent nor received, in which case
an empty RR is sent, and even if the only other RTCP packet in the compound
packet is a BYE."

Now when a compound RTCP packet arrives in the server's RTCP port,  if RC
(Report Count) is not zero, the server's SSRC must be present. Hence server
is able to uniquely determine client's identity (since server is capable of
asisgn unique Sender SSRC for each client).
You are right that there is a trick involved: even if no data is received
from the server, RC is set to 1 (instead of zero which seems to be
recommended but not required by 1889), a report block is included in the RR
with zero bytes/packets.
Do you see any problem?

- thomas


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



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Date: Wed, 19 Mar 2003 17:11:07 -0800
To: multicast@lists.uoregon.edu
From: Ross Finlayson <finlayson@live.com>
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Subject: [AVT] Bad RTCP in IETF multicast sessions
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FYI, I have noticed a couple of problems (one minor; one major) with RTCP 
packets being sent on at least two of the current IETF multicast sessions: 
"channel 1 MPEG-1" and "channel 2 MPEG-1".  (I haven't checked the H.261 
sessions.   Note also that the separate, high-bitrate, NCast sessions do 
not seem to have the problem.)

First, the minor problem: Two of the participants (receivers):
         128.223.214.24 (==tesla.uoregon.edu)
         158.38.62.70 (==tikneppen.uninett.no)
on the "channel 1 MPEG-1" session have been sending RTCP packets like the 
following:
         81ca0002 e8e9e765 010e3132
These appear to be RTCP "CNAME" "SDES" packets, but there are a couple of 
problems with them:
1/ Because these RTCP packets don't begin with a "RR" (or "SR"), they're 
illegal, according to section 6.1 of the current RTP spec: 
"draft-ietf-avt-rtp-new-12.txt".
2/ The "length" field in the "CNAME" SDES item (see section 6.5.1 of the 
RTP spec) is 14 (==0x0e), but there are only two following bytes (0x31 and 
0x32)!

These participants both appear to be using the same (broken) 
implementation.  Is this MIM?  In any case, it should be fixed.


Now, the major problem: The sending nodes for both "channel 1 MPEG-1" and 
"channel 2 MPEG-1"
         130.129.32.30 and 130.129.32.10, respectively
are sending RTCP "Sender Report" (SR) packets that contain a bad  RTP 
timestamp field.  Because of this, the RTP timestamp -> NTP time mapping is 
getting messed up, which in turn is messing up client audio/video 
synchronization.

Example:
1/ A RTP (video) packet with timestamp 0x2bfc3a46 is received at (Unix 
epoch) time 1048118445.555259
2/ Next, a RTCP SR packet is received (for the video session):
         80c80006 16a2d20b c2238330 69780000 3f0b1b5d 00160cf2 56d52c81 
81ca0005         16a2d20b 010d3133 302e3132 392e3332 2e333000 81cc0008 
16a2d20b 4d534f4b         1964eb29 00010010 ec1f67fd da1efce1 194ac2ed 
7d7ca166
(This RTCP packet actually contains a SDES and an APP packet, following the 
SR, but for the purpose of this message I care only about the SR.)
The NTP timestamp in this SR packet is
         0xc2238330,0x69780000
which maps to the (Unix epoch) time 1048118448.411987.  So far, so 
good.  However, the RTP timestamp in this SR is
         0x3f0b1b5d
which is *completely different* from the timestamps that have been 
appearing in RTP packets around this time.  In fact, it's more than 3000 
seconds later!  Note that the very next packet received is:
3/ A RTP (video) packet with timestamp 0x2bfc6932

It wouldn't be so bad if both the video and audio sessions had SRs with 
NTP->RTP timestamp mappings that were 3000+ seconds too 
high.  Unfortunately, though, this happens only with the video 
session.  The RTP timestamps in the audio session's SR's are OK.  Because 
of this, a receiver's A/V time synchronization can get badly messed up.

Please try to fix this (if not for the rest of this IETF, then in time for 
the next one).

         Ross.


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



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Hello all.

First let me start by saying... I'm a musician, with a MIDI 
keyboard, who is amazed by the fact that I've been around all 
these IETF's and hadn't wandered into one of your WG 
sessions... (pulled into other WG's and BOF's, more relative 
to my day job :-)Glad someone pointed me in your direction.

Oh well
I have lots of catching up to do. I have 2 questions, both 
very basic... 

Sometime ago(90's), when first doing recording from MIDI to 
MAC, I had to use a serial cable connected some sort of 
interface connected to a MIDI cable to the MIDI device.

With all of the developments with RTP and RTCP, being able to 
run the protocols over IPv4 and IPv6, from a MIDI device, do 
you have examples of what a network diagram would look like?
What types of cables and routers support this type of traffic?

The Q&A section on the Charter page is very informative!

Thanks,

Jai (aka Jessica)


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



From mailnull@www1.ietf.org  Wed Mar 19 23:05:27 2003
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To: avt@ietf.org
Date: Wed, 19 Mar 2003 20:01:21 -0800
From: Colin Perkins <csp@csperkins.org>
Subject: [AVT] Presentations from San Francisco meeting
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You can find copies of the presentation materials for both AVT sessions at
http://www.ietf.org/proceedings/03mar/index.html

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



From mailnull@www1.ietf.org  Wed Mar 19 23:29:59 2003
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Date: Wed, 19 Mar 2003 17:21:57 -0800 (PST)
From: "Lucy E. Lynch" <llynch@darkwing.uoregon.edu>
To: Ross Finlayson <finlayson@live.com>
cc: multicast@lists.uoregon.edu, <avt@ietf.org>
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Subject: [AVT] Re: multicast: Bad RTCP in IETF multicast sessions
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Ross -

yes, I'm running mim on tesla from the command line (using
an sdp file).

Lucy E. Lynch 				Academic User Services
Computing Center			University of Oregon
llynch@darkwing.uoregon.edu		(541) 346-1774/Cell: 912-7998

On Wed, 19 Mar 2003, Ross Finlayson wrote:

> FYI, I have noticed a couple of problems (one minor; one major) with RTCP
> packets being sent on at least two of the current IETF multicast sessions:
> "channel 1 MPEG-1" and "channel 2 MPEG-1".  (I haven't checked the H.261
> sessions.   Note also that the separate, high-bitrate, NCast sessions do
> not seem to have the problem.)
>
> First, the minor problem: Two of the participants (receivers):
>          128.223.214.24 (==tesla.uoregon.edu)
>          158.38.62.70 (==tikneppen.uninett.no)
> on the "channel 1 MPEG-1" session have been sending RTCP packets like the
> following:
>          81ca0002 e8e9e765 010e3132
> These appear to be RTCP "CNAME" "SDES" packets, but there are a couple of
> problems with them:
> 1/ Because these RTCP packets don't begin with a "RR" (or "SR"), they're
> illegal, according to section 6.1 of the current RTP spec:
> "draft-ietf-avt-rtp-new-12.txt".
> 2/ The "length" field in the "CNAME" SDES item (see section 6.5.1 of the
> RTP spec) is 14 (==0x0e), but there are only two following bytes (0x31 and
> 0x32)!
>
> These participants both appear to be using the same (broken)
> implementation.  Is this MIM?  In any case, it should be fixed.
>
>
> Now, the major problem: The sending nodes for both "channel 1 MPEG-1" and
> "channel 2 MPEG-1"
>          130.129.32.30 and 130.129.32.10, respectively
> are sending RTCP "Sender Report" (SR) packets that contain a bad  RTP
> timestamp field.  Because of this, the RTP timestamp -> NTP time mapping is
> getting messed up, which in turn is messing up client audio/video
> synchronization.
>
> Example:
> 1/ A RTP (video) packet with timestamp 0x2bfc3a46 is received at (Unix
> epoch) time 1048118445.555259
> 2/ Next, a RTCP SR packet is received (for the video session):
>          80c80006 16a2d20b c2238330 69780000 3f0b1b5d 00160cf2 56d52c81
> 81ca0005         16a2d20b 010d3133 302e3132 392e3332 2e333000 81cc0008
> 16a2d20b 4d534f4b         1964eb29 00010010 ec1f67fd da1efce1 194ac2ed
> 7d7ca166
> (This RTCP packet actually contains a SDES and an APP packet, following the
> SR, but for the purpose of this message I care only about the SR.)
> The NTP timestamp in this SR packet is
>          0xc2238330,0x69780000
> which maps to the (Unix epoch) time 1048118448.411987.  So far, so
> good.  However, the RTP timestamp in this SR is
>          0x3f0b1b5d
> which is *completely different* from the timestamps that have been
> appearing in RTP packets around this time.  In fact, it's more than 3000
> seconds later!  Note that the very next packet received is:
> 3/ A RTP (video) packet with timestamp 0x2bfc6932
>
> It wouldn't be so bad if both the video and audio sessions had SRs with
> NTP->RTP timestamp mappings that were 3000+ seconds too
> high.  Unfortunately, though, this happens only with the video
> session.  The RTP timestamps in the audio session's SR's are OK.  Because
> of this, a receiver's A/V time synchronization can get badly messed up.
>
> Please try to fix this (if not for the rest of this IETF, then in time for
> the next one).
>
>          Ross.
>
>

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



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From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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To: Thomas Zeng <zeng@PacketVideo.COM>
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Subject: Re: [AVT] Server RTCP RR demultiplexing when one port is used to receive RR s from lots of clients
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Hi Thomas,

When we are talking about demultiplexing of multiple RTP session on the 
same server port I have a different view. First the demultiplexing is 
not based on the SSRC but on the source IP address and port. An RTP 
session between two entities can be defined by the two end points 
address and ports. Assumes that data is sent and received on the same 
port. By strictly defining the RTP session with these two 4-tuples of 
 client address and port and server address and port for RTP and RTCP 
makes it possible to have separate SSRC spaces for each client. As not 
being one who have actually implemented this, I still guess that this 
how it has been done in a lot of servers out there. This solution of 
course does not work when one come to cases where the clients address is 
not known.


Thomas Zeng wrote:

>Hi Steve:
>
>As promised, I went back to  double checke RFC1889.
>
>It is said in section 6.1:
>
>	"SR or RR: The first RTCP packet in the compound packet must always
>be a report packet to facilitate header validation as described in Appendix
>A.2. This is true even if no data has been sent nor received, in which case
>an empty RR is sent, and even if the only other RTCP packet in the compound
>packet is a BYE."
>
>Now when a compound RTCP packet arrives in the server's RTCP port,  if RC
>(Report Count) is not zero, the server's SSRC must be present. Hence server
>is able to uniquely determine client's identity (since server is capable of
>asisgn unique Sender SSRC for each client).
>You are right that there is a trick involved: even if no data is received
>from the server, RC is set to 1 (instead of zero which seems to be
>recommended but not required by 1889), a report block is included in the RR
>with zero bytes/packets.
>Do you see any problem?
>

I see some problems with this. You assume that the client gets the SSRC of the server through signaling (RTSP). However RTP has not seen the SSRC yet. This creates problems with the implementations. A RTCP implementation creates the RTCP receiver report blocks based on the implementations internal state. This state is normally populated based on incoming traffic. Therefore having the client create a state for the server that is not based on received traffic will requires implementation work. 

Secondly, their is a problem with the report block. To include a report block you must include the extended sequence number as being the last packet seen plus the roll over counter. This creates problems as the client RTP implementation has so far not seen any packet. So sending a report block does in fact lie. I see this as an problem. This may certainly have issues with the RTP/RTCP implementations.


Best Regards


Magnus Westerlund 

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



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



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From: "Stepping, Michael" <michael.stepping@FernUni-Hagen.de>
To: "Nükhet Özbek \(Dijital Arge\)" <nukhet.ozbek@vestel.com.tr>,
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Date: Thu, 20 Mar 2003 10:14:24 +0100
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Hi,

Your questions are related to Video codecs.

You will find a lot of information if you google for MPEG-4, MPEG-4 AVC,
H.264 or simply look in ISO/IEC 14496-10 or ITU-T H.264, respectively.

Yours,
Michael

> -----Original Message-----
> To: 'avt@ietf.org'
> Subject: [AVT] Hello
> 
> 
> Hi,
> 
> I am anew subscriber of the list.
> And I have a question about H26.L and H.324 framework.
> What is H.324 framework? And what is the relation with H26.L?
> 
> Thanks & Regards,
> Nükhet.
> _______________________________________________
> Audio/Video Transport Working Group

_______________________________________________
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From: Thomas Zeng <zeng@PacketVideo.COM>
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>,
        Thomas Zeng
	 <zeng@PacketVideo.COM>
Cc: "'avt@ietf.org'" <avt@ietf.org>
Subject: RE: [AVT] Server RTCP RR demultiplexing when one port is used to 
	receive RR s from lots of clients
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Thanks for your comments, Magnus. My answers inlined.

-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@era.ericsson.se]
Sent: Wednesday, March 19, 2003 9:16 PM
To: Thomas Zeng
Cc: 'casner@acm.org'; 'avt@ietf.org'
Subject: Re: [AVT] Server RTCP RR demultiplexing when one port is used
to receive RR s from lots of clients


Hi Thomas,

When we are talking about demultiplexing of multiple RTP session on the 
same server port I have a different view. First the demultiplexing is 
not based on the SSRC but on the source IP address and port. An RTP 
session between two entities can be defined by the two end points 
address and ports. Assumes that data is sent and received on the same 
port. By strictly defining the RTP session with these two 4-tuples of 
 client address and port and server address and port for RTP and RTCP 
makes it possible to have separate SSRC spaces for each client. As not 
being one who have actually implemented this, I still guess that this 
how it has been done in a lot of servers out there. This solution of 
course does not work when one come to cases where the clients address is 
not known.

[TZ] The last case (when Server is in the open, and Clinet is  behind a NAT)
is what I am talking about. 


Thomas Zeng wrote:

>Hi Steve:
>
>As promised, I went back to  double checke RFC1889.
>
>It is said in section 6.1:
>
>	"SR or RR: The first RTCP packet in the compound packet must always
>be a report packet to facilitate header validation as described in Appendix
>A.2. This is true even if no data has been sent nor received, in which case
>an empty RR is sent, and even if the only other RTCP packet in the compound
>packet is a BYE."
>
>Now when a compound RTCP packet arrives in the server's RTCP port,  if RC
>(Report Count) is not zero, the server's SSRC must be present. Hence server
>is able to uniquely determine client's identity (since server is capable of
>asisgn unique Sender SSRC for each client).
>You are right that there is a trick involved: even if no data is received
>from the server, RC is set to 1 (instead of zero which seems to be
>recommended but not required by 1889), a report block is included in the RR
>with zero bytes/packets.
>Do you see any problem?
>

I see some problems with this. You assume that the client gets the SSRC of
the server through signaling (RTSP). However RTP has not seen the SSRC yet.
This creates problems with the implementations. A RTCP implementation
creates the RTCP receiver report blocks based on the implementations
internal state. This state is normally populated based on incoming traffic.
Therefore having the client create a state for the server that is not based
on received traffic will requires implementation work. 

[TZ]  Here Server SSRC happens to be in the RTP header. Hence client can use
RTP header to obtain server SSRC if client has received at least one RTP for
the session. If it has not but has received Setup response, it then can use
SSRC obtained from there. Yes, it is not nice because RTP implementation is
prefered not to have dependencies on the control protocol. However, RTP
implementation must somehow obtain server's RTCP receive address and port,
anyway,
from no where but the control protocol.  Adding SSRC to the list of
dependencies isn't a new breach of the rule.

Secondly, their is a problem with the report block. To include a report
block you must include the extended sequence number as being the last packet
seen plus the roll over counter. This creates problems as the client RTP
implementation has so far not seen any packet. So sending a report block
does in fact lie. I see this as an problem. This may certainly have issues
with the RTP/RTCP implementations.

[TZ] Yes, I agree with you on this one. Fortunately, since not only the
highest SeqNum but also the total number bytes received are  0, there is no
confusion.




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



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From: Eric Cheung <cheung@research.att.com>
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Subject: Re: [AVT] Question on RTP sequence number in RFC2833 and bis-02
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Henning Schulzrinne and I discussed this at IETF 56, and I am sending 
this to close this thread.

The RTP packets sent out at the end an event should have increasing 
sequence number.  This is primarily so that the receiver can detect if 
one or more of these packets are lost.  If they all have the same 
sequence number, then there is no way for the receiver to tell.

Some implementations I have tested send out 3 packets with same sequence 
number.  This should not present a major problem in terms of receiver 
recognizing the DTMF event, but may cause inaccurate RTCP reports.

Thanks
Eric

Eric Cheung wrote:
> Hi, I am still hoping that I can get some clarification from the WG on 
> this question.  Apologies if this was discussed before, but I could not 
> find anything by searching on the archive.
> 
> Thanks
> Eric
> 
> Eric Cheung wrote:
> 
>> Hi Henning, Scott
>>
>> Want to clarify a fine point.
>>
>> In RFC2833:
>>
>>    If there has been no new event
>>    in the last interval, the event SHOULD be retransmitted three times
>>    or until the next event is recognized.
>>
>> Q1: For a device conforming to RFC2833, should the RTP sequence number 
>> in the 3 packets be incremented, or should they be the same?
>>
>>
>> In RFC2833bis-02, this seems to be clarified:
>>
>>    If an event has ended and
>>    there has been no new event in the last interval, the event SHOULD be
>>    retransmitted three times at the interval used by the source for
>>    updates.
>>
>>    In all cases, the RTP sequence number MUST be incremented by one in
>>    each RTP packet.
>>
>> Q2: Can you confirm that if a device conforms to bis-02, the sequence 
>> number in the 3 packets will be monotonically increasing?
>>
>> Q3: In writing this, I realize that I am not sure whether there should 
>> be 4 packets (1 original transmission + 3 retransmission) or 3 
>> packets.  Can you clarify this?
>>
>>
>> Sorry for the nit-picking, but when testing it seems many devices do 
>> many different things when it comes to 2833.
>>
>> Thanks
>> Eric
> 

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



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From: Greg Herlein <gherlein@herlein.com>
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Subject: [AVT] Speex: Apologies for Missed Meeting
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To those expecting my presentation yesterday on the Speex codec,
I apologize for not being there.  I was caught out in the traffic
mess that occurred due to the anti-war protests - I could not get
to the meeting in time.

I hope that you all had a chance to look at the codec and the
profile anyway, and that if you have comments you will make them
to the list or privately.

Version 1.0 of the codec specification and implementation will be
released on Monday.  We ask that if you have concerns about the
actual codec portion (as opposed to the RTP Profile) that you
make those concerns known immediatley if you think it should
delay the 1.0 release.

Again, sorry I could not make it.  

Greg

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



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Subject: Re: [AVT] Speex: Apologies for Missed Meeting
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Greg,

> To those expecting my presentation yesterday on the Speex codec,
> I apologize for not being there.  I was caught out in the traffic
> mess that occurred due to the anti-war protests - I could not get
> to the meeting in time.

I did step through your slides so that the audience could read them,
although that is not the same as the presentation you would have
made.

> I hope that you all had a chance to look at the codec and the
> profile anyway, and that if you have comments you will make them
> to the list or privately.

One quick correction:  your document is a "payload format" not a
"profile".  The the introduction of the RTP spec for the explanation
of the difference.

                                                        -- Steve

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



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Date: Fri, 21 Mar 2003 18:55:15 -0800
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From: Ross Finlayson <finlayson@live.com>
Subject: Re: [AVT] Speex: Apologies for Missed Meeting
Cc: Greg Herlein <gherlein@herlein.com>
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Greg,

Sorry you couldn't make the meeting.  Here were some comments that I had 
planned to make about the Speex RTP packet format:

1/ It's quite common in audio payload formats to use the RTP 'M' bit to 
indicate the start of a talkspurt 
<http://www.cs.columbia.edu/~hgs/rtp/faq.html#marker>.  You might want to 
do the same (assuming, of course, that you plan to allow for senders to 
avoid sending packets during periods of silence).

2/ Having the number of frames per RTP packet indicated 'out of 
band'  (i.e., using SDP) is unorthodox, and rather inflexible.  The 
alternative, more common approach is to indicate that 'in band' - i.e., in 
the RTP packet itself.  Because Speex frames are variable-length (correct?) 
and don't have a 'sync word' to indicate their start (correct?), you could 
do this by having a 'packet length' descriptor precede each Speex frame in 
a RTP packet.

3/ Similarly, what happens if a Speex frame is too large to fit in a RTP 
packet?  Although Speex frames may be quite small, you should also allow 
for the possibility of RTP packets being even smaller.

To deal with 2/ and 3/, you might consider having a descriptor - preceding 
each Speex frame in a RTP packet - something like the following:

+-+-+-+-+-+-+-+-+
| S|E|D|    L    |
+-+-+-+-+-+-+-+-+

where:
- S (1 bit): Set to 1 iff the packet contains the start of a Speex frame 
(or one or more complete Speex frames)
- E (1 bit): Set to 1 iff the packet contains the end of a Speex frame (or 
one or more complete Speex frames)
- D (1 bit): Set to 0 if the descriptor is 1 octet (i.e., 8 bits) 
long.  Set to 1 if the descriptor is 2 octets (i.e., 16 bits) long.
- L (5 bits, or 13 bits, depending on "D"): The length of the following 
Speex frame.  As a special case, a value of zero could mean: The entire 
remainder of the RTP packet is a single Speex frame.  (Thus, if having just 
a single Speex frame per RTP packet is going to be the common case, then 
you could indicate this using just a single 0xC0 descriptor byte.)

	Ross.

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



From mailnull@www1.ietf.org  Sat Mar 22 10:15:39 2003
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Date: Sat, 22 Mar 2003 07:14:11 -0800 (PST)
From: Greg Herlein <gherlein@herlein.com>
To: Ross Finlayson <finlayson@live.com>
cc: Jean-Marc Valin <jean-marc.valin@hermes.usherb.ca>,
        Roger Hardiman <roger@telepresence.dmem.strath.ac.uk>,
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Subject: [AVT] Speex payload discussion
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The overwhelming design goal for the speex codec implementation
in RTP is "keep it as simple as possible" - at least so far.  I
know that the speex community will embrace anything that makes
sense though, so comments are very welcome.

> 1/ It's quite common in audio payload formats to use the RTP 'M' bit to 
> indicate the start of a talkspurt 
> <http://www.cs.columbia.edu/~hgs/rtp/faq.html#marker>.  You might want to 
> do the same (assuming, of course, that you plan to allow for senders to 
> avoid sending packets during periods of silence).

If speek is used in VBR with VAD enabled, then I don't think this
is needed.  The "silent frame" is only 5 bits long, so while it's
not "avoiding sending packets" it is certainly saving lots of
bandwidth.  And it's totally transparent - the codec library
takes care of it.  Marker bits used this way are only hints -
there is always the timestamp too.  And, relying on a bit in a
packet that may be lost seems unwise.  The codec deals with it
internally seemingly fine.  Why make it harder?

Sending such a "silent packet" is good too, since (assuming it
arrives) it tells the decoder that there is in facta silent frame
allowing it to play CN if it wants, and to keep the jitter
playback buffer happily syncronized.

> 2/ Having the number of frames per RTP packet indicated 'out
> of band' (i.e., using SDP) is unorthodox, and rather
> inflexible.  The alternative, more common approach is to
> indicate that 'in band' - i.e., in the RTP packet itself.  
> Because Speex frames are variable-length (correct?)  and
> don't have a 'sync word' to indicate their start (correct?),
> you could do this by having a 'packet length' descriptor
> precede each Speex frame in a RTP packet.

Well, that was argued at length on the speex list.  The normal
way to use speex will be one frame per packet.  That being the
case, sending an extra octet in every frame only adds overhead
that serves no useful purpose.  Most of the speex folks seemed in
favor of requiring out-of-band means to signal the unusual.

> 3/ Similarly, what happens if a Speex frame is too large to
> fit in a RTP packet?  Although Speex frames may be quite
> small, you should also allow for the possibility of RTP
> packets being even smaller.

Hmmmm.  We'll have to think about that.  I don't think you can
split a G.723.1 frame across multiple RTP packets, so why have
that capability in Speex?  The largest single Speex frame is 110
bytes long (that's 44kbps rate sampled at 32kHz) ... so even on
networks with small MTUs that should not be a problem.  The
nominal 8000kbps (8kHz sample rate) VoIP type frame would be only
20 bytes long.  I susepct that no MTU will ever be smaller than
that!

> 
> To deal with 2/ and 3/, you might consider having a descriptor - preceding 
> each Speex frame in a RTP packet - something like the following:

<snip>

Well, same concern as above - your suggestion has come up before
(remarkably similar to other proposals too - we all tend to think
a bit alike it seems) but is it really needed to add that
constant extra bits of overhead?

One argument is that it's *only* a single octet.  The other is
that any overhead not needed in a majority of the cases is
wasteful bloat.  

I'd love to hear comments from the wider AVT community on this.  

Greg

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



From mailnull@www1.ietf.org  Sat Mar 22 10:24:42 2003
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Subject: Re: [AVT] Speex payload discussion
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> If speek is used in VBR with VAD enabled, then I don't think this
> is needed.  The "silent frame" is only 5 bits long, so while it's
> not "avoiding sending packets" it is certainly saving lots of
> bandwidth.  And it's totally transparent - the codec library
> takes care of it.  Marker bits used this way are only hints -
> there is always the timestamp too.  And, relying on a bit in a
> packet that may be lost seems unwise.  The codec deals with it
> internally seemingly fine.  Why make it harder?

Because M bits are useful hints to other applications that may not know 
the details of the codec format. Thus, all voice codecs should use them 
unless there's a really, really good reason not to.

One of the design goals of RTP is to allow codec-unaware entities by 
providing hints about clock rates, silence periods and the like. I see 
no reason why Speex should deviate from this well-proven design.

Henning

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



From mailnull@www1.ietf.org  Sun Mar 23 16:41:20 2003
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Date: Sun, 23 Mar 2003 13:37:42 -0800 (PST)
From: Greg Herlein <gherlein@herlein.com>
To: Henning Schulzrinne <hgs@cs.columbia.edu>
cc: Ross Finlayson <finlayson@live.com>,
        Jean-Marc Valin <jean-marc.valin@hermes.usherb.ca>,
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        Simon Morlat <simon.morlat@linphone.org>, avt@ietf.org
Subject: Re: [AVT] Speex payload discussion
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> Because M bits are useful hints to other applications that may not know 
> the details of the codec format. Thus, all voice codecs should use them 
> unless there's a really, really good reason not to.

Well, I would argue (with all due respect) that other
applications that don't know how to decode a codec should not set
up that stream in the first place.  :)  I would argue that the
marker bit is defined in the codec profile and if you don't know
the details of the codec, you probably won't know what the marker
bit means.  It might mean something else entirely. :)

But, seriously, RFC1890 states: 

"For applications which send no packets during silence, the first
packet of a talkspurt (first packet after a silence period) is
distinguished by setting the marker bit in the RTP data header.  
Applications without silence suppression set the bit to zero.' "

It turns out that using libspeex it's implementation-specific
whether you send 5-bit frames (padded to one octet) during
silence, or whether you send nothing during silence.  You *do*
get the 5 bit value in the byte returned when you encode the
frame.  However, contrary to at least my understanding, you do
*NOT* have to send that frame for libspeex to happily cruise
along on the decoder side.  If you don't send it, libspeex still
does the right thing.  If you do send it, libspeex still does the
right thing.

But, to retain the behavior of other "well-proven designs" I
propose that we add language to the speex payload format that
states that if an implementaion sends the 5-bit silence frame (as
a one octet payload) that it SHOULD set the marker bit in the
first subsequent packet that contains valid audio (non-silence)
data.  If the implementation does not send silence frames, then
it MUST set the marker bit in the first subsequent packet that
contains valid audio (non-silence) data.

> One of the design goals of RTP is to allow codec-unaware entities by 
> providing hints about clock rates, silence periods and the like. I see 
> no reason why Speex should deviate from this well-proven design.

I certainly did not intend to deviate from anything.  I was only
saying that if we send a (short) frame that is effectively a
placeholder meaning "silence" then there was no need for the
marker bit, and if there was no need, then it would be redundant
to require it to be included.  But, if it breaks other stuff to
not set that bit, then by all means let's state it must be set.

Greg

_______________________________________________
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From mailnull@www1.ietf.org  Sun Mar 23 16:58:10 2003
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Among other reasons, the M bit allows codec-unaware applications to 
perform better playout delay adaptation (since you want to do those 
during silence, not talk spurts). Thus, I'm glad to hear that Speex will 
make this possible.

Also, this is *not* an 'extra' bit. You already paid for it :-)


> I certainly did not intend to deviate from anything.  I was only
> saying that if we send a (short) frame that is effectively a
> placeholder meaning "silence" then there was no need for the
> marker bit, and if there was no need, then it would be redundant

Assuming you know the codec. My premise was that you didn't.

> to require it to be included.  But, if it breaks other stuff to
> not set that bit, then by all means let's state it must be set.
> 
> Greg

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I haven't got the full context, so I hope what I'm saying makes sense.

> If speek is used in VBR with VAD enabled, then I don't think this
> is needed.  The "silent frame" is only 5 bits long, so while it's
> not "avoiding sending packets" it is certainly saving lots of
> bandwidth.  And it's totally transparent - the codec library
> takes care of it.  Marker bits used this way are only hints -
> there is always the timestamp too.  And, relying on a bit in a
> packet that may be lost seems unwise.  The codec deals with it
> internally seemingly fine.  Why make it harder?

Actually, the "silence frames" can only happen when DTX is enabled (just
VBR or VAD is not enough). Also, while in Ogg files they are encoded
with 5 bits, in network applications, there's really _no transmission_.
Note also that the way it works is that there's already a flag in the
last packet before silence that says "from now on, if you don't recieve
anything, it's noise not packet loss". BTW, although for now we're not
using the marker bit, it may still be good to set it.

> Well, that was argued at length on the speex list.  The normal
> way to use speex will be one frame per packet.  That being the
> case, sending an extra octet in every frame only adds overhead
> that serves no useful purpose.  Most of the speex folks seemed in
> favor of requiring out-of-band means to signal the unusual.

I think I agree on your point. While if would be possible to get clever
tricks to find the number of packets from the packet length, I don't
think it's really clean.

	Jean-Marc

--=20
Jean-Marc Valin, M.Sc.A.
LABORIUS (http://www.gel.usherb.ca/laborius)
Universit=E9 de Sherbrooke, Qu=E9bec, Canada


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Article today on www.slashdot.org

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Slashdot article today...
 
 http://www.xiph.org/press/2003/nonprofitspeex1/

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To: Greg Herlein <gherlein@herlein.com>
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Subject: Re: [AVT] Speex payload discussion 
In-Reply-To: Your message of "Sat, 22 Mar 2003 07:14:11 PST."
             <Pine.LNX.4.21.0303220643250.32171-100000@tux.herlein.com> 
Date: Mon, 24 Mar 2003 14:47:55 -0500
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A few comments inline - I'll skip the M bit issue, since I agree with
Henning's comments.

--> Greg Herlein writes:
...
>> 2/ Having the number of frames per RTP packet indicated 'out
>> of band' (i.e., using SDP) is unorthodox, and rather
>> inflexible.  The alternative, more common approach is to
>> indicate that 'in band' - i.e., in the RTP packet itself.  
>> Because Speex frames are variable-length (correct?)  and
>> don't have a 'sync word' to indicate their start (correct?),
>> you could do this by having a 'packet length' descriptor
>> precede each Speex frame in a RTP packet.
>
>Well, that was argued at length on the speex list.  The normal
>way to use speex will be one frame per packet.  That being the
>case, sending an extra octet in every frame only adds overhead
>that serves no useful purpose.  Most of the speex folks seemed in
>favor of requiring out-of-band means to signal the unusual.

The EVRC payload format addresses this by defining a simple format where
only a single frame is included in the packet, and a more complex format
which has a header and multiple frames. The signalling then selects which
of the two formats is used. This may be an acceptable solution here, too.

Alternatively, if you can tell the length of the frame by parsing the frame
data, there is no need to signal the number of frames per packet.

>> 3/ Similarly, what happens if a Speex frame is too large to
>> fit in a RTP packet?  Although Speex frames may be quite
>> small, you should also allow for the possibility of RTP
>> packets being even smaller.
>
>Hmmmm.  We'll have to think about that.  I don't think you can
>split a G.723.1 frame across multiple RTP packets, so why have
>that capability in Speex?  The largest single Speex frame is 110
>bytes long (that's 44kbps rate sampled at 32kHz) ... so even on
>networks with small MTUs that should not be a problem.  The
>nominal 8000kbps (8kHz sample rate) VoIP type frame would be only
>20 bytes long.  I susepct that no MTU will ever be smaller than
>that!

Given the small size of the packets, it's probably acceptable to leave
fragementation to the IP layer. 

Cheers,
Colin
_______________________________________________
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Date: Mon, 24 Mar 2003 12:05:13 -0800 (PST)
From: Greg Herlein <gherlein@herlein.com>
To: Colin Perkins <csp@isi.edu>
cc: Ross Finlayson <finlayson@live.com>,
        Jean-Marc Valin <jean-marc.valin@hermes.usherb.ca>,
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Subject: Re: [AVT] Speex payload discussion 
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> The EVRC payload format addresses this by defining a simple format where
> only a single frame is included in the packet, and a more complex format
> which has a header and multiple frames. The signalling then selects which
> of the two formats is used. This may be an acceptable solution here, too.

Is it *that* bad a thing to just let the signalling define the
number of frames in a packet in the first place, the way it's
currently defined?

> Alternatively, if you can tell the length of the frame by parsing the frame
> data, there is no need to signal the number of frames per packet.

That's not possible, unfortunately.

Greg

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        Roger Hardiman <roger@telepresence.dmem.strath.ac.uk>,
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Subject: Re: [AVT] Speex payload discussion 
In-Reply-To: Your message of "Mon, 24 Mar 2003 12:05:13 PST."
             <Pine.LNX.4.21.0303241204020.23141-100000@tux.herlein.com> 
Date: Mon, 24 Mar 2003 15:55:01 -0500
From: Colin Perkins <csp@isi.edu>
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--> Greg Herlein writes:
>> The EVRC payload format addresses this by defining a simple format where
>> only a single frame is included in the packet, and a more complex format
>> which has a header and multiple frames. The signalling then selects which
>> of the two formats is used. This may be an acceptable solution here, too.
>
>Is it *that* bad a thing to just let the signalling define the
>number of frames in a packet in the first place, the way it's
>currently defined?

No, it's just a little unusual.  If there's no reason why a sender would
want to change the number of frames per packet, then signaling it out of
band is entirely appropriate. 

Colin
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From: Timur Friedman <timur.friedman@lip6.fr>
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Subject: Re: [AVT] Comments on draft-ietf-avt-rtcp-report-extns-03.txt
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Magnus Westerlund wrote :
> 24. Section A.1: Why is the extended sequence number initiated to the
> middle of the space? Normally one would start with an extended sequence
> number of 0. Shouldn't one try to be consistent with extended highest
> sequence number received in the receiver reporting.

For the detailed packet traces, reported upon by the Loss RLE blocks defined 
in the RTCP Extended Reports draft, we wish to make as few assumptions as 
possible regarding the packets that are observed.

In particular, we wish to avoid making the assumptions made for basic RTCP: 
for instance, that a certain number of packets must be received in order for 
a source to be considered valid, or that a certain number of packets must be 
received in order to consider that a large jump in sequence numbers has 
occurred.  These assumptions aid the functionality of RTCP, helping avoid 
errors in the receiver count, for instance.  But they might mask the sort of 
anomalous events that one would wish to catch with extended reports.

To address the particular question raised above, it is entirely possible that 
the sequence numbers could jump backwards.  This could result in rollover in 
the 16-bit RTP sequence number space.  To avoid rollover in the larger 32-bit 
extended sequence number space, we suggest starting roughly in the middle of 
that space.

- Timur Friedman

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To: Greg Herlein <gherlein@herlein.com>
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        Roger Hardiman
 <roger@telepresence.dmem.strath.ac.uk>,
        Simon Morlat
 <simon.morlat@linphone.org>, avt@ietf.org
Subject: Re: [AVT] Speex payload discussion
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Colin Perkins wrote:

>--> Greg Herlein writes:
>  
>
>>>The EVRC payload format addresses this by defining a simple format where
>>>only a single frame is included in the packet, and a more complex format
>>>which has a header and multiple frames. The signalling then selects which
>>>of the two formats is used. This may be an acceptable solution here, too.
>>>      
>>>
>>Is it *that* bad a thing to just let the signalling define the
>>number of frames in a packet in the first place, the way it's
>>currently defined?
>>    
>>
>
>No, it's just a little unusual.  If there's no reason why a sender would
>want to change the number of frames per packet, then signaling it out of
>band is entirely appropriate. 
>  
>

Yes, but there normally exist one very good reason why one wants to 
change the number of frames per packet, congestion control. For most low 
bit-rate codecs the simplest way of reducing the bit-rate is simple to 
reduce the number of packets sent. As the IP/UDP/RTP headers often 
represents a bit-rate  of one or more complete audio streams. By sending 
two instead of one frame per packet the rate can often be reduced by 
30%. However as this trade off often comes at the expense of less 
robustness and increased end-to-end delay one tries to avoid it if 
possible. Therefore it is reasonable to be able to change this on the 
fly. However if the Speex codec contains other mechanisms that are 
useful to change the bit-rate then it might not be necessary.

An alternative way if to actually define different payload types when 
there might be need for different packetizations.

Another point, if you have variable frame length how does one 
distinguish the end of one frame from the beginning of the next? If you 
don't have such a difference it will not be possible to demultiplex at 
the receiver.

Best Regards

Magnus Westerlund 

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



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



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Subject: [AVT] SDP signaling for RTCP XR format
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Hi,

Below is an proposal for an chapter on SDP signaling in the RTCP XR 
format. Please provide any comments as soon as possible. This will be 
included in the next revision of the draft.

Best Regards

Magnus Westerlund

------------------------------------------------------

5. SDP Signaling

   This section defines a SDP attribute that can be used for signaling
   in applications utilizing SDP [s3]. This is signaling is defined to
   be used with Offer-Answer [s1] and for applications using SDP to
   describe media configurations, like in SAP [s4] and RTSP [s5]. There
   exist other potential signaling methods, which are not defined here.

   The RTCP XR formats MAY be used without prior signaling. This is in
   line with other RTCP packet types and follows the normal rules as
   written in [7]. All the reporting blocks defined in this document can
   be used without prior signaling. If any future defined report block
   does not allow usage without signaling, so should be noted and
   motivated in its specification.

5.1 The SDP attribute

   This section defines one new SDP attribute "rtcp-xr" that can be used
   to signal the participants in a media session that it should utilize
   any specified reporting format(s). This attribute can in the future
   be easily extended with new parameters to cover any new reporting
   formats.

   The RTCP XR format SDP attribute is defined as below in ABNF [s2]. It
   is an session and media level attribute. When specified at session
   level it applies to all media blocks part of the session. Any media
   level specification MUST replace an session level specification, if
   present, for that media block.

   rtcp-xr-attrib = "a=" "rtcp-xr" ":" xr-format *(SP xr-format) CRLF

   xr-format = pckt-loss-rle
             / dupp-rle
             / rcv-ts
             / stat-sum
             / rcv-rtt
             / voip-stat
             / format-ext

   pckt-loss-rle = "rle-pckt" ["=" 1*DIGIT]
   dupp-rle      = "rle-dup" ["=" 1*DIGIT]
   rcv-ts        = "recv-ts" ["=" 1*DIGIT]
   stat-sum      = "stat-sum"
   recv-rtt      = "recv-rtt"
   voip-stat     = "voip-sum"
   format-ext    = non-ws-string

   non-ws-string = 1*(%x21-FF)

   The "rtcp-xr" attribute contains one or more XR format related
   parameters. Each parameter defines its own interpretation as this
   will depend on the format(s). The attribute is extensible so that any
   future XR format block can define its own parameter(s). There may be
   formats where a "rtcp-xr" parameter is not applicable. However, for
   any future defined formats the need of a parameter should be
   evaluated.

   The defined "rtcp-xr" parameters belong to two categories. The first
   category corresponds to reporting formats that simple reports on the
   RTP stream and related things. The second category is formats that
   perform some type of functionality, like the RTT measuring
   functionality for a RTP receiver. This functionality uses two
   different XR formats, that together provide the functionality.

   For category 1, the reporting formats, this document defines the
   following parameters. The parameter names and their corresponding XR
   format is given in table 1.

   Parameter name    XR format (Nr and Name)
   --------------------------
   rle-pckt          1  Loss RLE Report Block
   rle-dup           2  Duplicate RLE Report Block
   recv-ts           3  Timestamp Report Block
   stat-sum          4  Statistics Summary Report Block
   voip-sum          7  VoIP Metrics Report Block

   Table 1. The category 1 parameters and their XR formats.


   The "rle-pckt", "rle-dup", and "recv-ts" parameters MAY all take
   specify an integer value. This value indicates the largest size the
   whole report block SHOULD have in bytes. This shall be seen as an
   indication that thinning shall be applied if necessary to meet the
   target size.

   Category 2, consist of the RTT measuring formats using the XR formats
   "Receiver Timestamp Report Block" (5) and "DLRR Report Block" (6).
   These two formats and their functionality is represented by the
   "recv-rtt" parameter.

5.2 Usage in Offer-Answer

   When having reporting formats, it is in the interest of the sender of
   media, to request that the receiver sends reports on this media
   stream. However the interpretation will be dependent on any direction
   attribute. In the below definition, text applying for media stream
   with "sendrecv" attributes does also apply to media streams without
   any direction attribute. The definitions apply for media streams with
   a unicast destination, unless otherwise noted.

   For media streams in offers with a "sendonly" or a "sendrecv"
   attribute the inclusion of "rtcp-xr" attributes with parameters
   belonging to category 1, has the interpretation that the receiver of
   the offer SHOULD send the XR formats corresponding to the parameter
   to the sender of these streams. When doing the answer the answerer
   MAY include the rtcp-xr attribute with category 1 parameters for
   media streams with the "sendrecv" attribute to request that the
   offerer SHOULD send the corresponding XR formats from the offerer to
   the answerer.

   For media streams in offers with "recvonly" attribute the offerer MAY
   include a parameter to declare its capability of sending the
   corresponding XR formats. If the answerer includes a "rtcp-xr"
   attribute with an category 1 parameter the offerer SHOULD send the
   corresponding XR formats.

   For media streams with multicast destinations the inclusion of an
   rtcp-xr attribute with category 1 parameters means that any media
   recipient SHOULD send the corresponding XR formats in RTCP.

   For an SDP offer with media streams containing the "rtcp-xr"
   attribute with the "recv-rtt" parameter the offerer declares itself
   capable of receiving the "Receiver Timestamp Report Block" and send
   responses using the "DLRR Report Block". The answerer MAY also
   include the parameter (recv-rtt) and thereby declare its capability
   to receive and respond to the RTT measuring XR formats. The inclusion
   of the parameter does not require the receiver of an offer or answer
   to actually send any "Receiver Timestamp Report Block". Note that the
   usage of the Receiver Timestamp Report Block is only useful for a
   receiver only, as any media sender will have this functionality in
   the RTCP Sender Report packet type.

   For an SDP offer with media streams with multicast destinations
   containing the "rtcp-xr" attribute with the "recv-rtt" parameter the
   participants SHOULD respond with a "DLRR Report Block" to any
   received "Receiver Timestamp Report Block". A participant SHOULD NOT
   respond to a "Receiver Timestamp Report Block" if the parameter are
   not present in the offer. The answerer SHALL NOT add the "recv-rtt"
   parameter in a "rtcp-xr" attribute in any response for a media stream
   with multicast destination.


5.3 Usage out-side of Offer/Answer

   This applies to the usage of SDP that is currently defined in SAP and
   RTSP where SDP is used to declare what applies for a multi-media
   presentation. This SDP applies to how a receiver of these media
   streams shall behave to receive the media and under what rules.
   Therefore the usage of the "rtcp-xr" format is simpler in this case
   then compared to offer/answer, however the functionality goal is the
   same.

   For media streams having the "rtcp-xr" attribute with any category 1
   parameter the receiver of this media stream SHOULD send the
   corresponding reports. If the attribute contains the "recv-rtt"
   parameter the recipient of the media stream MAY send "Receiver
   Timestamp Report Block". The receiver of an "Receiver Timestamp
   Report Block" SHOULD send a "DLRR Report Block" in response if and
   only if the "recv-rtt" parameter is present.


6. IANA Considerations

   This document defines a new RTCP packet type, the extended report
   (XR) type, within the existing Internet Assigned Numbers Authority
   (IANA) registry of RTP RTCP Control Packet Types.  This document also
   defines a new IANA registry: the registry of RTP XR Block Types.
   Within this new registry, this document defines an initial set of
   seven block types and describes how the remaining types are to be
   allocated.  Further this document defines a new SDP attribute
   "a=rtcp-xr" to be registered, and for its parameters a IANA registry
   is defined.

6.1 XR Packet Type

   The IANA SHALL register the RTP extended report (XR) packet defined
   by this document as packet type 207 in the registry of RTP RTCP
   Control Packet types (PT).

6.2 RTP XR Block Type Registry

   The IANA SHALL create the RTP XR Block Type Registry to cover the
   name space of the extended report block type (BT) field specified in
   Section 3 of this document.  The BT field contains eight bits,
   allowing 256 values.  The IANA SHALL manage the RTP XR Block Type
   Registry according to the Specification Required policy of RFC 2434
   [6].  Future specifications SHOULD attribute block type values in
   strict numeric order following the values attributed in this
   document:


      BT  name
      --  ----
       1  Loss RLE Report Block
       2  Duplicate RLE Report Block
       3  Timestamp Report Block
       4  Statistics Summary Report Block
       5  Receiver Timestamp Report Block
       6  DLRR Report Block
       7  VoIP Metrics Report Block


   The BT value 255 is reserved for future extensions.

   Furthermore, future specifications SHOULD avoid the value 0.  Doing
   so facilitates packet validity checking, since an all-zeros field
   might commonly be found in an ill-formed packet.

6.3 The "rtcp-xr" SDP attribute

   One SDP attribute is to be registered using the below information:

   SDP Attribute ("att-field"):

     Attribute name:     rtcp-xr
     Long form:          RTP Extended Report Parameters
     Type of name:       att-field
     Type of attribute:  Session and Media level
     Subject to charset: No
     Purpose:            See this document
     Reference:          This document
     Values:             See this document and registrations below

   The attribute has an extensible parameter field and therefore a
   registry for these parameters are needed. IANA is requested to create
   one RTCP-XR SDP parameter registry and register the defined values.
   This registry is managed according to the  Specification Required
   policy of RFC 2434. Any registration MUST contain the following
   information:

   - Contact information of the one doing the registration, including at
   least name, address, and email.

   - An ABNF definition of the parameter, in accordance with the
   "format-ext" definition.

   - A description of what the parameter represents and how it shall be
   interpreted, both normally and in Offer/Answer.

   This document registers the six parameters defined in chapter 5.1,
   "rle-pckt", "rle-dup", "recv-ts", "stat-sum", "voip-sum", and "recv-
   rtt".


7. Security Considerations

   This document extends the RTCP reporting mechanism.  The security
   considerations that apply to RTCP reports also apply to XR reports.
   This section details the additional security considerations that
   apply to the extensions.

   The extensions introduce heightened confidentiality concerns.
   Standard RTCP reports contain a limited number of summary statistics.
   The information contained in XR reports is both more detailed and
   more extensive (covering a larger number of parameters).  The per-
   packet information contained in Loss RLE, Duplicate RLE, and
   Timestamp Report Blocks facilitates MINC inference of multicast
   distribution trees for RTP data packets, and inference of link
   characteristics such as loss and delay.  This inference reveals
   information that might otherwise be considered confidential to
   autonomous system administrators.  The VoIP Metrics Report Block
   provides information on the quality of ongoing voice calls.  Though
   such information might be carried in application specific format in
   standard RTP sessions, making it available in a standard format here
   makes it more available to potential eavesdroppers.

   No new mechanisms are introduced in this document to ensure
   confidentiality.  Standard encryption procedures can be used when
   confidentiality is a concern to end hosts.  Autonomous system
   administrators concerned about the loss of confidentiality regarding
   their networks can encrypt traffic, or filter it to exclude RTCP
   packets containing the XR report blocks concerned.

   Any encryption or filtering of XR report blocks entails a loss of
   monitoring information to third parties.  For example, a network that
   establishes a tunnel to encrypt VoIP Report Blocks denies that
   information to the service providers traversed by the tunnel.  The
   service providers cannot then monitor or respond to the quality of
   the VoIP calls that they carry, potentially creating problems for the
   network's users.  As a default, XR packets SHOULD NOT be encrypted or
   filtered.

   The extensions also make certain denial of service attacks easier.
   This is because of the potential to create RTCP packets much larger
   than average with the per packet reporting capabilities of the Loss
   RLE, Duplicate RLE, and Timestamp Report Blocks.  Because of the
   automatic bandwidth adjustment mechanisms in RTCP, if some session
   participants are sending large RTCP packets, all participants will
   see their RTCP reporting intervals lengthened, meaning they will be
   able to report less frequently.  To limit the effects of large
   packets, even in the absence of denial of service attacks,
   applications SHOULD limit the size of XR report blocks and employ the
   thinning techniques described in this document in order to fit
   reports into the space available.

   The usage of the SDP parameters defined affect the behavior of the
   RTCP XR format blocks. However the security implications are
   dependent on the report formats. Due to the fact that behavior can be
   changed by the SDP signaling, the SDP is RECOMMENDED to be
   authenticate to prevent tampering by intermediates.


References

   The references are divided into normative references and non-
   normative references.

Normative References

   [1] S. Bradner, "Key words for use in RFCs to indicate requirement
   levels," BCP 14, RFC 2119, IETF, March 1997.

   [s2] D. Crocker, P. Overell, "Augmented BNF for Syntax
   Specifications: ABNF", RFC 2234, Internet Engineering Task Force,
   November 1997.

   [2] ETSI, "Quality of Service (QoS) measurement methodologies," ETSI
   TS 101 329-5 V1.1.1 (2000-11), November 2000.

   [s3] M. Handley, V. Jacobson, "SDP: Session Description Protocol",
   RFC 2327, Internet Engineering Task Force, April 1998.

   [3] R. Hovey and S. Bradner, "The Organizations Involved in the IETF
   Standards Process," best current practice BCP 11, RFC 2028, IETF,
   October 1996.

   [4] ITU-T, "The E-Model, a computational model for use in
   transmission planning," Recommendation G.107 (05/00), May 2000.

   [5] J. Reynolds and J. Postel, "Assigned Numbers," standard STD 2,
   RFC 1700, IETF, October 1994.

   [6] T. Narten and H. Alvestrand, "Guidelines for Writing an IANA
   Considerations Section in RFCs," best current practice BCP 26, RFC
   2434, IETF, October 1998.

   [s1] J. Rosenberg, H. Schulzrinne, "An Offer/Answer Model with the
   Session Description Protocol (SDP)", RFC 3264, Internet Engineering
   Task Force, June 2002.

   [7] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A
   transport protocol for real-time applications," RFC 1889, IETF,
   February 1996.

Non-Normative References

   [8] A. Adams, T. Bu, R. Caceres, N. G. Duffield, T. Friedman, J.
   Horowitz, F. Lo Presti, S. B. Moon, V. Paxson, and D. Towsley, "The
   Use of End-to-End Multicast Measurements for Characterizing Internal
   Network Behavior," IEEE Communications Magazine, May 2000.

   [9] R. Caceres, N. G. Duffield, and T. Friedman, "Impromptu
   measurement infrastructures using RTP," Proc. IEEE Infocom 2002.

   [10] A. D. Clark, "Modeling the Effects of Burst Packet Loss and
   Recency on Subjective Voice Quality," Proc. IP Telephony Workshop
   2001.

   [s4] M. Handley, C. Perkins, E. Whelan, "Session Announcement
   Protocol", RFC 2974, Internet Engineering Task Force, October 2000.

   [s5] H. Schulzrinne, R. Lanphier, and A. Rao, "Real time streaming
   protocol (RTSP)," RFC 2326, Internet Engineering Task Force, Apr.
   1998.

   [11] D. Sisalem and A. Wolisz, "MLDA: A TCP-friendly Congestion
   Control Framework for Heterogeneous Multicast Environments", Proc.



-- 

Magnus Westerlund 

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


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



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Date: Wed, 26 Mar 2003 17:07:30 +0100
From: Giuseppe Giordano <Giuseppe.Giordano@italtel.it>
Organization: Italtel
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Hi, all
I have a doubt about RTCP.
Rfc 1889 says that RTCP reports should be sent even if
a endpoint is receive only, or sendonly, that is in the case of a
monodirectional
connection.
But,
if I created a connection between 2 media gateways (unicast RTP session)
by means of
MGCP protocol in the following way:

1) CRCX message  to the first gateway (endpoint A)
cx mode = rcvonly
remote session description: absent

2) Answer to the CRCX message with endpoin A local session description

3) CRCX message to the second gateway (endpoint B)
cx mode = sendonly
remote session description: present

In this case, endpoint A must send RTCP reports to B?
Endpoint A receives RTP packets from B. Can it use the RTP sender IP
address
to deliver RTCP packets?

Thanks for the attention,
best regards,

Giuseppe Giordano


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Date: Thu, 27 Mar 2003 16:23:24 +0100
From: Jose Rey <rey@panasonic.de>
Subject: RE: [AVT] SDP signaling for RTCP XR format
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Hi Magnus,

I've got some comments/questions on your proposal...

1.- The term "RTCP XR format" to address the Loss RLE Report Block, Duplicate RLE ..., is not consistent with the rest of the draft. I prefer "XR blocks" which is used throghout the text.
2.- Just curious, why are the report blocks used without signalling? It would be helpful to include the reason for this in the draft (if it's not there already and I missed it)
3.- I like your classification of parameters because is needed afterwards to define the usage, however I don't understand your reasoning since both types of parameters define a functionality. Why not rename it to unilateral (category 1) and collaborative (category 2) or something alike? 
4.- Section 5.2 par. 5: This is a bit confusing. The receiver may declare its capability for the XR blocks and not send the report blocks themselves? Shouldn't it be that by declaring its capability is also declaring its willingness to use them? Or is it maybe because the receiver decides itself when to send (them dynamically during the session) them and thus might be that they are not needed at all? Does this make sense? Maybe is the text that confuses me.


thanks,

JosÃ©

> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf 
> Of Magnus
> Westerlund
> Sent: Wednesday, March 26, 2003 4:36 PM
> To: IETF AVT WG
> Subject: [AVT] SDP signaling for RTCP XR format
> 
> 
> Hi,
> 
> Below is an proposal for an chapter on SDP signaling in the RTCP XR 
> format. Please provide any comments as soon as possible. This will be 
> included in the next revision of the draft.
> 
> Best Regards
> 
> Magnus Westerlund
> 
> ------------------------------------------------------
> 
> 5. SDP Signaling
> 
>    This section defines a SDP attribute that can be used for signaling
>    in applications utilizing SDP [s3]. This is signaling is defined to
>    be used with Offer-Answer [s1] and for applications using SDP to
>    describe media configurations, like in SAP [s4] and RTSP 
> [s5]. There
>    exist other potential signaling methods, which are not 
> defined here.
> 
>    The RTCP XR formats MAY be used without prior signaling. This is in
>    line with other RTCP packet types and follows the normal rules as
>    written in [7]. All the reporting blocks defined in this 
> document can
>    be used without prior signaling. If any future defined report block
>    does not allow usage without signaling, so should be noted and
>    motivated in its specification.
> 
> 5.1 The SDP attribute
> 
>    This section defines one new SDP attribute "rtcp-xr" that 
> can be used
>    to signal the participants in a media session that it 
> should utilize
>    any specified reporting format(s). This attribute can in the future
>    be easily extended with new parameters to cover any new reporting
>    formats.
> 
>    The RTCP XR format SDP attribute is defined as below in 
> ABNF [s2]. It
>    is an session and media level attribute. When specified at session
>    level it applies to all media blocks part of the session. Any media
>    level specification MUST replace an session level specification, if
>    present, for that media block.
> 
>    rtcp-xr-attrib = "a=" "rtcp-xr" ":" xr-format *(SP xr-format) CRLF
> 
>    xr-format = pckt-loss-rle
>              / dupp-rle
>              / rcv-ts
>              / stat-sum
>              / rcv-rtt
>              / voip-stat
>              / format-ext
> 
>    pckt-loss-rle = "rle-pckt" ["=" 1*DIGIT]
>    dupp-rle      = "rle-dup" ["=" 1*DIGIT]
>    rcv-ts        = "recv-ts" ["=" 1*DIGIT]
>    stat-sum      = "stat-sum"
>    recv-rtt      = "recv-rtt"
>    voip-stat     = "voip-sum"
>    format-ext    = non-ws-string
> 
>    non-ws-string = 1*(%x21-FF)
> 
>    The "rtcp-xr" attribute contains one or more XR format related
>    parameters. Each parameter defines its own interpretation as this
>    will depend on the format(s). The attribute is extensible 
> so that any
>    future XR format block can define its own parameter(s). 
> There may be
>    formats where a "rtcp-xr" parameter is not applicable. However, for
>    any future defined formats the need of a parameter should be
>    evaluated.
> 
>    The defined "rtcp-xr" parameters belong to two categories. 
> The first
>    category corresponds to reporting formats that simple 
> reports on the
>    RTP stream and related things. The second category is formats that
>    perform some type of functionality, like the RTT measuring
>    functionality for a RTP receiver. This functionality uses two
>    different XR formats, that together provide the functionality.
> 
>    For category 1, the reporting formats, this document defines the
>    following parameters. The parameter names and their 
> corresponding XR
>    format is given in table 1.
> 
>    Parameter name    XR format (Nr and Name)
>    --------------------------
>    rle-pckt          1  Loss RLE Report Block
>    rle-dup           2  Duplicate RLE Report Block
>    recv-ts           3  Timestamp Report Block
>    stat-sum          4  Statistics Summary Report Block
>    voip-sum          7  VoIP Metrics Report Block
> 
>    Table 1. The category 1 parameters and their XR formats.
> 
> 
>    The "rle-pckt", "rle-dup", and "recv-ts" parameters MAY all take
>    specify an integer value. This value indicates the largest size the
>    whole report block SHOULD have in bytes. This shall be seen as an
>    indication that thinning shall be applied if necessary to meet the
>    target size.
> 
>    Category 2, consist of the RTT measuring formats using the 
> XR formats
>    "Receiver Timestamp Report Block" (5) and "DLRR Report Block" (6).
>    These two formats and their functionality is represented by the
>    "recv-rtt" parameter.
> 
> 5.2 Usage in Offer-Answer
> 
>    When having reporting formats, it is in the interest of 
> the sender of
>    media, to request that the receiver sends reports on this media
>    stream. However the interpretation will be dependent on 
> any direction
>    attribute. In the below definition, text applying for media stream
>    with "sendrecv" attributes does also apply to media streams without
>    any direction attribute. The definitions apply for media 
> streams with
>    a unicast destination, unless otherwise noted.
> 
>    For media streams in offers with a "sendonly" or a "sendrecv"
>    attribute the inclusion of "rtcp-xr" attributes with parameters
>    belonging to category 1, has the interpretation that the 
> receiver of
>    the offer SHOULD send the XR formats corresponding to the parameter
>    to the sender of these streams. When doing the answer the answerer
>    MAY include the rtcp-xr attribute with category 1 parameters for
>    media streams with the "sendrecv" attribute to request that the
>    offerer SHOULD send the corresponding XR formats from the 
> offerer to
>    the answerer.
> 
>    For media streams in offers with "recvonly" attribute the 
> offerer MAY
>    include a parameter to declare its capability of sending the
>    corresponding XR formats. If the answerer includes a "rtcp-xr"
>    attribute with an category 1 parameter the offerer SHOULD send the
>    corresponding XR formats.
> 
>    For media streams with multicast destinations the inclusion of an
>    rtcp-xr attribute with category 1 parameters means that any media
>    recipient SHOULD send the corresponding XR formats in RTCP.
> 
>    For an SDP offer with media streams containing the "rtcp-xr"
>    attribute with the "recv-rtt" parameter the offerer declares itself
>    capable of receiving the "Receiver Timestamp Report Block" and send
>    responses using the "DLRR Report Block". The answerer MAY also
>    include the parameter (recv-rtt) and thereby declare its capability
>    to receive and respond to the RTT measuring XR formats. 
> The inclusion
>    of the parameter does not require the receiver of an offer 
> or answer
>    to actually send any "Receiver Timestamp Report Block". 
> Note that the
>    usage of the Receiver Timestamp Report Block is only useful for a
>    receiver only, as any media sender will have this functionality in
>    the RTCP Sender Report packet type.
> 
>    For an SDP offer with media streams with multicast destinations
>    containing the "rtcp-xr" attribute with the "recv-rtt" 
> parameter the
>    participants SHOULD respond with a "DLRR Report Block" to any
>    received "Receiver Timestamp Report Block". A participant 
> SHOULD NOT
>    respond to a "Receiver Timestamp Report Block" if the parameter are
>    not present in the offer. The answerer SHALL NOT add the "recv-rtt"
>    parameter in a "rtcp-xr" attribute in any response for a 
> media stream
>    with multicast destination.
> 
> 
> 5.3 Usage out-side of Offer/Answer
> 
>    This applies to the usage of SDP that is currently defined 
> in SAP and
>    RTSP where SDP is used to declare what applies for a multi-media
>    presentation. This SDP applies to how a receiver of these media
>    streams shall behave to receive the media and under what rules.
>    Therefore the usage of the "rtcp-xr" format is simpler in this case
>    then compared to offer/answer, however the functionality 
> goal is the
>    same.
> 
>    For media streams having the "rtcp-xr" attribute with any 
> category 1
>    parameter the receiver of this media stream SHOULD send the
>    corresponding reports. If the attribute contains the "recv-rtt"
>    parameter the recipient of the media stream MAY send "Receiver
>    Timestamp Report Block". The receiver of an "Receiver Timestamp
>    Report Block" SHOULD send a "DLRR Report Block" in response if and
>    only if the "recv-rtt" parameter is present.
> 
> 
> 6. IANA Considerations
> 
>    This document defines a new RTCP packet type, the extended report
>    (XR) type, within the existing Internet Assigned Numbers Authority
>    (IANA) registry of RTP RTCP Control Packet Types.  This 
> document also
>    defines a new IANA registry: the registry of RTP XR Block Types.
>    Within this new registry, this document defines an initial set of
>    seven block types and describes how the remaining types are to be
>    allocated.  Further this document defines a new SDP attribute
>    "a=rtcp-xr" to be registered, and for its parameters a 
> IANA registry
>    is defined.
> 
> 6.1 XR Packet Type
> 
>    The IANA SHALL register the RTP extended report (XR) packet defined
>    by this document as packet type 207 in the registry of RTP RTCP
>    Control Packet types (PT).
> 
> 6.2 RTP XR Block Type Registry
> 
>    The IANA SHALL create the RTP XR Block Type Registry to cover the
>    name space of the extended report block type (BT) field 
> specified in
>    Section 3 of this document.  The BT field contains eight bits,
>    allowing 256 values.  The IANA SHALL manage the RTP XR Block Type
>    Registry according to the Specification Required policy of RFC 2434
>    [6].  Future specifications SHOULD attribute block type values in
>    strict numeric order following the values attributed in this
>    document:
> 
> 
>       BT  name
>       --  ----
>        1  Loss RLE Report Block
>        2  Duplicate RLE Report Block
>        3  Timestamp Report Block
>        4  Statistics Summary Report Block
>        5  Receiver Timestamp Report Block
>        6  DLRR Report Block
>        7  VoIP Metrics Report Block
> 
> 
>    The BT value 255 is reserved for future extensions.
> 
>    Furthermore, future specifications SHOULD avoid the value 0.  Doing
>    so facilitates packet validity checking, since an all-zeros field
>    might commonly be found in an ill-formed packet.
> 
> 6.3 The "rtcp-xr" SDP attribute
> 
>    One SDP attribute is to be registered using the below information:
> 
>    SDP Attribute ("att-field"):
> 
>      Attribute name:     rtcp-xr
>      Long form:          RTP Extended Report Parameters
>      Type of name:       att-field
>      Type of attribute:  Session and Media level
>      Subject to charset: No
>      Purpose:            See this document
>      Reference:          This document
>      Values:             See this document and registrations below
> 
>    The attribute has an extensible parameter field and therefore a
>    registry for these parameters are needed. IANA is 
> requested to create
>    one RTCP-XR SDP parameter registry and register the defined values.
>    This registry is managed according to the  Specification Required
>    policy of RFC 2434. Any registration MUST contain the following
>    information:
> 
>    - Contact information of the one doing the registration, 
> including at
>    least name, address, and email.
> 
>    - An ABNF definition of the parameter, in accordance with the
>    "format-ext" definition.
> 
>    - A description of what the parameter represents and how 
> it shall be
>    interpreted, both normally and in Offer/Answer.
> 
>    This document registers the six parameters defined in chapter 5.1,
>    "rle-pckt", "rle-dup", "recv-ts", "stat-sum", "voip-sum", 
> and "recv-
>    rtt".
> 
> 
> 7. Security Considerations
> 
>    This document extends the RTCP reporting mechanism.  The security
>    considerations that apply to RTCP reports also apply to XR reports.
>    This section details the additional security considerations that
>    apply to the extensions.
> 
>    The extensions introduce heightened confidentiality concerns.
>    Standard RTCP reports contain a limited number of summary 
> statistics.
>    The information contained in XR reports is both more detailed and
>    more extensive (covering a larger number of parameters).  The per-
>    packet information contained in Loss RLE, Duplicate RLE, and
>    Timestamp Report Blocks facilitates MINC inference of multicast
>    distribution trees for RTP data packets, and inference of link
>    characteristics such as loss and delay.  This inference reveals
>    information that might otherwise be considered confidential to
>    autonomous system administrators.  The VoIP Metrics Report Block
>    provides information on the quality of ongoing voice calls.  Though
>    such information might be carried in application specific format in
>    standard RTP sessions, making it available in a standard 
> format here
>    makes it more available to potential eavesdroppers.
> 
>    No new mechanisms are introduced in this document to ensure
>    confidentiality.  Standard encryption procedures can be used when
>    confidentiality is a concern to end hosts.  Autonomous system
>    administrators concerned about the loss of confidentiality 
> regarding
>    their networks can encrypt traffic, or filter it to exclude RTCP
>    packets containing the XR report blocks concerned.
> 
>    Any encryption or filtering of XR report blocks entails a loss of
>    monitoring information to third parties.  For example, a 
> network that
>    establishes a tunnel to encrypt VoIP Report Blocks denies that
>    information to the service providers traversed by the tunnel.  The
>    service providers cannot then monitor or respond to the quality of
>    the VoIP calls that they carry, potentially creating 
> problems for the
>    network's users.  As a default, XR packets SHOULD NOT be 
> encrypted or
>    filtered.
> 
>    The extensions also make certain denial of service attacks easier.
>    This is because of the potential to create RTCP packets much larger
>    than average with the per packet reporting capabilities of the Loss
>    RLE, Duplicate RLE, and Timestamp Report Blocks.  Because of the
>    automatic bandwidth adjustment mechanisms in RTCP, if some session
>    participants are sending large RTCP packets, all participants will
>    see their RTCP reporting intervals lengthened, meaning they will be
>    able to report less frequently.  To limit the effects of large
>    packets, even in the absence of denial of service attacks,
>    applications SHOULD limit the size of XR report blocks and 
> employ the
>    thinning techniques described in this document in order to fit
>    reports into the space available.
> 
>    The usage of the SDP parameters defined affect the behavior of the
>    RTCP XR format blocks. However the security implications are
>    dependent on the report formats. Due to the fact that 
> behavior can be
>    changed by the SDP signaling, the SDP is RECOMMENDED to be
>    authenticate to prevent tampering by intermediates.
> 
> 
> References
> 
>    The references are divided into normative references and non-
>    normative references.
> 
> Normative References
> 
>    [1] S. Bradner, "Key words for use in RFCs to indicate requirement
>    levels," BCP 14, RFC 2119, IETF, March 1997.
> 
>    [s2] D. Crocker, P. Overell, "Augmented BNF for Syntax
>    Specifications: ABNF", RFC 2234, Internet Engineering Task Force,
>    November 1997.
> 
>    [2] ETSI, "Quality of Service (QoS) measurement 
> methodologies," ETSI
>    TS 101 329-5 V1.1.1 (2000-11), November 2000.
> 
>    [s3] M. Handley, V. Jacobson, "SDP: Session Description Protocol",
>    RFC 2327, Internet Engineering Task Force, April 1998.
> 
>    [3] R. Hovey and S. Bradner, "The Organizations Involved 
> in the IETF
>    Standards Process," best current practice BCP 11, RFC 2028, IETF,
>    October 1996.
> 
>    [4] ITU-T, "The E-Model, a computational model for use in
>    transmission planning," Recommendation G.107 (05/00), May 2000.
> 
>    [5] J. Reynolds and J. Postel, "Assigned Numbers," standard STD 2,
>    RFC 1700, IETF, October 1994.
> 
>    [6] T. Narten and H. Alvestrand, "Guidelines for Writing an IANA
>    Considerations Section in RFCs," best current practice BCP 26, RFC
>    2434, IETF, October 1998.
> 
>    [s1] J. Rosenberg, H. Schulzrinne, "An Offer/Answer Model with the
>    Session Description Protocol (SDP)", RFC 3264, Internet Engineering
>    Task Force, June 2002.
> 
>    [7] H. Schulzrinne, S. Casner, R. Frederick, and V. 
> Jacobson, "RTP: A
>    transport protocol for real-time applications," RFC 1889, IETF,
>    February 1996.
> 
> Non-Normative References
> 
>    [8] A. Adams, T. Bu, R. Caceres, N. G. Duffield, T. Friedman, J.
>    Horowitz, F. Lo Presti, S. B. Moon, V. Paxson, and D. Towsley, "The
>    Use of End-to-End Multicast Measurements for 
> Characterizing Internal
>    Network Behavior," IEEE Communications Magazine, May 2000.
> 
>    [9] R. Caceres, N. G. Duffield, and T. Friedman, "Impromptu
>    measurement infrastructures using RTP," Proc. IEEE Infocom 2002.
> 
>    [10] A. D. Clark, "Modeling the Effects of Burst Packet Loss and
>    Recency on Subjective Voice Quality," Proc. IP Telephony Workshop
>    2001.
> 
>    [s4] M. Handley, C. Perkins, E. Whelan, "Session Announcement
>    Protocol", RFC 2974, Internet Engineering Task Force, October 2000.
> 
>    [s5] H. Schulzrinne, R. Lanphier, and A. Rao, "Real time streaming
>    protocol (RTSP)," RFC 2326, Internet Engineering Task Force, Apr.
>    1998.
> 
>    [11] D. Sisalem and A. Wolisz, "MLDA: A TCP-friendly Congestion
>    Control Framework for Heterogeneous Multicast Environments", Proc.
> 
> 
> 
> -- 
> 
> Magnus Westerlund 
> 
> Multimedia Technologies, Ericsson Research ERA/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
> 
> 
> _______________________________________________
> 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 mailnull@www1.ietf.org  Mon Mar 31 03:03:13 2003
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To: Jose Rey <rey@panasonic.de>
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Subject: Re: [AVT] SDP signaling for RTCP XR format
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Hi Jose,

Thanks for giving some comments, it helps improving the text quickly. 
Responses below:

Jose Rey wrote:

>Hi Magnus,
>
>I've got some comments/questions on your proposal...
>
>1.- The term "RTCP XR format" to address the Loss RLE Report Block, Duplicate RLE ..., is not consistent with the rest of the draft. I prefer "XR blocks" which is used throghout the text.
>

Yes, I agree that I may not have used a totally aligned language. It 
should be corrected.

>2.- Just curious, why are the report blocks used without signaling? It would be helpful to include the reason for this in the draft (if it's not there already and I missed it)
>

As I understand it the use case for having report blocks without 
signaling is that you know that you are using application foo. Foo has 
defined that we will use certain report blocks. Then it is not necessary 
to use the signaling, which may in fact not exist. Certain report blocks 
may be used to discover if the other end parts understand them. For 
example the RTT measurement can be tried a few times to see if any 
receiver responds to them.

>3.- I like your classification of parameters because is needed afterwards to define the usage, however I don't understand your reasoning since both types of parameters define a functionality. Why not rename it to unilateral (category 1) and collaborative (category 2) or something alike? 
>

Yes, good proposal. I will see what I can do about the text.

>4.- Section 5.2 par. 5: This is a bit confusing. The receiver may declare its capability for the XR blocks and not send the report blocks themselves? Shouldn't it be that by declaring its capability is also declaring its willingness to use them? Or is it maybe because the receiver decides itself when to send (them dynamically during the session) them and thus might be that they are not needed at all? Does this make sense? Maybe is the text that confuses me.
>
>  
>

The text might not be the best. However the functionality provided is 
that only the receiver can choice to use the Receiver Timestamp Report 
Block or not. The other end does only need to know if the other end part 
actually understands them and will respond with a DLRR Report block. If 
the other ends does not understand it is wasted bandwidth.

Here one issue actually surface. Is it the right thing to restrict a 
receiver to not send DLRR report block unless the usage has been 
explicitly signaled in multicast? The reason why I see a need to 
restrict it is the potential to create enormous report blocks in large 
multicast groups. And the only result is that the different receivers 
will determine there RTT to other receivers which seems to be rather 
pointless in most application. It seems that you would in fact like to 
control this operation to a few different operation modes in multicast.
1. fully meshed, both senders and receivers return DLRR blocks.
2. Sender only, only senders return DLRR blocks.
3. functionality off, no one responds with DLRR blocks.

Should we define such a distinction in the signaling?

Best Regards

Magnus Westerlund 

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



_______________________________________________
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From: Jose Rey <rey@panasonic.de>
Subject: RE: [AVT] SDP signaling for RTCP XR format
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Hi Magnus,

please see inline:

> >2.- Just curious, why are the report blocks used without 
> signaling? It would be helpful to include the reason for this 
> in the draft (if it's not there already and I missed it)
> >
> 
> As I understand it the use case for having report blocks without 
> signaling is that you know that you are using application 
> foo. Foo has 
> defined that we will use certain report blocks. Then it is 
> not necessary 
> to use the signaling, which may in fact not exist. Certain 
> report blocks 
> may be used to discover if the other end parts understand them. For 
> example the RTT measurement can be tried a few times to see if any 
> receiver responds to them.

I think such a reasoning should be included in the text. The problem I see is that if someone is not aware of this fact, he could just download the SDP description and start a session. Since the offerer might "secretly" need some category1 parameters which the answerer is (of course) not providing as he wasn't requested to do so, the reception would be probably not of good quality, thus making a total waste of bandwidth out of the whole procedure...

To solve this, why not declare some kind of attribute in SDP that defines the set of feedback messages that the application needs of?
> 
> >3.- I like your classification of parameters because is 
> needed afterwards to define the usage, however I don't 
> understand your reasoning since both types of parameters 
> define a functionality. Why not rename it to unilateral 
> (category 1) and collaborative (category 2) or something alike? 
> >
> 
> Yes, good proposal. I will see what I can do about the text.

Good.

> 
> >4.- Section 5.2 par. 5: This is a bit confusing. The 
> receiver may declare its capability for the XR blocks and not 
> send the report blocks themselves? Shouldn't it be that by 
> declaring its capability is also declaring its willingness to 
> use them? Or is it maybe because the receiver decides itself 
> when to send (them dynamically during the session) them and 
> thus might be that they are not needed at all? Does this make 
> sense? Maybe is the text that confuses me.
> >
> >  
> >
> 
> The text might not be the best. However the functionality provided is 
> that only the receiver can choice to use the Receiver 
> Timestamp Report 
> Block or not. The other end does only need to know if the 
> other end part 
> actually understands them and will respond with a DLRR Report 
> block. If 
> the other ends does not understand it is wasted bandwidth.
> 

I agree with the above. My comment was more that I think declarations in answers to category1-offers should be binding: if declared -> MUST be sent; if not declared -> MUST not be sent. Declarations in the offers just indicate the offerer SHOULD send them if supported, as it is now.
For the category2 parameters it is different, since it is the receiver that benefits from such metrics, thus: if declared in the answer-> MAY be sent; if not declared -> MUST NOT be sent because the answerer does not wish to have such metrics. This is valid both unicast and multicast. Is this also your view? Would this work with the Offer/Answer model?


> Here one issue actually surface. Is it the right thing to restrict a 
> receiver to not send DLRR report block unless the usage has been 
> explicitly signaled in multicast? The reason why I see a need to 
> restrict it is the potential to create enormous report blocks 
> in large 
> multicast groups.

I disagree with your last sentence: RRs only report on RTP packets, not on RTCP. Thus, the size of the RRs would be equal, whether XRs are sent or not. It is only the number of RTCP packets ("traditional" and XR) that would be different. Still... a waste of bandwidth if it is not needed for every session member to know the RTT to every other one ...

> And the only result is that the different receivers 
> will determine there RTT to other receivers which seems to be rather 
> pointless in most application. It seems that you would in 
> fact like to 
> control this operation to a few different operation modes in 
> multicast.
> 1. fully meshed, both senders and receivers return DLRR blocks.
> 2. Sender only, only senders return DLRR blocks.
> 3. functionality off, no one responds with DLRR blocks.
> 
> Should we define such a distinction in the signaling?

Yes, I find this useful. The third one would be the one without rtcp-xr attribute in the offer (if you assume declaration/omission is binding).


One more thing: how about the RTCP bandwidth sharing of the XR packets? how is, for example, the order of transmission defined relative to the "traditional" RTCP packets? I think this is not specified, a short comment would do like "the profile in use SHALL ensure that the RTCP bandwidth is equally shared by packets sent by this RTP server or sender".

cheers,

JosÃ©

_______________________________________________
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From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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To: Jose Rey <rey@panasonic.de>
CC: IETF AVT WG <avt@ietf.org>
Subject: Re: [AVT] SDP signaling for RTCP XR format
References: <NGBBJIBNJOKHDHFKEOMHKEJIELAA.rey@panasonic.de>
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Hi Jose,

Jose Rey wrote:

>Hi Magnus,
>
>please see inline:
>
>  
>
>>>2.- Just curious, why are the report blocks used without 
>>>      
>>>
>>signaling? It would be helpful to include the reason for this 
>>in the draft (if it's not there already and I missed it)
>>    
>>
>>As I understand it the use case for having report blocks without 
>>signaling is that you know that you are using application 
>>foo. Foo has 
>>defined that we will use certain report blocks. Then it is 
>>not necessary 
>>to use the signaling, which may in fact not exist. Certain 
>>report blocks 
>>may be used to discover if the other end parts understand them. For 
>>example the RTT measurement can be tried a few times to see if any 
>>receiver responds to them.
>>    
>>
>
>I think such a reasoning should be included in the text. The problem I see is that if someone is not aware of this fact, he could just download the SDP description and start a session. Since the offerer might "secretly" need some category1 parameters which the answerer is (of course) not providing as he wasn't requested to do so, the reception would be probably not of good quality, thus making a total waste of bandwidth out of the whole procedure...
>  
>

Yes, I will try to clarify this. My intention is that the SDP parameters 
should be used if there is SDP signaling available for this application. 
The one exception is that application  foo always uses these reports. 
And since you are running the client for application foo it knows that 
it shall use the report formats.

>To solve this, why not declare some kind of attribute in SDP that defines the set of feedback messages that the application needs of?
>

You mean in the general case for like SDES information types, etc.?

>>>4.- Section 5.2 par. 5: This is a bit confusing. The 
>>>      
>>>
>>receiver may declare its capability for the XR blocks and not 
>>send the report blocks themselves? Shouldn't it be that by 
>>declaring its capability is also declaring its willingness to 
>>use them? Or is it maybe because the receiver decides itself 
>>when to send (them dynamically during the session) them and 
>>thus might be that they are not needed at all? Does this make 
>>sense? Maybe is the text that confuses me.
>>    
>>
>>> 
>>>
>>>      
>>>
>>The text might not be the best. However the functionality provided is 
>>that only the receiver can choice to use the Receiver 
>>Timestamp Report 
>>Block or not. The other end does only need to know if the 
>>other end part 
>>actually understands them and will respond with a DLRR Report 
>>block. If 
>>the other ends does not understand it is wasted bandwidth.
>>
>>    
>>
>
>I agree with the above. My comment was more that I think declarations in answers to category1-offers should be binding: if declared -> MUST be sent; if not declared -> MUST not be sent. Declarations in the offers just indicate the offerer SHOULD send them if supported, as it is now.
>

I think SHOULD is a good requirement levels for something that is simply 
information. If you have very good reason not to send a report format 
you should not be forced to do it. However you should have really good 
reasons before you don't. When it comes to the other direction I might 
be more willingly to adopt a MUST NOT send policy. The question is how 
you can make it work with the MAY at top level. Some better language may 
solve this. I also need to consider what implications this have in 
interactions with clients not understanding the attribute. Probably no 
problem as they will not send it. 

>For the category2 parameters it is different, since it is the receiver that benefits from such metrics, thus: if declared in the answer-> MAY be sent; if not declared -> MUST NOT be sent because the answerer does not wish to have such metrics. This is valid both unicast and multicast. Is this also your view? Would this work with the Offer/Answer model?
>  
>

Yes, it is my view. So far I don't see any problems in Offer/Answer 
usage. Please think this trough one more time.

>
>  
>
>>Here one issue actually surface. Is it the right thing to restrict a 
>>receiver to not send DLRR report block unless the usage has been 
>>explicitly signaled in multicast? The reason why I see a need to 
>>restrict it is the potential to create enormous report blocks 
>>in large 
>>multicast groups.
>>    
>>
>
>I disagree with your last sentence: RRs only report on RTP packets, not on RTCP. Thus, the size of the RRs would be equal, whether XRs are sent or not. It is only the number of RTCP packets ("traditional" and XR) that would be different. Still... a waste of bandwidth if it is not needed for every session member to know the RTT to every other one ...
>  
>

No, I think you are missing one thing where the DLRR blocks can become 
enormous. If you are part of an ASM group with 100 participants and they 
all send the Receiver Timestamp block in there RTCP reports. Normally 
you will send on average at the same pace as all others. Therefore your 
DLRR response will contain 100 sources. As everybody is doing this the 
average RTCP size will be large.  Therefore the usage of  option 1 below 
seems to be limited to small groups. When using option 2 only a few out 
of all sending the Receiver Timestamp block actually answers with the 
delay information. Therefore only the sender reporting will be effected 
by this.

>  
>
>>And the only result is that the different receivers 
>>will determine there RTT to other receivers which seems to be rather 
>>pointless in most application. It seems that you would in 
>>fact like to 
>>control this operation to a few different operation modes in 
>>multicast.
>>1. fully meshed, both senders and receivers return DLRR blocks.
>>2. Sender only, only senders return DLRR blocks.
>>3. functionality off, no one responds with DLRR blocks.
>>
>>Should we define such a distinction in the signaling?
>>    
>>
>
>Yes, I find this useful. The third one would be the one without rtcp-xr attribute in the offer (if you assume declaration/omission is binding).
>
>
>One more thing: how about the RTCP bandwidth sharing of the XR packets? how is, for example, the order of transmission defined relative to the "traditional" RTCP packets? I think this is not specified, a short comment would do like "the profile in use SHALL ensure that the RTCP bandwidth is equally shared by packets sent by this RTP server or sender".
>  
>
Are you meaning the sharing of bandwidth between different RTCP packet 
types? And also within a XR packet that the different blocks are sent 
not all the time? The first problem is that the RTCP compound packet can 
reach the MTU and some packets has to be excluded. I think round robin 
in that case is a reasonable strategy if not defined elsewhere. This 
would mean equal share for all non mandatory packets. I think the same 
strategy can be used within a XR packet to ensure that all types of 
report blocks may send. However looking at the block types that can be 
sent that has variable size I has proposed a way of limiting the packet 
size. I think such a limitation should also be possible to defined for 
the DLRR packet.
 I hesitant to write policy that actually is application or profile 
dependent. However defining a default behavior I find acceptable with a 
side note that this can be replaced by application or profile 
specifications.

Best Regards


Magnus Westerlund 

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



_______________________________________________
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To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
cc: Jose Rey <rey@panasonic.de>, IETF AVT WG <avt@ietf.org>
Subject: Re: [AVT] SDP signaling for RTCP XR format 
In-Reply-To: Your message of "Mon, 31 Mar 2003 09:57:00 +0200."
             <3E87F4CC.7080704@era.ericsson.se> 
Date: Mon, 31 Mar 2003 10:46:44 -0500
From: Colin Perkins <csp@csperkins.org>
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--> Magnus Westerlund writes:
...
>>2.- Just curious, why are the report blocks used without signaling? It
>>would be helpful to include the reason for this in the draft (if it's not
>>there already and I missed it)
>
>As I understand it the use case for having report blocks without 
>signaling is that you know that you are using application foo. Foo has 
>defined that we will use certain report blocks. Then it is not necessary 
>to use the signaling, which may in fact not exist. Certain report blocks 
>may be used to discover if the other end parts understand them. For 
>example the RTT measurement can be tried a few times to see if any 
>receiver responds to them.

RTP allows applications to send new types of RTCP packet, even if these are
not explicitly signalled. RTCP XR shouldn't impose additional restrictions.

Colin
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Subject: [AVT] Interesting effort out of P1394 ...
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Hi everyone,

        New effort in the IEEE of possible interest to AVT
folks:

http://www.businesswire.com/cgi-bin/f_headline.cgi?bw.033103/230905415

        Running IEEE 1394 over the 1000baseT physical layer,
with negotiations to switch the wire between Ethernet and
IEEE P1394 layer 2.  Intended for audio and video, but since 1394
has a MIDI payload already, this would be a third way to
send MIDI out of a RJ-45 socket (MWPP over RTP over IP, 
DMIDI over Layer 2 Ethernet, and IEEE 1394 over the physical
layer for 1000baseT). 

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------
_______________________________________________
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To: "Magnus Westerlund" <magnus.westerlund@era.ericsson.se>,
        "Colin Perkins" <csp@csperkins.org>
Cc: "Jose Rey" <rey@panasonic.de>, "IETF AVT WG" <avt@ietf.org>
References: <200303311546.h2VFkivm002320@purple.nge.isi.edu>
Subject: Re: [AVT] SDP signaling for RTCP XR format 
Date: Mon, 31 Mar 2003 21:00:57 +0200
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Colin Perkins wrote:

> --> Magnus Westerlund writes:
> ...
> >>2.- Just curious, why are the report blocks used without signaling? It
> >>would be helpful to include the reason for this in the draft (if it's
not
> >>there already and I missed it)
> >
> >As I understand it the use case for having report blocks without
> >signaling is that you know that you are using application foo. Foo has
> >defined that we will use certain report blocks. Then it is not necessary
> >to use the signaling, which may in fact not exist. Certain report blocks
> >may be used to discover if the other end parts understand them. For
> >example the RTT measurement can be tried a few times to see if any
> >receiver responds to them.
>
> RTP allows applications to send new types of RTCP packet, even if these
are
> not explicitly signalled. RTCP XR shouldn't impose additional
restrictions.

The draft will be sure to specify that SDP signalling is an optional
mechanism, to avoid any confusion.

- Timur

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Subject: [AVT] Couple of clarifications on  RFC2833
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Question-1

the RFC 2833, section 3.6 states the following

"If an event continues for more than one period, the source generating the events should send a new event packet with the RTP timestamp value corresponding to the beginning of the event and the duration of the event increased correspondingly."

What this "corresponding" mean?  Does it mean that all the RTP packets related to that event should use the SAME timestamp as that of its first packet or does it mean that timestamp for next packet= timestamp of the first packet + duration of event?

I have seen some implementations where all the RTP packets for one specific DTMF event use the timestamp as that of first packet. But this creates an interesting situation where audio RTP packets will have latest time stamp and then we have DTMF RTP packet carrying a previous timestamp. Since sequence number are incremented sequentially for both audio as well dtmf RTP packets, the dtmf  packets with higher sequence number will have smaller timestamp value compared to an audio RTP packet with  smaller sequence number.


Question-2

While using inband DTMF detection the sequence numbers are incremented by "one" for every packet sent by the media gateway for that call whether they are dtmf or audio. That means in an RTP stream for example ,  sequence numbers 5,8,9,10,11 could belong to a DTMF event RTP packets while 1,2,3,4,6,7,12,13 to audio RTP packets. This is perfectly alright with in-band DTMF detection.

How should the  sequence numbers be assigned  for out-of-band DTMF detection?  As in this case there would be two RTP stream going to different IP address /port, one for audio and another one for DTMF. If sequence numbers are assigned as above then both audio receivers as well as DTMF receiver are going to get gaps in sequence numbers ? Or should there be one set of sequence numbers for audio stream and another for DTMF stream?


Will appreciate any clarification on these 


Thanks

Hadi

 

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To: "Hashmi, Mohamed Hadi" <Hadi.Hashmi@Level3.com>
cc: avt@ietf.org
Subject: Re: [AVT] Couple of clarifications on RFC2833 
In-Reply-To: Your message of "Mon, 31 Mar 2003 15:15:16 MST."
             <3DB099919328BA41905F651A6EF8397608CD5A@idc1exc0004.corp.global.level3.com> 
Date: Mon, 31 Mar 2003 18:06:51 -0500
From: Colin Perkins <csp@isi.edu>
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--> "Hashmi, Mohamed Hadi" writes:
>Question-1
>
>the RFC 2833, section 3.6 states the following
>
>"If an event continues for more than one period, the source generating the
>events should send a new event packet with the RTP timestamp value
>corresponding to the beginning of the event and the duration of the event
>increased correspondingly."
>
>What this "corresponding" mean?  Does it mean that all the RTP packets
>related to that event should use the SAME timestamp as that of its first
>packet or does it mean that timestamp for next packet timestamp of the
>first packet + duration of event?

My understanding is that all the packets comprising a single event have the
same timestamp. The second (and subsequent) packets have the same timestamp
as the first packet, but increased duration fields. 

>I have seen some implementations where all the RTP packets for one specific
>DTMF event use the timestamp as that of first packet. But this creates an
>interesting situation where audio RTP packets will have latest time stamp
>and then we have DTMF RTP packet carrying a previous timestamp. Since
>sequence number are incremented sequentially for both audio as well dtmf
>RTP packets, the dtmf  packets with higher sequence number will have
>smaller timestamp value compared to an audio RTP packet with  smaller
>sequence number.

So? RTP does not mandate that packets are sent in order of increasing
timestamp.

>Question-2
>
>While using inband DTMF detection the sequence numbers are incremented by
>"one" for every packet sent by the media gateway for that call whether they
>are dtmf or audio. That means in an RTP stream for example ,  sequence
>numbers 5,8,9,10,11 could belong to a DTMF event RTP packets while 1,2,3,
>4,6,7,12,13 to audio RTP packets. This is perfectly alright with in-band
>DTMF detection.
>
>How should the  sequence numbers be assigned  for out-of-band DTMF
>detection?  As in this case the re would be two RTP stream going to
>different IP address /port, one for audio and another one for DTMF. If
>sequence numbers are assigned as above then both audio receivers as well as
>DTMF receiver are going to get gaps in sequence numbers ? Or should there
>be one set of sequence numbers for audio stream and another for DTMF
>stream?  

If there are two separate RTP streams, they each have their own sequence
number space. Accordingly, you'd use one set of sequence numbers for audio
stream and another for DTMF stream.

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



From mailnull@www1.ietf.org  Mon Mar 31 18:32:52 2003
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Subject: RE: [AVT] Couple of clarifications on RFC2833 
Date: Mon, 31 Mar 2003 16:30:01 -0700
Message-ID: <3DB099919328BA41905F651A6EF839767A478B@idc1exc0004.corp.global.level3.com>
Thread-Topic: [AVT] Couple of clarifications on RFC2833 
Thread-Index: AcL32jo1UB0A5NkmSYirn65o6kCz0wAAo3Qw
From: "Hashmi, Mohamed Hadi" <Hadi.Hashmi@Level3.com>
To: "Colin Perkins" <csp@isi.edu>
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Colin,


Thanks for you quick answers. 

>So? RTP does not mandate that packets are sent in order of increasing
>timestamp.

This can potentially impact the inter arrival jitter calculation and also latency. The way to overcome this problem will be to ignore DTMF packets

>If there are two separate RTP streams, they each have their own sequence
>number space. Accordingly, you'd use one set of sequence numbers for audio
>.stream and another for DTMF stream.

Is this observation based on RFC1889 or is this explicitly stated in RFC2833? I mean how do I convince a vendor that this is how it is supposed to be done?


Thanks

Hadi 

-----Original Message-----
From: Colin Perkins [mailto:csp@isi.edu]
Sent: Monday, March 31, 2003 4:07 PM
To: Hashmi, Mohamed Hadi
Cc: avt@ietf.org
Subject: Re: [AVT] Couple of clarifications on RFC2833 


--> "Hashmi, Mohamed Hadi" writes:
>Question-1
>
>the RFC 2833, section 3.6 states the following
>
>"If an event continues for more than one period, the source generating the
>events should send a new event packet with the RTP timestamp value
>corresponding to the beginning of the event and the duration of the event
>increased correspondingly."
>
>What this "corresponding" mean?  Does it mean that all the RTP packets
>related to that event should use the SAME timestamp as that of its first
>packet or does it mean that timestamp for next packet timestamp of the
>first packet + duration of event?

My understanding is that all the packets comprising a single event have the
same timestamp. The second (and subsequent) packets have the same timestamp
as the first packet, but increased duration fields. 

>I have seen some implementations where all the RTP packets for one specific
>DTMF event use the timestamp as that of first packet. But this creates an
>interesting situation where audio RTP packets will have latest time stamp
>and then we have DTMF RTP packet carrying a previous timestamp. Since
>sequence number are incremented sequentially for both audio as well dtmf
>RTP packets, the dtmf  packets with higher sequence number will have
>smaller timestamp value compared to an audio RTP packet with  smaller
>sequence number.

So? RTP does not mandate that packets are sent in order of increasing
timestamp.

This can potentially impact the inter arrival jitter calculation and also latency


>Question-2
>
>While using inband DTMF detection the sequence numbers are incremented by
>"one" for every packet sent by the media gateway for that call whether they
>are dtmf or audio. That means in an RTP stream for example ,  sequence
>numbers 5,8,9,10,11 could belong to a DTMF event RTP packets while 1,2,3,
>4,6,7,12,13 to audio RTP packets. This is perfectly alright with in-band
>DTMF detection.
>
>How should the  sequence numbers be assigned  for out-of-band DTMF
>detection?  As in this case the re would be two RTP stream going to
>different IP address /port, one for audio and another one for DTMF. If
>sequence numbers are assigned as above then both audio receivers as well as
>DTMF receiver are going to get gaps in sequence numbers ? Or should there
>be one set of sequence numbers for audio stream and another for DTMF
>stream?  

If there are two separate RTP streams, they each have their own sequence
number space. Accordingly, you'd use one set of sequence numbers for audio
stream and another for DTMF stream.

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



From mailnull@www1.ietf.org  Mon Mar 31 22:48:37 2003
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To: "Hashmi, Mohamed Hadi" <Hadi.Hashmi@Level3.com>
Cc: <avt@ietf.org>
References: <3DB099919328BA41905F651A6EF8397608CD5A@idc1exc0004.corp.global.level3.com>
Subject: Re: [AVT] Couple of clarifications on  RFC2833
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Hello Hashmi:
This issue is discussed a lot in this mailing list. Please scan through the
mailing list or google for "RFC2833 + Confusions"
Ex:
http://www1.ietf.org/mail-archive/working-groups/avt/current/msg01693.html

Finally, Please read my comments  also.
----- Original Message -----
From: Hashmi, Mohamed Hadi <Hadi.Hashmi@Level3.com>
To: <avt@ietf.org>
Sent: Tuesday, April 01, 2003 3:45 AM
Subject: [AVT] Couple of clarifications on RFC2833


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----------------------------------------------------------------------------
----


Question-1

the RFC 2833, section 3.6 states the following

"If an event continues for more than one period, the source generating the
events should send a new event packet with the RTP timestamp value
corresponding to the beginning of the event and the duration of the event
increased correspondingly."

What this "corresponding" mean?  Does it mean that all the RTP packets
related to that event should use the SAME timestamp as that of its first
packet or does it mean that timestamp for next packet= timestamp of the
first packet + duration of event?

Bino: Please visualize the DTMF events starting from a RTP TS and after that
the DTMF event packet  will represent its duration. If you really want a
notion of Current RTP TS you can add up  Events Starting RTP TS + Duration.

I have seen some implementations where all the RTP packets for one specific
DTMF event use the timestamp as that of first packet. But this creates an
interesting situation where audio RTP packets will have latest time stamp
and then we have DTMF RTP packet carrying a previous timestamp. Since
sequence number are incremented sequentially for both audio as well dtmf RTP
packets, the dtmf  packets with higher sequence number will have smaller
timestamp value compared to an audio RTP packet with  smaller sequence
number.

Bino: Logic is very simple. Read  session 3.2 ( Simultaneous Generation of
Audio and Events) and read RFC 2198 (RTP Payload for Redundant Audio Data).
Otherwise go ahead with a simple solution, support only outofband DTMF or
inband DTMF. This will dramatically improve the interoperability of your
device.


Question-2

While using inband DTMF detection the sequence numbers are incremented by
"one" for every packet sent by the media gateway for that call whether they
are dtmf or audio. That means in an RTP stream for example ,  sequence
numbers 5,8,9,10,11 could belong to a DTMF event RTP packets while
1,2,3,4,6,7,12,13 to audio RTP packets. This is perfectly alright with
in-band DTMF detection.

How should the  sequence numbers be assigned  for out-of-band DTMF
detection?  As in this case there would be two RTP stream going to different
IP address /port, one for audio and another one for DTMF. If sequence
numbers are assigned as above then both audio receivers as well as DTMF
receiver are going to get gaps in sequence numbers ? Or should there be one
set of sequence numbers for audio stream and another for DTMF stream?

Will appreciate any clarification on these

Bino: Please  read RFC 2198 (RTP Payload for Redundant Audio Data).


Thanks

Hadi



_______________________________________________
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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To: "Hashmi, Mohamed Hadi" <Hadi.Hashmi@Level3.com>
cc: avt@ietf.org
Subject: Re: [AVT] Couple of clarifications on RFC2833 
In-Reply-To: Your message of "Mon, 31 Mar 2003 16:30:01 MST."
             <3DB099919328BA41905F651A6EF839767A478B@idc1exc0004.corp.global.level3.com> 
Date: Mon, 31 Mar 2003 22:58:24 -0500
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--> "Hashmi, Mohamed Hadi" writes:
>Colin,
>
>
>Thanks for you quick answers. 
>
>>So? RTP does not mandate that packets are sent in order of increasing
>>timestamp.
>
>This can potentially impact the inter arrival jitter calculation and also
>latency. The way to over come this problem will be to ignore DTMF packets

It can affect the jitter calculation (there is a similar issue when video
frames are split across multiple RTP packets).  The latency should not be
affected, if you design the playout buffer appropriately. 

>>If there are two separate RTP streams, they each have their own sequence
>>number space. Accordingly, you'd use one set of sequence numbers for audio
>>.stream and another for DTMF stream.
>
>Is this observation based on RFC1889 or is this explicitly stated in
>RFC2833? I mean how do I convince a vendor that this is how it is supposed
>to be done?

It's standard RTP behaviour. If you share a sequence number space across
multiple sessions, sent to different locations, you cannot tell if there
was loss.  Hence you should use separate sequence number spaces for each
RTP session.

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



