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From: Richard Barnes <rlb@ipv.sx>
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Cc: draft-ietf-avt-rtp-isac@tools.ietf.org
Subject: [payload] AD Evaluation: draft-ietf-avt-rtp-isac-04
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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

--e89a8f839d516b1ef204d94fa35c
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Overall, this document looks in pretty good shape. =A0A couple of comments =
I would like to get resolved before IETF LC:<div><br></div><div>Section 3.1=
., &quot;<span style=3D"line-height:15.59375px">Consult source code for det=
ails</span>&quot;</div>
<div>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 Alte=
rnatively, if the codec really does expect a combined BEI/FL field, you cou=
ld specify such a field here, opaque to the RTP layer. =A0However, you woul=
d at least need to say how the recipient knows where this field starts and =
stops.</div>
<div><br></div><div>Section 3.2., &quot;<span style=3D"font-size:13px;line-=
height:1.2em">The length of the encoded data is variable and depends on=A0<=
/span><span style=3D"font-size:13px;line-height:1.2em">the signal character=
istics and the target bit rate.</span>&quot;</div>
<div>However, there&#39;s nothing in this section that tells a recipient ho=
w to determine what this length is. =A0</div><div><br></div><div>Section 3.=
3., &quot;...=A0<span style=3D"font-size:13px;line-height:1.2em">verifying =
the CRC checksum ...</span>&quot;</div>
<div>Please specify which CRC function is to be applied, and how the CRC va=
lue field is formatted.</div><div><br></div><div>Section 3.3., &quot;<span =
style=3D"font-size:13px;line-height:1.2em">If this value would exceed 255 a=
t encoding...</span>&quot;</div>
<div>It sounds like the LEN field is a single octet unsigned integer. =A0Pl=
ease state that explicitly.=A0</div><div><br></div><div>Section 9.2. Inform=
ative References.</div><div>Better path to source code would be &quot;webrt=
c/<span style=3D"font-size:13px;line-height:1.2em">modules/audio_coding/cod=
ecs/isac&quot;</span></div>
<div><span style=3D"font-size:13px;line-height:1.2em"><br></span></div><div=
><span style=3D"font-size:13px;line-height:1.2em">Thanks,</span></div><div>=
<span style=3D"font-size:13px;line-height:1.2em">--Richard</span></div>

--e89a8f839d516b1ef204d94fa35c--

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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Richard Barnes'" <rlb@ipv.sx>, <payload@ietf.org>
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Cc: draft-ietf-avt-rtp-isac@tools.ietf.org
Subject: Re: [payload] AD Evaluation: draft-ietf-avt-rtp-isac-04
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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


------=_NextPart_000_062F_01CE2F1C.58E38AA0
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<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
xmlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" =
xmlns=3D"http://www.w3.org/TR/REC-html40"><head><META =
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<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Hi Richard,<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";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.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Still in this case more information can be =
supplied.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>As for the comments the authors should address =
them<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Roni Even<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] <b>On Behalf =
Of </b>Richard Barnes<br><b>Sent:</b> 01 April, 2013 8:06 =
PM<br><b>To:</b> payload@ietf.org<br><b>Cc:</b> =
draft-ietf-avt-rtp-isac@tools.ietf.org<br><b>Subject:</b> [payload] AD =
Evaluation: draft-ietf-avt-rtp-isac-04<o:p></o:p></span></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Overall, =
this document looks in pretty good shape. &nbsp;A couple of comments I =
would like to get resolved before IETF LC:<o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>Section 3.1., &quot;Consult source code for =
details&quot;<o:p></o:p></p></div><div><p class=3DMsoNormal>If the =
details are only specified in the source code, then the source needs to =
be a normative reference. &nbsp;And I would prefer to avoid that =
&nbsp;:) &nbsp;Better to specify here how the BEI and FL fields are =
encoded. &nbsp; Alternatively, if the codec really does expect a =
combined BEI/FL field, you could specify such a field here, opaque to =
the RTP layer. &nbsp;However, you would at least need to say how the =
recipient knows where this field starts and =
stops.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>Section 3.2., &quot;<span =
style=3D'font-size:10.0pt'>The length of the encoded data is variable =
and depends on&nbsp;the signal characteristics and the target bit =
rate.</span>&quot;<o:p></o:p></p></div><div><p =
class=3DMsoNormal>However, there's nothing in this section that tells a =
recipient how to determine what this length is. =
&nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>Section 3.3., &quot;...&nbsp;<span =
style=3D'font-size:10.0pt'>verifying the CRC checksum =
...</span>&quot;<o:p></o:p></p></div><div><p class=3DMsoNormal>Please =
specify which CRC function is to be applied, and how the CRC value field =
is formatted.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>Section 3.3., &quot;<span =
style=3D'font-size:10.0pt'>If this value would exceed 255 at =
encoding...</span>&quot;<o:p></o:p></p></div><div><p =
class=3DMsoNormal>It sounds like the LEN field is a single octet =
unsigned integer. &nbsp;Please state that =
explicitly.&nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>Section 9.2. Informative =
References.<o:p></o:p></p></div><div><p class=3DMsoNormal>Better path to =
source code would be &quot;webrtc/<span =
style=3D'font-size:10.0pt'>modules/audio_coding/codecs/isac&quot;</span><=
o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:10.0pt'>Thanks,</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:10.0pt'>--Richard</span><o:p></o:p></p></div></div></b=
ody></html>
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From: Paul Coverdale <coverdale@sympatico.ca>
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Subject: [payload] FW: Request for G.711 RTP Payload Format to be adopted as a PAYLOAD work item
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I also support this as a new work item the Payload WG.

 

Paul Coverdale

From: Michael Ramalho (mramalho) 
Sent: Thursday, March 28, 2013 2:04 PM
To: payload@ietf.org
Subject: Request for G.711 RTP Payload Format to be adopted as a PAYLOAD
work item

 

Ali Begen, Roni Even (PAYLOAD WG Chairs) and Payload WG,

 

This email requests that the RTP payload format for ITU-T Rec. G.711.0 be a
formal work item for the PAYLOAD WG.

 

The G.711.0 RTP payload draft (draft-ramalho-payload-g7110) has been
discussed at past IETFs and is listed as a "related document" for the
PAYLOAD working group (see: http://datatracker.ietf.org/wg/payload/).

 

The history/chronology of G.711.0 work in the IETF Payload Working Group
follows after my signature for those interested.

 

If adopted, a milestone submit date of December 2013 is suggested.

 

Regards,

 

Michael Ramalho

 

>>Chronology of G.711.0 Work in the IETF PAYLOAD WG<<

 

>> IETF 81 Quebec, Canada, July 24-29, 2011

 

I presented the G.711.0 "Compression Segments" draft at the PAYLOAD meeting
held within the AVTEXT timeslot. This draft was a combination of a
"G.711-like" RTP payload specification and text describing how G.711.0 could
be used as a lossless compression mechanism for "G.711.0 segments" of an
end-to-end G.711 session.

 

In discussion that ensued at that meeting it was decided that this draft
should be split into two drafts:

 

Draft 1: "G.711.0 RTP Payload Format" draft (targeted as standards track
RFC), and

Draft 2: "G.711.0 Use Cases" draft (targeted as informational RFC).

 

Soon after IETF 81 this was accomplished by the following drafts:
draft-ramalho-payload-g7110-01.txt (G.711.0 RTP payload format) and
draft-ramalho-g7110-segments-00.txt (G.711.0 "use cases").

 

As the G.711.0 payload draft is nearly identical to the G.711 RTP payload
specification, there was little debate on the mailing lists about it outside
of the storage mode definition (the G.711 RTP payload specification did not
have a storage mode defined).

 

>>IETF 83 Paris, France, March 27, 2012

 

I presented: draft-ramalho-g7110-segments-00 during the PAYLOAD segment
inside of the AVTEXT meeting slot
(http://www.ietf.org/proceedings/83/slides/slides-83-avtext-6.pdf ).

 

I did not present on the G.711.0 payload format draft, as the only issue
being debated on the list for this draft was the storage mode - and the
storage mode agreements were converging to a solution on the mailing list.

 

I renewed the G.711.0 RTP payload specification with a new version
(draft-ramalho-payload-g7110-02.txt) which captured the email discussions on
the storage mode issues.

 

I let the use case draft (draft-ramalho-g7110-segments-00) expire due to
lack of interest on the mailing list. I believe interest will revive when
the payload specification is complete.

 

>>Summary: This email requests the G.711.0 RTP Payload Format be a formal
working group item with an associated milestone (December 2013 suggested).


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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:#1F497D=
'>I also support this as a new work item the Payload =
WG.<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'>Paul =
Coverdale<o:p></o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
Michael Ramalho (mramalho) <br><b>Sent:</b> Thursday, March 28, 2013 =
2:04 PM<br><b>To:</b> <a =
href=3D"mailto:payload@ietf.org">payload@ietf.org</a><br><b>Subject:</b> =
Request for G.711 RTP Payload Format to be adopted as a PAYLOAD work =
item<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Ali Begen, =
Roni Even (PAYLOAD WG Chairs) and Payload WG,<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>This email =
requests that the RTP payload format for ITU-T Rec. G.711.0 be a formal =
work item for the PAYLOAD WG.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>The G.711.0 =
RTP payload draft (draft-ramalho-payload-g7110) has been discussed at =
past IETFs and is listed as a &#8220;related document&#8221; for the =
PAYLOAD working group (see: <a =
href=3D"http://datatracker.ietf.org/wg/payload/">http://datatracker.ietf.=
org/wg/payload/</a>).<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>The =
history/chronology of G.711.0 work in the IETF Payload Working Group =
follows after my signature for those interested.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>If adopted, =
a milestone submit date of December 2013 is suggested.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Regards,<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Michael =
Ramalho<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>&gt;&gt;Chronology of G.711.0 Work in the IETF PAYLOAD =
WG&lt;&lt;<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>&gt;&gt; IETF 81 Quebec, Canada, July 24-29, =
2011<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>I presented the G.711.0 &#8220;Compression =
Segments&#8221; draft at the PAYLOAD meeting held within the AVTEXT =
timeslot. This draft was a combination of a &#8220;G.711-like&#8221; RTP =
payload specification and text describing how G.711.0 could be used as a =
lossless compression mechanism for &#8220;G.711.0 segments&#8221; of an =
end-to-end G.711 session.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>In =
discussion that ensued at that meeting it was decided that this draft =
should be split into two drafts:<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Draft 1: =
&#8220;G.711.0 RTP Payload Format&#8221; draft (targeted as standards =
track RFC), and<o:p></o:p></p><p class=3DMsoNormal>Draft 2: =
&#8220;G.711.0 Use Cases&#8221; draft (targeted as informational =
RFC).<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Soon after IETF 81 this was accomplished by the =
following drafts: draft-ramalho-payload-g7110-01.txt (G.711.0 RTP =
payload format) and draft-ramalho-g7110-segments-00.txt (G.711.0 =
&#8220;use cases&#8221;).<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>As the =
G.711.0 payload draft is nearly identical to the G.711 RTP payload =
specification, there was little debate on the mailing lists about it =
outside of the storage mode definition (the G.711 RTP payload =
specification did not have a storage mode defined).<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>&gt;&gt;IETF =
83 Paris, France, March 27, 2012<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>I presented: =
draft-ramalho-g7110-segments-00 during the PAYLOAD segment inside of the =
AVTEXT meeting slot (<a =
href=3D"http://www.ietf.org/proceedings/83/slides/slides-83-avtext-6.pdf"=
>http://www.ietf.org/proceedings/83/slides/slides-83-avtext-6.pdf</a> =
).<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>I did not present on the G.711.0 payload format draft, =
as the only issue being debated on the list for this draft was the =
storage mode &#8211; and the storage mode agreements were converging to =
a solution on the mailing list.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>I renewed =
the G.711.0 RTP payload specification with a new version =
(draft-ramalho-payload-g7110-02.txt) which captured the email =
discussions on the storage mode issues.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>I let the =
use case draft (draft-ramalho-g7110-segments-00) expire due to lack of =
interest on the mailing list. I believe interest will revive when the =
payload specification is complete.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>&gt;&gt;Summary: This email requests the G.711.0 RTP =
Payload Format be a formal working group item with an associated =
milestone (December 2013 suggested).<o:p></o:p></p></div></body></html>
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Michael Ramalho (mramalho)" <mramalho@cisco.com>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: [payload] Request for G.711 RTP Payload Format to be adopted as a PAYLOAD work item
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Date: Mon, 1 Apr 2013 14:46:35 -0400
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From: Richard Barnes <rlb@ipv.sx>
To: Roni Even <ron.even.tlv@gmail.com>
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Cc: draft-ietf-avt-rtp-isac@tools.ietf.org, payload@ietf.org
Subject: Re: [payload] AD Evaluation: draft-ietf-avt-rtp-isac-04
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--e89a8fb1ef1651d27a04d9510af2
Content-Type: text/plain; charset=ISO-8859-1

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****
>

--e89a8fb1ef1651d27a04d9510af2
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

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 par=
se out the fields I would feed into the black box of the codec. =A0They des=
cribe the fields, but it seems to me that they need more detail in order fo=
r people to know how to parse them out of the payload.<div>
<br></div><div>--Richard</div><div><br><br><div class=3D"gmail_quote">On Mo=
n, Apr 1, 2013 at 2:03 PM, Roni Even <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:ron.even.tlv@gmail.com" 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;border-left:1p=
x #ccc solid;padding-left:1ex"><div lang=3D"EN-US" link=3D"blue" vlink=3D"p=
urple"><div><p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi Richard,<u=
></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">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 t=
o 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.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Still in this case more i=
nformation can be supplied.<u></u><u></u></span></p><p class=3D"MsoNormal">=
<span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-=
serif&quot;;color:#1f497d"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">As for the comments the a=
uthors should address them<u></u><u></u></span></p><p class=3D"MsoNormal"><=
span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-s=
erif&quot;;color:#1f497d"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Roni Even<u></u><u></u></=
span></p><p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family=
:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u=
></span></p>
<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;"> <a href=
=3D"mailto:payload-bounces@ietf.org" target=3D"_blank">payload-bounces@ietf=
.org</a> [mailto:<a href=3D"mailto:payload-bounces@ietf.org" target=3D"_bla=
nk">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.org" 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.ietf.org</a><br>
<b>Subject:</b> [payload] AD Evaluation: draft-ietf-avt-rtp-isac-04<u></u><=
u></u></span></p><div><div class=3D"h5"><p class=3D"MsoNormal"><u></u>=A0<u=
></u></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=
:<u></u><u></u></p>
<div><p class=3D"MsoNormal"><u></u>=A0<u></u></p></div><div><p class=3D"Mso=
Normal">Section 3.1., &quot;Consult source code for details&quot;<u></u><u>=
</u></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. =A0=
And I would prefer to avoid that =A0:) =A0Better to specify here how the BE=
I and FL fields are encoded. =A0 Alternatively, if the codec really does ex=
pect a combined BEI/FL field, you could specify such a field here, opaque t=
o the RTP layer. =A0However, you would at least need to say how the recipie=
nt knows where this field starts and stops.<u></u><u></u></p>
</div><div><p class=3D"MsoNormal"><u></u>=A0<u></u></p></div><div><p class=
=3D"MsoNormal">Section 3.2., &quot;<span style=3D"font-size:10.0pt">The len=
gth of the encoded data is variable and depends on=A0the signal characteris=
tics and the target bit rate.</span>&quot;<u></u><u></u></p>
</div><div><p class=3D"MsoNormal">However, there&#39;s nothing in this sect=
ion that tells a recipient how to determine what this length is. =A0<u></u>=
<u></u></p></div><div><p class=3D"MsoNormal"><u></u>=A0<u></u></p></div><di=
v><p class=3D"MsoNormal">
Section 3.3., &quot;...=A0<span style=3D"font-size:10.0pt">verifying the CR=
C checksum ...</span>&quot;<u></u><u></u></p></div><div><p class=3D"MsoNorm=
al">Please specify which CRC function is to be applied, and how the CRC val=
ue field is formatted.<u></u><u></u></p>
</div><div><p class=3D"MsoNormal"><u></u>=A0<u></u></p></div><div><p class=
=3D"MsoNormal">Section 3.3., &quot;<span style=3D"font-size:10.0pt">If this=
 value would exceed 255 at encoding...</span>&quot;<u></u><u></u></p></div>=
<div>
<p class=3D"MsoNormal">It sounds like the LEN field is a single octet unsig=
ned integer. =A0Please state that explicitly.=A0<u></u><u></u></p></div><di=
v><p class=3D"MsoNormal"><u></u>=A0<u></u></p></div><div><p class=3D"MsoNor=
mal">Section 9.2. Informative References.<u></u><u></u></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&q=
uot;</span><u></u><u></u></p></div><div><p class=3D"MsoNormal"><u></u>=A0<u=
></u></p>
</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt">Thanks,<=
/span><u></u><u></u></p></div><div><p class=3D"MsoNormal"><span style=3D"fo=
nt-size:10.0pt">--Richard</span><u></u><u></u></p></div></div></div></div><=
/div>
</blockquote></div><br></div>

--e89a8fb1ef1651d27a04d9510af2--

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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Richard Barnes'" <rlb@ipv.sx>
References: <CAL02cgS_0GYw0is8rW5tMhDmsb9A8Mng6oEVwxGks-OdZYi3Gg@mail.gmail.com>	<062e01ce2f03$33956840$9ac038c0$@gmail.com> <CAL02cgTdFbOb=A4HbEd=rJAL8wae_fD7yUsAgfHJZ1AuwNFvkg@mail.gmail.com>
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Date: Tue, 2 Apr 2013 00:44:43 +0300
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Cc: draft-ietf-avt-rtp-isac@tools.ietf.org, payload@ietf.org
Subject: Re: [payload] AD Evaluation: draft-ietf-avt-rtp-isac-04
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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

 


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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Hi Richard,<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>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.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Roni<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
Richard Barnes [mailto:rlb@ipv.sx] <br><b>Sent:</b> 01 April, 2013 9:47 =
PM<br><b>To:</b> Roni Even<br><b>Cc:</b> payload@ietf.org; =
draft-ietf-avt-rtp-isac@tools.ietf.org<br><b>Subject:</b> Re: [payload] =
AD Evaluation: draft-ietf-avt-rtp-isac-04<o:p></o:p></span></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Sure. I =
didn't mean to imply that we need a reference to the codec, in fact the =
opposite. &nbsp;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. &nbsp;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.<o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>--Richard<o:p></o:p></p></div><div><p =
class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal>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:<o:p></o:p></p><div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Hi Richard,</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";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><o:p></o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Still in this case more information can be =
supplied.</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>As for the comments the authors should address =
them</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Roni Even</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
<a href=3D"mailto:payload-bounces@ietf.org" =
target=3D"_blank">payload-bounces@ietf.org</a> [mailto:<a =
href=3D"mailto:payload-bounces@ietf.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.org" =
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.ietf.org</a><br><b>Subjec=
t:</b> [payload] AD Evaluation: =
draft-ietf-avt-rtp-isac-04</span><o:p></o:p></p><div><div><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Overall, =
this document looks in pretty good shape. &nbsp;A couple of comments I =
would like to get resolved before IETF LC:<o:p></o:p></p><div><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p></div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Section =
3.1., &quot;Consult source code for =
details&quot;<o:p></o:p></p></div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>If the =
details are only specified in the source code, then the source needs to =
be a normative reference. &nbsp;And I would prefer to avoid that =
&nbsp;:) &nbsp;Better to specify here how the BEI and FL fields are =
encoded. &nbsp; Alternatively, if the codec really does expect a =
combined BEI/FL field, you could specify such a field here, opaque to =
the RTP layer. &nbsp;However, you would at least need to say how the =
recipient knows where this field starts and =
stops.<o:p></o:p></p></div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p></div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Section =
3.2., &quot;<span style=3D'font-size:10.0pt'>The length of the encoded =
data is variable and depends on&nbsp;the signal characteristics and the =
target bit rate.</span>&quot;<o:p></o:p></p></div><div><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>However, =
there's nothing in this section that tells a recipient how to determine =
what this length is. &nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p></div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Section =
3.3., &quot;...&nbsp;<span style=3D'font-size:10.0pt'>verifying the CRC =
checksum ...</span>&quot;<o:p></o:p></p></div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Please =
specify which CRC function is to be applied, and how the CRC value field =
is formatted.<o:p></o:p></p></div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p></div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Section =
3.3., &quot;<span style=3D'font-size:10.0pt'>If this value would exceed =
255 at encoding...</span>&quot;<o:p></o:p></p></div><div><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>It sounds =
like the LEN field is a single octet unsigned integer. &nbsp;Please =
state that explicitly.&nbsp;<o:p></o:p></p></div><div><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p></div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Section =
9.2. Informative References.<o:p></o:p></p></div><div><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Better path =
to source code would be &quot;webrtc/<span =
style=3D'font-size:10.0pt'>modules/audio_coding/codecs/isac&quot;</span><=
o:p></o:p></p></div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p></div><div><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:10.0pt'>Thanks,</span><o:p></o:p></p></div><div><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span =
style=3D'font-size:10.0pt'>--Richard</span><o:p></o:p></p></div></div></d=
iv></div></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></body></html>
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Subject: Re: [payload] [AVTCORE] [Technical Errata Reported] RFC4733 (3489)
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AVTCORE and Payload WG,

There appear to exist an interoperability issue around the the need to
use the same timestamp rate for the DTMF and the audio. Thus, I think we
should serioulsy consider to approve this erratum. And if the text is
not correct, propose a better text.

I personally believe the proposal is significantly clearer than the old
text and also reflect what must have been intended.

This issues a 2-week last call on input on this erratum. After the 17th
of April I will issue a final recommendation on this erratum.


For previous discussion that resulted in this updated erratum proposal
see this AVTCORE mailing list discussion:
http://www.ietf.org/mail-archive/web/avt/current/msg15669.html

Cheers

Magnus Westerlund
AVTCORE WG chair

On 2013-02-19 08:42, RFC Errata System wrote:
> 
> The following errata report has been submitted for RFC4733,
> "RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals".
> 
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=4733&eid=3489
> 
> --------------------------------------
> Type: Technical
> Reported by: Xavier Marjou <xavier.marjou@orange.com>
> 
> Section: 2.1
> 
> Original Text
> -------------
> Named telephone events are carried as part of the audio stream and MUST use the same sequence number and timestamp base as the regular audio channel to simplify the generation of audio waveforms at a gateway.
> 
> Corrected Text
> --------------
> Named telephone events are carried as part of the audio stream and if they use the same SSRC (therefore the same timing and sequence number space), they MUST use the same timestamp clock rate as the regular audio channel.
> 
> Notes
> -----
> RFC4733 was written in a way to avoid the multiple clock-rate problem by mandating the use of the same SSRC for DTMF and audio. However it's not explicitly written, which brings an ambiguity. It was commented that RFC4733 needs to be updated. (c.f. http://tools.ietf.org/wg/avtext/minutes?item=minutes-85-avtext.html). This errata obsoletes the previous proposed errata (ID: 3451).
> 
> Instructions:
> -------------
> This errata is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary. 
> 
> --------------------------------------
> RFC4733 (draft-ietf-avt-rfc2833bis-15)
> --------------------------------------
> Title               : RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals
> Publication Date    : December 2006
> Author(s)           : H. Schulzrinne, T. Taylor
> Category            : PROPOSED STANDARD
> Source              : Audio/Video Transport
> Area                : Real-time Applications and Infrastructure
> Stream              : IETF
> Verifying Party     : IESG
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 
> 


-- 

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
----------------------------------------------------------------------


From petithug@acm.org  Wed Apr  3 11:23:43 2013
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Subject: Re: [payload] [AVTCORE] [Technical Errata Reported] RFC4733 (3489)
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256

On 04/03/2013 07:55 AM, Magnus Westerlund wrote:
> AVTCORE and Payload WG,
> 
> There appear to exist an interoperability issue around the the need to use
> the same timestamp rate for the DTMF and the audio. Thus, I think we should
> serioulsy consider to approve this erratum. And if the text is not correct,
> propose a better text.
> 
> I personally believe the proposal is significantly clearer than the old 
> text and also reflect what must have been intended.
> 
> This issues a 2-week last call on input on this erratum. After the 17th of
> April I will issue a final recommendation on this erratum.
> 

I support this text.

BTW I would like to applaud the way this erratum was handled. i.e. discussed
and refined first in the WG, as it should be[1].

Thanks.

[1] http://www.rfc-editor.org/pipermail/rfc-interest/2012-August/004839.html

> 
> For previous discussion that resulted in this updated erratum proposal see
> this AVTCORE mailing list discussion: 
> http://www.ietf.org/mail-archive/web/avt/current/msg15669.html
> 
> Cheers
> 
> Magnus Westerlund AVTCORE WG chair
> 
> On 2013-02-19 08:42, RFC Errata System wrote:
>> 
>> The following errata report has been submitted for RFC4733, "RTP Payload
>> for DTMF Digits, Telephony Tones, and Telephony Signals".
>> 
>> -------------------------------------- You may review the report below
>> and at: http://www.rfc-editor.org/errata_search.php?rfc=4733&eid=3489
>> 
>> -------------------------------------- Type: Technical Reported by:
>> Xavier Marjou <xavier.marjou@orange.com>
>> 
>> Section: 2.1
>> 
>> Original Text ------------- Named telephone events are carried as part of
>> the audio stream and MUST use the same sequence number and timestamp base
>> as the regular audio channel to simplify the generation of audio
>> waveforms at a gateway.
>> 
>> Corrected Text -------------- Named telephone events are carried as part
>> of the audio stream and if they use the same SSRC (therefore the same
>> timing and sequence number space), they MUST use the same timestamp clock
>> rate as the regular audio channel.
>> 
>> Notes ----- RFC4733 was written in a way to avoid the multiple clock-rate
>> problem by mandating the use of the same SSRC for DTMF and audio. However
>> it's not explicitly written, which brings an ambiguity. It was commented
>> that RFC4733 needs to be updated. (c.f.
>> http://tools.ietf.org/wg/avtext/minutes?item=minutes-85-avtext.html).
>> This errata obsoletes the previous proposed errata (ID: 3451).
>> 
>> Instructions: ------------- This errata is currently posted as
>> "Reported". If necessary, please use "Reply All" to discuss whether it
>> should be verified or rejected. When a decision is reached, the verifying
>> party (IESG) can log in to change the status and edit the report, if
>> necessary.
>> 
>> -------------------------------------- RFC4733
>> (draft-ietf-avt-rfc2833bis-15) -------------------------------------- 
>> Title               : RTP Payload for DTMF Digits, Telephony Tones, and
>> Telephony Signals Publication Date    : December 2006 Author(s)
>> : H. Schulzrinne, T. Taylor Category            : PROPOSED STANDARD 
>> Source              : Audio/Video Transport Area                :
>> Real-time Applications and Infrastructure Stream              : IETF 
>> Verifying Party     : IESG

- -- 
Marc Petit-Huguenin
Email: marc@petit-huguenin.org
Blog: http://blog.marc.petit-huguenin.org
Profile: http://www.linkedin.com/in/petithug
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From internet-drafts@ietf.org  Fri Apr 12 06:33:37 2013
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Subject: [payload] I-D Action: draft-ietf-payload-rtp-howto-03.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           : How to Write an RTP Payload Format
	Author(s)       : Magnus Westerlund
	Filename        : draft-ietf-payload-rtp-howto-03.txt
	Pages           : 51
	Date            : 2013-04-12

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-03

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-payload-rtp-howto-03


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


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Subject: Re: [payload] I-D Action: draft-ietf-payload-rtp-howto-03.txt
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WG,

Sorry for being tardy in updating this document. I have finally done a
rewrite of the text on security consideration, especially the template
text. I now intended to request review of this from the sec-dir and SEC
ADs if they think this works.

I have also done a complete review and taken care of any detected
published documents, concluded WG and other things that has changed
since last years submission.

If the security considerations are okay I do think this document is
ready for publications. I would really appreciate if some could take the
time to review it and see if they agree that it is ready for WG last call.

Cheers

Magnus Westerlund

On 2013-04-12 15:33, internet-drafts@ietf.org wrote:
> 
> 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           : How to Write an RTP Payload Format
> 	Author(s)       : Magnus Westerlund
> 	Filename        : draft-ietf-payload-rtp-howto-03.txt
> 	Pages           : 51
> 	Date            : 2013-04-12
> 
> 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-03
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-payload-rtp-howto-03
> 
> 
> 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
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [payload] Security consideration template text for RTP payload formats
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Secdir and Sec ADs,

I have just updated the draft on Howto write RTP payload formats:
https://datatracker.ietf.org/wg/payload/

I would like to request review from both some Secdir persons and at
least one of the ADs regarding the security aspects of this document.
The reasons are that this document contains both recommendations on what
to think about when writing Security Consideration sections and template
text. As this is a document that has existed a long time we have
determined that the template text will be commonly used by authors of
RTP payload formats.

Thus I want to ensure that we have at least now have agreement on this
text, especially in light of the discussion around
https://datatracker.ietf.org/doc/draft-ietf-avt-srtp-not-mandatory/
and that we don't want secdir reviewers raising issues with this
template text when you see it in the last call reviews in future years.

The relevant security sections from the Howto draft are the following:

7.2.  Security Considerations

   All Internet drafts require a Security Considerations section.  The
   security considerations section in an RTP payload format needs to
   concentrate on the security properties this particular format has.
   Some payload formats have very few specific issues or properties and
   can fully fall back on the security considerations for RTP in general
   and those of the profile being used.  Because those documents are
   always applicable, a reference to these is normally placed first in
   the security considerations section.  There is suggested text in the
   template below.

   The security issues of confidentiality, integrity protection and
   source authentication are common issue for all payload formats.
   These should be solved by mechanisms external to the payload and do
   not need any special consideration in the payload format except for
   an reminder on these issues.  Reasons for this division is further
   documented in "Securing the RTP Protocol Framework: Why RTP Does Not
   Mandate a Single Media Security Solution"
   [I-D.ietf-avt-srtp-not-mandatory].  For a survey of available
   mechanisms to meet these goals, please review "Options for Securing
   RTP Sessions" [I-D.ietf-avtcore-rtp-security-options].  Suitable
   stock text to inform people about this is included in the template.

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

   1.  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.  For example a generic non-
       destructive compression algorithm may provide an output of almost
       an infinite size for a very limited input, thus consuming memory
       or storage space out of proportion with what the receiving
       application expected.  Such inputs can cause some sort of
       disruption, i.e.  a denial of service attack on the receiver side
       by preventing that host from producing any goodput.  Certain
       decoding operations may also vary in the amount of processing
       needed to perform those operations depending on the input.  This
       may also be a security risk if it is possible to raise processing
       load significantly above nominal simply by designing a malicious
       input sequence.  If such potential attacks exist, this must be
       made clear in the security considerations section to make
       implementers aware of the need to take precautions against such
       behavior.

   2.  The inclusion of active content in the media format or its
       transport.  "Active content" means scripts etc.  that allow an
       attacker to perform potentially arbitrary operations on the
       receiver.  Most active contents has limited possibility to access
       the system or perform operations outside a protected sandbox.
       RFC 4855 [RFC4855] has a requirement that it be noted in the
       media types registration if the payload format contains active
       content or not.  If the payload format has active content it is
       strongly recommended that references to any security model
       applicable for such content are provided.  A boilerplate text for
       "no active content" is included in the template.  This must be
       changed if the format actually carries active content.

   3.  Some media formats allow for the carrying of "user data", or
       types of data which are not known at the time of the
       specification of the payload format.  Such data may be a security
       risk and should be mentioned.

   4.  Audio or Speech codecs supporting variable bit-rate based on
       audio/speech input or having discontinuous transmission support
       must consider the issues discussed in Guidelines for the Use of
       Variable Bit Rate Audio with Secure RTP [RFC6562].

   Suitable stock text for the security considerations section is
   provided in the template in the appendix.  However, authors do need
   to actively consider any security issues from the start.  Failure to
   address these issues may block approval and publication.


---- Next Section ----

A.13.  Security Considerations

   [See Section 7.2]

   RTP packets using the payload format defined in this specification
   are subject to the security considerations discussed in the RTP
   specification [RFC3550] , and in any applicable RTP profile such as
   RTP/AVP [RFC3551], RTP/AVPF [RFC4585], RTP/SAVP [RFC3711] or RTP/
   SAVPF [RFC5124].  However, as "Securing the RTP Protocol Framework:
   Why RTP Does Not Mandate a Single Media Security Solution"
   [I-D.ietf-avt-srtp-not-mandatory] discusses it is not an RTP payload
   formats responsibility to discuss or mandate what solutions are used
   to meet the basic security goals like confiedenitality, integrity and
   source authenticity for RTP in general.  This responsibility lays on
   anyone using RTP in an application.  They can find guidance on
   available security mechanisms and important considerations in Options
   for Securing RTP Sessions [I-D.ietf-avtcore-rtp-security-options].
   The rest of the this security consideration discusses the security
   impacting properties of the payload format itself.

   This RTP payload format and its media decoder do not exhibit any
   significant non-uniformity in the receiver-side computational
   complexity for packet processing, and thus are unlikely to pose a
   denial-of-service threat due to the receipt of pathological data.
   Nor does the RTP payload format contain any active content.

   [The previous paragraph may need editing due to the format breaking
   either of the statements.  Fill in here any further potential
   security threats created by the payload format itself.]


Thanks


Magnus Westerlund

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--14dae93996073611bf04da81c9a0
Content-Type: text/plain; charset=ISO-8859-1

I reviewed this draft and have a couple of questions before IETF LC:

Major:

>From the current text, it seems like the frame reconstruction algorithm is
mostly clear:
1. Collect all packets with a given PictureID
2. Order packets by PartID
3. Go through packets in order, updating PartID as you go through each
packet.
4. If you get to the end of a packet with an incomplete partition, and the
PartID of the next packet is different from the current PartID (indicating
a lost packet), then discard the incomplete partition and:
4.1. If S=1, start a new partition at the beginning of the new packet
4.2. If S=0, skip to the next partition boundary in the new packet and
start a new partition starting there
Is this understanding correct?  In any case, it would really help
interoperability if you could write this algorithm down explicitly in the
document, probably around Section 4.4.  Note that this assumes that the
bitstream format has partition markers.


Minor:

The document refers several times to the "golden" and "altref" frames.
 However, I don't see it defined anywhere how these frames are set.  Is
this in the bit stream definition?

Section 4.2. says that the PartID MUST NOT be larger than 8.  But the field
is 4 bits long.  Is this correct?

In Section 4.2., there are 8- and 16-bit versions of the PictureID.  The
definition says that when the maximum ID is reached, the ID rolls over.
 MUST the length remain the same?

In the congestion control section, please add a note that the discarding of
non-reference frames cannot be done if the stream is encrypted (because the
non-reference marker is encrypted).

--14dae93996073611bf04da81c9a0
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I reviewed this draft and have a couple of questions before IETF LC:<div><b=
r></div><div>Major:</div><div><br></div><div>From the current text, it seem=
s like the frame reconstruction algorithm is mostly clear:</div><div>1. Col=
lect all packets with a given PictureID</div>
<div>2. Order packets by PartID</div><div>3. Go through packets in order, u=
pdating PartID as you go through each packet. =A0</div><div>4. If you get t=
o the end of a packet with an incomplete partition, and the PartID of the n=
ext packet is different from the current PartID (indicating a lost packet),=
 then discard the incomplete partition and:</div>
<div>4.1. If S=3D1, start a new partition at the beginning of the new packe=
t</div><div>4.2. If S=3D0, skip to the next partition boundary in the new p=
acket and start a new partition starting there</div><div>Is this understand=
ing correct? =A0In any case, it would really help interoperability if you c=
ould write this algorithm down explicitly in the document, probably around =
Section 4.4. =A0Note that this assumes that the bitstream format has partit=
ion markers.</div>
<div><br></div><div><br></div><div>Minor:=A0</div><div><br></div><div>The d=
ocument refers several times to the &quot;golden&quot; and &quot;altref&quo=
t; frames. =A0However, I don&#39;t see it defined anywhere how these frames=
 are set. =A0Is this in the bit stream definition?</div>
<div><br></div><div>Section 4.2. says that the PartID MUST NOT be larger th=
an 8. =A0But the field is 4 bits long. =A0Is this correct?</div><div><br></=
div><div>In Section 4.2., there are 8- and 16-bit versions of the PictureID=
. =A0The definition says that when the maximum ID is reached, the ID rolls =
over. =A0MUST the length remain the same?</div>
<br class=3D"Apple-interchange-newline">In the congestion control section, =
please add a note that the discarding of non-reference frames cannot be don=
e if the stream is encrypted (because the non-reference marker is encrypted=
).

--14dae93996073611bf04da81c9a0--

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Subject: Re: [payload] [AVTCORE] [Technical Errata Reported] RFC4733 (3489)
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AD,

I recommend that you verify this errata as there has been no objection
during this last call period and the text was developed in the context
of the AVTCORE WG.

Cheers

Magnus Westerlund
as WG chair of AVTCORE WG


On 2013-04-03 16:55, Magnus Westerlund wrote:
> AVTCORE and Payload WG,
> 
> There appear to exist an interoperability issue around the the need to
> use the same timestamp rate for the DTMF and the audio. Thus, I think we
> should serioulsy consider to approve this erratum. And if the text is
> not correct, propose a better text.
> 
> I personally believe the proposal is significantly clearer than the old
> text and also reflect what must have been intended.
> 
> This issues a 2-week last call on input on this erratum. After the 17th
> of April I will issue a final recommendation on this erratum.
> 
> 
> For previous discussion that resulted in this updated erratum proposal
> see this AVTCORE mailing list discussion:
> http://www.ietf.org/mail-archive/web/avt/current/msg15669.html
> 
> Cheers
> 
> Magnus Westerlund
> AVTCORE WG chair
> 
> On 2013-02-19 08:42, RFC Errata System wrote:
>>
>> The following errata report has been submitted for RFC4733,
>> "RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals".
>>
>> --------------------------------------
>> You may review the report below and at:
>> http://www.rfc-editor.org/errata_search.php?rfc=4733&eid=3489
>>
>> --------------------------------------
>> Type: Technical
>> Reported by: Xavier Marjou <xavier.marjou@orange.com>
>>
>> Section: 2.1
>>
>> Original Text
>> -------------
>> Named telephone events are carried as part of the audio stream and MUST use the same sequence number and timestamp base as the regular audio channel to simplify the generation of audio waveforms at a gateway.
>>
>> Corrected Text
>> --------------
>> Named telephone events are carried as part of the audio stream and if they use the same SSRC (therefore the same timing and sequence number space), they MUST use the same timestamp clock rate as the regular audio channel.
>>
>> Notes
>> -----
>> RFC4733 was written in a way to avoid the multiple clock-rate problem by mandating the use of the same SSRC for DTMF and audio. However it's not explicitly written, which brings an ambiguity. It was commented that RFC4733 needs to be updated. (c.f. http://tools.ietf.org/wg/avtext/minutes?item=minutes-85-avtext.html). This errata obsoletes the previous proposed errata (ID: 3451).
>>
>> Instructions:
>> -------------
>> This errata is currently posted as "Reported". If necessary, please
>> use "Reply All" to discuss whether it should be verified or
>> rejected. When a decision is reached, the verifying party (IESG)
>> can log in to change the status and edit the report, if necessary. 
>>
>> --------------------------------------
>> RFC4733 (draft-ietf-avt-rfc2833bis-15)
>> --------------------------------------
>> Title               : RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals
>> Publication Date    : December 2006
>> Author(s)           : H. Schulzrinne, T. Taylor
>> Category            : PROPOSED STANDARD
>> Source              : Audio/Video Transport
>> Area                : Real-time Applications and Infrastructure
>> Stream              : IETF
>> Verifying Party     : IESG
>> _______________________________________________
>> Audio/Video Transport Core Maintenance
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>>
>>
> 
> 


-- 

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
----------------------------------------------------------------------


From gonzalo.camarillo@ericsson.com  Thu Apr 18 01:09:45 2013
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Date: Thu, 18 Apr 2013 11:09:41 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Cc: schulzrinne@cs.columbia.edu, even.roni@huawei.com, avt@ietf.org, "payload@ietf.org" <payload@ietf.org>, tom.taylor.stds@gmail.com
Subject: Re: [payload] [AVTCORE] [Technical Errata Reported] RFC4733 (3489)
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Hi,

I have just verified it.

Thanks,

Gonzalo

On 18/04/2013 10:50 AM, Magnus Westerlund wrote:
> AD,
> 
> I recommend that you verify this errata as there has been no objection
> during this last call period and the text was developed in the context
> of the AVTCORE WG.
> 
> Cheers
> 
> Magnus Westerlund
> as WG chair of AVTCORE WG
> 
> 
> On 2013-04-03 16:55, Magnus Westerlund wrote:
>> AVTCORE and Payload WG,
>>
>> There appear to exist an interoperability issue around the the need to
>> use the same timestamp rate for the DTMF and the audio. Thus, I think we
>> should serioulsy consider to approve this erratum. And if the text is
>> not correct, propose a better text.
>>
>> I personally believe the proposal is significantly clearer than the old
>> text and also reflect what must have been intended.
>>
>> This issues a 2-week last call on input on this erratum. After the 17th
>> of April I will issue a final recommendation on this erratum.
>>
>>
>> For previous discussion that resulted in this updated erratum proposal
>> see this AVTCORE mailing list discussion:
>> http://www.ietf.org/mail-archive/web/avt/current/msg15669.html
>>
>> Cheers
>>
>> Magnus Westerlund
>> AVTCORE WG chair
>>
>> On 2013-02-19 08:42, RFC Errata System wrote:
>>>
>>> The following errata report has been submitted for RFC4733,
>>> "RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals".
>>>
>>> --------------------------------------
>>> You may review the report below and at:
>>> http://www.rfc-editor.org/errata_search.php?rfc=4733&eid=3489
>>>
>>> --------------------------------------
>>> Type: Technical
>>> Reported by: Xavier Marjou <xavier.marjou@orange.com>
>>>
>>> Section: 2.1
>>>
>>> Original Text
>>> -------------
>>> Named telephone events are carried as part of the audio stream and MUST use the same sequence number and timestamp base as the regular audio channel to simplify the generation of audio waveforms at a gateway.
>>>
>>> Corrected Text
>>> --------------
>>> Named telephone events are carried as part of the audio stream and if they use the same SSRC (therefore the same timing and sequence number space), they MUST use the same timestamp clock rate as the regular audio channel.
>>>
>>> Notes
>>> -----
>>> RFC4733 was written in a way to avoid the multiple clock-rate problem by mandating the use of the same SSRC for DTMF and audio. However it's not explicitly written, which brings an ambiguity. It was commented that RFC4733 needs to be updated. (c.f. http://tools.ietf.org/wg/avtext/minutes?item=minutes-85-avtext.html). This errata obsoletes the previous proposed errata (ID: 3451).
>>>
>>> Instructions:
>>> -------------
>>> This errata is currently posted as "Reported". If necessary, please
>>> use "Reply All" to discuss whether it should be verified or
>>> rejected. When a decision is reached, the verifying party (IESG)
>>> can log in to change the status and edit the report, if necessary. 
>>>
>>> --------------------------------------
>>> RFC4733 (draft-ietf-avt-rfc2833bis-15)
>>> --------------------------------------
>>> Title               : RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals
>>> Publication Date    : December 2006
>>> Author(s)           : H. Schulzrinne, T. Taylor
>>> Category            : PROPOSED STANDARD
>>> Source              : Audio/Video Transport
>>> Area                : Real-time Applications and Infrastructure
>>> Stream              : IETF
>>> Verifying Party     : IESG
>>> _______________________________________________
>>> Audio/Video Transport Core Maintenance
>>> avt@ietf.org
>>> https://www.ietf.org/mailman/listinfo/avt
>>>
>>>
>>
>>
> 
> 


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Subject: [payload] Errata report on RFC6184 - H.264 payload
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Hi,

An errata report was filed on RFC6184
http://www.rfc-editor.org/errata_search.php?eid=3318  it is given bellow.

I would like to get some feedback about this errata, is it correct?

"Section 8.1 says:

On page 46 start from line 7: 

"When max-dpb is signaled, the receiver MUST be able to decode

NAL unit streams that conform to the signaled highest level,

with the exception that the MaxDpbMbs value in Table A-1 of [1]

for the signaled highest level is replaced with the value of

max-dpb * 3 / 8. Consequently, a receiver that signals max-dpb

          ^^^^^

MUST be capable of storing the following number of decoded

frames, complementary field pairs, and non-paired fields in its

decoded picture buffer:

Min(max-dpb * 3 / 8 / ( PicWidthInMbs * FrameHeightInMbs),

              ^^^^^

16)"

It should say:

When max-dpb is signaled, the receiver MUST be able to decode

NAL unit streams that conform to the signaled highest level,

with the exception that the MaxDpbMbs value in Table A-1 of [1]

for the signaled highest level is replaced with the value of

max-dpb * 8 / 3. Consequently, a receiver that signals max-dpb

          ^^^^^

MUST be capable of storing the following number of decoded

frames, complementary field pairs, and non-paired fields in its

decoded picture buffer:

Min(max-dpb * 8 / 3 / ( PicWidthInMbs * FrameHeightInMbs),

              ^^^^^

16)

"

 

 

 

My personal view is that  since MaxDpbMbs= max-dpb * 8 / 3 the errata is
correct.

 

 

Roni Even

 

 

 

 


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New"'>&#8220;<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:24.1pt'><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:24.1pt'><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:24.1pt'><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:24.1pt'>My personal view is that &nbsp;since =
MaxDpbMbs=3D max-dpb * 8 / 3 the errata is correct.<o:p></o:p></p><p =
class=3DMsoNormal style=3D'margin-left:24.1pt'><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal style=3D'margin-left:24.1pt'><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal style=3D'margin-left:24.1pt'>Roni =
Even<o:p></o:p></p><p class=3DMsoNormal =
style=3D'margin-left:24.1pt'><o:p>&nbsp;</o:p></p><p class=3DMsoNormal =
style=3D'margin-left:24.1pt'><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></body></html>
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Roni Even <ron.even.tlv@gmail.com>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: Errata report on RFC6184 - H.264 payload
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Date: Fri, 19 Apr 2013 16:35:00 +0000
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Subject: Re: [payload] Errata report on RFC6184 - H.264 payload
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The errata looks correct. For further clarity, it may also help to add pare=
nthesis
around (max-dpb*8/3).

NEW:
When max-dpb is signaled, the receiver MUST be able to decode
NAL unit streams that conform to the signaled highest level,
with the exception that the MaxDpbMbs value in Table A-1 of [1]
for the signaled highest level is replaced with the value of
max-dpb * 8/3. Consequently, a receiver that signals max-dpb
MUST be capable of storing the following number of decoded
frames, complementary field pairs, and non-paired fields in its
decoded picture buffer:
Min((max-dpb * 8/3) / (PicWidthInMbs * FrameHeightInMbs),16)

Mo


From: Roni Even [mailto:ron.even.tlv@gmail.com]
Sent: Friday, April 19, 2013 3:41 AM
To: payload@ietf.org
Cc: 'Botzko, Stephen'; Mo Zanaty (mzanaty); Wang, Ye-Kui; Tom Kristensen (t=
omkrist)
Subject: Errata report on RFC6184 - H.264 payload

Hi,
An errata report was filed on RFC6184 http://www.rfc-editor.org/errata_sear=
ch.php?eid=3D3318  it is given bellow.
I would like to get some feedback about this errata, is it correct?
"Section 8.1 says:
On page 46 start from line 7:
"When max-dpb is signaled, the receiver MUST be able to decode
NAL unit streams that conform to the signaled highest level,
with the exception that the MaxDpbMbs value in Table A-1 of [1]
for the signaled highest level is replaced with the value of
max-dpb * 3 / 8. Consequently, a receiver that signals max-dpb
          ^^^^^
MUST be capable of storing the following number of decoded
frames, complementary field pairs, and non-paired fields in its
decoded picture buffer:
Min(max-dpb * 3 / 8 / ( PicWidthInMbs * FrameHeightInMbs),
              ^^^^^
16)"
It should say:
When max-dpb is signaled, the receiver MUST be able to decode
NAL unit streams that conform to the signaled highest level,
with the exception that the MaxDpbMbs value in Table A-1 of [1]
for the signaled highest level is replaced with the value of
max-dpb * 8 / 3. Consequently, a receiver that signals max-dpb
          ^^^^^
MUST be capable of storing the following number of decoded
frames, complementary field pairs, and non-paired fields in its
decoded picture buffer:
Min(max-dpb * 8 / 3 / ( PicWidthInMbs * FrameHeightInMbs),
              ^^^^^
16)
"



My personal view is that  since MaxDpbMbs=3D max-dpb * 8 / 3 the errata is =
correct.


Roni Even





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<p class=3D"MsoNormal">The errata looks correct. For further clarity, it ma=
y also help to add parenthesis<o:p></o:p></p>
<p class=3D"MsoNormal">around (max-dpb*8/3).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">NEW:<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">When max-dpb is signale=
d, the receiver MUST be able to decode<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">NAL unit streams that c=
onform to the signaled highest level,<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">with the exception that=
 the MaxDpbMbs value in Table A-1 of [1]<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">for the signaled highes=
t level is replaced with the value of<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt"><span style=3D"backgrou=
nd:yellow;mso-highlight:yellow">max-dpb * 8/3</span>. Consequently, a recei=
ver that signals max-dpb<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">MUST be capable of stor=
ing the following number of decoded<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">frames, complementary f=
ield pairs, and non-paired fields in its<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">decoded picture buffer:=
<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">Min(<span style=3D"back=
ground:yellow;mso-highlight:yellow">(max-dpb * 8/3)</span> / (PicWidthInMbs=
 * FrameHeightInMbs),16)<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Mo<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<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;"> Roni Eve=
n [mailto:ron.even.tlv@gmail.com]
<br>
<b>Sent:</b> Friday, April 19, 2013 3:41 AM<br>
<b>To:</b> payload@ietf.org<br>
<b>Cc:</b> 'Botzko, Stephen'; Mo Zanaty (mzanaty); Wang, Ye-Kui; Tom Kriste=
nsen (tomkrist)<br>
<b>Subject:</b> Errata report on RFC6184 - H.264 payload<o:p></o:p></span><=
/p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal">An errata report was filed on RFC6184 <a href=3D"htt=
p://www.rfc-editor.org/errata_search.php?eid=3D3318">
http://www.rfc-editor.org/errata_search.php?eid=3D3318</a> &nbsp;it is give=
n bellow.<o:p></o:p></p>
<p class=3D"MsoNormal">I would like to get some feedback about this errata,=
 is it correct?<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&#8220;Section 8.1 says:<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">On page 46 start from l=
ine 7: <o:p>
</o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">&quot;When max-dpb is s=
ignaled, the receiver MUST be able to decode<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">NAL unit streams that c=
onform to the signaled highest level,<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">with the exception that=
 the MaxDpbMbs value in Table A-1 of [1]<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">for the signaled highes=
t level is replaced with the value of<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">max-dpb * 3 / 8. Conseq=
uently, a receiver that signals max-dpb<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ^^^^^<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">MUST be capable of stor=
ing the following number of decoded<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">frames, complementary f=
ield pairs, and non-paired fields in its<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">decoded picture buffer:=
<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">Min(max-dpb * 3 / 8 / (=
 PicWidthInMbs * FrameHeightInMbs),<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ^^^^^<o:p></o:p></p=
>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">16)&quot;<o:p></o:p></p=
>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">It should say:<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">When max-dpb is signale=
d, the receiver MUST be able to decode<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">NAL unit streams that c=
onform to the signaled highest level,<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">with the exception that=
 the MaxDpbMbs value in Table A-1 of [1]<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">for the signaled highes=
t level is replaced with the value of<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">max-dpb * 8 / 3. Conseq=
uently, a receiver that signals max-dpb<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ^^^^^<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">MUST be capable of stor=
ing the following number of decoded<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">frames, complementary f=
ield pairs, and non-paired fields in its<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">decoded picture buffer:=
<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">Min(max-dpb * 8 / 3 / (=
 PicWidthInMbs * FrameHeightInMbs),<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ^^^^^<o:p></o:p></p=
>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">16)<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt"><span style=3D"font-siz=
e:10.0pt;font-family:&quot;Courier New&quot;">&#8220;<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt"><span style=3D"font-siz=
e:10.0pt;font-family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt"><span style=3D"font-siz=
e:10.0pt;font-family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt"><span style=3D"font-siz=
e:10.0pt;font-family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">My personal view is tha=
t &nbsp;since MaxDpbMbs=3D max-dpb * 8 / 3 the errata is correct.<o:p></o:p=
></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt">Roni Even<o:p></o:p></p=
>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:24.1pt"><span style=3D"font-siz=
e:10.0pt;font-family:&quot;Courier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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Thread-Topic: [payload] I-D Action: draft-ietf-payload-rtp-aptx-00.txt
Thread-Index: Ac4gDnCa3Fo7/g6xRW2P9VT6PrKdAQgXDU0g
From: "Hans-Heinrich Hansen" <hhhansen@MAYAH.com>
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MAYAH and other companies have already implemented the following =
optional media type parameters defined in previous apt-X =
drafts/discussions:
1. A media type attribute embedded-sync=3Doff/on that applies to both =
variants of apt-X=20
2. Media type attributes embedded-aux-left=3Doff/on and =
embedded-aux-right=3Doff/on that apply to the Standard apt-X variant =
only.
3. A Media type attributes embedded-aux=3Doff/on that applies to the =
Enhanced apt-X variant only.

Is it possible that the new draft will also contain these parameters as =
an option for mono and stereo connections?

The new optional parameters embedded-autosync-channels and =
embedded-aux-channels could be defined for multi channel audio (more =
than 2 channels).

From mzanaty@cisco.com  Thu Apr 25 11:57:49 2013
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: "Dale R. Worley" <worley@ariadne.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "payload@ietf.org" <payload@ietf.org>, "Ali C. Begen (abegen)" <abegen@cisco.com>
Thread-Topic: [MMUSIC] Should we update the IANA registry to reflect RFC 5761?
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Date: Thu, 25 Apr 2013 18:57:46 +0000
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Subject: Re: [payload] [MMUSIC] Should we update the IANA registry to reflect RFC 5761?
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I agree with the IANA update below for clarity. But practically speaking, P=
AYLOAD will never assign another static payload, so we are effectively limi=
ted to the dynamic range for all modern codecs unless PAYLOAD decides to re=
allocate 35-63 from static to dynamic. I could only see that happening if p=
roducts really need to support >32 codecs. (True codecs, not demux IDs for =
different streams in bundles.)

Mo

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Dale R. Worley
Sent: Thursday, April 25, 2013 1:26 PM
To: mmusic@ietf.org
Subject: [MMUSIC] Should we update the IANA registry to reflect RFC 5761?

I was looking at the IANA registry "RTP Payload types (PT) for
standard audio and video encodings"
(http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xml#rtp-para=
meters-1)
to see what the allowed dynamic payload types are, and what PTs are
reserved to avoid confusion with RTCP.  The unassigned range is:

35-71   Unassigned
72-76   Reserved for RTCP conflict avoidance    [RFC3551]
77-95   Unassigned
96-127  dynamic                                 [RFC3551]

Unfortunately, this doesn't reflect RFC 5761 ("Multiplexing RTP Data
and Control Packets on a Single Port"), which gives these
instructions in section 4:

   Given these constraints, it is RECOMMENDED to follow the guidelines
   in the RTP/AVP profile [7] for the choice of RTP payload type values,
   with the additional restriction that payload type values in the range
   64-95 MUST NOT be used.  Specifically, dynamic RTP payload types
   SHOULD be chosen in the range 96-127 where possible.  Values below 64
   MAY be used if that is insufficient, in which case it is RECOMMENDED
   that payload type numbers that are not statically assigned by [7] be
   used first.

Since the entire range 64 to 95 is now reserved, the table should
read:

35-63   Unassigned
64-71   Reserved for RTCP conflict avoidance    [RFC5761]
72-76   Reserved for RTCP conflict avoidance    [RFC3551]
77-95   Reserved for RTCP conflict avoidance    [RFC5761]
96-127  dynamic                                 [RFC3551]

IMHO, it would be helpful to get this table up to date with regard to
the RFCs, even though it is marked "Closed".

Do people agree with me on this?  (And how do we go about it?  I
assume that the AD can route it to IANA.)

(This also limits us to 61 PTs, which is another matter.)

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

From adam@nostrum.com  Thu Apr 25 12:29:40 2013
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Subject: Re: [payload] [MMUSIC] Should we update the IANA registry to reflect RFC 5761?
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On 4/25/13 13:57, Mo Zanaty (mzanaty) wrote:
> I agree with the IANA update below for clarity. But practically speaking, PAYLOAD will never assign another static payload, so we are effectively limited to the dynamic range for all modern codecs unless PAYLOAD decides to reallocate 35-63 from static to dynamic.

I think RFC 3551, section 3 is still the governing text on how RTP/AVP 
(and its related profiles) allocates payload types:

 >  "This profile reserves payload type numbers in the range 96-127
 >   exclusively for dynamic assignment.  Applications SHOULD first use
 >   values in this range for dynamic payload types.  Those applications
 >   which need to define more than 32 dynamic payload types MAY bind
 >   codes below 96, in which case it is RECOMMENDED that unassigned
 >   payload type numbers be used first.  However, the statically assigned
 >   payload types are default bindings and MAY be dynamically bound to
 >   new encodings if needed."

(With particular emphasis on the final quoted sentence)

So, _practically_ speaking, the proposed registry update does make a big 
difference: it clarifies that 64-71 and 77-95 are off limits. Also, 
_practically_ speaking, there's room to talk about 32 + 64 = 96 distinct 
codecs, since even the statically assigned payload types can be rebound.

If you work with teams that are under the mistaken impression that you 
cannot dynamically re-bind the first 64 RTP payload types, I would beg 
you to disabuse them of that notion.

/a

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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 4/25/13 13:57, Mo Zanaty (mzanaty)
      wrote:<br>
    </div>
    <blockquote
cite="mid:3879D71E758A7E4AA99A35DD8D41D3D90F6DC561@xmb-rcd-x14.cisco.com"
      type="cite">
      <pre wrap="">I agree with the IANA update below for clarity. But practically speaking, PAYLOAD will never assign another static payload, so we are effectively limited to the dynamic range for all modern codecs unless PAYLOAD decides to reallocate 35-63 from static to dynamic.</pre>
    </blockquote>
    <br>
    I think
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    RFC 3551, section 3 is still the governing text on how RTP/AVP (and
    its related profiles) allocates payload types:<br>
    <br>
    &gt;&nbsp; "This profile reserves payload type numbers in the range
    96-127<br>
    &gt;&nbsp;&nbsp; exclusively for dynamic assignment.&nbsp; Applications SHOULD
    first use<br>
    &gt;&nbsp;&nbsp; values in this range for dynamic payload types.&nbsp; Those
    applications<br>
    &gt;&nbsp;&nbsp; which need to define more than 32 dynamic payload types MAY
    bind<br>
    &gt;&nbsp;&nbsp; codes below 96, in which case it is RECOMMENDED that
    unassigned<br>
    &gt;&nbsp;&nbsp; payload type numbers be used first.&nbsp; However, the statically
    assigned<br>
    &gt;&nbsp;&nbsp; payload types are default bindings and MAY be dynamically
    bound to<br>
    &gt;&nbsp;&nbsp; new encodings if needed."<br>
    <br>
    (With particular emphasis on the final quoted sentence)<br>
    <br>
    So, _practically_ speaking, the proposed registry update does make a
    big difference: it clarifies that 64-71 and 77-95 are off limits.
    Also, _practically_ speaking, there's room to talk about 32 + 64 =
    96 distinct codecs, since even the statically assigned payload types
    can be rebound.<br>
    <br>
    If you work with teams that are under the mistaken impression that
    you cannot dynamically re-bind the first 64 RTP payload types, I
    would beg you to disabuse them of that notion.<br>
    <br>
    /a<br>
  </body>
</html>

--------------060404010905000303020204--

From magnus.westerlund@ericsson.com  Thu Apr 25 23:50:38 2013
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Cc: "payload@ietf.org" <payload@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [payload] [MMUSIC] Should we update the IANA registry to reflect RFC 5761?
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Hi,

And to add to that. These limitations are only present given that you
actually are using RTP and RTCP multiplexing. You can regain the whole
128 numbers by having RTP and RTCP separated.

This quote from RFC 5761:

   Given these constraints, it is RECOMMENDED to follow the guidelines
   in the RTP/AVP profile [7] for the choice of RTP payload type values,
   with the additional restriction that payload type values in the range
   64-95 MUST NOT be used.  Specifically, dynamic RTP payload types
   SHOULD be chosen in the range 96-127 where possible.  Values below 64
   MAY be used if that is insufficient, in which case it is RECOMMENDED
   that payload type numbers that are not statically assigned by [7] be
   used first.

I think is clear that you have 96 values for use. The MUST NOT is really
to ensure that you will not have issues. Assuming tighter signalling
control could have allowed dynamic blocking of the space, but that was
deemed to complex and error prone.

Regarding the IANA, Mo you have correctly identified the registry as
closed and Adam pointed to the relevant text. Is this level of
indirection so problematic that this registry needs a Note to the effect?

Cheers

Magnus

On 2013-04-25 21:29, Adam Roach wrote:
> On 4/25/13 13:57, Mo Zanaty (mzanaty) wrote:
>> I agree with the IANA update below for clarity. But practically speaking, PAYLOAD will never assign another static payload, so we are effectively limited to the dynamic range for all modern codecs unless PAYLOAD decides to reallocate 35-63 from static to dynamic.
> 
> I think RFC 3551, section 3 is still the governing text on how RTP/AVP
> (and its related profiles) allocates payload types:
> 
>>  "This profile reserves payload type numbers in the range 96-127
>>   exclusively for dynamic assignment.  Applications SHOULD first use
>>   values in this range for dynamic payload types.  Those applications
>>   which need to define more than 32 dynamic payload types MAY bind
>>   codes below 96, in which case it is RECOMMENDED that unassigned
>>   payload type numbers be used first.  However, the statically assigned
>>   payload types are default bindings and MAY be dynamically bound to
>>   new encodings if needed."
> 
> (With particular emphasis on the final quoted sentence)
> 
> So, _practically_ speaking, the proposed registry update does make a big
> difference: it clarifies that 64-71 and 77-95 are off limits. Also,
> _practically_ speaking, there's room to talk about 32 + 64 = 96 distinct
> codecs, since even the statically assigned payload types can be rebound.
> 
> If you work with teams that are under the mistaken impression that you
> cannot dynamically re-bind the first 64 RTP payload types, I would beg
> you to disabuse them of that notion.
> 
> /a
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> 


-- 

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
----------------------------------------------------------------------


From worley@shell01.TheWorld.com  Fri Apr 26 11:22:05 2013
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From: worley@ariadne.com (Dale R. Worley)
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Subject: Re: [payload] [MMUSIC] Should we update the IANA registry to reflect RFC 5761?
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> From: Magnus Westerlund <magnus.westerlund@ericsson.com>
> 
> Regarding the IANA, Mo you have correctly identified the registry as
> closed and Adam pointed to the relevant text. Is this level of
> indirection so problematic that this registry needs a Note to the effect?

(cleaning up my proposal)

I use the IANA registries as the reference for how the various number
spaces are managed.  The current final rows of the Payload Types table
read:

    35-71   Unassigned
    72-76   Reserved for RTCP conflict avoidance    [RFC3551]
    77-95   Unassigned
    96-127  dynamic                                 [RFC3551]

I find these to be problematic in several ways:

1) RFC 5761 is not mentioned at all, despite that it provides
important modifications of the governing text in RFC 3551.  This is a
practical problem:  Note that Adam quoted the text in RFC 3551, not
the text in RFC 5761, and the 3551 text is now incorrect.

2) The range that is reserved for RTCP avoidance is not specified
correctly.  It's true that the rest of the RTCP avoidance range is
marked "Unassigned", but in the context of RFC 3551, that suggests
that they can be used as a secondary dynamic assignment area.

3) The range 35-71 should be marked more clearly as the secondary
dynamic assignment area.

Because of this, I suggest the following changes to this registry:

1) The "Reference" section should be changed from "[RFC3551]" to
"[RFC5761][RFC3551]".

2) The final rows should be changed to

    35-63   Unassigned/secondary dynamic area       [RFC5761]
    64-71   Reserved for RTCP conflict avoidance    [RFC5761]
    72-76   Reserved for RTCP conflict avoidance    [RFC3551]
    77-95   Reserved for RTCP conflict avoidance    [RFC5761]
    96-127  Dynamic                                 [RFC3551]

Dale

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Subject: [payload] Biggest Fake Conference in Computer Science
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Biggest Fake Conference in Computer Science


We are researchers from different parts of the world and conducted a study on  
the worldâ€™s biggest bogus computer science conference WORLDCOMP 
( http://sites.google.com/site/worlddump1 ) organized by Prof. Hamid Arabnia 
from University of Georgia, USA.


We submitted a fake paper to WORLDCOMP 2011 and again (the same paper 
with a modified title) to WORLDCOMP 2012. This paper had numerous 
fundamental mistakes. Sample statements from that paper include: 

(1). Binary logic is fuzzy logic and vice versa
(2). Pascal developed fuzzy logic
(3). Object oriented languages do not exhibit any polymorphism or inheritance
(4). TCP and IP are synonyms and are part of OSI model 
(5). Distributed systems deal with only one computer
(6). Laptop is an example for a super computer
(7). Operating system is an example for computer hardware


Also, our paper did not express any conceptual meaning.  However, it 
was accepted both the times without any modifications (and without 
any reviews) and we were invited to submit the final paper and a 
payment of $500+ fee to present the paper. We decided to use the 
fee for better purposes than making Prof. Hamid Arabnia (Chairman 
of WORLDCOMP) rich. After that, we received few reminders from 
WORLDCOMP to pay the fee but we never responded. 


We MUST say that you should look at the above website if you have any thoughts 
to submit a paper to WORLDCOMP.  DBLP and other indexing agencies have stopped 
indexing WORLDCOMPâ€™s proceedings since 2011 due to its fakeness. See 
http://www.informatik.uni-trier.de/~ley/db/conf/icai/index.html for of one of the 
conferences of WORLDCOMP and notice that there is no listing after 2010. See Section 2 of
http://sites.google.com/site/dumpconf for comments from well-known researchers 
about WORLDCOMP. 


The status of your WORLDCOMP papers can be changed from scientific
to other (i.e., junk or non-technical) at any time. Better not to have a paper than 
having it in WORLDCOMP and spoil the resume and peace of mind forever!


Our study revealed that WORLDCOMP is a money making business, 
using University of Georgia mask, for Prof. Hamid Arabnia. He is throwing 
out a small chunk of that money (around 20 dollars per paper published 
in WORLDCOMPâ€™s proceedings) to his puppet (Mr. Ashu Solo or A.M.G. Solo) 
who publicizes WORLDCOMP and also defends it at various forums, using 
fake/anonymous names. The puppet uses fake names and defames other conferences
to divert traffic to WORLDCOMP. He also makes anonymous phone calls and tries to 
threaten the critiques of WORLDCOMP (See Item 7 of Section 5 of above website). 
That is, the puppet does all his best to get a maximum number of papers published 
at WORLDCOMP to get more money into his (and Prof. Hamid Arabniaâ€™s) pockets. 


Monte Carlo Resort (the venue of WORLDCOMP for more than 10 years, until 2012) has 
refused to provide the venue for WORLDCOMPâ€™13 because of the fears of their image 
being tarnished due to WORLDCOMPâ€™s fraudulent activities. That is why WORLDCOMPâ€™13 
is taking place at a different resort. WORLDCOMP will not be held after 2013. 


The draft paper submission deadline is over but still there are no committee 
members, no reviewers, and there is no conference Chairman. The only contact 
details available on WORLDCOMPâ€™s website is just an email address! 

Let us make a direct request to Prof. Hamid arabnia: publish all reviews for 
all the papers (after blocking identifiable details) since 2000 conference. Reveal 
the names and affiliations of all the reviewers (for each year) and how many 
papers each reviewer had reviewed on average. We also request him to look at 
the Open Challenge (Section 6) at https://sites.google.com/site/moneycomp1 


Sorry for posting to multiple lists. Spreading the word is the only way to stop 
this bogus conference. Please forward this message to other mailing lists and people. 


We are shocked with Prof. Hamid Arabnia and his puppetâ€™s activities 
http://worldcomp-fake-bogus.blogspot.com   Search Google using the 
keyword worldcomp fake for additional links.

