
From Peter.Stevens@bbc.co.uk  Wed Sep  4 06:09:17 2013
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From: Peter Stevens <Peter.Stevens@bbc.co.uk>
To: "John Lindsay (Lindsay@worldcastsystems.com)" <Lindsay@worldcastsystems.com>, "'payload@ietf.org'" <payload@ietf.org>
Thread-Topic: Re: [payload] I-D Action: draft-ietf-payload-rtp-aptx-00.txt
Thread-Index: Ac6pb+yV9qXp2OBbR9a6jPbN2JKFbw==
Date: Wed, 4 Sep 2013 13:09:01 +0000
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Subject: Re: [payload] I-D Action: draft-ietf-payload-rtp-aptx-00.txt
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Subject: [payload] I-D Action: draft-ietf-payload-rtp-aptx-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Audio/Video Transport Payloads Working Gr=
oup of the IETF.

	Title           : RTP Payload Format for Standard apt-X and Enhanced apt-X=
 Codecs
	Author(s)       : John Lindsay
                          Hartmut Foerster
	Filename        : draft-ietf-payload-rtp-aptx-01.txt
	Pages           : 22
	Date            : 2013-09-05

Abstract:
   This document specifies a scheme for packetizing Standard apt-X, or
   Enhanced apt-X, encoded audio data into Real-time Transport Protocol
   (RTP) packets.  The document describes a payload format that permits
   transmission of multiple related audio channels in a single RTP
   payload, and a means of establishing Standard apt-X and Enhanced
   apt-X connections through the Session Description Protocol (SDP).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-aptx

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-payload-rtp-aptx-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-payload-rtp-aptx-01


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Subject: [payload] Comment to draft-ietf-payload-rtp-opus-01
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--20cf303345836448c304e5b5fe5f
Content-Type: text/plain; charset=ISO-8859-1

In section 3.1.3 Discontinuous Transmission (DTX), I'm missing some text
that explains how to update the RTP packets correctly after a silent
period. How should rtp timestamp and sequence number be updated while not
sending rtp packets (silent period), to prepare for when we eventually have
data to transmit again? My thinking is that rtp timestamp should be updated
with the number of samples that were not sent, while sequence number should
only be updated when sending packets. Correct?

/Tina le Grand.

--20cf303345836448c304e5b5fe5f
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">In section 3.1.3 Discontinuous Transmission (DTX), I&#39;m=
 missing some text that explains how to update the RTP packets correctly af=
ter a silent period. How should rtp timestamp and sequence number be update=
d while not sending rtp packets (silent period), to prepare for when we eve=
ntually have data to transmit again? My thinking is that rtp timestamp shou=
ld be updated with the number of samples that were not sent, while sequence=
 number should only be updated when sending packets. Correct?<div>

<br></div><div>/Tina le Grand.</div></div>

--20cf303345836448c304e5b5fe5f--

From internet-drafts@ietf.org  Fri Sep  6 08:53:39 2013
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Subject: [payload] I-D Action: draft-ietf-payload-rtp-h265-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Audio/Video Transport Payloads Working Gr=
oup of the IETF.

	Title           : RTP Payload Format for High Efficiency Video Coding
	Author(s)       : Ye-Kui Wang
                          Yago Sanchez
                          Thomas Schierl
                          Stephan Wenger
                          Miska M. Hannuksela
	Filename        : draft-ietf-payload-rtp-h265-01.txt
	Pages           : 76
	Date            : 2013-09-06

Abstract:
   This memo describes an RTP payload format for the video coding
   standard  ITU-T  Recommendation  H.265  and  ISO/IEC  International
   Standard 23008-2, both also known as High Efficiency Video Coding
   (HEVC) [HEVC], developed by the Joint Collaborative Team on Video
   Coding (JCT-VC).  The RTP payload format allows for packetization of
   one or more Network Abstraction Layer (NAL) units in each RTP packet
   payload, as well as fragmentation of a NAL unit into multiple RTP
   packets.  Furthermore, it supports transmission of an HEVC stream
   over a single as well as multiple RTP flows.  The payload format has
   wide applicability in videoconferencing, Internet video streaming,
   and high bit-rate entertainment-quality video, among others.




The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-h265

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-payload-rtp-h265-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-payload-rtp-h265-01


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until the htmlized version and diff are available at tools.ietf.org.

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ftp://ftp.ietf.org/internet-drafts/


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From: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com>
To: "payload@ietf.org" <payload@ietf.org>, "avt@ietf.org" <avt@ietf.org>
Thread-Topic: [payload] I-D Action: draft-ietf-payload-rtp-h265-01.txt
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Date: Fri, 6 Sep 2013 16:06:41 +0000
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Dear All,

The main changes (in addition to some other, basically editorial, changes) =
compared to -00 are:
- Added the media type parameter dec-parallel-cap that was agreed in Berlin
- Added media type parameters max-fps, max-tr, and max-tc as agreed on the =
reflector
- Added an informative note on detection of the last NAL unit of an AU afte=
r the semantics of the M bit as agreed on the reflector

Comments are welcome!

BR, YK

> -----Original Message-----
> From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On
> Behalf Of internet-drafts@ietf.org
> Sent: Friday, September 06, 2013 8:54 AM
> To: i-d-announce@ietf.org
> Cc: payload@ietf.org
> Subject: [payload] I-D Action: draft-ietf-payload-rtp-h265-01.txt
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
>  This draft is a work item of the Audio/Video Transport Payloads Working
> Group of the IETF.
>=20
> 	Title           : RTP Payload Format for High Efficiency Video Coding
> 	Author(s)       : Ye-Kui Wang
>                           Yago Sanchez
>                           Thomas Schierl
>                           Stephan Wenger
>                           Miska M. Hannuksela
> 	Filename        : draft-ietf-payload-rtp-h265-01.txt
> 	Pages           : 76
> 	Date            : 2013-09-06
>=20
> Abstract:
>    This memo describes an RTP payload format for the video coding
>    standard  ITU-T  Recommendation  H.265  and  ISO/IEC  International
>    Standard 23008-2, both also known as High Efficiency Video Coding
>    (HEVC) [HEVC], developed by the Joint Collaborative Team on Video
>    Coding (JCT-VC).  The RTP payload format allows for packetization of
>    one or more Network Abstraction Layer (NAL) units in each RTP packet
>    payload, as well as fragmentation of a NAL unit into multiple RTP
>    packets.  Furthermore, it supports transmission of an HEVC stream
>    over a single as well as multiple RTP flows.  The payload format has
>    wide applicability in videoconferencing, Internet video streaming,
>    and high bit-rate entertainment-quality video, among others.
>=20
>=20
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-h265
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-payload-rtp-h265-01
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-payload-rtp-h265-01
>=20
>=20
> Please note that it may take a couple of minutes from the time of submiss=
ion
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload

From jmvalin@mozilla.com  Fri Sep  6 12:19:48 2013
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Date: Fri, 06 Sep 2013 15:19:39 -0400
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 09/06/2013 08:15 AM, Tina le Grand wrote:
> In section 3.1.3 Discontinuous Transmission (DTX), I'm missing some
> text that explains how to update the RTP packets correctly after a
> silent period. How should rtp timestamp and sequence number be
> updated while not sending rtp packets (silent period), to prepare
> for when we eventually have data to transmit again? My thinking is
> that rtp timestamp should be updated with the number of samples
> that were not sent, while sequence number should only be updated
> when sending packets. Correct?

That's also my understanding, but I'm far from an expert on this. Can
anyone with good knowledge of RTP confirm that it's the right thing to do?

	Jean-Marc

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Date: Fri, 6 Sep 2013 15:46:50 -0400
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From: Roman Shpount <roman@telurix.com>
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--f46d043be1e0b8083504e5bc4c29
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On Fri, Sep 6, 2013 at 3:19 PM, Jean-Marc Valin <jmvalin@mozilla.com> wrote:

> On 09/06/2013 08:15 AM, Tina le Grand wrote:
> > In section 3.1.3 Discontinuous Transmission (DTX), I'm missing some
> > text that explains how to update the RTP packets correctly after a
> > silent period. How should rtp timestamp and sequence number be
> > updated while not sending rtp packets (silent period), to prepare
> > for when we eventually have data to transmit again? My thinking is
> > that rtp timestamp should be updated with the number of samples
> > that were not sent, while sequence number should only be updated
> > when sending packets. Correct?
>
> That's also my understanding, but I'm far from an expert on this. Can
> anyone with good knowledge of RTP confirm that it's the right thing to do?
>
>
In my experience and according to RFC 3551 this is the right thing to do.
Furthermore, the first packet of a talkspurt, that is, the first packet
after a silence period during which packets have not been transmitted
contiguously, SHOULD be distinguished by setting the marker bit in the RTP
data header to one. Also, since this is already standard behavior for all
audio codecs according to RFC 3551 there is probably no need to define this
behavior specifically for OPUS.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Fri, Sep 6, 2013 at 3:19 PM, Jean-Marc Valin <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:jmvalin@mozilla.com" target=3D"_blank">jmvalin@mozilla.com</a>=
&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">On 09/06/2013 08:15 AM, T=
ina le Grand wrote:<br>
&gt; In section 3.1.3 Discontinuous Transmission (DTX), I&#39;m missing som=
e<br>
&gt; text that explains how to update the RTP packets correctly after a<br>
&gt; silent period. How should rtp timestamp and sequence number be<br>
&gt; updated while not sending rtp packets (silent period), to prepare<br>
&gt; for when we eventually have data to transmit again? My thinking is<br>
&gt; that rtp timestamp should be updated with the number of samples<br>
&gt; that were not sent, while sequence number should only be updated<br>
&gt; when sending packets. Correct?<br>
<br>
That&#39;s also my understanding, but I&#39;m far from an expert on this. C=
an<br>
anyone with good knowledge of RTP confirm that it&#39;s the right thing to =
do?<br>
<br></blockquote></div><br></div>In my experience and according to RFC 3551=
 this is the right thing to do. Furthermore, the first packet of a talkspur=
t, that is, the first packet after a silence period during which packets ha=
ve not been transmitted contiguously, SHOULD be distinguished by setting th=
e marker bit in the RTP data header to one. Also, since this is already sta=
ndard behavior for all audio codecs according to RFC 3551 there is probably=
 no need to define this behavior specifically for OPUS.<br>
<div class=3D"gmail_extra"><div>_____________<br>Roman Shpount</div>
<br></div></div>

--f46d043be1e0b8083504e5bc4c29--

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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Jean-Marc Valin <jmvalin@mozilla.com>, Tina le Grand <tina.legrand@webrtc.org>
Thread-Topic: [payload] Comment to draft-ietf-payload-rtp-opus-01
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Subject: Re: [payload] Comment to draft-ietf-payload-rtp-opus-01
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That's correct. See RFC 3551 section 4.1.

Since Opus DTX mode uses receiver concealment and comfort noise, it may be =
good to specify that the generic CN payload (RFC 3389) should not be used w=
ith Opus.

Mo


-----Original Message-----
From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On Behalf =
Of Jean-Marc Valin
Sent: Friday, September 06, 2013 3:20 PM
To: Tina le Grand
Cc: payload@ietf.org
Subject: Re: [payload] Comment to draft-ietf-payload-rtp-opus-01

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On 09/06/2013 08:15 AM, Tina le Grand wrote:
> In section 3.1.3 Discontinuous Transmission (DTX), I'm missing some
> text that explains how to update the RTP packets correctly after a
> silent period. How should rtp timestamp and sequence number be
> updated while not sending rtp packets (silent period), to prepare
> for when we eventually have data to transmit again? My thinking is
> that rtp timestamp should be updated with the number of samples
> that were not sent, while sequence number should only be updated
> when sending packets. Correct?

That's also my understanding, but I'm far from an expert on this. Can
anyone with good knowledge of RTP confirm that it's the right thing to do?

	Jean-Marc

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On 09/06/2013 03:46 PM, Roman Shpount wrote:
> In my experience and according to RFC 3551 this is the right thing
> to do. Furthermore, the first packet of a talkspurt, that is, the
> first packet after a silence period during which packets have not
> been transmitted contiguously, SHOULD be distinguished by setting
> the marker bit in the RTP data header to one. Also, since this is
> already standard behavior for all audio codecs according to RFC
> 3551 there is probably no need to define this behavior specifically
> for OPUS.

When in DTX mode, the Opus encoder will send a single "refresh" packet
every few seconds with the current noise level. Outside the encoder,
these packets are indistinguishable from normal Opus packets. Should
they have the marker bit set too or only the ones that correspond to
new voice activity. In the latter case, we'd have to figure out a way
to tell the refresh packets from the voice onset ones.

	Jean-Marc
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Jean-Marc Valin <jmvalin@mozilla.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [payload] Comment to draft-ietf-payload-rtp-opus-01
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Only voice onset after silence should set the marker bit, to aid in adjusti=
ng playout delay buffers. This, too, is covered in RFC 3551 section 4.1.

Mo

-----Original Message-----
From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On Behalf =
Of Jean-Marc Valin
Sent: Friday, September 06, 2013 5:19 PM
To: Roman Shpount
Cc: payload@ietf.org
Subject: Re: [payload] Comment to draft-ietf-payload-rtp-opus-01

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On 09/06/2013 03:46 PM, Roman Shpount wrote:
> In my experience and according to RFC 3551 this is the right thing
> to do. Furthermore, the first packet of a talkspurt, that is, the
> first packet after a silence period during which packets have not
> been transmitted contiguously, SHOULD be distinguished by setting
> the marker bit in the RTP data header to one. Also, since this is
> already standard behavior for all audio codecs according to RFC
> 3551 there is probably no need to define this behavior specifically
> for OPUS.

When in DTX mode, the Opus encoder will send a single "refresh" packet
every few seconds with the current noise level. Outside the encoder,
these packets are indistinguishable from normal Opus packets. Should
they have the marker bit set too or only the ones that correspond to
new voice activity. In the latter case, we'd have to figure out a way
to tell the refresh packets from the voice onset ones.

	Jean-Marc
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [payload] Comment to draft-ietf-payload-rtp-opus-01
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--f46d0442877a8e5f9404e5bdd0fc
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On Fri, Sep 6, 2013 at 5:19 PM, Jean-Marc Valin <jmvalin@mozilla.com> wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> On 09/06/2013 03:46 PM, Roman Shpount wrote:
> > In my experience and according to RFC 3551 this is the right thing
> > to do. Furthermore, the first packet of a talkspurt, that is, the
> > first packet after a silence period during which packets have not
> > been transmitted contiguously, SHOULD be distinguished by setting
> > the marker bit in the RTP data header to one. Also, since this is
> > already standard behavior for all audio codecs according to RFC
> > 3551 there is probably no need to define this behavior specifically
> > for OPUS.
>
> When in DTX mode, the Opus encoder will send a single "refresh" packet
> every few seconds with the current noise level. Outside the encoder,
> these packets are indistinguishable from normal Opus packets. Should
> they have the marker bit set too or only the ones that correspond to
> new voice activity. In the latter case, we'd have to figure out a way
> to tell the refresh packets from the voice onset ones.
>
>
Per RFC 3551 Section 4.1:

For applications which send either no packets or occasional comfort-noise
packets during silence, the first packet of a talkspur, that is, the first
packet after a silence period during which packets have not been
transmitted contiguously, SHOULD be distinguished by setting the marker bit
in the RTP data header to one.  The marker bit in all other packets is
zero.

So according to this only the first packet in a talkspur with voice onset
should have the marker bit set. Comfort noise packets should not have it.
_____________
Roman Shpount

--f46d0442877a8e5f9404e5bdd0fc
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On F=
ri, Sep 6, 2013 at 5:19 PM, Jean-Marc Valin <span dir=3D"ltr">&lt;<a href=
=3D"mailto:jmvalin@mozilla.com" target=3D"_blank">jmvalin@mozilla.com</a>&g=
t;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex"><div class=3D"im">-----BE=
GIN PGP SIGNED MESSAGE-----<br>
Hash: SHA1<br>
<br>
</div><div class=3D"im">On 09/06/2013 03:46 PM, Roman Shpount wrote:<br>
&gt; In my experience and according to RFC 3551 this is the right thing<br>
&gt; to do. Furthermore, the first packet of a talkspurt, that is, the<br>
&gt; first packet after a silence period during which packets have not<br>
&gt; been transmitted contiguously, SHOULD be distinguished by setting<br>
&gt; the marker bit in the RTP data header to one. Also, since this is<br>
&gt; already standard behavior for all audio codecs according to RFC<br>
&gt; 3551 there is probably no need to define this behavior specifically<br=
>
&gt; for OPUS.<br>
<br>
</div>When in DTX mode, the Opus encoder will send a single &quot;refresh&q=
uot; packet<br>
every few seconds with the current noise level. Outside the encoder,<br>
these packets are indistinguishable from normal Opus packets. Should<br>
they have the marker bit set too or only the ones that correspond to<br>
new voice activity. In the latter case, we&#39;d have to figure out a way<b=
r>
to tell the refresh packets from the voice onset ones.<br>
<div class=3D"im"><br></div></blockquote></div><br><div>Per RFC 3551 Sectio=
n 4.1:<br><br>For applications which send either
 no packets or occasional comfort-noise packets during silence, the=20
first packet of a talkspur, that is, the first packet after a silence=20
period during which packets have not been transmitted contiguously,=20
SHOULD be distinguished by setting the marker bit in the RTP data header
 to one. =A0The marker bit in all other packets is zero. <br><br></div>So a=
ccording to this only the first packet in a talkspur with voice onset shoul=
d have the marker bit set. Comfort noise packets should not have it.<br cle=
ar=3D"all">
<div>_____________<br>Roman Shpount</div>
<br></div></div>

--f46d0442877a8e5f9404e5bdd0fc--

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Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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WG,

As author I would really appreciate some reviews that the content are
appropriate. Thus I really appreciate responses even of the type. "Read
it, no issues spotted".

Don't assume that I am without flaws in my writing ;-).

Cheers

Magnus

On 2013-08-22 10:58, Ali C. Begen (abegen) wrote:
> (As a WG co-chair)
> 
> I am starting WGLC for the following draft (version 05). 
> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-howto/
> 
> This is an important document as it outlines how someone is supposed to write the payload specs.
> 
> If there are sections you don't understand or that are not clear enough, etc. please post them to the list. We need to get this right to minimize further headaches for future documents in our WG.
> 
> The deadline for the WGLC is September 20th.
> 
> -acbegen
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload
> 
> 


-- 

Magnus Westerlund

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Subject: Re: [payload] Comment to draft-ietf-payload-rtp-opus-01
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On 2013-09-06 23:35, Roman Shpount wrote:
> On Fri, Sep 6, 2013 at 5:19 PM, Jean-Marc Valin <jmvalin@mozilla.com
> <mailto:jmvalin@mozilla.com>> wrote:
> 
>     -----BEGIN PGP SIGNED MESSAGE-----
>     Hash: SHA1
> 
>     On 09/06/2013 03:46 PM, Roman Shpount wrote:
>     > In my experience and according to RFC 3551 this is the right thing
>     > to do. Furthermore, the first packet of a talkspurt, that is, the
>     > first packet after a silence period during which packets have not
>     > been transmitted contiguously, SHOULD be distinguished by setting
>     > the marker bit in the RTP data header to one. Also, since this is
>     > already standard behavior for all audio codecs according to RFC
>     > 3551 there is probably no need to define this behavior specifically
>     > for OPUS.
> 
>     When in DTX mode, the Opus encoder will send a single "refresh" packet
>     every few seconds with the current noise level. Outside the encoder,
>     these packets are indistinguishable from normal Opus packets. Should
>     they have the marker bit set too or only the ones that correspond to
>     new voice activity. In the latter case, we'd have to figure out a way
>     to tell the refresh packets from the voice onset ones.
> 
> 
> Per RFC 3551 Section 4.1:
> 
> For applications which send either no packets or occasional
> comfort-noise packets during silence, the first packet of a talkspur,
> that is, the first packet after a silence period during which packets
> have not been transmitted contiguously, SHOULD be distinguished by
> setting the marker bit in the RTP data header to one.  The marker bit in
> all other packets is zero.
> 
> So according to this only the first packet in a talkspur with voice
> onset should have the marker bit set. Comfort noise packets should not
> have it.

The behavior of the marker bit is up to every RTP Payload format to
define. However, I do recommend that you use this common behavior for
any codec carrying voice.

Cheers

Magnus Westerlund

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Subject: Re: [payload] Comment to draft-ietf-payload-rtp-opus-01
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On 2013-09-06 21:19, Jean-Marc Valin wrote:
> On 09/06/2013 08:15 AM, Tina le Grand wrote:
>> In section 3.1.3 Discontinuous Transmission (DTX), I'm missing
>> some text that explains how to update the RTP packets correctly
>> after a silent period. How should rtp timestamp and sequence
>> number be updated while not sending rtp packets (silent period),
>> to prepare for when we eventually have data to transmit again? My
>> thinking is that rtp timestamp should be updated with the number
>> of samples that were not sent, while sequence number should only
>> be updated when sending packets. Correct?
> 
> That's also my understanding, but I'm far from an expert on this.
> Can anyone with good knowledge of RTP confirm that it's the right
> thing to do?
> 

Yes, that is correct and should be easily to derive based on your
timestamp usage definition also. For voice the common rule is to
define the TS as the timestamp value corresponding to the earliest
sample present in the payload. Which is what you have done.

>From that follows in case of DTX that the packet after "skipped"
frames will have a TS field jumping to the start of the frame actually
included.

Cheers

Magnus Westerlund

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Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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Hi, Magnus:
I have read this document and it looks good.=20
Here are a few minor comments:
1. Section 7
Section 7 discusses four special consideration including
a. Media Format description
b. Security Considerations
c. Congestion Control
d. IANA Consideration
However it looks to me Media Format description or Congestion Control
Is not important or could be optional sections that need to be described=20
comparing with Security and IANA Considerations Section=20
Since  Section 7, 1st paragraph said:
"
These include the Security and IANA
 Considerations sections.
"
Also Section 7.1 seems just repeat what A.7 section said.

2.Section 7.2, 3rd paragraph:
Suggest the following change:
OLD TEXT:
"
That the decoding of the payload format or its media shows
substantial non-uniformity, either in output or in complexity to
perform the decoding operation. =20
"
NEW TEXT:
"
The decoding of the payload format or its media shows
substantial non-uniformity, either in output or in complexity to
perform the decoding operation. =20
"
3. Section 7.2, 3rd paragraph:
What does "goodput" mean? Good output?=20

4.Section 7.4, 2nd paragraph, 2nd sentence
It looks this sentence is disconnected from the 1st sentence.
Suggest the following change:
OLD TEXT:
"
Independently the format of these should be such that they can be
   included in the SDP attribute "a=3Dfmtp" string (See Section 6
   [RFC4566]) which is the common mapping. =20
"
NEW TEXT:
"
These parameters are independent of the format of these should be
such that they can be included in the SDP attribute "a=3Dfmtp" string (See =
Section 6
[RFC4566]) which is the common mapping. =20
"

Regards!
-Qin
-----Original Message-----
From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On Behalf =
Of Magnus Westerlund
Sent: Monday, September 09, 2013 10:08 PM
To: payload@ietf.org
Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05

WG,

As author I would really appreciate some reviews that the content are
appropriate. Thus I really appreciate responses even of the type. "Read
it, no issues spotted".

Don't assume that I am without flaws in my writing ;-).

Cheers

Magnus

On 2013-08-22 10:58, Ali C. Begen (abegen) wrote:
> (As a WG co-chair)
>=20
> I am starting WGLC for the following draft (version 05).=20
> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-howto/
>=20
> This is an important document as it outlines how someone is supposed to w=
rite the payload specs.
>=20
> If there are sections you don't understand or that are not clear enough, =
etc. please post them to the list. We need to get this right to minimize fu=
rther headaches for future documents in our WG.
>=20
> The deadline for the WGLC is September 20th.
>=20
> -acbegen
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload
>=20
>=20


--=20

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=E4r=F6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

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

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Subject: Re: [payload] Comment to draft-ietf-payload-rtp-opus-01
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Magnus Westerlund wrote:
> The behavior of the marker bit is up to every RTP Payload format to
> define. However, I do recommend that you use this common behavior for
> any codec carrying voice.

Okay, sounds like we should add a normative reference to RFC 3551 
Section 4.1, then.

From magnus.westerlund@ericsson.com  Tue Sep 10 01:51:05 2013
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Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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Many Thanks for the review.

See inline for comments.

On 2013-09-10 09:01, Qin Wu wrote:
> Hi, Magnus:
> I have read this document and it looks good. 
> Here are a few minor comments:
> 1. Section 7
> Section 7 discusses four special consideration including
> a. Media Format description
> b. Security Considerations
> c. Congestion Control
> d. IANA Consideration
> However it looks to me Media Format description or Congestion Control
> Is not important or could be optional sections that need to be described 
> comparing with Security and IANA Considerations Section 
> Since  Section 7, 1st paragraph said:
> "
> These include the Security and IANA
>  Considerations sections.
> "

So the Security and IANA are required for all drafts. The Media format
description I would recommend to ensure that people can understand the
important feature of the codec. Congestion control discussion is not
mandated by other procedures, but I at least thinks payload formats
really should include them. I would propose tweaking the section 7
introduction to say:

A number of sections in the payload format draft need special
consideration. These include the Security and IANA Considerations
sections that are required in all drafts. Payload formats are also
strongly recommended to have the media format description and congestion
control considerations.

Better?

> Also Section 7.1 seems just repeat what A.7 section said.

Ok, I think it is important to have the text that is present in A.7 as
it provides template instruction. What may need to consider is if the
text in 7.1 should contain anything additionally. Nothing springs to
mind immediately. I don't see the repetition as a major issue as it is
within the body vs within the template.

> 
> 2.Section 7.2, 3rd paragraph:
> Suggest the following change:
> OLD TEXT:
> "
> That the decoding of the payload format or its media shows
> substantial non-uniformity, either in output or in complexity to
> perform the decoding operation.  
> "
> NEW TEXT:
> "
> The decoding of the payload format or its media shows
> substantial non-uniformity, either in output or in complexity to
> perform the decoding operation.  
> "

Maybe this works better:

Potential security issues with an RTP payload format and the media
encoding that needs to be considered if they are applicable:

1. The decoding of the payload format or its media results in
substantial non-uniformity, either in output or in complexity to perform
the decoding operation.



> 3. Section 7.2, 3rd paragraph:
> What does "goodput" mean? Good output? 

Ahaa, goodput is how much actual usable (good) results that are produced.

See: http://en.wikipedia.org/wiki/Goodput

Maybe this is clearer:
Such inputs can cause some sort of disruption, i.e. a denial of service
attack on the receiver side by preventing that host from performing
usable work.


> 
> 4.Section 7.4, 2nd paragraph, 2nd sentence
> It looks this sentence is disconnected from the 1st sentence.
> Suggest the following change:
> OLD TEXT:
> "
> Independently the format of these should be such that they can be
>    included in the SDP attribute "a=fmtp" string (See Section 6
>    [RFC4566]) which is the common mapping.  
> "
> NEW TEXT:
> "
> These parameters are independent of the format of these should be
> such that they can be included in the SDP attribute "a=fmtp" string (See Section 6
> [RFC4566]) which is the common mapping.  
> "

Actually, independently are just confusing here. I think simplification to:

The format of these parameter should be such that they can be included
in the SDP attribute "a=fmtp" string ...

Is the right way to go.

Once more,

Thanks for the review.


Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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From: Qin Wu <bill.wu@huawei.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
Sent: Tuesday, September 10, 2013 4:51 PM
To: Qin Wu
Cc: payload@ietf.org
Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05


Many Thanks for the review.

See inline for comments.

On 2013-09-10 09:01, Qin Wu wrote:
> Hi, Magnus:
> I have read this document and it looks good.=20
> Here are a few minor comments:
> 1. Section 7
> Section 7 discusses four special consideration including
> a. Media Format description
> b. Security Considerations
> c. Congestion Control
> d. IANA Consideration
> However it looks to me Media Format description or Congestion Control
> Is not important or could be optional sections that need to be described=
=20
> comparing with Security and IANA Considerations Section=20
> Since  Section 7, 1st paragraph said:
> "
> These include the Security and IANA
>  Considerations sections.
> "

So the Security and IANA are required for all drafts. The Media format
description I would recommend to ensure that people can understand the
important feature of the codec. Congestion control discussion is not
mandated by other procedures, but I at least thinks payload formats
really should include them. I would propose tweaking the section 7
introduction to say:

A number of sections in the payload format draft need special
consideration. These include the Security and IANA Considerations
sections that are required in all drafts. Payload formats are also
strongly recommended to have the media format description and congestion
control considerations.

Better?

[Qin]: That is the change I am looking for, thanks.

> Also Section 7.1 seems just repeat what A.7 section said.

Ok, I think it is important to have the text that is present in A.7 as
it provides template instruction. What may need to consider is if the
text in 7.1 should contain anything additionally. Nothing springs to
mind immediately. I don't see the repetition as a major issue as it is
within the body vs within the template.

[Qin]: I can live with this, thanks for your clarification.
>=20
> 2.Section 7.2, 3rd paragraph:
> Suggest the following change:
> OLD TEXT:
> "
> That the decoding of the payload format or its media shows
> substantial non-uniformity, either in output or in complexity to
> perform the decoding operation. =20
> "
> NEW TEXT:
> "
> The decoding of the payload format or its media shows
> substantial non-uniformity, either in output or in complexity to
> perform the decoding operation. =20
> "

Maybe this works better:

Potential security issues with an RTP payload format and the media
encoding that needs to be considered if they are applicable:

1. The decoding of the payload format or its media results in
substantial non-uniformity, either in output or in complexity to perform
the decoding operation.

[Qin]:Your proposed change addresses my comment, thanks.

> 3. Section 7.2, 3rd paragraph:
> What does "goodput" mean? Good output?=20

Ahaa, goodput is how much actual usable (good) results that are produced.

See: http://en.wikipedia.org/wiki/Goodput

Maybe this is clearer:
Such inputs can cause some sort of disruption, i.e. a denial of service
attack on the receiver side by preventing that host from performing
usable work.

[Qin]: Looks good to me, thanks for taking my comment into account.
>=20
> 4.Section 7.4, 2nd paragraph, 2nd sentence
> It looks this sentence is disconnected from the 1st sentence.
> Suggest the following change:
> OLD TEXT:
> "
> Independently the format of these should be such that they can be
>    included in the SDP attribute "a=3Dfmtp" string (See Section 6
>    [RFC4566]) which is the common mapping. =20
> "
> NEW TEXT:
> "
> These parameters are independent of the format of these should be
> such that they can be included in the SDP attribute "a=3Dfmtp" string (Se=
e Section 6
> [RFC4566]) which is the common mapping. =20
> "

Actually, independently are just confusing here. I think simplification to:

The format of these parameter should be such that they can be included
in the SDP attribute "a=3Dfmtp" string ...

[Qin]: Make sense, thanks.


Is the right way to go.

Once more,

Thanks for the review.


Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=E4r=F6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From Lindsay@worldcastsystems.com  Tue Sep 10 06:50:16 2013
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Thread-Topic: [payload] I-D Action: draft-ietf-payload-rtp-aptx-01.txt
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From: "John Lindsay" <Lindsay@worldcastsystems.com>
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Cc: Foerster@worldcastsystems.com
Subject: [payload]  I-D Action: draft-ietf-payload-rtp-aptx-01.txt
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Hi

=20

A new draft has been uploaded to the IETF Payload workgroup at
https://datatracker.ietf.org/wg/payload/.

=20

The new document makes a few minor changes for IDNITS checking and also
takes account of the comments made by Peter Stevens last week.

=20

A summary of the changes are as follows.

=20

Page 1, IETF Copyright notice updated to 2013.

Page 10, updated formatting for IDNITS check.

Page 14, Section 6 updated to reflect RFC 6838 which supersedes RFC
4288.

Page 16, Section 6.2.1 type corrected An - A as per Peter Stevens
comments.

Page 18, Section 7 now reference 6.1 rather than 7.1, again thanks to
Peters proof reading.

Page 21, new RFC6338 referenced.

=20

Thanks to everyone who has read and commented on the draft.

=20

Regards

=20

John

=20

=20


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xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-GB link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>Hi<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>A new draft has been =
uploaded to the IETF Payload workgroup at <a =
href=3D"https://datatracker.ietf.org/wg/payload/">https://datatracker.iet=
f.org/wg/payload/</a>.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>The new document makes a =
few minor changes for IDNITS checking and also takes account of the =
comments made by Peter Stevens last week.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>A summary of the changes =
are as follows.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>Page 1, IETF Copyright =
notice updated to 2013.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>Page 10, updated formatting for IDNITS =
check.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>Page 14, Section 6 updated to reflect RFC 6838 =
which supersedes RFC 4288.<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>Page 16, Section 6.2.1 =
type corrected An &#8211; A as per Peter Stevens =
comments.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>Page 18, Section 7 now reference 6.1 rather than =
7.1, again thanks to Peters proof reading.<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>Page 21, new RFC6338 =
referenced.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>Thanks to everyone who =
has read and commented on the draft.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>Regards<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>John<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p></div></body></html>
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From hhhansen@MAYAH.com  Tue Sep 10 12:49:50 2013
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Thread-Topic: [payload] I-D Action: draft-ietf-payload-rtp-aptx-01.txt
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From: "Hans-Heinrich Hansen" <hhhansen@MAYAH.com>
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We will support and this draft in all our audio codecs!

Best regards,
Hans-Heinrich Hansen
MAYAH Communications GmbH


-----Original Message-----
From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org]On
Behalf Of internet-drafts@ietf.org
Sent: Friday, September 06, 2013 9:55 AM
To: i-d-announce@ietf.org
Cc: payload@ietf.org
Subject: [payload] I-D Action: draft-ietf-payload-rtp-aptx-01.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 Payloads Working =
Group of the IETF.

	Title           : RTP Payload Format for Standard apt-X and Enhanced =
apt-X Codecs
	Author(s)       : John Lindsay
                          Hartmut Foerster
	Filename        : draft-ietf-payload-rtp-aptx-01.txt
	Pages           : 22
	Date            : 2013-09-05

Abstract:
   This document specifies a scheme for packetizing Standard apt-X, or
   Enhanced apt-X, encoded audio data into Real-time Transport Protocol
   (RTP) packets.  The document describes a payload format that permits
   transmission of multiple related audio channels in a single RTP
   payload, and a means of establishing Standard apt-X and Enhanced
   apt-X connections through the Session Description Protocol (SDP).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-aptx

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-payload-rtp-aptx-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-payload-rtp-aptx-01


Please note that it may take a couple of minutes from the time of =
submission
until the htmlized version and diff are available at tools.ietf.org.

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

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


From prvs=959f67906=HeinzPeter.Reykers@wdr.de  Wed Sep 11 03:24:17 2013
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Date: Wed, 11 Sep 2013 12:23:40 +0200
From: "Heinz Peter Reykers" <HeinzPeter.Reykers@WDR.DE>
To: <payload@ietf.org>,"John Lindsay" <Lindsay@worldcastsystems.com>
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This is a MIME message. If you are reading this text, you may want to 
consider changing to a mail reader or gateway that understands how to 
properly handle MIME multipart messages.

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Dear John,
 
thank you for the corrections. Due to the fact that several
manufacturers already have implemented e-aptx as described in the draft
i hope instandly that it will become a rfc soon, so that all other
manufacturers will be compatible to each other! That's what we need for
our daily work in the master control rooms and also in the field.
 
 
Best regards
 
Heinz Peter 

 
____________________________________


Dipl.-Ing. Heinz Peter Reykers

Westdeutscher Rundfunk KÃ¶ln
Programmverbreitung / WAN 
Audio-Encoding - Kontribution 

Tel.: +49 (0)221 220 - 4183
Fax: +49 (0)221 220 - 774183
mobil: +49 (0)172 251 3066

e-mail: heinzpeter.reykers@wdr.de
 
Appellhofplatz 1
D-50600 KÃ¶ln
Archivhaus 1331
>>> "John Lindsay" <Lindsay@worldcastsystems.com> 10.09.13 15:50 >>>

Hi
 
A new draft has been uploaded to the IETF Payload workgroup at
https://datatracker.ietf.org/wg/payload/.
 
The new document makes a few minor changes for IDNITS checking and also
takes account of the comments made by Peter Stevens last week.
 
A summary of the changes are as follows.
 
Page 1, IETF Copyright notice updated to 2013.
Page 10, updated formatting for IDNITS check.
Page 14, Section 6 updated to reflect RFC 6838 which supersedes RFC
4288.
Page 16, Section 6.2.1 type corrected An â€“ A as per Peter Stevens
comments.
Page 18, Section 7 now reference 6.1 rather than 7.1, again thanks to
Peters proof reading.
Page 21, new RFC6338 referenced.
 
Thanks to everyone who has read and commented on the draft.
 
Regards
 
John
 
 

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crosoft-com:vml" xmlns:o =3D "urn:schemas-microsoft-com:office:office" =
xmlns:w =3D "urn:schemas-microsoft-com:office:word" xmlns:m =3D "http://sch=
emas.microsoft.com/office/2004/12/omml"><HEAD>
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</HEAD>
<BODY style=3D"MARGIN: 4px 4px 1px; FONT: 10pt Tahoma" lang=3DEN-GB =
link=3Dblue vLink=3Dpurple>
<DIV>Dear John,</DIV>
<DIV>&nbsp;</DIV>
<DIV>thank you for the corrections. Due to the fact that several manufactur=
ers already have implemented e-aptx as described in the draft i hope =
instandly that it will become a rfc soon, so that all other manufacturers =
will be compatible to each other! That's what we need for our daily work =
in the master control rooms and also in the field.</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>Best regards</DIV>
<DIV>&nbsp;</DIV>
<DIV>Heinz Peter&nbsp;<BR><BR></DIV>
<DIV>&nbsp;</DIV>
<DIV>
<DIV>____________________________________<BR><BR><BR>Dipl.-Ing. Heinz =
Peter Reykers<BR><BR>Westdeutscher Rundfunk K=C3=B6ln<BR>Programmverbreitun=
g /&nbsp;WAN=20
<DIV>Audio-Encoding&nbsp;- Kontribution&nbsp;<BR><BR>Tel.: +49 (0)221 220 =
- 4183<BR>Fax: +49 (0)221 220 - 774183<BR>mobil: +49 (0)172 251 3066<BR><BR=
>e-mail: <A href=3D"mailto:heinzpeter.reykers@wdr.de">heinzpeter.reykers@wd=
r.de</A></DIV></DIV>
<DIV>&nbsp;</DIV>
<DIV>Appellhofplatz 1</DIV>
<DIV>D-50600 K=C3=B6ln</DIV>
<DIV>Archivhaus 1331</DIV>&gt;&gt;&gt; "John Lindsay" &lt;Lindsay@worldcast=
systems.com&gt; 10.09.13 15:50 &gt;&gt;&gt;<BR></DIV>
<DIV class=3DWordSection1>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">Hi<o:p></o:p></SPAN></P=
>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">A new draft has been =
uploaded to the IETF Payload workgroup at <A href=3D"https://datatracker.ie=
tf.org/wg/payload/">https://datatracker.ietf.org/wg/payload/</A>.<o:p></o:p=
></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">The new document makes =
a few minor changes for IDNITS checking and also takes account of the =
comments made by Peter Stevens last week.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">A summary of the =
changes are as follows.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">Page 1, IETF Copyright =
notice updated to 2013.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">Page 10, updated =
formatting for IDNITS check.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">Page 14, Section 6 =
updated to reflect RFC 6838 which supersedes RFC 4288.<o:p></o:p></SPAN></P=
>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">Page 16, Section 6.2.1 =
type corrected An =E2=80=93 A as per Peter Stevens comments.<o:p></o:p></SP=
AN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">Page 18, Section 7 now =
reference 6.1 rather than 7.1, again thanks to Peters proof reading.<o:p></=
o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">Page 21, new RFC6338 =
referenced.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">Thanks to everyone who =
has read and commented on the draft.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">Regards<o:p></o:p></SPA=
N></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">John<o:p></o:p></SPAN><=
/P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P></DIV></BODY></HTML>

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From jonathan@vidyo.com  Thu Sep 12 08:49:43 2013
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From: Jonathan Lennox <jonathan@vidyo.com>
To: "payload@ietf.org" <payload@ietf.org>
Date: Thu, 12 Sep 2013 11:49:37 -0400
Thread-Topic: [payload] WGLC for draft-ietf-payload-rtp-howto-05
Thread-Index: Ac6vz0u/nejq6W0sTBKDc4Ob5jLQYA==
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Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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One point that's not mentioned in this document, but which might be worth b=
ringing up, is the conventions on how you choose a payload format's clock r=
ate.

I.e., the fact that clock rates are conventionally 90000 Hz for video (and =
we should give some background as to why), and they're conventionally the s=
ampling rate for audio.

It might also be worth discussing the benefits of the Opus clock rate appro=
ach (always use the maximum supported clock rate, i.e., 48000 Hz), vs. the =
conventional one, in the context of the multiple-clock-rates draft.

On Aug 22, 2013, at 4:58 AM, "Ali C. Begen (abegen)" <abegen@cisco.com> wro=
te:

> (As a WG co-chair)
>=20
> I am starting WGLC for the following draft (version 05).=20
> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-howto/
>=20
> This is an important document as it outlines how someone is supposed to w=
rite the payload specs.
>=20
> If there are sections you don't understand or that are not clear enough, =
etc. please post them to the list. We need to get this right to minimize fu=
rther headaches for future documents in our WG.
>=20
> The deadline for the WGLC is September 20th.
>=20
> -acbegen
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload
>=20

--
Jonathan Lennox
jonathan@vidyo.com



From csp@csperkins.org  Thu Sep 12 13:44:13 2013
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Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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On 22 Aug 2013, at 09:58, Ali C. Begen (abegen) <abegen@cisco.com> =
wrote:
> (As a WG co-chair)
>=20
> I am starting WGLC for the following draft (version 05).=20
> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-howto/
>=20
> This is an important document as it outlines how someone is supposed =
to write the payload specs.
>=20
> If there are sections you don't understand or that are not clear =
enough, etc. please post them to the list. We need to get this right to =
minimize further headaches for future documents in our WG.
>=20
> The deadline for the WGLC is September 20th.


I think this is an important draft, and I'm glad to see it progressing. =
It seems to be generally well written and containing good advice. I have =
some number of comments, but these are generally all minor points:

- In Section 3.1.2 the discussion of RFC 6015 mentions RFC 2733, but RFC =
2733 hasn=92t been cited or discussed

- In Section 3.1.2 the discussion of RFC 3611 might usefully point to =
the IANA registry for XR blocks and the XRBLOCK working group

- Section 3.2: the introduction to this section could be clearer that =
this is an overview of RTP features that are available independent of =
the payload format, and not a list of things that the payload format =
needs to explicitly opt in to/out of.

- Section 3.2.2 expands SSRC as =93Sender Source Identifier=94 but this =
should be =93Synchronisation Source Identifier=94.

- In Section 3.2.2 the statement =93[RFC5285] specifies how to extend =
the RTP header to carry metadata relating to the payload=94 is =
potentially misleading, since the type of information that usually goes =
into an RTP payload header is also =93metadata relating to the payload=94.=
 It might be useful to expand this discussion slightly, to make the =
distinction clear.

- In Section 3.2.2, it might be useful to explain that video payload =
formats are a common example of =93payload formats with a timestamp =
definition which results in no or little correlation between the media =
time instance and its transmission time=94; otherwise this sounds like =
such payload formats are something unusual.

- In Section 3.2.2, would it be helpful to explain typical RTP timestamp =
and sequence number behaviour when using discontinuous transmission?

- In Section 3.2.2, when talking about the RTP timestamp, it might be =
appropriate to reference to leap-seconds draft

- In Section 3.2.2, when talking about payload type numbers, it might be =
useful to be explicit that the binding is dynamic, and that other =
out-of-band configuration information in addition to the payload type to =
payload format binding is usually needed to understand a payload format. =
Reference Section 3.3.

- Should Section 3.2.3 reference the multiplex-guidelines or =
transport-multiplexing drafts?

- Section 3.2.4 says =93SSRCs with the same CNAME in different RTP =
sessions can be synchronized=94; this is also true of flows with =
different SSRC but the same CNAME within an RTP session.=20

- Section 3.4: should this say anything about AQM and/or QoS support?

- Section 4 might be clearer titled =93Standardisation Process for an =
RTP Payload Format=94

- Section 4 =93registration of the RTP payload format name=94 =96 how =
does this relate to the =93media type=94 discussed in Section 3.3?

- The discussion of SST and MST in Section 5.1.5 seems to confuse =
sessions with SSRCs. Isn=92t it more important whether the layers are =
sent within a single RTP media stream, or as separate RTP media streams =
with different SSRC values? Whether the different SSRCs are sent in one =
RTP session of many is then a separate issue, depending on how easily =
one wants to be able to separate the flows for QoS purposes.

- In Section 5.1.6, avoiding wrap-around within the RTCP reporting =
timeout interval is also important.

- The introduction to Section 7 could usefully reference the template =
payload format.

- Appendix A.10: mention that the RTP circuit breaker ought to be used =
and respected?

--=20
Colin Perkins
http://csperkins.org/




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Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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Hi Jonathan,

Thanks for the review.

On 2013-09-12 17:49, Jonathan Lennox wrote:
> One point that's not mentioned in this document, but which might be
> worth bringing up, is the conventions on how you choose a payload
> format's clock rate.
> 
> I.e., the fact that clock rates are conventionally 90000 Hz for video
> (and we should give some background as to why), and they're
> conventionally the sampling rate for audio.
> 
> It might also be worth discussing the benefits of the Opus clock rate
> approach (always use the maximum supported clock rate, i.e., 48000
> Hz), vs. the conventional one, in the context of the
> multiple-clock-rates draft.

Agreed, this should be better covered. I propose to add a new
sub-section in Section 5 covering this subject. I have therefore been
able to shorten the timestamp text in Section 3.2.2 to:

   Timestamp:  The RTP timestamp indicates the time instance the media
      sample belongs to.  For discrete media like video, it normally
      indicates when the media (frame) was sampled.  For continuous
      media it normally indicates the first time instance the media
      present in the payload represents.  For audio this is the sampling
      time of the first sample.  All RTP payload formats must specify
      the meaning of the timestamp value and the clock rates allowed.
      Selecting Timestamp rate is an active design choice and is further
      discussed in Section 5.2.

and remove a whole paragraph in Section 6.1: Audio Payloads



5.2.  Selecting Timestamp Definition

   The RTP Timestamp is an important part and have two design choices
   associated with it.  The first is the definition that determines what
   the timestamp value in a particular RTP packet will be, the second is
   which timestamp rate should be used.

   The timestamp definition needs to explicitly define what the
   timestamp value in the RTP packet represent for a particular payload
   format.  Two common definitions are used; for discretely sampled
   media, like video frames, the sampling time of the earliest included
   video frame which the data represent (fully or partially) is used;
   for continuous media like audio, the sampling time of the earliest
   sample which the payload data represent.  There exist cases where
   more elaborate or other definitions are used.

   RTP payload formats with a timestamp definition which results in no
   or little correlation between the media time instance and its
   transmission time cause the RTCP jitter calculation to become
   unusable due to the errors introduced on the sender side.  Such
   examples are for example payload formats for video that requires
   multiple packets, where a set of packets will have the same
   timestamp, another are payloads that includes alternating set of
   redundant information, if the definition used are for the earliest
   data included, the the timestamp may jump back and forth.  It should
   be noted if the payload format has this property or not.

   A RTP payload format also needs to define what timestamp rates, or
   clock rates (as it is also called), that may be used.  Depending on
   the RTP payload format this may be a single rate or multiple ones or
   theoretically any rate.  So what needs to be considered when
   selecting rate?

   The rate needs be selected so that one can determine where in the
   timeline of the media a particular sample (e.g. individual audio
   sample, or video frame) or set of samples (e.g. audio frames) belong.
   To enable correct synchronization of this data with previous frames,
   including over periods of discontinuous transmission or
   irregularities.

   For audio it is common to require audio sample accuracy.  Thus one
   commonly selects the input sampling rate to the code.  This can
   however be challenging for for audio codecs that support multiple
   different sampling frequencies, either as codec input or being used
   internally but effecting output, for example frame duration.
   Depending on how one expects to use these different sampling rates
   one can allow multiple timestamp rates, each matching a particular
   codec input or sampling rate.  However, due to the issues with using
   multiple different RTP timestamp rates for the same source (SSRC)
   [I-D.ietf-avtext-multiple-clock-rates] this should be avoided if one
   expects to need to switch between modes.

   An alternatives then is to find a common denominator frequency
   between the different modes, e.g. OPUS [I-D.ietf-payload-rtp-opus]
   that uses 48 KHz.  If the different modes uses or can use a common
   input/output frequency then selecting this also needs to be
   considered.  However, it is important to consider all aspects as the
   case of AMR-WB+ [RFC4352] illustrates.  AMR-WB+'s RTP timestamp rate
   has the very unusual value of 72 kHz, despite the fact that output
   normally is at a sample rate of 48kHz.  The design is motivated by
   the media codec's production of a large range of different frame
   lengths in time perspective.  The 72 kHz timestamp rate is the
   smallest found value that would make all of the frames the codec
   could produce result in an integer frame length in RTP timestamp
   ticks.  This way, a receiver can always correctly place the frames in
   relation to any other frame, even when the frame length changes.  The
   downside is that the decoder outputs for certain frame lengths is in
   fact partial samples.  The result is that the output in samples from
   the codec will vary from frame to frame, potentially making
   implementation more difficult.

   Video codecs has commonly been using 90 kHz, the reason is this is a
   common denominator between the usually used frame rates such as 24,
   25, 30, 50 and 60, and NTSC's odd 29.97 Hz.  There does however exist
   at least one exception in the payload format for SMPTE 292M video
   [RFC3497] that uses a clock rate of 148.5 Mhz. The reason here is
   that the timestamp then identify the exact start sample within a
   video frame.

   Timestamp rates below 1000 Hz are not appropriate because it will
   cause a too low resolution in the RTCP measurements that are
   expressed in RTP timestamps.  This is the main reason that the text
   RTP payload formats, like T.140 [RFC4103] uses 1000 Hz.
> 
> On Aug 22, 2013, at 4:58 AM, "Ali C. Begen (abegen)"
> <abegen@cisco.com> wrote:
> 
>> (As a WG co-chair)
>> 
>> I am starting WGLC for the following draft (version 05). 
>> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-howto/
>> 
>> This is an important document as it outlines how someone is
>> supposed to write the payload specs.
>> 
>> If there are sections you don't understand or that are not clear
>> enough, etc. please post them to the list. We need to get this
>> right to minimize further headaches for future documents in our
>> WG.
>> 
>> The deadline for the WGLC is September 20th.
>> 
>> -acbegen _______________________________________________ payload
>> mailing list payload@ietf.org 
>> https://www.ietf.org/mailman/listinfo/payload
>> 
> 
> -- Jonathan Lennox jonathan@vidyo.com
> 
> 
> _______________________________________________ payload mailing list 
> payload@ietf.org https://www.ietf.org/mailman/listinfo/payload
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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From: Richard Barnes <rlb@ipv.sx>
To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [payload] AD Evaluation: draft-ietf-avt-rtp-isac-04
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--089e0115e84a4c189104e64809dd
Content-Type: text/plain; charset=ISO-8859-1

Since Real Soon Now has turned into Not Very Soon At All, I'm moving this
one back to Active.  When you guys are ready to get around to it, please
re-request publication.
--Richard


On Thu, Jun 13, 2013 at 8:45 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

>  On 05/28/2013 08:34 PM, Richard Barnes wrote:
>
> Dear Authors,
>
>  Any update on these issues?
>
>
> Working on it, but as you know from other discussions, there isn't exactly
> a lack of things to work on at the moment - it dropped off my queue, and
> I'm picking it up again now.
>
> We'll get back to you Real Soon Now.
>
>
>  Thanks,
> --Richard
>
>
> On Mon, Apr 1, 2013 at 5:44 PM, Roni Even <ron.even.tlv@gmail.com> wrote:
>
>>  Hi Richard,
>>
>> These fields are not part of the RTP header but part of the iSAC payload
>> header and I agree that it will be better to clarify them.
>>
>> Roni
>>
>>
>>
>> *From:* Richard Barnes [mailto:rlb@ipv.sx]
>> *Sent:* 01 April, 2013 9:47 PM
>> *To:* Roni Even
>> *Cc:* payload@ietf.org; draft-ietf-avt-rtp-isac@tools.ietf.org
>> *Subject:* Re: [payload] AD Evaluation: draft-ietf-avt-rtp-isac-04
>>
>>
>>
>> Sure. I didn't mean to imply that we need a reference to the codec, in
>> fact the opposite.  I was looking for this document to tell me how to parse
>> out the fields I would feed into the black box of the codec.  They describe
>> the fields, but it seems to me that they need more detail in order for
>> people to know how to parse them out of the payload.
>>
>>
>>
>> --Richard
>>
>>
>>
>> On Mon, Apr 1, 2013 at 2:03 PM, Roni Even <ron.even.tlv@gmail.com> wrote:
>>
>> Hi Richard,
>>
>> As a general comment, for RTP payload specifications we do not require a
>> normative reference to the codec. We only require enough information that
>> will allow an RTP receiver to parse the information and move it to the
>> decoder that can be a black box. This allows us to publish RTP payload
>> specification also for codecs whose specifications are not publically
>> available.
>>
>> Still in this case more information can be supplied.
>>
>>
>>
>> As for the comments the authors should address them
>>
>>
>>
>> Roni Even
>>
>>
>>
>> *From:* payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] *On
>> Behalf Of *Richard Barnes
>> *Sent:* 01 April, 2013 8:06 PM
>> *To:* payload@ietf.org
>> *Cc:* draft-ietf-avt-rtp-isac@tools.ietf.org
>> *Subject:* [payload] AD Evaluation: draft-ietf-avt-rtp-isac-04
>>
>>
>>
>> Overall, this document looks in pretty good shape.  A couple of comments
>> I would like to get resolved before IETF LC:
>>
>>
>>
>> Section 3.1., "Consult source code for details"
>>
>> If the details are only specified in the source code, then the source
>> needs to be a normative reference.  And I would prefer to avoid that  :)
>>  Better to specify here how the BEI and FL fields are encoded.
>> Alternatively, if the codec really does expect a combined BEI/FL field, you
>> could specify such a field here, opaque to the RTP layer.  However, you
>> would at least need to say how the recipient knows where this field starts
>> and stops.
>>
>>
>>
>> Section 3.2., "The length of the encoded data is variable and depends
>> on the signal characteristics and the target bit rate."
>>
>> However, there's nothing in this section that tells a recipient how to
>> determine what this length is.
>>
>>
>>
>> Section 3.3., "... verifying the CRC checksum ..."
>>
>> Please specify which CRC function is to be applied, and how the CRC value
>> field is formatted.
>>
>>
>>
>> Section 3.3., "If this value would exceed 255 at encoding..."
>>
>> It sounds like the LEN field is a single octet unsigned integer.  Please
>> state that explicitly.
>>
>>
>>
>> Section 9.2. Informative References.
>>
>> Better path to source code would be "webrtc/
>> modules/audio_coding/codecs/isac"
>>
>>
>>
>> Thanks,
>>
>> --Richard
>>
>>
>>
>
>
>
> _______________________________________________
> payload mailing listpayload@ietf.orghttps://www.ietf.org/mailman/listinfo/payload
>
>
>

--089e0115e84a4c189104e64809dd
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Since Real Soon Now has turned into Not Very Soon At All, =
I&#39;m moving this one back to Active. =A0When you guys are ready to get a=
round to it, please re-request publication.<div>--Richard</div></div><div c=
lass=3D"gmail_extra">
<br><br><div class=3D"gmail_quote">On Thu, Jun 13, 2013 at 8:45 AM, Harald =
Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" ta=
rget=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">

 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF"><div class=3D"im">
    <div>On 05/28/2013 08:34 PM, Richard Barnes
      wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">Dear Authors,
        <div><br>
        </div>
        <div>Any update on these issues?</div>
      </div>
    </blockquote>
    <br></div>
    Working on it, but as you know from other discussions, there isn&#39;t
    exactly a lack of things to work on at the moment - it dropped off
    my queue, and I&#39;m picking it up again now.<br>
    <br>
    We&#39;ll get back to you Real Soon Now.<br>
    <br>
    <blockquote type=3D"cite"><div><div class=3D"h5">
      <div dir=3D"ltr">
        <div><br>
        </div>
        <div>Thanks,</div>
        <div>--Richard</div>
      </div>
      <div class=3D"gmail_extra"><br>
        <br>
        <div class=3D"gmail_quote">On Mon, Apr 1, 2013 at 5:44 PM, Roni
          Even <span dir=3D"ltr">&lt;<a href=3D"mailto:ron.even.tlv@gmail.c=
om" target=3D"_blank">ron.even.tlv@gmail.com</a>&gt;</span>
          wrote:<br>
          <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex">
            <div link=3D"blue" vlink=3D"purple" lang=3D"EN-US">
              <div>
                <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font=
-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi
                    Richard,</span></p>
                <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font=
-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">These
                    fields are not part of the RTP header but part of
                    the iSAC payload header and I agree that it will be
                    better to clarify them.</span></p>
                <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font=
-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Roni</spa=
n></p>
                <p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font=
-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">=A0</span=
></p>
                <p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;f=
ont-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span=
 style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&=
quot;">
                    Richard Barnes [mailto:<a href=3D"mailto:rlb@ipv.sx" ta=
rget=3D"_blank">rlb@ipv.sx</a>]
                    <br>
                    <b>Sent:</b> 01 April, 2013 9:47 PM<br>
                    <b>To:</b> Roni Even<br>
                    <b>Cc:</b> <a href=3D"mailto:payload@ietf.org" target=
=3D"_blank">payload@ietf.org</a>;
                    <a href=3D"mailto:draft-ietf-avt-rtp-isac@tools.ietf.or=
g" target=3D"_blank">draft-ietf-avt-rtp-isac@tools.ietf.org</a><br>
                    <b>Subject:</b> Re: [payload] AD Evaluation:
                    draft-ietf-avt-rtp-isac-04</span></p>
                <div>
                  <div>
                    <p class=3D"MsoNormal">=A0</p>
                    <p class=3D"MsoNormal">Sure. I didn&#39;t mean to imply
                      that we need a reference to the codec, in fact the
                      opposite. =A0I was looking for this document to tell
                      me how to parse out the fields I would feed into
                      the black box of the codec. =A0They describe the
                      fields, but it seems to me that they need more
                      detail in order for people to know how to parse
                      them out of the payload.</p>
                    <div>
                      <p class=3D"MsoNormal">=A0</p>
                    </div>
                    <div>
                      <p class=3D"MsoNormal">--Richard</p>
                    </div>
                    <div>
                      <p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"=
>=A0</p>
                      <div>
                        <p class=3D"MsoNormal">On Mon, Apr 1, 2013 at 2:03
                          PM, Roni Even &lt;<a href=3D"mailto:ron.even.tlv@=
gmail.com" target=3D"_blank">ron.even.tlv@gmail.com</a>&gt;
                          wrote:</p>
                        <div>
                          <div>
                            <p class=3D"MsoNormal"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497=
d">Hi
                                Richard,</span></p>
                            <p class=3D"MsoNormal"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497=
d">As
                                a general comment, for RTP payload
                                specifications we do not require a
                                normative reference to the codec. We
                                only require enough information that
                                will allow an RTP receiver to parse the
                                information and move it to the decoder
                                that can be a black box. This allows us
                                to publish RTP payload specification
                                also for codecs whose specifications are
                                not publically available.</span></p>
                            <p class=3D"MsoNormal"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497=
d">Still
                                in this case more information can be
                                supplied.</span></p>
                            <p class=3D"MsoNormal"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497=
d">=A0</span></p>
                            <p class=3D"MsoNormal"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497=
d">As
                                for the comments the authors should
                                address them</span></p>
                            <p class=3D"MsoNormal"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497=
d">=A0</span></p>
                            <p class=3D"MsoNormal"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497=
d">Roni
                                Even</span></p>
                            <p class=3D"MsoNormal"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497=
d">=A0</span></p>
                            <p class=3D"MsoNormal"><b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</sp=
an></b><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot=
;sans-serif&quot;">
                                <a href=3D"mailto:payload-bounces@ietf.org"=
 target=3D"_blank">payload-bounces@ietf.org</a>
                                [mailto:<a href=3D"mailto:payload-bounces@i=
etf.org" target=3D"_blank">payload-bounces@ietf.org</a>]
                                <b>On Behalf Of </b>Richard Barnes<br>
                                <b>Sent:</b> 01 April, 2013 8:06 PM<br>
                                <b>To:</b> <a href=3D"mailto:payload@ietf.o=
rg" target=3D"_blank">payload@ietf.org</a><br>
                                <b>Cc:</b> <a href=3D"mailto:draft-ietf-avt=
-rtp-isac@tools.ietf.org" target=3D"_blank">draft-ietf-avt-rtp-isac@tools.i=
etf.org</a><br>
                                <b>Subject:</b> [payload] AD Evaluation:
                                draft-ietf-avt-rtp-isac-04</span></p>
                            <div>
                              <div>
                                <p class=3D"MsoNormal">=A0</p>
                                <p class=3D"MsoNormal">Overall, this
                                  document looks in pretty good shape.
                                  =A0A couple of comments I would like to
                                  get resolved before IETF LC:</p>
                                <div>
                                  <p class=3D"MsoNormal">=A0</p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal">Section 3.1.,
                                    &quot;Consult source code for details&q=
uot;</p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal">If the details
                                    are only specified in the source
                                    code, then the source needs to be a
                                    normative reference. =A0And I would
                                    prefer to avoid that =A0:) =A0Better to
                                    specify here how the BEI and FL
                                    fields are encoded. =A0 Alternatively,
                                    if the codec really does expect a
                                    combined BEI/FL field, you could
                                    specify such a field here, opaque to
                                    the RTP layer. =A0However, you would
                                    at least need to say how the
                                    recipient knows where this field
                                    starts and stops.</p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal">=A0</p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal">Section 3.2., &quo=
t;<span style=3D"font-size:10.0pt">The
                                      length of the encoded data is
                                      variable and depends on=A0the signal
                                      characteristics and the target bit
                                      rate.</span>&quot;</p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal">However, there&#39=
;s
                                    nothing in this section that tells a
                                    recipient how to determine what this
                                    length is. =A0</p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal">=A0</p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal">
                                    Section 3.3., &quot;...=A0<span style=
=3D"font-size:10.0pt">verifying
                                      the CRC checksum ...</span>&quot;</p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal">Please specify
                                    which CRC function is to be applied,
                                    and how the CRC value field is
                                    formatted.</p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal">=A0</p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal">Section 3.3., &quo=
t;<span style=3D"font-size:10.0pt">If this
                                      value would exceed 255 at
                                      encoding...</span>&quot;</p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal">It sounds like
                                    the LEN field is a single octet
                                    unsigned integer. =A0Please state that
                                    explicitly.=A0</p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal">=A0</p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal">Section 9.2.
                                    Informative References.</p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal">Better path to
                                    source code would be &quot;webrtc/<span=
 style=3D"font-size:10.0pt">modules/audio_coding/codecs/isac&quot;</span></=
p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal">=A0</p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal"><span style=3D"fon=
t-size:10.0pt">Thanks,</span></p>
                                </div>
                                <div>
                                  <p class=3D"MsoNormal"><span style=3D"fon=
t-size:10.0pt">--Richard</span></p>
                                </div>
                              </div>
                            </div>
                          </div>
                        </div>
                      </div>
                      <p class=3D"MsoNormal">=A0</p>
                    </div>
                  </div>
                </div>
              </div>
            </div>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      </div></div><pre>_______________________________________________
payload mailing list
<a href=3D"mailto:payload@ietf.org" target=3D"_blank">payload@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/payload" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/payload</a>
</pre>
    </blockquote>
    <br>
  </div>

</blockquote></div><br></div>

--089e0115e84a4c189104e64809dd--

From ietf-ipr@ietf.org  Tue Sep 10 10:06:17 2013
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Dear Ye-Kui Wang, Yago Sanchez, Thomas Schierl, Stephan Wenger, Miska M. Ha=
nnuksela:

 An IPR disclosure that pertains to your Internet-Draft entitled "RTP Paylo=
ad
Format for High Efficiency Video Coding" (draft-ietf-payload-rtp-h265) was
submitted to the IETF Secretariat on 2013-09-09 and has been posted on the =
"IETF
Page of Intellectual Property Rights Disclosures"
(https://datatracker.ietf.org/ipr/2190/). The title of the IPR disclosure is
"Telefonaktiebolaget LM Ericsson (publ)'s Statement about IPR related to dr=
aft-
ietf-payload-rtp-h265-01."");

The IETF Secretariat


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "payload@ietf.org" <payload@ietf.org>
Thread-Topic: WGLC for draft-ietf-payload-rtp-aptx-01
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Subject: [payload] WGLC for draft-ietf-payload-rtp-aptx-01
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(As a WG co-chair)

I am starting WGLC for the following draft (version 01).=20
https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-aptx/

It is a short document. If you have comments, agreements or disagreements, =
please let the list know in either case.

The deadline for the WGLC is September 30th.

-acbegen=

From jonathan@vidyo.com  Mon Sep 16 11:36:59 2013
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From: Jonathan Lennox <jonathan@vidyo.com>
To: "Ali C. Begen (abegen)" <abegen@cisco.com>
Date: Mon, 16 Sep 2013 14:36:22 -0400
Thread-Topic: [payload] WGLC for draft-ietf-payload-rtp-aptx-01
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The document discusses that a number of items are dependent on the "deliver=
y method", and says that the delivery method can be negotiated through offe=
r/answer.  However, the delivery method parameter is never actually defined=
 anywhere.  Either delivery method needs to be defined, or discussion of it=
 needs to be removed, depending on whether or not delivery-method was actua=
lly meant to be included in the final spec.

The SDP mapping says that maxptime is encoded in the fmtp parameter.  This =
is incorrect; as specified by RFC 4566, maxptime is carried in its own SDP =
a=3D attribute.

The RTP header section says that the marker bit is not used.  Unless there'=
s a good reason for this, I suggest that instead the usual RFC 3551 marker =
bit semantic for audio be applied (i.e., the marker bit is set for end of s=
ilence / beginning of talkspurt).  If there's a reason this semantic is ina=
ppropriate, it should be explained.

The RTP header section also says that the RTP dynamic payload types can onl=
y be in the range 96 - 127.  This is incorrect; that's the primary range, b=
ut other values can be used once that range is exhausted.  See, e.g., the r=
ecent discussion on the avtcore list.

No information is given about the semantic content of the data carried in t=
he auxiliary data channel.  Is the semantic of this data defined by the cod=
ec, or would it need to be described or negotiated in SDP?  (Doing neither,=
 i.e. providing an arbitrary data channel with no mechanism for describing =
or negotiating its content, seems likely to lead to interoperability proble=
ms.)  Also, are there any security considerations introduced by the auxilia=
ry data channel?

It would be good to have a reference to some document by CSR plc describing=
 the apt-X codecs, even if they don't include full proprietary details of t=
he workings of the codec.

On Sep 16, 2013, at 9:08 AM, "Ali C. Begen (abegen)" <abegen@cisco.com> wro=
te:

> (As a WG co-chair)
>=20
> I am starting WGLC for the following draft (version 01).=20
> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-aptx/
>=20
> It is a short document. If you have comments, agreements or disagreements=
, please let the list know in either case.
>=20
> The deadline for the WGLC is September 30th.
>=20
> -acbegen
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload
>=20

--
Jonathan Lennox
jonathan@vidyo.com



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Subject: Re: [payload] I-D Action: draft-ietf-payload-rtp-aptx-01.txt
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Brief scan, one question:

Is it possible to offer a stable reference for what "Standard apt-x" and 
"Enhanced apt-x" means?

The closest I can get is "licensed by CSR plc", which presumably means 
that "it means what CSR plc thinks it means", but it would be great to 
have that stated explicitly.

Something like:

The codecs "apt-x" and "Extended apt-x" are defined by CSR plc, an UK 
company.

Section 3 of the draft *almost* says that, but isn't completely 
unambiguous ("licensed by" can mean both "CSR is a licensor" and "CSR is 
a licensee, someone else defines it").

I *think* this is a nit.


On 09/10/2013 03:50 PM, John Lindsay wrote:
>
> Hi
>
> A new draft has been uploaded to the IETF Payload workgroup at 
> https://datatracker.ietf.org/wg/payload/.
>
> The new document makes a few minor changes for IDNITS checking and 
> also takes account of the comments made by Peter Stevens last week.
>
> A summary of the changes are as follows.
>
> Page 1, IETF Copyright notice updated to 2013.
>
> Page 10, updated formatting for IDNITS check.
>
> Page 14, Section 6 updated to reflect RFC 6838 which supersedes RFC 4288.
>
> Page 16, Section 6.2.1 type corrected An -- A as per Peter Stevens 
> comments.
>
> Page 18, Section 7 now reference 6.1 rather than 7.1, again thanks to 
> Peters proof reading.
>
> Page 21, new RFC6338 referenced.
>
> Thanks to everyone who has read and commented on the draft.
>
> Regards
>
> John
>
>
>
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload


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    <div class="moz-cite-prefix">Brief scan, one question:<br>
      <br>
      Is it possible to offer a stable reference for what "Standard
      apt-x" and "Enhanced apt-x" means?<br>
      <br>
      The closest I can get is "licensed by CSR plc", which presumably
      means that "it means what CSR plc thinks it means", but it would
      be great to have that stated explicitly.<br>
      <br>
      Something like:<br>
      <br>
      The codecs "apt-x" and "Extended apt-x" are defined by CSR plc, an
      UK company.<br>
      <br>
      Section 3 of the draft *almost* says that, but isn't completely
      unambiguous ("licensed by" can mean both "CSR is a licensor" and
      "CSR is a licensee, someone else defines it").<br>
      <br>
      I *think* this is a nit.<br>
      <br>
      <br>
      On 09/10/2013 03:50 PM, John Lindsay wrote:<br>
    </div>
    <blockquote
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	font-family:"Calibri","sans-serif";
	color:windowtext;}
span.EmailStyle20
	{mso-style-type:personal-reply;
	font-family:"Calibri","sans-serif";
	color:#1F497D;}
.MsoChpDefault
	{mso-style-type:export-only;
	font-size:10.0pt;}
@page WordSection1
	{size:612.0pt 792.0pt;
	margin:72.0pt 72.0pt 72.0pt 72.0pt;}
div.WordSection1
	{page:WordSection1;}
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<o:shapedefaults v:ext="edit" spidmax="1028" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]-->
      <div class="WordSection1">
        <p class="MsoNormal"><span style="color:#1F497D">Hi<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">A new draft has
            been uploaded to the IETF Payload workgroup at <a
              moz-do-not-send="true"
              href="https://datatracker.ietf.org/wg/payload/">https://datatracker.ietf.org/wg/payload/</a>.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">The new
            document makes a few minor changes for IDNITS checking and
            also takes account of the comments made by Peter Stevens
            last week.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">A summary of
            the changes are as follows.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">Page 1, IETF
            Copyright notice updated to 2013.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">Page 10,
            updated formatting for IDNITS check.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">Page 14,
            Section 6 updated to reflect RFC 6838 which supersedes RFC
            4288.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">Page 16,
            Section 6.2.1 type corrected An &#8211; A as per Peter Stevens
            comments.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">Page 18,
            Section 7 now reference 6.1 rather than 7.1, again thanks to
            Peters proof reading.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">Page 21, new
            RFC6338 referenced.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">Thanks to
            everyone who has read and commented on the draft.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">Regards<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">John<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
payload mailing list
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</pre>
    </blockquote>
    <br>
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Hi,

Thanks for the review comments. I have tried addressing them. I am
submitting an updated draft including these edits. But, I do need
feedback on these edits.

See below.

On 2013-09-12 19:04, Colin Perkins wrote:
> On 22 Aug 2013, at 09:58, Ali C. Begen (abegen) <abegen@cisco.com>
> wrote:
>> (As a WG co-chair)
>> 
>> I am starting WGLC for the following draft (version 05). 
>> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-howto/
>> 
>> This is an important document as it outlines how someone is
>> supposed to write the payload specs.
>> 
>> If there are sections you don't understand or that are not clear
>> enough, etc. please post them to the list. We need to get this
>> right to minimize further headaches for future documents in our
>> WG.
>> 
>> The deadline for the WGLC is September 20th.
> 
> 
> I think this is an important draft, and I'm glad to see it
> progressing. It seems to be generally well written and containing
> good advice. I have some number of comments, but these are generally
> all minor points:
> 
> - In Section 3.1.2 the discussion of RFC 6015 mentions RFC 2733, but
> RFC 2733 hasn’t been cited or discussed

Correct, I added a reference in RFC 5109 text to state that this
replaced RFC 2733, and added as references in both places.

   RFC 5109:  The "RTP Payload Format for Generic Forward Error
      Correction (FEC)" [RFC5109] provides an XOR based FEC of the whole
      or parts of a number of RTP packets.  This specification replaced
      the previous specification for XOR based FEC [RFC2733].  These FEC
      packets are sent in a separate stream or as a redundant encoding
      using RFC 2198.  This FEC scheme has certain restrictions in the
      number of packets it can protect.  It is suitable for low-to-
      medium delay tolerant applications with limited amount of RTP
      packets.

   RFC 6015:  The "RTP Payload Format for 1-D Interleaved Parity Forward
      Error Correction (FEC)" [RFC6015] provides a variant of the XOR
      based Generic protection defined in [RFC2733].  The main
      difference is to use interleaving scheme on which packets gets
      included as source packets for a particular protection packet.
      The interleaving is defined by using every L packets as source
      data.  And then produce protection data over D number of packets.
      Thus each block of D x L source packets will result in L number of
      Repair packets, each capable of repairing one loss.  The goal is
      to provide better burst error robustness when the packet rate is
      higher.

> 
> - In Section 3.1.2 the discussion of RFC 3611 might usefully point to
> the IANA registry for XR blocks and the XRBLOCK working group

   RFC 3611:  The RTCP Extended Reports (RTCP XR) [RFC3611] consists of
      a framework for reports sent within RTCP.  It can easily be
      extended by defining new report formats, which has occurred.  The
      list of specified formats are available in IANA's RTCP XR Block
      Type registry (http://www.iana.org/assignments/rtcp-xr-block-types
      /rtcp-xr-block-types.xhtml).  The report formats that are defined
      in RFC3611 provide report information on packet loss, packet
      duplication, packet reception times, RTCP statistics summary and
      VoIP Quality.  [RFC3611] also defines a mechanism that allows
      receivers to calculate the RTT to other session participants when
      used.

> 
> - Section 3.2: the introduction to this section could be clearer that
> this is an overview of RTP features that are available independent of
> the payload format, and not a list of things that the payload format
> needs to explicitly opt in to/out of.

This section reviews a number of RTP features and concepts that are
available in RTP independent of the payload format. The RTP payload
format can make use of these when appropriate, and in some cases even
effect the behavior (RTP Timestamp and marker bit). This section does
not remove the necessity to read up on RTP. However it does point out a
few important details to remember when designing a payload format.


> 
> - Section 3.2.2 expands SSRC as “Sender Source Identifier” but this
> should be “Synchronisation Source Identifier”.

Note that I also change the fourth line to say "of multiple media
sources" rather than "from multiple senders.

   SSRC:  The Synchronisation Source Identifier (SSRC) is normally not
      used by a payload format other than to identify the RTP timestamp
      and sequence number space a packet belongs to, allowing
      simultaneously reception of multiple media sources.  However, some
      of the RTP mechanisms for improving resilience to packet loss uses
      multiple SSRCs to separate original data and repair or redundant
      data.


> 
> - In Section 3.2.2 the statement “[RFC5285] specifies how to extend
> the RTP header to carry metadata relating to the payload” is
> potentially misleading, since the type of information that usually
> goes into an RTP payload header is also “metadata relating to the
> payload”. It might be useful to expand this discussion slightly, to
> make the distinction clear.

Is this better?

   Header extensions:  Payload formats normally specifies the carrying
      of meta data related to the media payload in the payload header.
      In some cases the meta data may be more generic, e.g. relevant
      more to the media type (e.g. audio or video), or RTP packets in
      general.  In those cases the carrying of this additional meta data
      can be specified in a header extension.  One example is the
      transport of SMPTE time-codes in the RTP header [RFC5484].  As
      [RFC5285] specifies, header extensions must not contain
      information required in order to decode the payload successfully.

> 
> - In Section 3.2.2, it might be useful to explain that video payload
> formats are a common example of “payload formats with a timestamp
> definition which results in no or little correlation between the
> media time instance and its transmission time”; otherwise this sounds
> like such payload formats are something unusual.

This discussion got moved into the new section 5.2, where the related
text says:

   RTP payload formats with a timestamp definition which results in no
   or little correlation between the media time instance and its
   transmission time cause the RTCP jitter calculation to become
   unusable due to the errors introduced on the sender side.  Such
   examples are for example payload formats for video that requires
   multiple packets, where a set of packets will have the same
   timestamp, another are payloads that includes alternating set of
   redundant information, if the definition used are for the earliest
   data included, the the timestamp may jump back and forth.  It should
   be noted if the payload format has this property or not.

Good enough?


> 
> - In Section 3.2.2, would it be helpful to explain typical RTP
> timestamp and sequence number behaviour when using discontinuous
> transmission?

   Timestamp:  The RTP timestamp indicates the time instance the media
      sample belongs to.  For discrete media like video, it normally
      indicates when the media (frame) was sampled.  For continuous
      media it normally indicates the first time instance the media
      present in the payload represents.  For audio this is the sampling
      time of the first sample.  All RTP payload formats must specify
      the meaning of the timestamp value and the clock rates allowed.
      Selecting Timestamp rate is an active design choice and is further
      discussed in Section 5.2.

      Discontinuous transmissions (DTX) that is common among speech
      codecs, typically results in gaps or jumps in the timestamp values
      due that there are no media payload to transmit and the next used
      timestamp value represent the actual sampling time of the data
      transmitted.

   Sequence number:  The sequence number is monotonically increasing and
      is set as the packet is sent.  This property is used in many
      payload formats to recover the order of everything from the whole
      stream down to fragments of application data units (ADUs) and the
      order they need to be decoded.  Discontinuous transmissions does
      not result in gaps in the sequence number, as it is monotonically
      increasing for each sent RTP packet.

> 
> - In Section 3.2.2, when talking about the RTP timestamp, it might be
> appropriate to reference to leap-seconds draft

I chose to add a paragraph on this in Section 3.2.4

   Leap Seconds are extra seconds added or seconds removed to keep our
   clocks in sync with earths rotation.  Adding or removing seconds can
   impact first of all the reference clock as discussed in "RTP and Leap
   Seconds" [I-D.ietf-avtcore-leap-second].  But also in cases where the
   RTP timestamp value are derived using the wall clock during the leap
   second event errors can occur.  Implementations needs to consider
   leap seconds and should consider the recommendations in
   [I-D.ietf-avtcore-leap-second].

> 
> - In Section 3.2.2, when talking about payload type numbers, it might
> be useful to be explicit that the binding is dynamic, and that other
> out-of-band configuration information in addition to the payload type
> to payload format binding is usually needed to understand a payload
> format. Reference Section 3.3.

   Payload Type:  The payload type is used to indicate on a per packet
      basis which format is used.  The binding between a payload type
      number and a payload format and its configuration are dynamically
      bound and RTP session specific.  The configuration information can
      be bound to a payload type value by out-of-band signalling
      (Section 3.3).  An example of this would be video decoder
      configuration information.  Commonly the same payload type is used
      for a media stream for the whole duration of a session.  However
      in some cases it may be necessary to change the payload format or
      its configuration during the session.

> 
> - Should Section 3.2.3 reference the multiplex-guidelines or
> transport-multiplexing drafts?


I think the multiplexing guidelines makes sense. The transport
multiplexing being focused on providing a specific solution to one
combination of RTP sessions to transport mapping appears to specific.

I added the following last in the section.

   For more discussion and consideration of how and when to use the
   different RTP multiplexing points see
   [I-D.ietf-avtcore-multiplex-guidelines].

> 
> - Section 3.2.4 says “SSRCs with the same CNAME in different RTP
> sessions can be synchronized”; this is also true of flows with
> different SSRC but the same CNAME within an RTP session.

Correct, I changed this to:

SSRCs with the same CNAME sent in any of multiple RTP sessions can be
synchronized.

> 
> - Section 3.4: should this say anything about AQM and/or QoS
> support?

I tried writing something about this. I don't know how useful this is.
Feedback much appreciated.


3.4.2.  Different Queuing Algorithms

   Routers and Switches on the network path between an IP sender and a
   particular receiver can exhibit different behaviors affecting the end
   to end characteristics.  One of the more important aspects of this is
   queuing behavior.  Routers and Switches has some amount of queuing to
   handle temporary bursts of data that designated to leave the switch
   or router on the same egress link.  A queue when not empty results in
   an increased path delay.

   The implementation of the queuing effects the delay and also how
   congestion signals (Explicint Congestion Marking (ECN) or packet
   drops) are provided to the flow.  The other aspects are if the flow
   shares the queue with other flows and how the implementation affects
   the flow interaction.  This becomes important for example when real-
   time flows interacts with long lived TCP flows.  TCP have a built in
   behavior in its congestion control that strive to fill the buffer,
   thus all flows sharing the buffer experienced the delay build up.

   A common but quite poor queue handling mechanism is tail-drop, i.e.
   only drop packets when the incoming packet doesn't fit in the queue.
   Active Queue Management (AQM) is a term covering mechanism that tries
   to do something smarter buy actively managing the queue, for example
   by sending congestion signals earlier by droping packets earlier in
   the queue.  The behavior also affects the flow interactions.  For
   example Random Early Drop (RED) selects which packet to drop
   randomly.  This to give flows that have more packets in the queue a
   higher probability to experience the packet loss (congestion signal).
   There is ongoing work to find suitable mechanisms to recommend for
   implementation and reduce the use of tail-drop.

3.4.3.  Quality of Service

   Using best effort Internet one have no guaratees for the paths
   properties.  Quality of Service (QoS) mechanism are intended to
   provide the possility to bound the path properties.  Where Diffserv
   [RFC2475] markings effects the queing and forwarding behaviors of
   routers, the mechanism provides only statistical guarantees and care
   in how much marked packets of different types that are entering the
   network.  Flow based QoS like IntServ [RFC1663] has the potential for
   stricter guarantees as the properties are agreed on by each hop on
   the path.


> 
> - Section 4 might be clearer titled “Standardisation Process for an
> RTP Payload Format”

Agreed


> 
> - Section 4 “registration of the RTP payload format name” – how does
> this relate to the “media type” discussed in Section 3.3?

It really should be referencing the Media Type as this is the registry
which matters. Thus changed sentence to:

For specifications that are defined by standards bodies other than the
IETF the primary milestone is registration of the Media Type for the RTP
payload format.

> 
> - The discussion of SST and MST in Section 5.1.5 seems to confuse
> sessions with SSRCs. Isn’t it more important whether the layers are
> sent within a single RTP media stream, or as separate RTP media
> streams with different SSRC values? Whether the different SSRCs are
> sent in one RTP session of many is then a separate issue, depending
> on how easily one wants to be able to separate the flows for QoS
> purposes.

The issue is that the SVC payload RFC is a bit confused on this matter.
I agree that there are two separate important distinctions here. First
if the layers are sent in one RTP packet stream or used multiple ones.
In the second case one gets the question of how one distributes that
over RTP sessions and underlying transports. Especially as some that
implements SVC MST clearly uses a single RTP session and are after the
possibility on a SSRC level to control transmission of layers in RTP
middleboxes.

I have tried to address this in the text, please review.

5.1.5.  Media Scalability

   Some codecs support various types of media scalability, i.e. some
   data of a media stream may be removed to adapt the media's
   properties, such as bitrate and quality.  The adaptation may be
   applied in the following dimensions of the media:

   Temporal:  For video codecs it is possible to adapt the frame rate,
      e.g. for H.264 [RFC6184].

   Spatial:  Video codecs supporting scalability may adapt the
      resolution, e.g. in SVC [RFC6190].

   Quality:  The quality of the media stream may be scaled by adapting
      the accuracy of the coding process, as, e.g.  possible with Signal
      to Noise Ratio (SNR) fidelity scalability of SVC [RFC6190].

   At the time of writing this document, codecs that support scalability
   have a bit of revival.  It has been realized that getting the
   required functionality for supporting the features of the media
   stream into the RTP framework is quite challenging.  One of the
   recent examples for layered and scalable codecs is Scalable Video
   Coding [RFC6190] (SVC).

   SVC is a good example for a payload format supporting media
   scalability features, which have been in its basic form already
   included in RTP.  A layered codec supports the dropping of data parts
   of a media stream, i.e. RTP packets may be not transmitted or
   forwarded to a client in order to adapt the media stream rate as well
   as the media stream quality, while still providing a decodable subset
   of the media stream to a client.  One example for using the
   scalability feature may be an RTP Mixer (Multipoint Control Unit)
   which controls the rate and quality sent out to participants in a
   communication based on dropping RTP packets or removing part of the
   payload.  Another example may be an transport channel which allows
   for differentiation in Quality of Service (QoS) parameters based on
   RTP sessions in a multicast session.  In such a case, the more
   important packets of the scalable media stream (base layer) may get
   better QoS parameters, then the less important packets (enhancement
   layer) in order to provide some kind of graceful degradation.  The
   scalability features required for allowing an adaptive transport as
   described in the two examples above are based on RTP multiplexing in
   order to identify the packets to be dropped or transmitted/forwarded.
   The multiplexing features defined for Scalable Video Coding [RFC6190]
   are:

      single session transmission (SST), where all media layers of the
      media are transported as single source (SSRC) in a single RTP
      session; as well as

      multi session transmission (MST), which should more accurately be
      called multi stream transmission, where different media layers or
      a set of media layers are transported in different RTP streams,
      i.e. using multiple sources (SSRCs).

   In the first case (SST), additional in-band as well as out-of-band
   signaling is required in order to allow identification of packets
   belonging to a specific media layer.  Furthermore, an adaptation of
   the media stream requires dropping of specific packets in order to
   provide the client with a compliant media stream.  In case of using
   encryption, it is typically required for an adapting network device
   to be in the security context to allow packet dropping and providing
   an intact RTP session to the client.  This typically requires the
   network device to be an RTP mixer.

   In general having a media unaware network device dropping excessive
   packets will be more problematic than having a Media Aware Network
   Entity (MANE).  First is the need to understand the media format and
   know which ADUs or payloads that belongs to the layers, that no other
   layer will be dependent on after the dropping.  Secondly, if the MANE
   can work as RTP mixer or translator it can rewrite the RTP and RTCP
   in such a way that the receiver will not suspect non-intentional RTP
   packet losses needing repair actions.  This as the receiver can't
   determine if a lost packet was an important base layer packet or one
   of the less important extension layers.

   In the second case (MST), the RTP packet streams can be sent using a
   single or multiple RTP sessions, and thus transport flows, e.g. on
   differnt multicast groups.  Transmitting the streams in different RTP
   sessions then the out-of-band signaling typically provides enough
   information to identify the media layers and its properties.  The
   decision for dropping packets is based on the Network Address which
   identifies the RTP session to be dropped.  In order to allow correct
   data provision to a decoder after reception from different sessions,
   data re-alignment mechanisms are described for Scalable Video Coding
   [RFC6190].  A more generic one is also described in Rapid Sync for
   RTP flows [RFC6051], which is purely based on existing RTP

   mechanisms, i.e. the NTP timestamp, for inter-session
   synchronization.  Another signaling feature is the generic indication
   of dependencies of RTP sessions in SDP, as defined in the Media
   Decoding Dependency Grouping in SDP [RFC5583].

   Using MST within a single RTP session is also possible and allow
   stream level handling instead of looking deeper into the packets by a
   MANE.  However, transport flow level properties will be the same
   unless packet based mechanisms like DiffServ is used.

   When QoS settings, e.g. DiffServ markings, are used to ensure that
   the extension layers are dropped prior the base layer the receiving
   end-point has the benefit in MST to know which layer or set of layers
   the missing packets belong as it will be bound to different RTP
   sessions.  Thus explicitly indicating the importance of the loss.




> 
> - In Section 5.1.6, avoiding wrap-around within the RTCP reporting
> timeout interval is also important.

If I understand this correctly, this is a minor issue with the RTCP SR
and RR packet types as they all uses 32-bit counters. However, other
packet types may have issues if they not use extended sequence number
counters.

This might be more relevant to considerations for configuring RTCP
bandwidth and design of RTCP extensions, rather than being an RTP
Payload issue. There exists a configuration goal to ensure that RTCP
reporting doesn't wrap its fields within multiple reporting intervals.

I propose that the following paragraph is added in this section:

RTCP is also affected by high packet rates. For RTCP mechanism that do
not use extended counters there is significant risk that they wrap
multiple times between RTCP reporting or feedback, thus producing
uncertainty which packet(s) that are referenced. The payload designer
can't effect the RTCP packet formats used and their design, but can note
this considerations when configuring RTCP bandwidth and reporting
intervals to avoid to wrapping issues.


> 
> - The introduction to Section 7 could usefully reference the template
> payload format.

Yes, is this ok?

   A number of sections in the payload format draft need special
   consideration.  These include the Security and IANA Considerations
   sections that are required in all drafts.  Payload formats are also
   strongly recommended to have the media format description and
   congestion control considerations.  The included RTP Payload format
   template (Appendix A) contains draft text for some of these sections.

> 
> - Appendix A.10: mention that the RTP circuit breaker ought to be
> used and respected?
> 

Okay, added text, I have avoided RFC 2119 language. I think circuit
breakers should make clear as an update to RFC 3550 when they are
required. The proposed text is:

   Congestion control for RTP SHALL be used in accordance with RFC 3550
   [RFC3550], and with any applicable RTP profile; e.g., RFC 3551
   [RFC3551].  An additional requirement if best-effort service is being
   used is: users of this payload format MUST monitor packet loss to
   ensure that the packet loss rate is within acceptable parameters.
   Circuit Breakers [I-D.ietf-avtcore-rtp-circuit-breakers] is an update
   to RTP [RFC3550] that defines criteria for when one is required to
   stop sending RTP Packet Streams.  The circuit breakers is to be
   implemented and followed.



Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Audio/Video Transport Payloads Working Gr=
oup of the IETF.

	Title           : How to Write an RTP Payload Format
	Author(s)       : Magnus Westerlund
	Filename        : draft-ietf-payload-rtp-howto-06.txt
	Pages           : 56
	Date            : 2013-09-21

Abstract:
   This document contains information on how to best write an RTP
   payload format.  It provides reading tips, design practices, and
   practical tips on how to produce an RTP payload format specification
   quickly and with good results.  A template is also included with
   instructions that can be used when writing an RTP payload format.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-howto

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-payload-rtp-howto-06

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-payload-rtp-howto-06


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


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Hi Magnus,

On 21 Sep 2013, at 12:56, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:
> Thanks for the review comments. I have tried addressing them. I am
> submitting an updated draft including these edits. But, I do need
> feedback on these edits.
>=20
> See below.
>=20
> On 2013-09-12 19:04, Colin Perkins wrote:
>> On 22 Aug 2013, at 09:58, Ali C. Begen (abegen) <abegen@cisco.com>
>> wrote:
>>> (As a WG co-chair)
>>>=20
>>> I am starting WGLC for the following draft (version 05).
>>> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-howto/
>>>=20
>>> This is an important document as it outlines how someone is
>>> supposed to write the payload specs.
>>>=20
>>> If there are sections you don't understand or that are not clear
>>> enough, etc. please post them to the list. We need to get this
>>> right to minimize further headaches for future documents in our
>>> WG.
>>>=20
>>> The deadline for the WGLC is September 20th.
>>=20
>>=20
>> I think this is an important draft, and I'm glad to see it
>> progressing. It seems to be generally well written and containing
>> good advice. I have some number of comments, but these are generally
>> all minor points:
>>=20
>> - In Section 3.1.2 the discussion of RFC 6015 mentions RFC 2733, but
>> RFC 2733 hasn=92t been cited or discussed
>=20
> Correct, I added a reference in RFC 5109 text to state that this
> replaced RFC 2733, and added as references in both places.
>=20
>   RFC 5109:  The "RTP Payload Format for Generic Forward Error
>      Correction (FEC)" [RFC5109] provides an XOR based FEC of the =
whole
>      or parts of a number of RTP packets.  This specification replaced
>      the previous specification for XOR based FEC [RFC2733].  These =
FEC
>      packets are sent in a separate stream or as a redundant encoding
>      using RFC 2198.  This FEC scheme has certain restrictions in the
>      number of packets it can protect.  It is suitable for low-to-
>      medium delay tolerant applications with limited amount of RTP
>      packets.
>=20
>   RFC 6015:  The "RTP Payload Format for 1-D Interleaved Parity =
Forward
>      Error Correction (FEC)" [RFC6015] provides a variant of the XOR
>      based Generic protection defined in [RFC2733].  The main
>      difference is to use interleaving scheme on which packets gets
>      included as source packets for a particular protection packet.
>      The interleaving is defined by using every L packets as source
>      data.  And then produce protection data over D number of packets.
>      Thus each block of D x L source packets will result in L number =
of
>      Repair packets, each capable of repairing one loss.  The goal is
>      to provide better burst error robustness when the packet rate is
>      higher.

This looks good.

>> - In Section 3.1.2 the discussion of RFC 3611 might usefully point to
>> the IANA registry for XR blocks and the XRBLOCK working group
>=20
>   RFC 3611:  The RTCP Extended Reports (RTCP XR) [RFC3611] consists of
>      a framework for reports sent within RTCP.  It can easily be
>      extended by defining new report formats, which has occurred.  The
>      list of specified formats are available in IANA's RTCP XR Block
>      Type registry =
(http://www.iana.org/assignments/rtcp-xr-block-types
>      /rtcp-xr-block-types.xhtml).  The report formats that are defined
>      in RFC3611 provide report information on packet loss, packet
>      duplication, packet reception times, RTCP statistics summary and
>      VoIP Quality.  [RFC3611] also defines a mechanism that allows
>      receivers to calculate the RTT to other session participants when
>      used.

Good.

>> - Section 3.2: the introduction to this section could be clearer that
>> this is an overview of RTP features that are available independent of
>> the payload format, and not a list of things that the payload format
>> needs to explicitly opt in to/out of.
>=20
> This section reviews a number of RTP features and concepts that are
> available in RTP independent of the payload format. The RTP payload
> format can make use of these when appropriate, and in some cases even
> effect the behavior (RTP Timestamp and marker bit). This section does
> not remove the necessity to read up on RTP. However it does point out =
a
> few important details to remember when designing a payload format.

It might be worth expanding this to say "...even affect the behaviour =
(RTP timestamp and marker bit), but it is important to note that not all =
features and concepts are relevant to every payload format"?

>> - Section 3.2.2 expands SSRC as =93Sender Source Identifier=94 but =
this
>> should be =93Synchronisation Source Identifier=94.
>=20
> Note that I also change the fourth line to say "of multiple media
> sources" rather than "from multiple senders.
>=20
>   SSRC:  The Synchronisation Source Identifier (SSRC) is normally not
>      used by a payload format other than to identify the RTP timestamp
>      and sequence number space a packet belongs to, allowing
>      simultaneously reception of multiple media sources.  However, =
some
>      of the RTP mechanisms for improving resilience to packet loss =
uses
>      multiple SSRCs to separate original data and repair or redundant
>      data.

Ok.

>> - In Section 3.2.2 the statement =93[RFC5285] specifies how to extend
>> the RTP header to carry metadata relating to the payload=94 is
>> potentially misleading, since the type of information that usually
>> goes into an RTP payload header is also =93metadata relating to the
>> payload=94. It might be useful to expand this discussion slightly, to
>> make the distinction clear.
>=20
> Is this better?
>=20
>   Header extensions:  Payload formats normally specifies the carrying
>      of meta data related to the media payload in the payload header.
>      In some cases the meta data may be more generic, e.g. relevant
>      more to the media type (e.g. audio or video), or RTP packets in
>      general.  In those cases the carrying of this additional meta =
data
>      can be specified in a header extension.  One example is the
>      transport of SMPTE time-codes in the RTP header [RFC5484].  As
>      [RFC5285] specifies, header extensions must not contain
>      information required in order to decode the payload successfully.

That's better, but I might suggest even stronger wording: "RTP payload =
formats often need to include metadata relating to the payload data =
being transported. Such metadata is sent as a payload header, at the =
start of the payload section of the RTP packet. The RTP packet also =
includes space for a header extension [RFC5275]; this can be used to =
transport payload format independent metadata, for example a SMPTE time =
code for the packet [RFC5484]. The RTP header extensions is not intended =
to carry headers that relate to a particular payload format., and must =
not contain information needed in order to decode the payload."

The last sentence of the first paragraph of Section 3.2.2 also needs =
updating. I suggest "Finally, [RFC5285] specifies how to transport =
payload format independent metadata relating to the RTP packet".

>> - In Section 3.2.2, it might be useful to explain that video payload
>> formats are a common example of =93payload formats with a timestamp
>> definition which results in no or little correlation between the
>> media time instance and its transmission time=94; otherwise this =
sounds
>> like such payload formats are something unusual.
>=20
> This discussion got moved into the new section 5.2, where the related
> text says:
>=20
>   RTP payload formats with a timestamp definition which results in no
>   or little correlation between the media time instance and its
>   transmission time cause the RTCP jitter calculation to become
>   unusable due to the errors introduced on the sender side.  Such
>   examples are for example payload formats for video that requires
>   multiple packets, where a set of packets will have the same
>   timestamp, another are payloads that includes alternating set of
>   redundant information, if the definition used are for the earliest
>   data included, the the timestamp may jump back and forth.  It should
>   be noted if the payload format has this property or not.
>=20
> Good enough?

Better, but could still be more explicit. How about changing the "Such =
examples are..." sentence to "A common example is a payload format for a =
video codec where the RTP timestamp represents the capture time of the =
video frame, but frames are large enough that multiple RTP packets need =
to be sent for each frame spread across the framing interval".

>> - In Section 3.2.2, would it be helpful to explain typical RTP
>> timestamp and sequence number behaviour when using discontinuous
>> transmission?
>=20
>   Timestamp:  The RTP timestamp indicates the time instance the media
>      sample belongs to.  For discrete media like video, it normally
>      indicates when the media (frame) was sampled.  For continuous
>      media it normally indicates the first time instance the media
>      present in the payload represents.  For audio this is the =
sampling
>      time of the first sample.  All RTP payload formats must specify
>      the meaning of the timestamp value and the clock rates allowed.
>      Selecting Timestamp rate is an active design choice and is =
further
>      discussed in Section 5.2.
>=20
>      Discontinuous transmissions (DTX) that is common among speech
>      codecs, typically results in gaps or jumps in the timestamp =
values
>      due that there are no media payload to transmit and the next used
>      timestamp value represent the actual sampling time of the data
>      transmitted.
>=20
>   Sequence number:  The sequence number is monotonically increasing =
and
>      is set as the packet is sent.  This property is used in many
>      payload formats to recover the order of everything from the whole
>      stream down to fragments of application data units (ADUs) and the
>      order they need to be decoded.  Discontinuous transmissions does
>      not result in gaps in the sequence number, as it is monotonically
>      increasing for each sent RTP packet.

Good.

>> - In Section 3.2.2, when talking about the RTP timestamp, it might be
>> appropriate to reference to leap-seconds draft
>=20
> I chose to add a paragraph on this in Section 3.2.4
>=20
>   Leap Seconds are extra seconds added or seconds removed to keep our
>   clocks in sync with earths rotation.  Adding or removing seconds can
>   impact first of all the reference clock as discussed in "RTP and =
Leap
>   Seconds" [I-D.ietf-avtcore-leap-second].  But also in cases where =
the
>   RTP timestamp value are derived using the wall clock during the leap
>   second event errors can occur.  Implementations needs to consider
>   leap seconds and should consider the recommendations in
>   [I-D.ietf-avtcore-leap-second].
>=20

Okay.

>> - In Section 3.2.2, when talking about payload type numbers, it might
>> be useful to be explicit that the binding is dynamic, and that other
>> out-of-band configuration information in addition to the payload type
>> to payload format binding is usually needed to understand a payload
>> format. Reference Section 3.3.
>=20
>   Payload Type:  The payload type is used to indicate on a per packet
>      basis which format is used.  The binding between a payload type
>      number and a payload format and its configuration are dynamically
>      bound and RTP session specific.  The configuration information =
can
>      be bound to a payload type value by out-of-band signalling
>      (Section 3.3).  An example of this would be video decoder
>      configuration information.  Commonly the same payload type is =
used
>      for a media stream for the whole duration of a session.  However
>      in some cases it may be necessary to change the payload format or
>      its configuration during the session.

Good.

>> - Should Section 3.2.3 reference the multiplex-guidelines or
>> transport-multiplexing drafts?
>=20
>=20
> I think the multiplexing guidelines makes sense. The transport
> multiplexing being focused on providing a specific solution to one
> combination of RTP sessions to transport mapping appears to specific.
>=20
> I added the following last in the section.
>=20
>   For more discussion and consideration of how and when to use the
>   different RTP multiplexing points see
>   [I-D.ietf-avtcore-multiplex-guidelines].

Okay.

>> - Section 3.2.4 says =93SSRCs with the same CNAME in different RTP
>> sessions can be synchronized=94; this is also true of flows with
>> different SSRC but the same CNAME within an RTP session.
>=20
> Correct, I changed this to:
>=20
> SSRCs with the same CNAME sent in any of multiple RTP sessions can be
> synchronized.

Okay.

>> - Section 3.4: should this say anything about AQM and/or QoS
>> support?
>=20
> I tried writing something about this. I don't know how useful this is.
> Feedback much appreciated.
>=20
>=20
> 3.4.2.  Different Queuing Algorithms
>=20
>   Routers and Switches on the network path between an IP sender and a
>   particular receiver can exhibit different behaviors affecting the =
end
>   to end characteristics.  One of the more important aspects of this =
is
>   queuing behavior.  Routers and Switches has some amount of queuing =
to
>   handle temporary bursts of data that designated to leave the switch
>   or router on the same egress link.  A queue when not empty results =
in
>   an increased path delay.
>=20
>   The implementation of the queuing effects the delay and also how
>   congestion signals (Explicint Congestion Marking (ECN) or packet
>   drops) are provided to the flow.

Reference RFC 6679 here?

>  The other aspects are if the flow
>   shares the queue with other flows and how the implementation affects
>   the flow interaction.  This becomes important for example when real-
>   time flows interacts with long lived TCP flows.  TCP have a built in
>   behavior in its congestion control that strive to fill the buffer,
>   thus all flows sharing the buffer experienced the delay build up.
>=20
>   A common but quite poor queue handling mechanism is tail-drop, i.e.
>   only drop packets when the incoming packet doesn't fit in the queue.
>   Active Queue Management (AQM) is a term covering mechanism that =
tries
>   to do something smarter buy actively managing the queue, for example
>   by sending congestion signals earlier by droping packets earlier in
>   the queue.  The behavior also affects the flow interactions.  For
>   example Random Early Drop (RED) selects which packet to drop
>   randomly.  This to give flows that have more packets in the queue a
>   higher probability to experience the packet loss (congestion =
signal).
>   There is ongoing work to find suitable mechanisms to recommend for
>   implementation and reduce the use of tail-drop.
>=20
> 3.4.3.  Quality of Service
>=20
>   Using best effort Internet one have no guaratees for the paths
>   properties.  Quality of Service (QoS) mechanism are intended to
>   provide the possility to bound the path properties.  Where Diffserv
>   [RFC2475] markings effects the queing and forwarding behaviors of
>   routers, the mechanism provides only statistical guarantees and care
>   in how much marked packets of different types that are entering the
>   network.  Flow based QoS like IntServ [RFC1663] has the potential =
for
>   stricter guarantees as the properties are agreed on by each hop on
>   the path.

I think this is useful. It highlights that there are issues to consider, =
which is the goal.

>> - Section 4 might be clearer titled =93Standardisation Process for an
>> RTP Payload Format=94
>=20
> Agreed
>=20
>=20
>>=20
>> - Section 4 =93registration of the RTP payload format name=94 =96 how =
does
>> this relate to the =93media type=94 discussed in Section 3.3?
>=20
> It really should be referencing the Media Type as this is the registry
> which matters. Thus changed sentence to:
>=20
> For specifications that are defined by standards bodies other than the
> IETF the primary milestone is registration of the Media Type for the =
RTP
> payload format.

Ok.

>> - The discussion of SST and MST in Section 5.1.5 seems to confuse
>> sessions with SSRCs. Isn=92t it more important whether the layers are
>> sent within a single RTP media stream, or as separate RTP media
>> streams with different SSRC values? Whether the different SSRCs are
>> sent in one RTP session of many is then a separate issue, depending
>> on how easily one wants to be able to separate the flows for QoS
>> purposes.
>=20
> The issue is that the SVC payload RFC is a bit confused on this =
matter.
> I agree that there are two separate important distinctions here. First
> if the layers are sent in one RTP packet stream or used multiple ones.
> In the second case one gets the question of how one distributes that
> over RTP sessions and underlying transports. Especially as some that
> implements SVC MST clearly uses a single RTP session and are after the
> possibility on a SSRC level to control transmission of layers in RTP
> middleboxes.
>=20
> I have tried to address this in the text, please review.
>=20
> 5.1.5.  Media Scalability
>=20
>   Some codecs support various types of media scalability, i.e. some
>   data of a media stream may be removed to adapt the media's
>   properties, such as bitrate and quality.  The adaptation may be
>   applied in the following dimensions of the media:
>=20
>   Temporal:  For video codecs it is possible to adapt the frame rate,
>      e.g. for H.264 [RFC6184].
>=20
>   Spatial:  Video codecs supporting scalability may adapt the
>      resolution, e.g. in SVC [RFC6190].
>=20
>   Quality:  The quality of the media stream may be scaled by adapting
>      the accuracy of the coding process, as, e.g.  possible with =
Signal
>      to Noise Ratio (SNR) fidelity scalability of SVC [RFC6190].
>=20
>   At the time of writing this document, codecs that support =
scalability
>   have a bit of revival.  It has been realized that getting the
>   required functionality for supporting the features of the media
>   stream into the RTP framework is quite challenging.  One of the
>   recent examples for layered and scalable codecs is Scalable Video
>   Coding [RFC6190] (SVC).
>=20
>   SVC is a good example for a payload format supporting media
>   scalability features, which have been in its basic form already
>   included in RTP.  A layered codec supports the dropping of data =
parts
>   of a media stream, i.e. RTP packets may be not transmitted or
>   forwarded to a client in order to adapt the media stream rate as =
well
>   as the media stream quality, while still providing a decodable =
subset
>   of the media stream to a client.  One example for using the
>   scalability feature may be an RTP Mixer (Multipoint Control Unit)
>   which controls the rate and quality sent out to participants in a
>   communication based on dropping RTP packets or removing part of the
>   payload.  Another example may be an transport channel which allows
>   for differentiation in Quality of Service (QoS) parameters based on
>   RTP sessions in a multicast session.  In such a case, the more
>   important packets of the scalable media stream (base layer) may get
>   better QoS parameters, then the less important packets (enhancement
>   layer) in order to provide some kind of graceful degradation.  The
>   scalability features required for allowing an adaptive transport as
>   described in the two examples above are based on RTP multiplexing in
>   order to identify the packets to be dropped or =
transmitted/forwarded.
>   The multiplexing features defined for Scalable Video Coding =
[RFC6190]
>   are:
>=20
>      single session transmission (SST), where all media layers of the
>      media are transported as single source (SSRC) in a single RTP
>      session; as well as
>=20
>      multi session transmission (MST), which should more accurately be
>      called multi stream transmission, where different media layers or
>      a set of media layers are transported in different RTP streams,
>      i.e. using multiple sources (SSRCs).
>=20
>   In the first case (SST), additional in-band as well as out-of-band
>   signaling is required in order to allow identification of packets
>   belonging to a specific media layer.  Furthermore, an adaptation of
>   the media stream requires dropping of specific packets in order to
>   provide the client with a compliant media stream.  In case of using
>   encryption, it is typically required for an adapting network device
>   to be in the security context to allow packet dropping and providing
>   an intact RTP session to the client.  This typically requires the
>   network device to be an RTP mixer.
>=20
>   In general having a media unaware network device dropping excessive
>   packets will be more problematic than having a Media Aware Network
>   Entity (MANE).  First is the need to understand the media format and
>   know which ADUs or payloads that belongs to the layers, that no =
other
>   layer will be dependent on after the dropping.  Secondly, if the =
MANE
>   can work as RTP mixer or translator it can rewrite the RTP and RTCP
>   in such a way that the receiver will not suspect non-intentional RTP
>   packet losses needing repair actions.  This as the receiver can't
>   determine if a lost packet was an important base layer packet or one
>   of the less important extension layers.
>=20
>   In the second case (MST), the RTP packet streams can be sent using a
>   single or multiple RTP sessions, and thus transport flows, e.g. on
>   differnt multicast groups.  Transmitting the streams in different =
RTP
>   sessions then the out-of-band signaling typically provides enough
>   information to identify the media layers and its properties.  The
>   decision for dropping packets is based on the Network Address which
>   identifies the RTP session to be dropped.  In order to allow correct
>   data provision to a decoder after reception from different sessions,
>   data re-alignment mechanisms are described for Scalable Video Coding
>   [RFC6190].  A more generic one is also described in Rapid Sync for
>   RTP flows [RFC6051], which is purely based on existing RTP
>=20
>   mechanisms, i.e. the NTP timestamp, for inter-session
>   synchronization.  Another signaling feature is the generic =
indication
>   of dependencies of RTP sessions in SDP, as defined in the Media
>   Decoding Dependency Grouping in SDP [RFC5583].
>=20
>   Using MST within a single RTP session is also possible and allow
>   stream level handling instead of looking deeper into the packets by =
a
>   MANE.  However, transport flow level properties will be the same
>   unless packet based mechanisms like DiffServ is used.
>=20
>   When QoS settings, e.g. DiffServ markings, are used to ensure that
>   the extension layers are dropped prior the base layer the receiving
>   end-point has the benefit in MST to know which layer or set of =
layers
>   the missing packets belong as it will be bound to different RTP
>   sessions.  Thus explicitly indicating the importance of the loss.

I think this is okay, but it really needs an implementer of the SVC =
format to comment too.

>> - In Section 5.1.6, avoiding wrap-around within the RTCP reporting
>> timeout interval is also important.
>=20
> If I understand this correctly, this is a minor issue with the RTCP SR
> and RR packet types as they all uses 32-bit counters. However, other
> packet types may have issues if they not use extended sequence number
> counters.
>=20
> This might be more relevant to considerations for configuring RTCP
> bandwidth and design of RTCP extensions, rather than being an RTP
> Payload issue. There exists a configuration goal to ensure that RTCP
> reporting doesn't wrap its fields within multiple reporting intervals.
>=20
> I propose that the following paragraph is added in this section:
>=20
> RTCP is also affected by high packet rates. For RTCP mechanism that do
> not use extended counters there is significant risk that they wrap
> multiple times between RTCP reporting or feedback, thus producing
> uncertainty which packet(s) that are referenced. The payload designer
> can't effect the RTCP packet formats used and their design, but can =
note
> this considerations when configuring RTCP bandwidth and reporting
> intervals to avoid to wrapping issues.

That's fine, although I was actually suggesting something simpler, which =
is to change "As rule of thumb, it must not be possible to wrap the =
sequence number space in less than 2 minutes (TCP maximum segment =
lifetime)" to "...must not be possible to wrap the sequence number space =
in less than an RTCP reporting interval".

>> - The introduction to Section 7 could usefully reference the template
>> payload format.
>=20
> Yes, is this ok?
>=20
>   A number of sections in the payload format draft need special
>   consideration.  These include the Security and IANA Considerations
>   sections that are required in all drafts.  Payload formats are also
>   strongly recommended to have the media format description and
>   congestion control considerations.  The included RTP Payload format
>   template (Appendix A) contains draft text for some of these =
sections.

Yes.

>> - Appendix A.10: mention that the RTP circuit breaker ought to be
>> used and respected?
>>=20
>=20
> Okay, added text, I have avoided RFC 2119 language. I think circuit
> breakers should make clear as an update to RFC 3550 when they are
> required. The proposed text is:
>=20
>   Congestion control for RTP SHALL be used in accordance with RFC 3550
>   [RFC3550], and with any applicable RTP profile; e.g., RFC 3551
>   [RFC3551].  An additional requirement if best-effort service is =
being
>   used is: users of this payload format MUST monitor packet loss to
>   ensure that the packet loss rate is within acceptable parameters.
>   Circuit Breakers [I-D.ietf-avtcore-rtp-circuit-breakers] is an =
update
>   to RTP [RFC3550] that defines criteria for when one is required to
>   stop sending RTP Packet Streams.  The circuit breakers is to be
>   implemented and followed.


Looks good.

Cheers,
Colin

--=20
Colin Perkins
http://csperkins.org/




From Lindsay@worldcastsystems.com  Tue Sep 24 10:03:19 2013
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From: "John Lindsay" <Lindsay@worldcastsystems.com>
To: <payload@ietf.org>
Cc: Jonathan Lennox <jonathan@vidyo.com>, Foerster@worldcastsystems.com
Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-aptx-01
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Hi Jonathon

Thanks for reviewing this and the points raised.

I agree on re-reading that the delivery method test is a bit abstract
and maybe doesn't add anything, by delivery method we mean the algorithm
selected e.g. Standard or Enhanced APTX.
As I think this is clear to the reader I think these lines could be
removed.

Hence I propose changing some sections to remove these references.

6.2
From
      The required parameters "variant" and "bitresolution" MUST be
      included in the SDP "a=3Dfmtp" attribute and MUST follow the
      delivery-method that applies.=20
To
      The required parameters "variant" and "bitresolution" MUST be
      included in the SDP "a=3Dfmtp" attribute.

From
      The optional parameters "stereo-channel-pairs", "embedded-
      autosync-channels", "embedded-aux-channels", and "maxptime" when
      present, MUST be included in the SDP "a=3Dfmtp" attribute and MUST
      follow the delivery-method that applies.=20
To
      The optional parameters "stereo-channel-pairs", "embedded-
      autosync-channels", "embedded-aux-channels",  MUST be=20
      included in the SDP "a=3Dfmtp" attribute.


SDP-Mapping

Agreed this should have its own SDP a=3Dattribute, suggest changing
section 6.2 from

     The parameter "maxptime" when present, MUST be included in the=20
      SDP "a=3Dmaxptime" attribute.

Marker bit.

Again this can be changed as per RFC3551, as we have no use for this in
our profile (RFC3550) it can follow the recommendations of RFC 3551
where it will be zero for most of the intended applications that we can
see.=20

Suggest changing section 5.1 from

  o  M - This payload format defines no use for this bit. Senders
      SHOULD set this bit to zero in each outgoing packet.

  o  M - As per [RFC3551]


Payload types
As I understand it the avtcore group has been discussing a new RFC that
will update RFC3551 due to conflicts but it is currently unpublished
work. What recommendations does the group have for the doc on this?
Currently I feel we reflect what is stated in RFC3551?

Aux data
The Aux data channel can be used for whatever the codec user requires
and is not defined by this draft, it provides a point to point channel
and is embedded within the Audio payload at 1/4 of the Audio sampling
rate?
To get access to this you would have to decode the audio payload, for
which you would need either a standard of Enhanced apt-X codec. Hence we
feel any security issues are minimal? What is the feeling of the
workgroup?

Section 3 of the doc provides an overview of the codecs operation. The
aim of the draft is how to packetize Standard or Enhanced apt-X over RTP
not to explain the operation of the codec. Further details are likely to
be proprietary to CSR and interested parties would need to talk to CSR
directly about obtaining licenses and setting up NDA's for further
information.=20

Please feel free to comment on the above.

Thanks

John


-----Original Message-----
From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On
Behalf Of Jonathan Lennox
Sent: 16 September 2013 19:36
To: Ali C. Begen (abegen)
Cc: payload@ietf.org
Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-aptx-01

The document discusses that a number of items are dependent on the
"delivery method", and says that the delivery method can be negotiated
through offer/answer.  However, the delivery method parameter is never
actually defined anywhere.  Either delivery method needs to be defined,
or discussion of it needs to be removed, depending on whether or not
delivery-method was actually meant to be included in the final spec.

The SDP mapping says that maxptime is encoded in the fmtp parameter.
This is incorrect; as specified by RFC 4566, maxptime is carried in its
own SDP a=3D attribute.

The RTP header section says that the marker bit is not used.  Unless
there's a good reason for this, I suggest that instead the usual RFC
3551 marker bit semantic for audio be applied (i.e., the marker bit is
set for end of silence / beginning of talkspurt).  If there's a reason
this semantic is inappropriate, it should be explained.

The RTP header section also says that the RTP dynamic payload types can
only be in the range 96 - 127.  This is incorrect; that's the primary
range, but other values can be used once that range is exhausted.  See,
e.g., the recent discussion on the avtcore list.

No information is given about the semantic content of the data carried
in the auxiliary data channel.  Is the semantic of this data defined by
the codec, or would it need to be described or negotiated in SDP?
(Doing neither, i.e. providing an arbitrary data channel with no
mechanism for describing or negotiating its content, seems likely to
lead to interoperability problems.)  Also, are there any security
considerations introduced by the auxiliary data channel?

It would be good to have a reference to some document by CSR plc
describing the apt-X codecs, even if they don't include full proprietary
details of the workings of the codec.

On Sep 16, 2013, at 9:08 AM, "Ali C. Begen (abegen)" <abegen@cisco.com>
wrote:

> (As a WG co-chair)
>=20
> I am starting WGLC for the following draft (version 01).=20
> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-aptx/
>=20
> It is a short document. If you have comments, agreements or
disagreements, please let the list know in either case.
>=20
> The deadline for the WGLC is September 30th.
>=20
> -acbegen
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload
>=20

--
Jonathan Lennox
jonathan@vidyo.com


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

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From: "John Lindsay" <Lindsay@worldcastsystems.com>
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Hi Harald

=20

Thanks for reviewing this and the points raised.

=20

To increase clarity we can change the statement in section 3

=20

From

Standard apt-X and Enhanced apt-X are proprietary audio coding

   algorithms, licensed by CSR plc and widely deployed in a variety of

   audio processing equipment.

=20

To

Standard apt-X and Enhanced apt-X are proprietary audio coding

   algorithms, which can be licensed from CSR plc and are widely=20

  deployed in a variety of audio processing equipment.

=20

Section 3 of the doc provides an overview of the codecs operation. The
aim of the draft is how to packetize Standard or Enhanced apt-X over RTP
not to explain the operation of the codec. Further details are likely to
be proprietary to CSR and interested parties would need to talk to CSR
directly about obtaining licenses and setting up NDA's etc for further
information.=20

=20

Regards

=20

John

=20

From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On
Behalf Of Harald Alvestrand
Sent: 17 September 2013 13:53
To: payload@ietf.org
Subject: Re: [payload] I-D Action: draft-ietf-payload-rtp-aptx-01.txt

=20

Brief scan, one question:

Is it possible to offer a stable reference for what "Standard apt-x" and
"Enhanced apt-x" means?

The closest I can get is "licensed by CSR plc", which presumably means
that "it means what CSR plc thinks it means", but it would be great to
have that stated explicitly.

Something like:

The codecs "apt-x" and "Extended apt-x" are defined by CSR plc, an UK
company.

Section 3 of the draft *almost* says that, but isn't completely
unambiguous ("licensed by" can mean both "CSR is a licensor" and "CSR is
a licensee, someone else defines it").

I *think* this is a nit.


On 09/10/2013 03:50 PM, John Lindsay wrote:

	Hi

	=20

	A new draft has been uploaded to the IETF Payload workgroup at
https://datatracker.ietf.org/wg/payload/.

	=20

	The new document makes a few minor changes for IDNITS checking
and also takes account of the comments made by Peter Stevens last week.

	=20

	A summary of the changes are as follows.

	=20

	Page 1, IETF Copyright notice updated to 2013.

	Page 10, updated formatting for IDNITS check.

	Page 14, Section 6 updated to reflect RFC 6838 which supersedes
RFC 4288.

	Page 16, Section 6.2.1 type corrected An - A as per Peter
Stevens comments.

	Page 18, Section 7 now reference 6.1 rather than 7.1, again
thanks to Peters proof reading.

	Page 21, new RFC6338 referenced.

	=20

	Thanks to everyone who has read and commented on the draft.

	=20

	Regards

	=20

	John

	=20

	=20

=09
=09
=09
=09

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

=20


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<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body bgcolor=3Dwhite =
lang=3DEN-GB link=3Dblue vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoPlainText><span style=3D'color:windowtext'>Hi =
Harald<o:p></o:p></span></p><p class=3DMsoPlainText><span =
style=3D'color:windowtext'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span style=3D'color:windowtext'>Thanks for =
reviewing this and the points raised.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:windowtext'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:windowtext'>To increase clarity =
we can change the statement in section 3<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:windowtext'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:windowtext'>From<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:windowtext'>Standard apt-X and =
Enhanced apt-X are proprietary audio coding<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:windowtext'>&nbsp;&nbsp; =
algorithms, licensed by CSR plc and widely deployed in a variety =
of<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:windowtext'>&nbsp;&nbsp; audio processing =
equipment.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:windowtext'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:windowtext'>To<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:windowtext'>Standard apt-X and =
Enhanced apt-X are proprietary audio coding<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:windowtext'>&nbsp;&nbsp; =
algorithms, which can be licensed from CSR plc and are widely =
<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:windowtext'>&nbsp;&nbsp;deployed in a variety of audio =
processing equipment.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:windowtext'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span style=3D'color:windowtext'>Section 3 of the =
doc provides an overview of the codecs operation. The aim of the draft =
is how to packetize Standard or Enhanced apt-X over RTP not to explain =
the operation of the codec. Further details are likely to be proprietary =
to CSR and interested parties would need to talk to CSR directly about =
obtaining licenses and setting up NDA's etc for further information. =
<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:windowtext'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:windowtext'>Regards<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:windowtext'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:windowtext'>John<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowt=
ext;mso-fareast-language:EN-GB'>From:</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowt=
ext;mso-fareast-language:EN-GB'> payload-bounces@ietf.org =
[mailto:payload-bounces@ietf.org] <b>On Behalf Of </b>Harald =
Alvestrand<br><b>Sent:</b> 17 September 2013 13:53<br><b>To:</b> =
payload@ietf.org<br><b>Subject:</b> Re: [payload] I-D Action: =
draft-ietf-payload-rtp-aptx-01.txt<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal>Brief =
scan, one question:<br><br>Is it possible to offer a stable reference =
for what &quot;Standard apt-x&quot; and &quot;Enhanced apt-x&quot; =
means?<br><br>The closest I can get is &quot;licensed by CSR plc&quot;, =
which presumably means that &quot;it means what CSR plc thinks it =
means&quot;, but it would be great to have that stated =
explicitly.<br><br>Something like:<br><br>The codecs &quot;apt-x&quot; =
and &quot;Extended apt-x&quot; are defined by CSR plc, an UK =
company.<br><br>Section 3 of the draft *almost* says that, but isn't =
completely unambiguous (&quot;licensed by&quot; can mean both &quot;CSR =
is a licensor&quot; and &quot;CSR is a licensee, someone else defines =
it&quot;).<br><br>I *think* this is a nit.<br><br><br>On 09/10/2013 =
03:50 PM, John Lindsay wrote:<o:p></o:p></p></div><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>Hi</span><o:p></o:p></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>A new draft has been =
uploaded to the IETF Payload workgroup at <a =
href=3D"https://datatracker.ietf.org/wg/payload/">https://datatracker.iet=
f.org/wg/payload/</a>.</span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>The new document makes a =
few minor changes for IDNITS checking and also takes account of the =
comments made by Peter Stevens last week.</span><o:p></o:p></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>A summary of the changes =
are as follows.</span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>Page 1, IETF Copyright =
notice updated to 2013.</span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>Page 10, updated formatting for IDNITS =
check.</span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>Page 14, Section 6 updated to reflect RFC 6838 =
which supersedes RFC 4288.</span><o:p></o:p></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>Page 16, Section 6.2.1 =
type corrected An &#8211; A as per Peter Stevens =
comments.</span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>Page 18, Section 7 now reference 6.1 rather than =
7.1, again thanks to Peters proof reading.</span><o:p></o:p></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>Page 21, new RFC6338 =
referenced.</span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>Thanks to everyone who =
has read and commented on the draft.</span><o:p></o:p></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>Regards</span><o:p></o:p></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>John</span><o:p></o:p></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif";mso-fareast-language:EN-GB'><br><br><br><o:p></o:p></span>=
</p><pre>_______________________________________________<o:p></o:p></pre>=
<pre>payload mailing list<o:p></o:p></pre><pre><a =
href=3D"mailto:payload@ietf.org">payload@ietf.org</a><o:p></o:p></pre><pr=
e><a =
href=3D"https://www.ietf.org/mailman/listinfo/payload">https://www.ietf.o=
rg/mailman/listinfo/payload</a><o:p></o:p></pre></blockquote><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif";mso-fareast-language:EN-GB'><o:p>&nbsp;</o:p></span></p></=
div></body></html>
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From: Jonathan Lennox <jonathan@vidyo.com>
To: John Lindsay <Lindsay@worldcastsystems.com>
Thread-Topic: [payload] WGLC for draft-ietf-payload-rtp-aptx-01
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Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-aptx-01
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On Sep 24, 2013, at 1:03 PM, John Lindsay <Lindsay@worldcastsystems.com>
 wrote:

> Hi Jonathon
>=20
> Thanks for reviewing this and the points raised.
>=20
> I agree on re-reading that the delivery method test is a bit abstract
> and maybe doesn't add anything, by delivery method we mean the algorithm
> selected e.g. Standard or Enhanced APTX.
> As I think this is clear to the reader I think these lines could be
> removed.

Okay.

> Hence I propose changing some sections to remove these references.
>=20
> 6.2
> From
>      The required parameters "variant" and "bitresolution" MUST be
>      included in the SDP "a=3Dfmtp" attribute and MUST follow the
>      delivery-method that applies.=20
> To
>      The required parameters "variant" and "bitresolution" MUST be
>      included in the SDP "a=3Dfmtp" attribute.

Seems good.

> From
>      The optional parameters "stereo-channel-pairs", "embedded-
>      autosync-channels", "embedded-aux-channels", and "maxptime" when
>      present, MUST be included in the SDP "a=3Dfmtp" attribute and MUST
>      follow the delivery-method that applies.=20
> To
>      The optional parameters "stereo-channel-pairs", "embedded-
>      autosync-channels", "embedded-aux-channels",  MUST be=20
>      included in the SDP "a=3Dfmtp" attribute.

You lost the "when present" in your edit, which I assume should still be th=
ere.

> SDP-Mapping
>=20
> Agreed this should have its own SDP a=3Dattribute, suggest changing
> section 6.2 from
>=20
>     The parameter "maxptime" when present, MUST be included in the=20
>      SDP "a=3Dmaxptime" attribute.

That seems good.

> Marker bit.
>=20
> Again this can be changed as per RFC3551, as we have no use for this in
> our profile (RFC3550) it can follow the recommendations of RFC 3551
> where it will be zero for most of the intended applications that we can
> see.=20
>=20
> Suggest changing section 5.1 from
>=20
>  o  M - This payload format defines no use for this bit. Senders
>      SHOULD set this bit to zero in each outgoing packet.
>=20
>  o  M - As per [RFC3551]

A bit vague, since RFC 3551 has other usages of the M bit for video.  I'd s=
uggest "As per [RFC3551] Section 4.1".


> Payload types
> As I understand it the avtcore group has been discussing a new RFC that
> will update RFC3551 due to conflicts but it is currently unpublished
> work. What recommendations does the group have for the doc on this?
> Currently I feel we reflect what is stated in RFC3551?

I'd just say "A dynamic payload type MUST be used", and drop the reference =
to the range.

> Aux data
> The Aux data channel can be used for whatever the codec user requires
> and is not defined by this draft, it provides a point to point channel
> and is embedded within the Audio payload at 1/4 of the Audio sampling
> rate?

The problem with "whatever the codec user requires" is that it's "whatever =
the sender wants", not "whatever the receiver wants" -- so the receiver mig=
ht have no idea what the sender is putting in this channel.  Unless you're =
assuming that both sides are under the same administrative control, this se=
ems to me like it can to interoperability problems.

What is this channel commonly used for, and how is it configured?

> To get access to this you would have to decode the audio payload, for
> which you would need either a standard of Enhanced apt-X codec. Hence we
> feel any security issues are minimal? What is the feeling of the
> workgroup?

Again, it's what it's used for, and whether you can always guarantee that t=
he other end isn't sending you something hostile if the channel is used for=
 some sort of active data.  (Don't assume security-through-obscurity; i.e.,=
 the apt-X codec being proprietary doesn't mean that a bad guy can't encode=
 or decode it.)

>=20
> Section 3 of the doc provides an overview of the codecs operation. The
> aim of the draft is how to packetize Standard or Enhanced apt-X over RTP
> not to explain the operation of the codec. Further details are likely to
> be proprietary to CSR and interested parties would need to talk to CSR
> directly about obtaining licenses and setting up NDA's for further
> information.=20

That's fine -- but does CSR have any public (non-technical) data sheets tha=
t can be cited, or anything else that an interested party can use to figure=
 out how to contact CSR to decide whether they want to begin this process? =
 I see for instance <http://www.csr.com/sites/default/files/white-papers/ap=
tx_lossless_whitepaper1.pdf> -- citing that (or something better) as an inf=
ormative reference for background on apt-X would seem reasonable.

>=20
> Please feel free to comment on the above.
>=20
> Thanks
>=20
> John
>=20
>=20
> -----Original Message-----
> From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On
> Behalf Of Jonathan Lennox
> Sent: 16 September 2013 19:36
> To: Ali C. Begen (abegen)
> Cc: payload@ietf.org
> Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-aptx-01
>=20
> The document discusses that a number of items are dependent on the
> "delivery method", and says that the delivery method can be negotiated
> through offer/answer.  However, the delivery method parameter is never
> actually defined anywhere.  Either delivery method needs to be defined,
> or discussion of it needs to be removed, depending on whether or not
> delivery-method was actually meant to be included in the final spec.
>=20
> The SDP mapping says that maxptime is encoded in the fmtp parameter.
> This is incorrect; as specified by RFC 4566, maxptime is carried in its
> own SDP a=3D attribute.
>=20
> The RTP header section says that the marker bit is not used.  Unless
> there's a good reason for this, I suggest that instead the usual RFC
> 3551 marker bit semantic for audio be applied (i.e., the marker bit is
> set for end of silence / beginning of talkspurt).  If there's a reason
> this semantic is inappropriate, it should be explained.
>=20
> The RTP header section also says that the RTP dynamic payload types can
> only be in the range 96 - 127.  This is incorrect; that's the primary
> range, but other values can be used once that range is exhausted.  See,
> e.g., the recent discussion on the avtcore list.
>=20
> No information is given about the semantic content of the data carried
> in the auxiliary data channel.  Is the semantic of this data defined by
> the codec, or would it need to be described or negotiated in SDP?
> (Doing neither, i.e. providing an arbitrary data channel with no
> mechanism for describing or negotiating its content, seems likely to
> lead to interoperability problems.)  Also, are there any security
> considerations introduced by the auxiliary data channel?
>=20
> It would be good to have a reference to some document by CSR plc
> describing the apt-X codecs, even if they don't include full proprietary
> details of the workings of the codec.
>=20
> On Sep 16, 2013, at 9:08 AM, "Ali C. Begen (abegen)" <abegen@cisco.com>
> wrote:
>=20
>> (As a WG co-chair)
>>=20
>> I am starting WGLC for the following draft (version 01).=20
>> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-aptx/
>>=20
>> It is a short document. If you have comments, agreements or
> disagreements, please let the list know in either case.
>>=20
>> The deadline for the WGLC is September 30th.
>>=20
>> -acbegen
>> _______________________________________________
>> payload mailing list
>> payload@ietf.org
>> https://www.ietf.org/mailman/listinfo/payload
>>=20
>=20
> --
> Jonathan Lennox
> jonathan@vidyo.com
>=20
>=20
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload


From Lindsay@worldcastsystems.com  Wed Sep 25 05:09:18 2013
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Thread-Topic: [payload] WGLC for draft-ietf-payload-rtp-aptx-01
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From: "John Lindsay" <Lindsay@worldcastsystems.com>
To: "Jonathan Lennox" <jonathan@vidyo.com>
Cc: Hartmut Foerster <Foerster@worldcastsystems.com>, payload@ietf.org
Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-aptx-01
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Hi Jonathon

Editorial comments noted and will be included in next draft.
Specifically the missing "when present"  and "As per [RFC3551] Section
4.1" notes.
Payload types, ok, agreed, again will be added to next draft.

In terms of Aux data the majority of users in the EBU ACIP workgroup
(where the requests for an apt-X RFC originated ) are codec
manufacturers typically aiming at a studio to transmitter (STL)
application, (however apt-x over RTP obviously has other uses). Hence
the majority of the current users we know of will uses the aux data
channel to transmit RS232 data  up to 1/4 sampling rate typically
containing RDS data. It will be up to the implementer of the solution to
decide what data the channel is used for and typically the units would
be used in a complete network hence all the equipment is generally
controlled by one broadcaster.

>From a security point of view the aux channel travels in the compressed
audio stream and encapsulated in an RTP packet hence security concerns
for the aux channel are exactly the same as the audio channel and pose
no different risk? As above its generally an RS232 data channel with a
Protocol, hence if malicious data were to be sent in this channel it
would be up to end equipment to decide if it was protocol compliant
legitimate data and hence to decode or not.=20

I have taken a look at CSR's website, CSR's main focus is Bluetooth
audio and they operate in a more commercial environment than apt-X's
previous 25 year history in broadcast audio. Hence the data on apt-X and
Enhanced apt-X now more reflects this focus. However some useful links
may be.

http://www.csr.com/products/60/aptx
http://www.csr.com/products/58/aptx-enhanced

Regards

John


-----Original Message-----
From: Jonathan Lennox [mailto:jonathan@vidyo.com]=20
Sent: 24 September 2013 21:36
To: John Lindsay
Cc: <payload@ietf.org>; Ali C. Begen (abegen); Hartmut Foerster
Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-aptx-01


On Sep 24, 2013, at 1:03 PM, John Lindsay <Lindsay@worldcastsystems.com>
 wrote:

> Hi Jonathon
>=20
> Thanks for reviewing this and the points raised.
>=20
> I agree on re-reading that the delivery method test is a bit abstract=20
> and maybe doesn't add anything, by delivery method we mean the=20
> algorithm selected e.g. Standard or Enhanced APTX.
> As I think this is clear to the reader I think these lines could be=20
> removed.

Okay.

> Hence I propose changing some sections to remove these references.
>=20
> 6.2
> From
>      The required parameters "variant" and "bitresolution" MUST be
>      included in the SDP "a=3Dfmtp" attribute and MUST follow the
>      delivery-method that applies.=20
> To
>      The required parameters "variant" and "bitresolution" MUST be
>      included in the SDP "a=3Dfmtp" attribute.

Seems good.

> From
>      The optional parameters "stereo-channel-pairs", "embedded-
>      autosync-channels", "embedded-aux-channels", and "maxptime" when
>      present, MUST be included in the SDP "a=3Dfmtp" attribute and =
MUST
>      follow the delivery-method that applies.=20
> To
>      The optional parameters "stereo-channel-pairs", "embedded-
>      autosync-channels", "embedded-aux-channels",  MUST be=20
>      included in the SDP "a=3Dfmtp" attribute.

You lost the "when present" in your edit, which I assume should still be
there.

> SDP-Mapping
>=20
> Agreed this should have its own SDP a=3Dattribute, suggest changing=20
> section 6.2 from
>=20
>     The parameter "maxptime" when present, MUST be included in the=20
>      SDP "a=3Dmaxptime" attribute.

That seems good.

> Marker bit.
>=20
> Again this can be changed as per RFC3551, as we have no use for this=20
> in our profile (RFC3550) it can follow the recommendations of RFC 3551

> where it will be zero for most of the intended applications that we=20
> can see.
>=20
> Suggest changing section 5.1 from
>=20
>  o  M - This payload format defines no use for this bit. Senders
>      SHOULD set this bit to zero in each outgoing packet.
>=20
>  o  M - As per [RFC3551]

A bit vague, since RFC 3551 has other usages of the M bit for video.
I'd suggest "As per [RFC3551] Section 4.1".


> Payload types
> As I understand it the avtcore group has been discussing a new RFC=20
> that will update RFC3551 due to conflicts but it is currently=20
> unpublished work. What recommendations does the group have for the doc
on this?
> Currently I feel we reflect what is stated in RFC3551?

I'd just say "A dynamic payload type MUST be used", and drop the
reference to the range.

> Aux data
> The Aux data channel can be used for whatever the codec user requires=20
> and is not defined by this draft, it provides a point to point channel

> and is embedded within the Audio payload at 1/4 of the Audio sampling=20
> rate?

The problem with "whatever the codec user requires" is that it's
"whatever the sender wants", not "whatever the receiver wants" -- so the
receiver might have no idea what the sender is putting in this channel.
Unless you're assuming that both sides are under the same administrative
control, this seems to me like it can to interoperability problems.

What is this channel commonly used for, and how is it configured?

> To get access to this you would have to decode the audio payload, for=20
> which you would need either a standard of Enhanced apt-X codec. Hence=20
> we feel any security issues are minimal? What is the feeling of the=20
> workgroup?

Again, it's what it's used for, and whether you can always guarantee
that the other end isn't sending you something hostile if the channel is
used for some sort of active data.  (Don't assume
security-through-obscurity; i.e., the apt-X codec being proprietary
doesn't mean that a bad guy can't encode or decode it.)

>=20
> Section 3 of the doc provides an overview of the codecs operation. The

> aim of the draft is how to packetize Standard or Enhanced apt-X over=20
> RTP not to explain the operation of the codec. Further details are=20
> likely to be proprietary to CSR and interested parties would need to=20
> talk to CSR directly about obtaining licenses and setting up NDA's for

> further information.

That's fine -- but does CSR have any public (non-technical) data sheets
that can be cited, or anything else that an interested party can use to
figure out how to contact CSR to decide whether they want to begin this
process?  I see for instance
<http://www.csr.com/sites/default/files/white-papers/aptx_lossless_white
paper1.pdf> -- citing that (or something better) as an informative
reference for background on apt-X would seem reasonable.

>=20
> Please feel free to comment on the above.
>=20
> Thanks
>=20
> John
>=20
>=20
> -----Original Message-----
> From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On=20
> Behalf Of Jonathan Lennox
> Sent: 16 September 2013 19:36
> To: Ali C. Begen (abegen)
> Cc: payload@ietf.org
> Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-aptx-01
>=20
> The document discusses that a number of items are dependent on the=20
> "delivery method", and says that the delivery method can be negotiated

> through offer/answer.  However, the delivery method parameter is never

> actually defined anywhere.  Either delivery method needs to be=20
> defined, or discussion of it needs to be removed, depending on whether

> or not delivery-method was actually meant to be included in the final
spec.
>=20
> The SDP mapping says that maxptime is encoded in the fmtp parameter.
> This is incorrect; as specified by RFC 4566, maxptime is carried in=20
> its own SDP a=3D attribute.
>=20
> The RTP header section says that the marker bit is not used.  Unless=20
> there's a good reason for this, I suggest that instead the usual RFC
> 3551 marker bit semantic for audio be applied (i.e., the marker bit is

> set for end of silence / beginning of talkspurt).  If there's a reason

> this semantic is inappropriate, it should be explained.
>=20
> The RTP header section also says that the RTP dynamic payload types=20
> can only be in the range 96 - 127.  This is incorrect; that's the=20
> primary range, but other values can be used once that range is=20
> exhausted.  See, e.g., the recent discussion on the avtcore list.
>=20
> No information is given about the semantic content of the data carried

> in the auxiliary data channel.  Is the semantic of this data defined=20
> by the codec, or would it need to be described or negotiated in SDP?
> (Doing neither, i.e. providing an arbitrary data channel with no=20
> mechanism for describing or negotiating its content, seems likely to=20
> lead to interoperability problems.)  Also, are there any security=20
> considerations introduced by the auxiliary data channel?
>=20
> It would be good to have a reference to some document by CSR plc=20
> describing the apt-X codecs, even if they don't include full=20
> proprietary details of the workings of the codec.
>=20
> On Sep 16, 2013, at 9:08 AM, "Ali C. Begen (abegen)"=20
> <abegen@cisco.com>
> wrote:
>=20
>> (As a WG co-chair)
>>=20
>> I am starting WGLC for the following draft (version 01).=20
>> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-aptx/
>>=20
>> It is a short document. If you have comments, agreements or
> disagreements, please let the list know in either case.
>>=20
>> The deadline for the WGLC is September 30th.
>>=20
>> -acbegen
>> _______________________________________________
>> payload mailing list
>> payload@ietf.org
>> https://www.ietf.org/mailman/listinfo/payload
>>=20
>=20
> --
> Jonathan Lennox
> jonathan@vidyo.com
>=20
>=20
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload


From mramalho@cisco.com  Wed Sep 25 14:16:45 2013
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From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
To: "payload@ietf.org" <payload@ietf.org>
Thread-Topic: Toward WGLC on G.711.0 RTP Payload Spec
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Subject: [payload] Toward WGLC on G.711.0 RTP Payload Spec
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All,

The payload WG has a milestone to submit the RTP Payload Format for G.711.0=
 for proposed standard in December 2013 (http://datatracker.ietf.org/wg/pay=
load/charter/ ).

The current working group draft is draft-ietf-payload-g7110-00 ( http://dat=
atracker.ietf.org/doc/draft-ietf-payload-g7110/ ).

Although this working group document is a "00" draft, it spent much time an=
d prior re-writes as an individual contribution draft known as draft-ramalh=
o-payload-g7110-0X.

I have not received any requests to change this document since the conversi=
on of the individual contribution draft to the working group draft (which o=
ccurred in June 2013).

As I would like to ask the Payload chairs to last call this document at the=
 Vancouver IETF, please provide any input you have on this document to this=
 list (including if you think it is ready as it is).

Thanks in advance,

Michael A. Ramalho

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<p class=3D"MsoNormal">All,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The payload WG has a milestone to submit the RTP Pay=
load Format for G.711.0 for proposed standard in December 2013 (http://data=
tracker.ietf.org/wg/payload/charter/ ).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The current working group draft is draft-ietf-payloa=
d-g7110-00 (
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-payload-g7110/">http:=
//datatracker.ietf.org/doc/draft-ietf-payload-g7110/</a> ).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Although this working group document is a &#8220;00&=
#8221; draft, it spent much time and prior re-writes as an individual contr=
ibution draft known as draft-ramalho-payload-g7110-0X.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I have not received any requests to change this docu=
ment since the conversion of the individual contribution draft to the worki=
ng group draft (which occurred in June 2013).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">As I would like to ask the Payload chairs to last ca=
ll this document at the Vancouver IETF, please provide any input you have o=
n this document to this list (including if you think it is ready as it is).=
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks in advance,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Michael A. Ramalho<o:p></o:p></p>
</div>
</body>
</html>

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Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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Hi,

Removing everything except questions / comments.

Feedback especially requested on the last comment. Anyone with SVC
experience should review the changes for Section 5.1.5.

Inline

On 2013-09-22 18:31, Colin Perkins wrote:
> Hi Magnus,
> 
> On 21 Sep 2013, at 12:56, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
>> Thanks for the review comments. I have tried addressing them. I am
> 
> 
>>> - Section 3.2: the introduction to this section could be clearer
>>> that this is an overview of RTP features that are available
>>> independent of the payload format, and not a list of things that
>>> the payload format needs to explicitly opt in to/out of.
>> 
>> This section reviews a number of RTP features and concepts that
>> are available in RTP independent of the payload format. The RTP
>> payload format can make use of these when appropriate, and in some
>> cases even effect the behavior (RTP Timestamp and marker bit). This
>> section does not remove the necessity to read up on RTP. However it
>> does point out a few important details to remember when designing a
>> payload format.
> 
> It might be worth expanding this to say "...even affect the behaviour
> (RTP timestamp and marker bit), but it is important to note that not
> all features and concepts are relevant to every payload format"?

Good suggestion: The resulting text now reads:

This section reviews a number of RTP features and concepts that are
available in RTP independent of the payload format. The RTP payload
format can make use of these when appropriate, and even affect the
behaviour (RTP timestamp and marker bit), but it is important to note
that not all features and concepts are relevant to every payload format.
This section does not remove the necessity to read up on RTP. However it
does point out a few important details to remember when designing a
payload format.



> 
>>> - In Section 3.2.2 the statement “[RFC5285] specifies how to
>>> extend the RTP header to carry metadata relating to the payload”
>>> is potentially misleading, since the type of information that
>>> usually goes into an RTP payload header is also “metadata
>>> relating to the payload”. It might be useful to expand this
>>> discussion slightly, to make the distinction clear.
>> 
>> Is this better?
>> 
>> Header extensions:  Payload formats normally specifies the
>> carrying of meta data related to the media payload in the payload
>> header. In some cases the meta data may be more generic, e.g.
>> relevant more to the media type (e.g. audio or video), or RTP
>> packets in general.  In those cases the carrying of this additional
>> meta data can be specified in a header extension.  One example is
>> the transport of SMPTE time-codes in the RTP header [RFC5484].  As 
>> [RFC5285] specifies, header extensions must not contain information
>> required in order to decode the payload successfully.
> 
> That's better, but I might suggest even stronger wording: "RTP
> payload formats often need to include metadata relating to the
> payload data being transported. Such metadata is sent as a payload
> header, at the start of the payload section of the RTP packet. The
> RTP packet also includes space for a header extension [RFC5275]; this
> can be used to transport payload format independent metadata, for
> example a SMPTE time code for the packet [RFC5484]. The RTP header
> extensions is not intended to carry headers that relate to a
> particular payload format., and must not contain information needed
> in order to decode the payload."
> 
> The last sentence of the first paragraph of Section 3.2.2 also needs
> updating. I suggest "Finally, [RFC5285] specifies how to transport
> payload format independent metadata relating to the RTP packet".
> 

I like this, I will include this.


>>> - In Section 3.2.2, it might be useful to explain that video
>>> payload formats are a common example of “payload formats with a
>>> timestamp definition which results in no or little correlation
>>> between the media time instance and its transmission time”;
>>> otherwise this sounds like such payload formats are something
>>> unusual.
>> 
>> This discussion got moved into the new section 5.2, where the
>> related text says:
>> 
>> RTP payload formats with a timestamp definition which results in
>> no or little correlation between the media time instance and its 
>> transmission time cause the RTCP jitter calculation to become 
>> unusable due to the errors introduced on the sender side.  Such 
>> examples are for example payload formats for video that requires 
>> multiple packets, where a set of packets will have the same 
>> timestamp, another are payloads that includes alternating set of 
>> redundant information, if the definition used are for the earliest 
>> data included, the the timestamp may jump back and forth.  It
>> should be noted if the payload format has this property or not.
>> 
>> Good enough?
> 
> Better, but could still be more explicit. How about changing the
> "Such examples are..." sentence to "A common example is a payload
> format for a video codec where the RTP timestamp represents the
> capture time of the video frame, but frames are large enough that
> multiple RTP packets need to be sent for each frame spread across the
> framing interval".

Yes, that is more focused.

> 
>>> - Section 3.4: should this say anything about AQM and/or QoS 
>>> support?
>> 
>> I tried writing something about this. I don't know how useful this
>> is. Feedback much appreciated.
>> 
>> 
>> 3.4.2.  Different Queuing Algorithms
>> 
>> Routers and Switches on the network path between an IP sender and
>> a particular receiver can exhibit different behaviors affecting the
>> end to end characteristics.  One of the more important aspects of
>> this is queuing behavior.  Routers and Switches has some amount of
>> queuing to handle temporary bursts of data that designated to leave
>> the switch or router on the same egress link.  A queue when not
>> empty results in an increased path delay.
>> 
>> The implementation of the queuing effects the delay and also how 
>> congestion signals (Explicint Congestion Marking (ECN) or packet 
>> drops) are provided to the flow.
> 
> Reference RFC 6679 here?

Yes, for people that like to dig into ECN, going to the RTP spec is a
good first step.

> 
>> The other aspects are if the flow shares the queue with other flows
>> and how the implementation affects the flow interaction.  This
>> becomes important for example when real- time flows interacts with
>> long lived TCP flows.  TCP have a built in behavior in its
>> congestion control that strive to fill the buffer, thus all flows
>> sharing the buffer experienced the delay build up.
>> 
>> A common but quite poor queue handling mechanism is tail-drop,
>> i.e. only drop packets when the incoming packet doesn't fit in the
>> queue. Active Queue Management (AQM) is a term covering mechanism
>> that tries to do something smarter buy actively managing the queue,
>> for example by sending congestion signals earlier by droping
>> packets earlier in the queue.  The behavior also affects the flow
>> interactions.  For example Random Early Drop (RED) selects which
>> packet to drop randomly.  This to give flows that have more packets
>> in the queue a higher probability to experience the packet loss
>> (congestion signal). There is ongoing work to find suitable
>> mechanisms to recommend for implementation and reduce the use of
>> tail-drop.
>> 
>> 3.4.3.  Quality of Service
>> 
>> Using best effort Internet one have no guarantees for the paths 
>> properties.  Quality of Service (QoS) mechanism are intended to 
>> provide the possibility to bound the path properties.  Where
>> Diffserv [RFC2475] markings effects the queuing and forwarding
>> behaviors of routers, the mechanism provides only statistical
>> guarantees and care in how much marked packets of different types
>> that are entering the network.  Flow based QoS like IntServ
>> [RFC1663] has the potential for stricter guarantees as the
>> properties are agreed on by each hop on the path.
> 
> I think this is useful. It highlights that there are issues to
> consider, which is the goal.


Good, I was uncertain if this text said the right things.

> 
>>> - The discussion of SST and MST in Section 5.1.5 seems to
>>> confuse sessions with SSRCs. Isn’t it more important whether the
>>> layers are sent within a single RTP media stream, or as separate
>>> RTP media streams with different SSRC values? Whether the
>>> different SSRCs are sent in one RTP session of many is then a
>>> separate issue, depending on how easily one wants to be able to
>>> separate the flows for QoS purposes.
>> 
>> The issue is that the SVC payload RFC is a bit confused on this
>> matter. I agree that there are two separate important distinctions
>> here. First if the layers are sent in one RTP packet stream or used
>> multiple ones. In the second case one gets the question of how one
>> distributes that over RTP sessions and underlying transports.
>> Especially as some that implements SVC MST clearly uses a single
>> RTP session and are after the possibility on a SSRC level to
>> control transmission of layers in RTP middleboxes.
>> 
>> I have tried to address this in the text, please review.
>> 
>> 5.1.5.  Media Scalability
>> 
>> Some codecs support various types of media scalability, i.e. some 
>> data of a media stream may be removed to adapt the media's 
>> properties, such as bitrate and quality.  The adaptation may be 
>> applied in the following dimensions of the media:
>> 
>> Temporal:  For video codecs it is possible to adapt the frame
>> rate, e.g. for H.264 [RFC6184].
>> 
>> Spatial:  Video codecs supporting scalability may adapt the 
>> resolution, e.g. in SVC [RFC6190].
>> 
>> Quality:  The quality of the media stream may be scaled by
>> adapting the accuracy of the coding process, as, e.g.  possible
>> with Signal to Noise Ratio (SNR) fidelity scalability of SVC
>> [RFC6190].
>> 
>> At the time of writing this document, codecs that support
>> scalability have a bit of revival.  It has been realized that
>> getting the required functionality for supporting the features of
>> the media stream into the RTP framework is quite challenging.  One
>> of the recent examples for layered and scalable codecs is Scalable
>> Video Coding [RFC6190] (SVC).
>> 
>> SVC is a good example for a payload format supporting media 
>> scalability features, which have been in its basic form already 
>> included in RTP.  A layered codec supports the dropping of data
>> parts of a media stream, i.e. RTP packets may be not transmitted
>> or forwarded to a client in order to adapt the media stream rate as
>> well as the media stream quality, while still providing a decodable
>> subset of the media stream to a client.  One example for using the 
>> scalability feature may be an RTP Mixer (Multipoint Control Unit) 
>> which controls the rate and quality sent out to participants in a 
>> communication based on dropping RTP packets or removing part of
>> the payload.  Another example may be an transport channel which
>> allows for differentiation in Quality of Service (QoS) parameters
>> based on RTP sessions in a multicast session.  In such a case, the
>> more important packets of the scalable media stream (base layer)
>> may get better QoS parameters, then the less important packets
>> (enhancement layer) in order to provide some kind of graceful
>> degradation.  The scalability features required for allowing an
>> adaptive transport as described in the two examples above are based
>> on RTP multiplexing in order to identify the packets to be dropped
>> or transmitted/forwarded. The multiplexing features defined for
>> Scalable Video Coding [RFC6190] are:
>> 
>> single session transmission (SST), where all media layers of the 
>> media are transported as single source (SSRC) in a single RTP 
>> session; as well as
>> 
>> multi session transmission (MST), which should more accurately be 
>> called multi stream transmission, where different media layers or a
>> set of media layers are transported in different RTP streams, i.e.
>> using multiple sources (SSRCs).
>> 
>> In the first case (SST), additional in-band as well as out-of-band 
>> signaling is required in order to allow identification of packets 
>> belonging to a specific media layer.  Furthermore, an adaptation
>> of the media stream requires dropping of specific packets in order
>> to provide the client with a compliant media stream.  In case of
>> using encryption, it is typically required for an adapting network
>> device to be in the security context to allow packet dropping and
>> providing an intact RTP session to the client.  This typically
>> requires the network device to be an RTP mixer.
>> 
>> In general having a media unaware network device dropping
>> excessive packets will be more problematic than having a Media
>> Aware Network Entity (MANE).  First is the need to understand the
>> media format and know which ADUs or payloads that belongs to the
>> layers, that no other layer will be dependent on after the
>> dropping.  Secondly, if the MANE can work as RTP mixer or
>> translator it can rewrite the RTP and RTCP in such a way that the
>> receiver will not suspect non-intentional RTP packet losses needing
>> repair actions.  This as the receiver can't determine if a lost
>> packet was an important base layer packet or one of the less
>> important extension layers.
>> 
>> In the second case (MST), the RTP packet streams can be sent using
>> a single or multiple RTP sessions, and thus transport flows, e.g.
>> on differnt multicast groups.  Transmitting the streams in
>> different RTP sessions then the out-of-band signaling typically
>> provides enough information to identify the media layers and its
>> properties.  The decision for dropping packets is based on the
>> Network Address which identifies the RTP session to be dropped.  In
>> order to allow correct data provision to a decoder after reception
>> from different sessions, data re-alignment mechanisms are described
>> for Scalable Video Coding [RFC6190].  A more generic one is also
>> described in Rapid Sync for RTP flows [RFC6051], which is purely
>> based on existing RTP
>> 
>> mechanisms, i.e. the NTP timestamp, for inter-session 
>> synchronization.  Another signaling feature is the generic
>> indication of dependencies of RTP sessions in SDP, as defined in
>> the Media Decoding Dependency Grouping in SDP [RFC5583].
>> 
>> Using MST within a single RTP session is also possible and allow 
>> stream level handling instead of looking deeper into the packets by
>> a MANE.  However, transport flow level properties will be the same 
>> unless packet based mechanisms like DiffServ is used.
>> 
>> When QoS settings, e.g. DiffServ markings, are used to ensure that 
>> the extension layers are dropped prior the base layer the
>> receiving end-point has the benefit in MST to know which layer or
>> set of layers the missing packets belong as it will be bound to
>> different RTP sessions.  Thus explicitly indicating the importance
>> of the loss.
> 
> I think this is okay, but it really needs an implementer of the SVC
> format to comment too.

Yes, that would be desirable.

> 
>>> - In Section 5.1.6, avoiding wrap-around within the RTCP
>>> reporting timeout interval is also important.
>> 
>> If I understand this correctly, this is a minor issue with the RTCP
>> SR and RR packet types as they all uses 32-bit counters. However,
>> other packet types may have issues if they not use extended
>> sequence number counters.
>> 
>> This might be more relevant to considerations for configuring RTCP 
>> bandwidth and design of RTCP extensions, rather than being an RTP 
>> Payload issue. There exists a configuration goal to ensure that
>> RTCP reporting doesn't wrap its fields within multiple reporting
>> intervals.
>> 
>> I propose that the following paragraph is added in this section:
>> 
>> RTCP is also affected by high packet rates. For RTCP mechanism that
>> do not use extended counters there is significant risk that they
>> wrap multiple times between RTCP reporting or feedback, thus
>> producing uncertainty which packet(s) that are referenced. The
>> payload designer can't effect the RTCP packet formats used and
>> their design, but can note this considerations when configuring
>> RTCP bandwidth and reporting intervals to avoid to wrapping
>> issues.
> 
> That's fine, although I was actually suggesting something simpler,
> which is to change "As rule of thumb, it must not be possible to wrap
> the sequence number space in less than 2 minutes (TCP maximum segment
> lifetime)" to "...must not be possible to wrap the sequence number
> space in less than an RTCP reporting interval".

That is difficult for someone that is new to quantify into meaningful
values. Especially as we have reporting intervals between milliseconds
(a scaled RTCP interval for the gigabit raw video) to many seconds for
large groups with small amounts of receiver reporting bandwidths.

In addition I don't think one reporting interval is really sufficient.
In absence of RTCP packet loss it provides the bare minimum to detect a
wrap in RTCP. But one really need to take the maximum interval between
two RTCP packets into account and considering loss I think 3-4 intervals
are needed to have reasonable saftey.

The motivation for using TCP MSL was the argument that TCP MSL is
selected as the time interval no emitted packet would still be possible
to arrive after. That is true also for RTP packets and thus a valid
considerations providing sufficient margins. It also covers almost all
RTCP usages that I can think of.

I have changed the text to now say:

Some media codecs require high packet rates, and in these cases the RTP
sequence number wraps too quickly. As rule of thumb, it must not be
possible to wrap the sequence number space within at least three RTCP
reporting intervals. As the reporting interval can vary widely due to
configuration and session properties, and also must take into account
the randomization of the interval, one can use the TCP maximum segment
lifetime, which is 2 minutes in ones calculations.


Cheers

Magnus Westerlund

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Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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WG,

As part of editing the comments I am proposing one additional change.
This is a clarification of text in Section 3.3.2.1 that used to say:

   A further issue with offer/answer which complicates things is that
   the answerer is allowed to renumber the payload types between offer
   and answer.  This is not recommended but allowed for support of
   gateways to the ITU conferencing suite.  This means that it must be
   possible to bind answers for payload types to the payload types in
   the offer even when the payload type number has been changed, and
   some of the proposed payload types have been removed.  This binding
   must normally be done by matching the configurations originally
   offered against those in the answer.

Which I propose to change to:

   A further issue with offer/answer which complicates things is that
   the answerer is allowed to renumber the payload types between offer
   and answer.  This is not recommended but allowed for support of
   gateways to the ITU conferencing suite.  This means that it must be
   possible to bind answers for payload types to the payload types in
   the offer even when the payload type number has been changed, and
   some of the proposed payload types have been removed.  This binding
   must normally be done by matching the configurations originally
   offered against those in the answer.  This may require specification
   in the payload format of which parameters that constitute a
   configuration, for example as done in Section 8.2.2 of the H.264 RTP
   Payload format [RFC6184]; "The parameters identifying a media format
   configuration for H.264 are profile-level-id and packetization-mode".

Making clear that one may have to write a definition of it. This has
become relevant in the discussion in MMUSIC regarding BUNDLE and what a
codec configuration really is. But the basic requirement for this
originates from Offer/Answer itself (RFC 3264).

I appreciate feedback on this change.

Cheers

Magnus

On 2013-08-22 10:58, Ali C. Begen (abegen) wrote:
> (As a WG co-chair)
> 
> I am starting WGLC for the following draft (version 05). 
> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-howto/
> 
> This is an important document as it outlines how someone is supposed to write the payload specs.
> 
> If there are sections you don't understand or that are not clear enough, etc. please post them to the list. We need to get this right to minimize further headaches for future documents in our WG.
> 
> The deadline for the WGLC is September 20th.
> 
> -acbegen
> _______________________________________________
> payload mailing list
> payload@ietf.org
> https://www.ietf.org/mailman/listinfo/payload
> 
> 


-- 

Magnus Westerlund

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From simao.campos@itu.int  Thu Sep 26 02:09:37 2013
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From: "Campos, Simao" <simao.campos@itu.int>
To: "Michael Ramalho (mramalho)" <mramalho@cisco.com>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: Toward WGLC on G.711.0 RTP Payload Spec
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Date: Thu, 26 Sep 2013 09:09:27 +0000
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Subject: Re: [payload] Toward WGLC on G.711.0 RTP Payload Spec
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--_000_47625A4A3006CE47BB72AF5BC1EBBEFDCB30C662TUCHM02TUECSPUN_
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Michael

I support your proposal of starting WGLC for this I-D.

Sim=E3o

From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On Behalf =
Of Michael Ramalho (mramalho)
Sent: Wednesday, September 25, 2013 11:17 PM
To: payload@ietf.org
Subject: [payload] Toward WGLC on G.711.0 RTP Payload Spec

All,

The payload WG has a milestone to submit the RTP Payload Format for G.711.0=
 for proposed standard in December 2013 (http://datatracker.ietf.org/wg/pay=
load/charter/ ).

The current working group draft is draft-ietf-payload-g7110-00 ( http://dat=
atracker.ietf.org/doc/draft-ietf-payload-g7110/ ).

Although this working group document is a "00" draft, it spent much time an=
d prior re-writes as an individual contribution draft known as draft-ramalh=
o-payload-g7110-0X.

I have not received any requests to change this document since the conversi=
on of the individual contribution draft to the working group draft (which o=
ccurred in June 2013).

As I would like to ask the Payload chairs to last call this document at the=
 Vancouver IETF, please provide any input you have on this document to this=
 list (including if you think it is ready as it is).

Thanks in advance,

Michael A. Ramalho

--_000_47625A4A3006CE47BB72AF5BC1EBBEFDCB30C662TUCHM02TUECSPUN_
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<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Michael<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">I support your proposa=
l of starting WGLC for this I-D.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Sim=E3o <o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> payload-=
bounces@ietf.org [mailto:payload-bounces@ietf.org]
<b>On Behalf Of </b>Michael Ramalho (mramalho)<br>
<b>Sent:</b> Wednesday, September 25, 2013 11:17 PM<br>
<b>To:</b> payload@ietf.org<br>
<b>Subject:</b> [payload] Toward WGLC on G.711.0 RTP Payload Spec<o:p></o:p=
></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">All,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The payload WG has a milestone to submit the RTP Pay=
load Format for G.711.0 for proposed standard in December 2013 (<a href=3D"=
http://datatracker.ietf.org/wg/payload/charter/">http://datatracker.ietf.or=
g/wg/payload/charter/</a> ).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The current working group draft is draft-ietf-payloa=
d-g7110-00 (
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-payload-g7110/">http:=
//datatracker.ietf.org/doc/draft-ietf-payload-g7110/</a> ).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Although this working group document is a &#8220;00&=
#8221; draft, it spent much time and prior re-writes as an individual contr=
ibution draft known as draft-ramalho-payload-g7110-0X.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I have not received any requests to change this docu=
ment since the conversion of the individual contribution draft to the worki=
ng group draft (which occurred in June 2013).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">As I would like to ask the Payload chairs to last ca=
ll this document at the Vancouver IETF, please provide any input you have o=
n this document to this list (including if you think it is ready as it is).=
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks in advance,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Michael A. Ramalho<o:p></o:p></p>
</div>
</body>
</html>

--_000_47625A4A3006CE47BB72AF5BC1EBBEFDCB30C662TUCHM02TUECSPUN_--

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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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Date: Thu, 26 Sep 2013 14:04:58 +0200
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Cc: "payload@ietf.org" <payload@ietf.org>
Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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On 2013-09-26 11:46, Ali C. Begen (abegen) wrote:
> Looks good to me except that there are bunch of lower-case 2119 keywords.
> Any possibility to avoid them or is it better to make them upper case?

No, they should not be interpreted in the RFC 2119 upper case meaning.
This is an informative document and regular English interpretation are
fine. I find it difficult to write without using may, should, recommend
and to some degree must. And removing these out of the document would be
a massive undertaking which I fail to see the benefit with.

Cheers

Magnus


> 
> -----Original Message-----
> From: Magnus Westerlund <magnus.westerlund@ericsson.com>
> Date: Thursday, September 26, 2013 10:28 AM
> To: "payload@ietf.org" <payload@ietf.org>
> Cc: "Ali C. Begen" <abegen@cisco.com>
> Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05
> 
>> WG,
>>
>> As part of editing the comments I am proposing one additional change.
>> This is a clarification of text in Section 3.3.2.1 that used to say:
>>
>>   A further issue with offer/answer which complicates things is that
>>   the answerer is allowed to renumber the payload types between offer
>>   and answer.  This is not recommended but allowed for support of
>>   gateways to the ITU conferencing suite.  This means that it must be
>>   possible to bind answers for payload types to the payload types in
>>   the offer even when the payload type number has been changed, and
>>   some of the proposed payload types have been removed.  This binding
>>   must normally be done by matching the configurations originally
>>   offered against those in the answer.
>>
>> Which I propose to change to:
>>
>>   A further issue with offer/answer which complicates things is that
>>   the answerer is allowed to renumber the payload types between offer
>>   and answer.  This is not recommended but allowed for support of
>>   gateways to the ITU conferencing suite.  This means that it must be
>>   possible to bind answers for payload types to the payload types in
>>   the offer even when the payload type number has been changed, and
>>   some of the proposed payload types have been removed.  This binding
>>   must normally be done by matching the configurations originally
>>   offered against those in the answer.  This may require specification
>>   in the payload format of which parameters that constitute a
>>   configuration, for example as done in Section 8.2.2 of the H.264 RTP
>>   Payload format [RFC6184]; "The parameters identifying a media format
>>   configuration for H.264 are profile-level-id and packetization-mode".
>>
>> Making clear that one may have to write a definition of it. This has
>> become relevant in the discussion in MMUSIC regarding BUNDLE and what a
>> codec configuration really is. But the basic requirement for this
>> originates from Offer/Answer itself (RFC 3264).
>>
>> I appreciate feedback on this change.
>>
>> Cheers
>>
>> Magnus
>>
>> On 2013-08-22 10:58, Ali C. Begen (abegen) wrote:
>>> (As a WG co-chair)
>>>
>>> I am starting WGLC for the following draft (version 05).
>>> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-howto/
>>>
>>> This is an important document as it outlines how someone is supposed to
>>> write the payload specs.
>>>
>>> If there are sections you don't understand or that are not clear
>>> enough, etc. please post them to the list. We need to get this right to
>>> minimize further headaches for future documents in our WG.
>>>
>>> The deadline for the WGLC is September 20th.
>>>
>>> -acbegen
>>> _______________________________________________
>>> payload mailing list
>>> payload@ietf.org
>>> https://www.ietf.org/mailman/listinfo/payload
>>>
>>>
>>
>>
>> -- 
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> FÃ¤rÃ¶gatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>
>>
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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On Sep 26, 2013, at 3:04 PM, Magnus Westerlund <magnus.westerlund@ericsson.=
com> wrote:

> On 2013-09-26 11:46, Ali C. Begen (abegen) wrote:
>> Looks good to me except that there are bunch of lower-case 2119 keywords=
.
>> Any possibility to avoid them or is it better to make them upper case?
>=20
> No, they should not be interpreted in the RFC 2119 upper case meaning.
> This is an informative document and regular English interpretation are
> fine.

yes, it is information and I keep forgetting that.

> I find it difficult to write without using may, should, recommend
> and to some degree must. And removing these out of the document would be
> a massive undertaking which I fail to see the benefit with.
>=20
I think putting a note somewhere around the intro part about the use of the=
se keywords in lower case would be beneficial.=20


> Cheers
>=20
> Magnus
>=20
>=20
>>=20
>> -----Original Message-----
>> From: Magnus Westerlund <magnus.westerlund@ericsson.com>
>> Date: Thursday, September 26, 2013 10:28 AM
>> To: "payload@ietf.org" <payload@ietf.org>
>> Cc: "Ali C. Begen" <abegen@cisco.com>
>> Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05
>>=20
>>> WG,
>>>=20
>>> As part of editing the comments I am proposing one additional change.
>>> This is a clarification of text in Section 3.3.2.1 that used to say:
>>>=20
>>>  A further issue with offer/answer which complicates things is that
>>>  the answerer is allowed to renumber the payload types between offer
>>>  and answer.  This is not recommended but allowed for support of
>>>  gateways to the ITU conferencing suite.  This means that it must be
>>>  possible to bind answers for payload types to the payload types in
>>>  the offer even when the payload type number has been changed, and
>>>  some of the proposed payload types have been removed.  This binding
>>>  must normally be done by matching the configurations originally
>>>  offered against those in the answer.
>>>=20
>>> Which I propose to change to:
>>>=20
>>>  A further issue with offer/answer which complicates things is that
>>>  the answerer is allowed to renumber the payload types between offer
>>>  and answer.  This is not recommended but allowed for support of
>>>  gateways to the ITU conferencing suite.  This means that it must be
>>>  possible to bind answers for payload types to the payload types in
>>>  the offer even when the payload type number has been changed, and
>>>  some of the proposed payload types have been removed.  This binding
>>>  must normally be done by matching the configurations originally
>>>  offered against those in the answer.  This may require specification
>>>  in the payload format of which parameters that constitute a
>>>  configuration, for example as done in Section 8.2.2 of the H.264 RTP
>>>  Payload format [RFC6184]; "The parameters identifying a media format
>>>  configuration for H.264 are profile-level-id and packetization-mode".
>>>=20
>>> Making clear that one may have to write a definition of it. This has
>>> become relevant in the discussion in MMUSIC regarding BUNDLE and what a
>>> codec configuration really is. But the basic requirement for this
>>> originates from Offer/Answer itself (RFC 3264).
>>>=20
>>> I appreciate feedback on this change.
>>>=20
>>> Cheers
>>>=20
>>> Magnus
>>>=20
>>> On 2013-08-22 10:58, Ali C. Begen (abegen) wrote:
>>>> (As a WG co-chair)
>>>>=20
>>>> I am starting WGLC for the following draft (version 05).
>>>> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-howto/
>>>>=20
>>>> This is an important document as it outlines how someone is supposed t=
o
>>>> write the payload specs.
>>>>=20
>>>> If there are sections you don't understand or that are not clear
>>>> enough, etc. please post them to the list. We need to get this right t=
o
>>>> minimize further headaches for future documents in our WG.
>>>>=20
>>>> The deadline for the WGLC is September 20th.
>>>>=20
>>>> -acbegen
>>>> _______________________________________________
>>>> payload mailing list
>>>> payload@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/payload
>>>>=20
>>>>=20
>>>=20
>>>=20
>>> --=20
>>>=20
>>> Magnus Westerlund
>>>=20
>>> ----------------------------------------------------------------------
>>> Multimedia Technologies, Ericsson Research EAB/TVM
>>> ----------------------------------------------------------------------
>>> Ericsson AB                | Phone  +46 10 7148287
>>> F=E4r=F6gatan 6                | Mobile +46 73 0949079
>>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>>> ----------------------------------------------------------------------
>>>=20
>>>=20
>>=20
>>=20
>=20
>=20
> --=20
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20


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From: Colin Perkins <csp@csperkins.org>
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Date: Thu, 26 Sep 2013 17:07:59 +0100
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Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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Hi Magnus,

This all looks fine.=20

Thanks,
Colin




On 26 Sep 2013, at 08:07, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:
> Hi,
>=20
> Removing everything except questions / comments.
>=20
> Feedback especially requested on the last comment. Anyone with SVC
> experience should review the changes for Section 5.1.5.
>=20
> Inline
>=20
> On 2013-09-22 18:31, Colin Perkins wrote:
>> Hi Magnus,
>>=20
>> On 21 Sep 2013, at 12:56, Magnus Westerlund
>> <magnus.westerlund@ericsson.com> wrote:
>>> Thanks for the review comments. I have tried addressing them. I am
>>=20
>>=20
>>>> - Section 3.2: the introduction to this section could be clearer
>>>> that this is an overview of RTP features that are available
>>>> independent of the payload format, and not a list of things that
>>>> the payload format needs to explicitly opt in to/out of.
>>>=20
>>> This section reviews a number of RTP features and concepts that
>>> are available in RTP independent of the payload format. The RTP
>>> payload format can make use of these when appropriate, and in some
>>> cases even effect the behavior (RTP Timestamp and marker bit). This
>>> section does not remove the necessity to read up on RTP. However it
>>> does point out a few important details to remember when designing a
>>> payload format.
>>=20
>> It might be worth expanding this to say "...even affect the behaviour
>> (RTP timestamp and marker bit), but it is important to note that not
>> all features and concepts are relevant to every payload format"?
>=20
> Good suggestion: The resulting text now reads:
>=20
> This section reviews a number of RTP features and concepts that are
> available in RTP independent of the payload format. The RTP payload
> format can make use of these when appropriate, and even affect the
> behaviour (RTP timestamp and marker bit), but it is important to note
> that not all features and concepts are relevant to every payload =
format.
> This section does not remove the necessity to read up on RTP. However =
it
> does point out a few important details to remember when designing a
> payload format.
>=20
>=20
>=20
>>=20
>>>> - In Section 3.2.2 the statement =93[RFC5285] specifies how to
>>>> extend the RTP header to carry metadata relating to the payload=94
>>>> is potentially misleading, since the type of information that
>>>> usually goes into an RTP payload header is also =93metadata
>>>> relating to the payload=94. It might be useful to expand this
>>>> discussion slightly, to make the distinction clear.
>>>=20
>>> Is this better?
>>>=20
>>> Header extensions:  Payload formats normally specifies the
>>> carrying of meta data related to the media payload in the payload
>>> header. In some cases the meta data may be more generic, e.g.
>>> relevant more to the media type (e.g. audio or video), or RTP
>>> packets in general.  In those cases the carrying of this additional
>>> meta data can be specified in a header extension.  One example is
>>> the transport of SMPTE time-codes in the RTP header [RFC5484].  As
>>> [RFC5285] specifies, header extensions must not contain information
>>> required in order to decode the payload successfully.
>>=20
>> That's better, but I might suggest even stronger wording: "RTP
>> payload formats often need to include metadata relating to the
>> payload data being transported. Such metadata is sent as a payload
>> header, at the start of the payload section of the RTP packet. The
>> RTP packet also includes space for a header extension [RFC5275]; this
>> can be used to transport payload format independent metadata, for
>> example a SMPTE time code for the packet [RFC5484]. The RTP header
>> extensions is not intended to carry headers that relate to a
>> particular payload format., and must not contain information needed
>> in order to decode the payload."
>>=20
>> The last sentence of the first paragraph of Section 3.2.2 also needs
>> updating. I suggest "Finally, [RFC5285] specifies how to transport
>> payload format independent metadata relating to the RTP packet".
>>=20
>=20
> I like this, I will include this.
>=20
>=20
>>>> - In Section 3.2.2, it might be useful to explain that video
>>>> payload formats are a common example of =93payload formats with a
>>>> timestamp definition which results in no or little correlation
>>>> between the media time instance and its transmission time=94;
>>>> otherwise this sounds like such payload formats are something
>>>> unusual.
>>>=20
>>> This discussion got moved into the new section 5.2, where the
>>> related text says:
>>>=20
>>> RTP payload formats with a timestamp definition which results in
>>> no or little correlation between the media time instance and its
>>> transmission time cause the RTCP jitter calculation to become
>>> unusable due to the errors introduced on the sender side.  Such
>>> examples are for example payload formats for video that requires
>>> multiple packets, where a set of packets will have the same
>>> timestamp, another are payloads that includes alternating set of
>>> redundant information, if the definition used are for the earliest
>>> data included, the the timestamp may jump back and forth.  It
>>> should be noted if the payload format has this property or not.
>>>=20
>>> Good enough?
>>=20
>> Better, but could still be more explicit. How about changing the
>> "Such examples are..." sentence to "A common example is a payload
>> format for a video codec where the RTP timestamp represents the
>> capture time of the video frame, but frames are large enough that
>> multiple RTP packets need to be sent for each frame spread across the
>> framing interval".
>=20
> Yes, that is more focused.
>=20
>>=20
>>>> - Section 3.4: should this say anything about AQM and/or QoS
>>>> support?
>>>=20
>>> I tried writing something about this. I don't know how useful this
>>> is. Feedback much appreciated.
>>>=20
>>>=20
>>> 3.4.2.  Different Queuing Algorithms
>>>=20
>>> Routers and Switches on the network path between an IP sender and
>>> a particular receiver can exhibit different behaviors affecting the
>>> end to end characteristics.  One of the more important aspects of
>>> this is queuing behavior.  Routers and Switches has some amount of
>>> queuing to handle temporary bursts of data that designated to leave
>>> the switch or router on the same egress link.  A queue when not
>>> empty results in an increased path delay.
>>>=20
>>> The implementation of the queuing effects the delay and also how
>>> congestion signals (Explicint Congestion Marking (ECN) or packet
>>> drops) are provided to the flow.
>>=20
>> Reference RFC 6679 here?
>=20
> Yes, for people that like to dig into ECN, going to the RTP spec is a
> good first step.
>=20
>>=20
>>> The other aspects are if the flow shares the queue with other flows
>>> and how the implementation affects the flow interaction.  This
>>> becomes important for example when real- time flows interacts with
>>> long lived TCP flows.  TCP have a built in behavior in its
>>> congestion control that strive to fill the buffer, thus all flows
>>> sharing the buffer experienced the delay build up.
>>>=20
>>> A common but quite poor queue handling mechanism is tail-drop,
>>> i.e. only drop packets when the incoming packet doesn't fit in the
>>> queue. Active Queue Management (AQM) is a term covering mechanism
>>> that tries to do something smarter buy actively managing the queue,
>>> for example by sending congestion signals earlier by droping
>>> packets earlier in the queue.  The behavior also affects the flow
>>> interactions.  For example Random Early Drop (RED) selects which
>>> packet to drop randomly.  This to give flows that have more packets
>>> in the queue a higher probability to experience the packet loss
>>> (congestion signal). There is ongoing work to find suitable
>>> mechanisms to recommend for implementation and reduce the use of
>>> tail-drop.
>>>=20
>>> 3.4.3.  Quality of Service
>>>=20
>>> Using best effort Internet one have no guarantees for the paths
>>> properties.  Quality of Service (QoS) mechanism are intended to
>>> provide the possibility to bound the path properties.  Where
>>> Diffserv [RFC2475] markings effects the queuing and forwarding
>>> behaviors of routers, the mechanism provides only statistical
>>> guarantees and care in how much marked packets of different types
>>> that are entering the network.  Flow based QoS like IntServ
>>> [RFC1663] has the potential for stricter guarantees as the
>>> properties are agreed on by each hop on the path.
>>=20
>> I think this is useful. It highlights that there are issues to
>> consider, which is the goal.
>=20
>=20
> Good, I was uncertain if this text said the right things.
>=20
>>=20
>>>> - The discussion of SST and MST in Section 5.1.5 seems to
>>>> confuse sessions with SSRCs. Isn=92t it more important whether the
>>>> layers are sent within a single RTP media stream, or as separate
>>>> RTP media streams with different SSRC values? Whether the
>>>> different SSRCs are sent in one RTP session of many is then a
>>>> separate issue, depending on how easily one wants to be able to
>>>> separate the flows for QoS purposes.
>>>=20
>>> The issue is that the SVC payload RFC is a bit confused on this
>>> matter. I agree that there are two separate important distinctions
>>> here. First if the layers are sent in one RTP packet stream or used
>>> multiple ones. In the second case one gets the question of how one
>>> distributes that over RTP sessions and underlying transports.
>>> Especially as some that implements SVC MST clearly uses a single
>>> RTP session and are after the possibility on a SSRC level to
>>> control transmission of layers in RTP middleboxes.
>>>=20
>>> I have tried to address this in the text, please review.
>>>=20
>>> 5.1.5.  Media Scalability
>>>=20
>>> Some codecs support various types of media scalability, i.e. some
>>> data of a media stream may be removed to adapt the media's
>>> properties, such as bitrate and quality.  The adaptation may be
>>> applied in the following dimensions of the media:
>>>=20
>>> Temporal:  For video codecs it is possible to adapt the frame
>>> rate, e.g. for H.264 [RFC6184].
>>>=20
>>> Spatial:  Video codecs supporting scalability may adapt the
>>> resolution, e.g. in SVC [RFC6190].
>>>=20
>>> Quality:  The quality of the media stream may be scaled by
>>> adapting the accuracy of the coding process, as, e.g.  possible
>>> with Signal to Noise Ratio (SNR) fidelity scalability of SVC
>>> [RFC6190].
>>>=20
>>> At the time of writing this document, codecs that support
>>> scalability have a bit of revival.  It has been realized that
>>> getting the required functionality for supporting the features of
>>> the media stream into the RTP framework is quite challenging.  One
>>> of the recent examples for layered and scalable codecs is Scalable
>>> Video Coding [RFC6190] (SVC).
>>>=20
>>> SVC is a good example for a payload format supporting media
>>> scalability features, which have been in its basic form already
>>> included in RTP.  A layered codec supports the dropping of data
>>> parts of a media stream, i.e. RTP packets may be not transmitted
>>> or forwarded to a client in order to adapt the media stream rate as
>>> well as the media stream quality, while still providing a decodable
>>> subset of the media stream to a client.  One example for using the
>>> scalability feature may be an RTP Mixer (Multipoint Control Unit)
>>> which controls the rate and quality sent out to participants in a
>>> communication based on dropping RTP packets or removing part of
>>> the payload.  Another example may be an transport channel which
>>> allows for differentiation in Quality of Service (QoS) parameters
>>> based on RTP sessions in a multicast session.  In such a case, the
>>> more important packets of the scalable media stream (base layer)
>>> may get better QoS parameters, then the less important packets
>>> (enhancement layer) in order to provide some kind of graceful
>>> degradation.  The scalability features required for allowing an
>>> adaptive transport as described in the two examples above are based
>>> on RTP multiplexing in order to identify the packets to be dropped
>>> or transmitted/forwarded. The multiplexing features defined for
>>> Scalable Video Coding [RFC6190] are:
>>>=20
>>> single session transmission (SST), where all media layers of the
>>> media are transported as single source (SSRC) in a single RTP
>>> session; as well as
>>>=20
>>> multi session transmission (MST), which should more accurately be
>>> called multi stream transmission, where different media layers or a
>>> set of media layers are transported in different RTP streams, i.e.
>>> using multiple sources (SSRCs).
>>>=20
>>> In the first case (SST), additional in-band as well as out-of-band
>>> signaling is required in order to allow identification of packets
>>> belonging to a specific media layer.  Furthermore, an adaptation
>>> of the media stream requires dropping of specific packets in order
>>> to provide the client with a compliant media stream.  In case of
>>> using encryption, it is typically required for an adapting network
>>> device to be in the security context to allow packet dropping and
>>> providing an intact RTP session to the client.  This typically
>>> requires the network device to be an RTP mixer.
>>>=20
>>> In general having a media unaware network device dropping
>>> excessive packets will be more problematic than having a Media
>>> Aware Network Entity (MANE).  First is the need to understand the
>>> media format and know which ADUs or payloads that belongs to the
>>> layers, that no other layer will be dependent on after the
>>> dropping.  Secondly, if the MANE can work as RTP mixer or
>>> translator it can rewrite the RTP and RTCP in such a way that the
>>> receiver will not suspect non-intentional RTP packet losses needing
>>> repair actions.  This as the receiver can't determine if a lost
>>> packet was an important base layer packet or one of the less
>>> important extension layers.
>>>=20
>>> In the second case (MST), the RTP packet streams can be sent using
>>> a single or multiple RTP sessions, and thus transport flows, e.g.
>>> on differnt multicast groups.  Transmitting the streams in
>>> different RTP sessions then the out-of-band signaling typically
>>> provides enough information to identify the media layers and its
>>> properties.  The decision for dropping packets is based on the
>>> Network Address which identifies the RTP session to be dropped.  In
>>> order to allow correct data provision to a decoder after reception
>>> from different sessions, data re-alignment mechanisms are described
>>> for Scalable Video Coding [RFC6190].  A more generic one is also
>>> described in Rapid Sync for RTP flows [RFC6051], which is purely
>>> based on existing RTP
>>>=20
>>> mechanisms, i.e. the NTP timestamp, for inter-session
>>> synchronization.  Another signaling feature is the generic
>>> indication of dependencies of RTP sessions in SDP, as defined in
>>> the Media Decoding Dependency Grouping in SDP [RFC5583].
>>>=20
>>> Using MST within a single RTP session is also possible and allow
>>> stream level handling instead of looking deeper into the packets by
>>> a MANE.  However, transport flow level properties will be the same
>>> unless packet based mechanisms like DiffServ is used.
>>>=20
>>> When QoS settings, e.g. DiffServ markings, are used to ensure that
>>> the extension layers are dropped prior the base layer the
>>> receiving end-point has the benefit in MST to know which layer or
>>> set of layers the missing packets belong as it will be bound to
>>> different RTP sessions.  Thus explicitly indicating the importance
>>> of the loss.
>>=20
>> I think this is okay, but it really needs an implementer of the SVC
>> format to comment too.
>=20
> Yes, that would be desirable.
>=20
>>=20
>>>> - In Section 5.1.6, avoiding wrap-around within the RTCP
>>>> reporting timeout interval is also important.
>>>=20
>>> If I understand this correctly, this is a minor issue with the RTCP
>>> SR and RR packet types as they all uses 32-bit counters. However,
>>> other packet types may have issues if they not use extended
>>> sequence number counters.
>>>=20
>>> This might be more relevant to considerations for configuring RTCP
>>> bandwidth and design of RTCP extensions, rather than being an RTP
>>> Payload issue. There exists a configuration goal to ensure that
>>> RTCP reporting doesn't wrap its fields within multiple reporting
>>> intervals.
>>>=20
>>> I propose that the following paragraph is added in this section:
>>>=20
>>> RTCP is also affected by high packet rates. For RTCP mechanism that
>>> do not use extended counters there is significant risk that they
>>> wrap multiple times between RTCP reporting or feedback, thus
>>> producing uncertainty which packet(s) that are referenced. The
>>> payload designer can't effect the RTCP packet formats used and
>>> their design, but can note this considerations when configuring
>>> RTCP bandwidth and reporting intervals to avoid to wrapping
>>> issues.
>>=20
>> That's fine, although I was actually suggesting something simpler,
>> which is to change "As rule of thumb, it must not be possible to wrap
>> the sequence number space in less than 2 minutes (TCP maximum segment
>> lifetime)" to "...must not be possible to wrap the sequence number
>> space in less than an RTCP reporting interval".
>=20
> That is difficult for someone that is new to quantify into meaningful
> values. Especially as we have reporting intervals between milliseconds
> (a scaled RTCP interval for the gigabit raw video) to many seconds for
> large groups with small amounts of receiver reporting bandwidths.
>=20
> In addition I don't think one reporting interval is really sufficient.
> In absence of RTCP packet loss it provides the bare minimum to detect =
a
> wrap in RTCP. But one really need to take the maximum interval between
> two RTCP packets into account and considering loss I think 3-4 =
intervals
> are needed to have reasonable saftey.
>=20
> The motivation for using TCP MSL was the argument that TCP MSL is
> selected as the time interval no emitted packet would still be =
possible
> to arrive after. That is true also for RTP packets and thus a valid
> considerations providing sufficient margins. It also covers almost all
> RTCP usages that I can think of.
>=20
> I have changed the text to now say:
>=20
> Some media codecs require high packet rates, and in these cases the =
RTP
> sequence number wraps too quickly. As rule of thumb, it must not be
> possible to wrap the sequence number space within at least three RTCP
> reporting intervals. As the reporting interval can vary widely due to
> configuration and session properties, and also must take into account
> the randomization of the interval, one can use the TCP maximum segment
> lifetime, which is 2 minutes in ones calculations.
>=20
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20



--=20
Colin Perkins
http://csperkins.org/




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Date: Fri, 27 Sep 2013 02:23:38 +0900
From: Noboru Harada <harada.noboru@lab.ntt.co.jp>
To: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
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Subject: Re: [payload] Toward WGLC on G.711.0 RTP Payload Spec
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Michael,

The document has been stable quite a while.
I support your proposal to move this draft to the next step.

Best Regards,
Noboru

> All,
> 
> The payload WG has a milestone to submit the RTP Payload Format for G.711.0 for proposed standard in December 2013 (http://datatracker.ietf.org/wg/payload/charter/ ).
> 
> The current working group draft is draft-ietf-payload-g7110-00 ( http://datatracker.ietf.org/doc/draft-ietf-payload-g7110/ ).
> 
> Although this working group document is a "00" draft, it spent much time and prior re-writes as an individual contribution draft known as draft-ramalho-payload-g7110-0X.
> 
> I have not received any requests to change this document since the conversion of the individual contribution draft to the working group draft (which occurred in June 2013).
> 
> As I would like to ask the Payload chairs to last call this document at the Vancouver IETF, please provide any input you have on this document to this list (including if you think it is ready as it is).
> 
> Thanks in advance,
> 
> Michael A. Ramalho

--------------------------------------
Noboru Harada
NTT Communication Science Laboratories
Tel: +81 46 240 3676
FAX: +81 46 240 3145
E-mail: harada.noboru@lab.ntt.co.jp
--------------------------------------


From stewe@stewe.org  Thu Sep 26 11:08:37 2013
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From: Stephan Wenger <stewe@stewe.org>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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Subject: Re: [payload] WGLC for draft-ietf-payload-rtp-howto-05
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Hi Magnus, group,

A few random comments.  None of them are showstoppers.  Yes, I should have
contributed those earlier.  Sorry.

Perhaps add a section about "writing style".  Aspects here:
(1) contrary to most other SDOs, the IETF has a history of spelling out
not only WHAT an implementation is to do, but also WHY.  As a lot of RTP
payload formats are written by media guys (who's home SDO is not
necessarily the IETF), mentioning this could help overcoming a culture
shock.
(2) OTOH, many/most implementers of RTP payload specs are the same people
that also implement the network centric parts of the media codec--at least
for well designed media codecs, systems, and payload formats.  Therefore,
it is also good to reflect the style and terminology of the media codings
pec in the payload format--even if that makes the payload spec a little
bit less accessible for folks working only in the IETF.  This is one
reason why I don't feel bad about the often lamented "density" of the
H.264 payload specs.  No compromises should be made on core IETF habits
like the use of 2119 keywords, though.
(3) It's also worth spelling out that the IETF has a historic (albeit
arguably not current) preference for lean and mean specs, even at the cost
of some functionality.  Most media codec SDOs operate under an assumption
that if there were a use-case/requirement (however exotic/weak it may be,
and I note that use/case and requirements discussions have quite often
nothing to do with the real world) and no competing tool, a proposal goes
in, pleasing certain IPR departments and leading to patent bonuses and
career advancements, at the cost of a bloated document.  Some RTP payload
formats written predominantly by media codec people arguably are bloated
as well for the same reason.  (And yes, I take personal blame for some of
this.)  While I do not see that we can change the situation (you want to
have the media competence of the media coding people), I think we can at
least point it out--hopefully in somewhat more diplomatic language--so to
explain a very common culture clash between IETF regulars and payload
format writers.

In 3.1.1 I would mention prominently the early and personal IPR disclosure
obligations in the IETF, as these differ from other SDOs which specify
media codecs.  Simply pointing to the policy docs may be not enough.

Section 3 (and section 1).  I think a lot of this description is written
for someone who wants to write an RTP payload format for an existing media
codec (with a stable and essentially unchangeable spec).  While this is
probably still the most common scenario, there are a number of cases where
the transport aspects of the media coder and the media aspects of the
transport (RTP payload format) have been developed concurrently, generally
with very pleasing results for both specs.  H.264 is one prominent
example, but there are others.  Arguably, joint development of codec and
RTP payload format is becoming a trend.  I think that a 2013 generation
RTP payload how-to should acknowledge this situation.  One way to do that
would be to include, in section 3, subsections "read and understand the
media coding spec", and "influence the media coding spec".  If people
think this is a good idea, I'm willing to draft a few paragraphs.

I think the 9000 byte path MTU is generally not available in home
environments.  Instead, MTU sizes larger than the 1500 bytes (from
Ethernet) are AFAICT, available only in a few selected (large/medium)
enterprises and in the academic world.  If that's true, it should be
stated.  And, in home environments, the limiting factor is often not even
the ISP, but those old $50 router boxes (and their NAT implementations)
that infest the world.  Mentioning this, perhaps with an informative
reference to a recent study, would send a message: if your codec will be
used a lot in home environments, don't go above 1500 bytes.  Not this
year, and not next.

Sections 5.1.1 and 5.1.2: one key use case of both aggregation and
fragmentation is the effective packetization of pre-recorded content,
where one cannot change the size of ADUs without transcoding.   The key
use case of aggregation is the packetization overhead for small ADUs such
as parameter sets or SEI messages in H.264.

In a different email, you asked for review of 5.1.5.  There are more
qualified people on this list to comment, but here are a few points.
(1) citing H.264 RFC 6184 for temporal scalability is a bit questionable,
as 6184 doe snot include a single mechanism to support temporal
scalability (although H.264 does).  So if people would go to 6184 as the
payload format for base H.264 and read it, expecting ideas on how to deal
with temporal scalability, they would come up blank after digging through
101 pages of not exactly easy to read spec language.  I fear that it would
be best to point to SVC and 6190 for temporal scalability, just as it is
done for spatial and SNR.
(2) with respect to MST: the doc could be read that re-alignment for
H.264-SVC could be implemented through 6051.  This is not the case, as
decoding time and NAL unit order are mostly decoupled (not only in SVC,
but in all modern video codecs).
(3) Perhaps write something to the extent that MST was unpopular because
of the impracticality of multiple transport addresses, and that this
reason is slowly going away through IETF-acceptance of RTP mux techniques.

Section 6.2, "recent" VC-1.  You probably wanted to cite VP8?  Calling
VC-1 and its payload spec "recent" is a bit of  a stretch.

Perhaps add a "Do and don't" sections listing key mistakes.  In the don't
section, list the handling of the timestamp in RFCs 2038 and 2250.

Stephan
=20


On 9.9.2013 07:07 , "Magnus Westerlund" <magnus.westerlund@ericsson.com>
wrote:

>WG,
>
>As author I would really appreciate some reviews that the content are
>appropriate. Thus I really appreciate responses even of the type. "Read
>it, no issues spotted".
>
>Don't assume that I am without flaws in my writing ;-).
>
>Cheers
>
>Magnus
>
>On 2013-08-22 10:58, Ali C. Begen (abegen) wrote:
>> (As a WG co-chair)
>>=20
>> I am starting WGLC for the following draft (version 05).
>> https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-howto/
>>=20
>> This is an important document as it outlines how someone is supposed to
>>write the payload specs.
>>=20
>> If there are sections you don't understand or that are not clear
>>enough, etc. please post them to the list. We need to get this right to
>>minimize further headaches for future documents in our WG.
>>=20
>> The deadline for the WGLC is September 20th.
>>=20
>> -acbegen
>> _______________________________________________
>> payload mailing list
>> payload@ietf.org
>> https://www.ietf.org/mailman/listinfo/payload
>>=20
>>=20
>
>
>--=20
>
>Magnus Westerlund
>
>----------------------------------------------------------------------
>Multimedia Technologies, Ericsson Research EAB/TVM
>----------------------------------------------------------------------
>Ericsson AB                | Phone  +46 10 7148287
>F=E4r=F6gatan 6                | Mobile +46 73 0949079
>SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>----------------------------------------------------------------------
>
>_______________________________________________
>payload mailing list
>payload@ietf.org
>https://www.ietf.org/mailman/listinfo/payload


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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Michael Ramalho (mramalho)" <mramalho@cisco.com>, "payload@ietf.org" <payload@ietf.org>
Date: Thu, 26 Sep 2013 18:19:37 +0000
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Subject: Re: [payload] Toward WGLC on G.711.0 RTP Payload Spec
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--------=_MB914453BF-6028-41CB-A74D-D6AD7C635494
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I support going for LC.

Paul

------ Original Message ------
From: "Michael Ramalho (mramalho)" <mramalho@cisco.com>
To: "payload@ietf.org" <payload@ietf.org>
Sent: 9/25/2013 5:16:37 PM
Subject: [payload] Toward WGLC on G.711.0 RTP Payload Spec
>All,
>
>
>
>The payload WG has a milestone to submit the RTP Payload Format for=20
>G.711.0 for proposed standard in December 2013=20
>(http://datatracker.ietf.org/wg/payload/charter/ ).
>
>
>
>The current working group draft is draft-ietf-payload-g7110-00 (=20
>http://datatracker.ietf.org/doc/draft-ietf-payload-g7110/ ).
>
>
>
>Although this working group document is a =E2=80=9C00=E2=80=9D draft, it=
 spent much=20
>time and prior re-writes as an individual contribution draft known as=20
>draft-ramalho-payload-g7110-0X.
>
>
>
>I have not received any requests to change this document since the=20
>conversion of the individual contribution draft to the working group=20
>draft (which occurred in June 2013).
>
>
>
>As I would like to ask the Payload chairs to last call this document at=
=20
>the Vancouver IETF, please provide any input you have on this document=20
>to this list (including if you think it is ready as it is).
>
>
>
>Thanks in advance,
>
>
>
>Michael A. Ramalho
>
--------=_MB914453BF-6028-41CB-A74D-D6AD7C635494
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<BODY lang=3DEN-US scroll=3Dauto link=3Dblue vLink=3Dpurple class>
<DIV>I support going for LC.</DIV>
<DIV>&nbsp;</DIV>
<DIV>Paul</DIV>
<DIV>&nbsp;</DIV>
<DIV>------ Original Message ------</DIV>
<DIV>From: "Michael Ramalho (mramalho)" &lt;<A href=3D"mailto:mramalho@cisc=
o.com">mramalho@cisco.com</A>&gt;</DIV>
<DIV>To: "payload@ietf.org" &lt;<A href=3D"mailto:payload@ietf.org">payload=
@ietf.org</A>&gt;</DIV>
<DIV>Sent: 9/25/2013 5:16:37 PM</DIV>
<DIV>Subject: [payload] Toward WGLC on G.711.0 RTP Payload Spec</DIV>
<BLOCKQUOTE class=3Dcite cite=3DD21571530BF9644D9A443D6BD95B91031564BB1C@xm=
b-rcd-x12.cisco.com type=3D"cite">
<DIV id=3D62b8c6de54a1457a8156b5d390bda7f6>
<DIV class=3DWordSection1>
<P class=3DMsoNormal>All,<o:p></o:p></P>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<P class=3DMsoNormal>The payload WG has a milestone to submit the RTP Paylo=
ad Format for G.711.0 for proposed standard in December 2013 (<A href=3D=
"http://datatracker.ietf.org/wg/payload/charter/">http://datatracker.ietf.o=
rg/wg/payload/charter/</A> ).<o:p></o:p></P>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<P class=3DMsoNormal>The current working group draft is draft-ietf-payload-=
g7110-00 ( <A href=3D"http://datatracker.ietf.org/doc/draft-ietf-payload-g7=
110/">http://datatracker.ietf.org/doc/draft-ietf-payload-g7110/</A> ).<o:p>=
</o:p></P>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<P class=3DMsoNormal>Although this working group document is a =E2=80=9C=
00=E2=80=9D draft, it spent much time and prior re-writes as an individual=
 contribution draft known as draft-ramalho-payload-g7110-0X.<o:p></o:p></P>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<P class=3DMsoNormal>I have not received any requests to change this docume=
nt since the conversion of the individual contribution draft to the working=
 group draft (which occurred in June 2013).<o:p></o:p></P>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<P class=3DMsoNormal>As I would like to ask the Payload chairs to last call=
 this document at the Vancouver IETF, please provide any input you have =
on this document to this list (including if you think it is ready as it =
is).<o:p></o:p></P>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<P class=3DMsoNormal>Thanks in advance,<o:p></o:p></P>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<P class=3DMsoNormal>Michael A. Ramalho<o:p></o:p></P></DIV></DIV></BLOCKQU=
OTE></BODY></HTML>
--------=_MB914453BF-6028-41CB-A74D-D6AD7C635494--


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From: Stephan Wenger <stewe@stewe.org>
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Cc: Jonathan Lennox <jonathan@vidyo.com>, Jill Boyce <jill@vidyo.com>
Subject: [payload] FW: New Version Notification for draft-wenger-payload-h265-payload-header-extension-00.txt
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Hi all,

In Berlin's payload session presentation of the H.265 payload format
(graciously hosted by AVTcore), I mentioned that we want to include
support for temporal scalability in the draft H.265 payload format.
Attached a quick draft showing how this could be done--basically a (we
believe) improved variant of the PACSI NAL unit of RFC 6190.  There are
two alternative proposals in the draft, trading off alignment with design
principles of H.265 one one side and efficiency on the other.

Vidyo has IPR on this draft, which we will make available under terms like
this: https://datatracker.ietf.org/ipr/1622/.  I believe that the concepts
of this draft should be included in the H.265 payload format itself,
rather than being stand-alone.  Once that happens, or if the group decides
the draft to stay stand-alone, we will file the official IPR disclosure
against the then-current and appropriate document, so not to clutter the
database. =20

Comments welcome.

Thanks,
Stephan



On 9.26.2013 13:08 , "internet-drafts@ietf.org" <internet-drafts@ietf.org>
wrote:

>
>A new version of I-D,
>draft-wenger-payload-h265-payload-header-extension-00.txt
>has been successfully submitted by Stephan Wenger and posted to the
>IETF repository.
>
>Filename:	 draft-wenger-payload-h265-payload-header-extension
>Revision:	 00
>Title:		 H.265 Payload Header Extension and Temporal Scalability Support
>Creation date:	 2013-09-26
>Group:		 Individual Submission
>Number of pages: 10
>URL:            =20
>http://www.ietf.org/internet-drafts/draft-wenger-payload-h265-payload-head
>er-extension-00.txt
>Status:         =20
>http://datatracker.ietf.org/doc/draft-wenger-payload-h265-payload-header-e
>xtension
>Htmlized:       =20
>http://tools.ietf.org/html/draft-wenger-payload-h265-payload-header-extens
>ion-00
>
>
>Abstract:
>   Proposed are the allocation of a NAL unit type from the
>   "unspecified" set of NAL units of H.265 [HEVC] for the purpose of
>   payload header extensions, a generic syntax for the use of that
>   pseudo NAL unit, and one application for it: the signaling of
>   temporal scalability support information.  Two alternative designs
>   are included.  The content of this draft is intended for
>   incorporation into the H.265 RTP payload [h265-payload].
>
>
>
>                 =20
>       =20
>
>
>Please note that it may take a couple of minutes from the time of
>submission
>until the htmlized version and diff are available at tools.ietf.org.
>
>The IETF Secretariat
>


From harald@alvestrand.no  Fri Sep 27 03:01:13 2013
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From: Harald Alvestrand <harald@alvestrand.no>
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References: <8C4E0C2409735E4FBC22D754A238F94D02F74C4C@APTSBS.apt.local> <52385092.5000000@alvestrand.no> <8C4E0C2409735E4FBC22D754A238F94D02F74FD5@APTSBS.apt.local>
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Cc: Foerster@worldcastsystems.com, payload@ietf.org
Subject: Re: [payload] I-D Action: draft-ietf-payload-rtp-aptx-01.txt
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On 09/24/2013 07:06 PM, John Lindsay wrote:
>
> Hi Harald
>
> Thanks for reviewing this and the points raised.
>
> To increase clarity we can change the statement in section 3
>
> From
>
> Standard apt-X and Enhanced apt-X are proprietary audio coding
>
> algorithms, licensed by CSR plc and widely deployed in a variety of
>
>    audio processing equipment.
>
> To
>
> Standard apt-X and Enhanced apt-X are proprietary audio coding
>
> algorithms, which can be licensed from CSR plc and are widely
>
>   deployed in a variety of audio processing equipment.
>

That looks good to me. It was a nit, but worth fixing!
>
> Section 3 of the doc provides an overview of the codecs operation. The 
> aim of the draft is how to packetize Standard or Enhanced apt-X over 
> RTP not to explain the operation of the codec. Further details are 
> likely to be proprietary to CSR and interested parties would need to 
> talk to CSR directly about obtaining licenses and setting up NDA's etc 
> for further information.
>
> Regards
>
> John
>
> *From:*payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] *On 
> Behalf Of *Harald Alvestrand
> *Sent:* 17 September 2013 13:53
> *To:* payload@ietf.org
> *Subject:* Re: [payload] I-D Action: draft-ietf-payload-rtp-aptx-01.txt
>
> Brief scan, one question:
>
> Is it possible to offer a stable reference for what "Standard apt-x" 
> and "Enhanced apt-x" means?
>
> The closest I can get is "licensed by CSR plc", which presumably means 
> that "it means what CSR plc thinks it means", but it would be great to 
> have that stated explicitly.
>
> Something like:
>
> The codecs "apt-x" and "Extended apt-x" are defined by CSR plc, an UK 
> company.
>
> Section 3 of the draft *almost* says that, but isn't completely 
> unambiguous ("licensed by" can mean both "CSR is a licensor" and "CSR 
> is a licensee, someone else defines it").
>
> I *think* this is a nit.
>
>
> On 09/10/2013 03:50 PM, John Lindsay wrote:
>
>     Hi
>
>     A new draft has been uploaded to the IETF Payload workgroup at
>     https://datatracker.ietf.org/wg/payload/.
>
>     The new document makes a few minor changes for IDNITS checking and
>     also takes account of the comments made by Peter Stevens last week.
>
>     A summary of the changes are as follows.
>
>     Page 1, IETF Copyright notice updated to 2013.
>
>     Page 10, updated formatting for IDNITS check.
>
>     Page 14, Section 6 updated to reflect RFC 6838 which supersedes
>     RFC 4288.
>
>     Page 16, Section 6.2.1 type corrected An -- A as per Peter Stevens
>     comments.
>
>     Page 18, Section 7 now reference 6.1 rather than 7.1, again thanks
>     to Peters proof reading.
>
>     Page 21, new RFC6338 referenced.
>
>     Thanks to everyone who has read and commented on the draft.
>
>     Regards
>
>     John
>
>
>
>
>     _______________________________________________
>
>     payload mailing list
>
>     payload@ietf.org  <mailto:payload@ietf.org>
>
>     https://www.ietf.org/mailman/listinfo/payload
>


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    <div class="moz-cite-prefix">On 09/24/2013 07:06 PM, John Lindsay
      wrote:<br>
    </div>
    <blockquote
      cite="mid:8C4E0C2409735E4FBC22D754A238F94D02F74FD5@APTSBS.apt.local"
      type="cite">
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      <div class="WordSection1">
        <p class="MsoPlainText"><span style="color:windowtext">Hi Harald<o:p></o:p></span></p>
        <p class="MsoPlainText"><span style="color:windowtext"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span style="color:windowtext">Thanks
            for reviewing this and the points raised.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext">To increase
            clarity we can change the statement in section 3<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext">From<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext">Standard
            apt-X and Enhanced apt-X are proprietary audio coding<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext">&nbsp;&nbsp;
            algorithms, licensed by CSR plc and widely deployed in a
            variety of<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext">&nbsp;&nbsp; audio
            processing equipment.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext">To<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext">Standard
            apt-X and Enhanced apt-X are proprietary audio coding<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext">&nbsp;&nbsp;
            algorithms, which can be licensed from CSR plc and are
            widely <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext">&nbsp;&nbsp;deployed
            in a variety of audio processing equipment.</span></p>
      </div>
    </blockquote>
    <br>
    That looks good to me. It was a nit, but worth fixing!<br>
    <blockquote
      cite="mid:8C4E0C2409735E4FBC22D754A238F94D02F74FD5@APTSBS.apt.local"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span style="color:windowtext"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoPlainText"><span style="color:windowtext">Section 3
            of the doc provides an overview of the codecs operation. The
            aim of the draft is how to packetize Standard or Enhanced
            apt-X over RTP not to explain the operation of the codec.
            Further details are likely to be proprietary to CSR and
            interested parties would need to talk to CSR directly about
            obtaining licenses and setting up NDA's etc for further
            information. <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext">Regards<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:windowtext">John<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <div>
          <div style="border:none;border-top:solid #B5C4DF
            1.0pt;padding:3.0pt 0cm 0cm 0cm">
            <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext;mso-fareast-language:EN-GB"
                  lang="EN-US">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext;mso-fareast-language:EN-GB"
                lang="EN-US"> <a class="moz-txt-link-abbreviated" href="mailto:payload-bounces@ietf.org">payload-bounces@ietf.org</a>
                [<a class="moz-txt-link-freetext" href="mailto:payload-bounces@ietf.org">mailto:payload-bounces@ietf.org</a>] <b>On Behalf Of </b>Harald
                Alvestrand<br>
                <b>Sent:</b> 17 September 2013 13:53<br>
                <b>To:</b> <a class="moz-txt-link-abbreviated" href="mailto:payload@ietf.org">payload@ietf.org</a><br>
                <b>Subject:</b> Re: [payload] I-D Action:
                draft-ietf-payload-rtp-aptx-01.txt<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <div>
          <p class="MsoNormal">Brief scan, one question:<br>
            <br>
            Is it possible to offer a stable reference for what
            "Standard apt-x" and "Enhanced apt-x" means?<br>
            <br>
            The closest I can get is "licensed by CSR plc", which
            presumably means that "it means what CSR plc thinks it
            means", but it would be great to have that stated
            explicitly.<br>
            <br>
            Something like:<br>
            <br>
            The codecs "apt-x" and "Extended apt-x" are defined by CSR
            plc, an UK company.<br>
            <br>
            Section 3 of the draft *almost* says that, but isn't
            completely unambiguous ("licensed by" can mean both "CSR is
            a licensor" and "CSR is a licensee, someone else defines
            it").<br>
            <br>
            I *think* this is a nit.<br>
            <br>
            <br>
            On 09/10/2013 03:50 PM, John Lindsay wrote:<o:p></o:p></p>
        </div>
        <blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
          <p class="MsoNormal"><span style="color:#1F497D">Hi</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">A new draft
              has been uploaded to the IETF Payload workgroup at <a
                moz-do-not-send="true"
                href="https://datatracker.ietf.org/wg/payload/">https://datatracker.ietf.org/wg/payload/</a>.</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">The new
              document makes a few minor changes for IDNITS checking and
              also takes account of the comments made by Peter Stevens
              last week.</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">A summary of
              the changes are as follows.</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">Page 1, IETF
              Copyright notice updated to 2013.</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">Page 10,
              updated formatting for IDNITS check.</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">Page 14,
              Section 6 updated to reflect RFC 6838 which supersedes RFC
              4288.</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">Page 16,
              Section 6.2.1 type corrected An &#8211; A as per Peter Stevens
              comments.</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">Page 18,
              Section 7 now reference 6.1 rather than 7.1, again thanks
              to Peters proof reading.</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">Page 21, new
              RFC6338 referenced.</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">Thanks to
              everyone who has read and commented on the draft.</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">Regards</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">John</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span
              style="font-size:12.0pt;font-family:&quot;Times New
              Roman&quot;,&quot;serif&quot;;mso-fareast-language:EN-GB"><br>
              <br>
              <br>
              <o:p></o:p></span></p>
          <pre>_______________________________________________<o:p></o:p></pre>
          <pre>payload mailing list<o:p></o:p></pre>
          <pre><a moz-do-not-send="true" href="mailto:payload@ietf.org">payload@ietf.org</a><o:p></o:p></pre>
          <pre><a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/payload">https://www.ietf.org/mailman/listinfo/payload</a><o:p></o:p></pre>
        </blockquote>
        <p class="MsoNormal"><span
            style="font-size:12.0pt;font-family:&quot;Times New
            Roman&quot;,&quot;serif&quot;;mso-fareast-language:EN-GB"><o:p>&nbsp;</o:p></span></p>
      </div>
    </blockquote>
    <br>
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