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From: Ross Finlayson <finlayson@live555.com>
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Subject: Re: [AVTCORE] Clarifying the H.264 RTP payload format specification's text on when to set the RTP "M" bit
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> If you are willing to detect the last NALU of a frame by waiting for =
the first VCL NALU of the next frame, that is very simple in both H.264 =
and H.265. The first bit after the VCL NALU header is set if it=92s the =
first VCL NALU of a frame. This works in H.265 because the slice header =
explicitly has such a flag.

Thanks.  This is exactly the kind of information that would be useful =
for implementers.


> Note that RTP receivers are technically forbidden to rely on the =
marker bit. But many implementations detect whether it is reliable, then =
use it as an optimization to minimize latency if reliable.

Yes, it's unfortunate that there may be implementations out there that =
do not set the RTP 'M' bit correctly.  But to reduce the likelihood of =
this happening, we should try to make our RTP payload format =
specifications as clear as is reasonable about when to set the 'M' bit.  =
(It is common - in non-interactive environments - for RTP transmitting =
applications to obtain their encoded data as a byte stream.)


>  I don=92t think the draft needs any more text beyond what it =
currently says.

On the contrary - I think you've explained that some additional text =
would be both useful and reasonable.  (Once again, because we're not =
MPEG, we should not feel the need to fully-specify only the behavior =
that's required by RTP *receivers*.)

So I suggest adding the following paragraph to the H.265 RTP payload =
format specification, just after the existing 'M bit' text:

----------
Unfortunately the contents of a NAL unit, alone, does not tell a RTP =
sender implementation whether or not the NAL unit ends an access unit.  =
Instead, the implementation can obtain this information separately, from =
the encoder.  If, however, this information is not directly available =
from the encoder (e.g., because the implementation is sending data that =
consists solely of a sequence of pre-encoded NAL units), then it must =
instead inspect the next NAL unit, to determine whether or not the =
current NAL unit ends an access unit.  The following rule can be used:
    The current NAL unit ends an access unit if it is a VCL NAL unit, =
and if either:
	- the next NAL unit is not a VCL NAL unit, or
	- the next NAL unit is a VCL NAL unit, and the high-order bit of =
the first byte after its NALU header is 1.
----------

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"><base href=3D"x-msg://971/"></head><body =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div><blockquote =
type=3D"cite"><div lang=3D"EN-US" link=3D"blue" vlink=3D"purple" =
style=3D"font-family: Helvetica; font-size: medium; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; =
"><div class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; ">If you are willing to detect the last NALU of a =
frame by waiting for the first VCL NALU of the next frame, that is very =
simple in both H.264 and H.265. The first bit after the VCL NALU header =
is set if it=92s the first VCL NALU of a frame. This works in H.265 =
because the slice header explicitly has such a =
flag.</span></div></div></div></blockquote><div><br></div>Thanks. =
&nbsp;This is exactly the kind of information that would be useful for =
implementers.</div><div><br></div><div><br><blockquote type=3D"cite"><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple" style=3D"font-family: =
Helvetica; font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div =
class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-family: Calibri, sans-serif; =
font-size: 11pt; ">Note that RTP receivers are technically forbidden to =
rely on the marker bit. But many implementations detect whether it is =
reliable, then use it as an optimization to minimize latency if =
reliable.</span></div></div></div></blockquote><div><br></div>Yes, it's =
unfortunate that there may be implementations out there that do not set =
the RTP 'M' bit correctly. &nbsp;But to reduce the likelihood of this =
happening, we should try to make our RTP payload format specifications =
as clear as is reasonable about when to set the 'M' bit. &nbsp;(It is =
common - in non-interactive environments - for RTP transmitting =
applications to obtain their encoded data as a byte =
stream.)</div><div><br></div><div><br><blockquote type=3D"cite"><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple" style=3D"font-family: =
Helvetica; font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div =
class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; ">&nbsp;</span><span style=3D"font-family: Calibri, =
sans-serif; font-size: 11pt; ">I don=92t think the draft needs any more =
text beyond what it currently =
says.</span></div></div></div></blockquote><div><br></div>On the =
contrary - I think you've explained that some additional text would be =
both useful and reasonable. &nbsp;(Once again, because we're not MPEG, =
we should not feel the need to fully-specify only the behavior that's =
required by RTP *receivers*.)</div><div><br></div><div><div>So I suggest =
adding the following paragraph to the H.265 RTP payload format =
specification, just after the existing 'M bit' =
text:</div><div><br></div><div>----------</div><div>Unfortunately the =
contents of a NAL unit, alone, does not tell a RTP sender implementation =
whether or not the NAL unit ends an access unit. &nbsp;Instead, the =
implementation can obtain this information separately, from the encoder. =
&nbsp;If, however, this information is not directly available from the =
encoder (e.g., because the implementation is sending data that consists =
solely of a sequence of pre-encoded NAL units), then it must instead =
inspect the next NAL unit, to determine whether or not the current NAL =
unit ends an access unit. &nbsp;The following rule can be =
used:</div><div><div>&nbsp; &nbsp; The current NAL unit ends an access =
unit if it is a VCL NAL unit, and if either:</div><div><span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">	</span>- the =
next NAL unit is not a VCL NAL unit, or</div><div><span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">	</span>- the =
next NAL unit is a VCL NAL unit, and the high-order bit of the first =
byte after its NALU header is =
1.</div><div>----------</div></div><div><br></div></div><div =
apple-content-edited=3D"true"><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: 2; =
text-align: -webkit-auto; text-indent: 0px; text-transform: none; =
white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; ">Ross =
Finlayson<br>Live Networks, Inc.<br><a =
href=3D"http://www.live555.com/">http://www.live555.com/</a></span></span>=

</div>
<br></body></html>=

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Subject: Re: [AVTCORE] Clarifying the H.265 RTP payload format specification's text on when to set the RTP "M" bit
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(I changed the "Subject:" line to make it clear that we're talking about =
the (future) H.265 RTP payload format specification here; not the =
(existing) H.264 RTP payload format specification.)


> First of all, implementers are expected to have basic knowledge of the =
video coding specification itself.

Yes.  However, in this case we seem to disagree on what constitutes =
'basic knowledge'.  Determining whether or not a NAL unit ends an =
'access unit' (in the situation - common for many implementations - =
where this is not available directly from the encoder) is far from =
'basic knowledge' of H.265.  Unless you can point to a specific section =
in the H.265 specification that describes this (in which case we could =
add a reference to that section), it seems prudent to add a little text =
to our RTP payload format specification that clarifies this.


> Of course, bad packetization/sender implementations [of the RTP 'M' =
bit] can make that unreliable =96 but bad packetization/sender =
implementations can make anything unreliable =96 thus they don=92t =
count.

I'm not sure what you're saying here.  I hope you're not saying that we =
shouldn't really care about whether or not implementations set the 'M' =
bit correctly, because in that case I definitely disagree; that's the =
whole point of this email thread.

=20
>  Lastly, the suggested text is too vague and even incorrect.

Like Mohamed, you're inadvertently making my point for me :-)


> For example, in the following copied part: what is =93the current NAL =
unit=94?

By this I mean the NAL unit that's being packed in the RTP packet for =
which we are considering whether or not to set the "M' bit (or the final =
such NAL unit, if there's more than one). =20


> Also, it is possible for non-VCL NAL unit to be between two VCL NAL =
units in an AU, the [suggested] language would suggest that the first of =
the two VCL NAL unit ends the AU =96 this is definitely incorrect.

Yes, good point - thanks.

Here, then, is new wording for the suggested paragraph (to be added to =
the H.265 RTP payload format specification, just after the existing 'M =
bit' text):

----------
Unfortunately the contents of a NAL unit, alone, does not tell a RTP =
sender implementation whether or not the NAL unit ends an access unit.  =
Instead, the implementation can obtain this information separately, from =
the encoder.  If, however, this information is not available directly =
from the encoder (e.g., because the implementation is sending data that =
consists solely of a sequence of pre-encoded NAL units), then it must =
instead inspect subsequent NAL units, to determine whether or not the =
NAL unit ends an access unit.  The following rule can be used:
    A NAL unit ends an access unit if it is a VCL NAL unit, and the =
next-occurring VCL NAL unit has the high-order bit of the first byte =
after its NALU header equal to 1.
----------

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/



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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"><base href=3D"x-msg://971/"></head><body =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div>(I changed the "Subject:" =
line to make it clear that we're talking about the (future) H.265 RTP =
payload format specification here; not the (existing) H.264 RTP payload =
format specification.)</div><div><br></div><div><br><blockquote =
type=3D"cite"><div lang=3D"EN-CA" link=3D"blue" vlink=3D"purple" =
style=3D"font-family: Helvetica; font-size: medium; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; =
"><div class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); ">First of all, =
implementers are expected to have basic knowledge of the video coding =
specification =
itself.</span></div></div></div></blockquote><div><br></div><div>Yes. =
&nbsp;However, in this case we seem to disagree on what constitutes =
'basic knowledge'. &nbsp;Determining whether or not a NAL unit ends an =
'access unit' (in the situation - common for many implementations - =
where this is not available directly from the encoder) is far from =
'basic knowledge' of H.265. &nbsp;Unless you can point to a specific =
section in the H.265 specification that describes this (in which case we =
could add a reference to that section), it seems prudent to add a little =
text to our RTP payload format specification that clarifies =
this.</div><div><br></div><div><br></div><blockquote type=3D"cite"><div =
lang=3D"EN-CA" link=3D"blue" vlink=3D"purple" style=3D"font-family: =
Helvetica; font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div =
class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); ">Of course, bad =
packetization/sender implementations [of the RTP 'M' bit] can make that =
unreliable =96 but bad packetization/sender implementations can make =
anything unreliable =96 thus they don=92t =
count.</span></div></div></div></blockquote><div><br></div>I'm not sure =
what you're saying here. &nbsp;I hope you're not saying that we =
shouldn't really care about whether or not implementations set the 'M' =
bit correctly, because in that case I definitely disagree; that's the =
whole point of this email =
thread.</div><div><br></div><div>&nbsp;<br><blockquote type=3D"cite"><div =
lang=3D"EN-CA" link=3D"blue" vlink=3D"purple" style=3D"font-family: =
Helvetica; font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div =
class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p></o:p></span></div><div style=3D"margin: 0in 0in 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); ">&nbsp;</span><span style=3D"color: rgb(31, 73, 125); =
font-family: Calibri, sans-serif; font-size: 11pt; ">Lastly, the =
suggested text is too vague and even =
incorrect.</span></div></div></div></blockquote><div><br></div>Like =
Mohamed, you're inadvertently making my point for me =
:-)</div><div><br></div><div><br></div><div><blockquote type=3D"cite"><div=
 lang=3D"EN-CA" link=3D"blue" vlink=3D"purple" style=3D"font-family: =
Helvetica; font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div =
class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"color: rgb(31, 73, 125); =
font-family: Calibri, sans-serif; font-size: 11pt; "> For example, in =
the following copied part: what is =93the current NAL =
unit=94?</span></div></div></div></blockquote><div><br></div>By this I =
mean the NAL unit that's being packed in the RTP packet for which we are =
considering whether or not to set the "M' bit (or the final such NAL =
unit, if there's more than one). =
&nbsp;</div><div><br></div><div><br><blockquote type=3D"cite"><div =
lang=3D"EN-CA" link=3D"blue" vlink=3D"purple" style=3D"font-family: =
Helvetica; font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div =
class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"color: rgb(31, 73, 125); =
font-family: Calibri, sans-serif; font-size: 11pt; "> Also, it is =
possible for non-VCL NAL unit to be between two VCL NAL units in an AU, =
the [suggested] language would suggest that the first of the two VCL NAL =
unit ends the AU =96 this is definitely =
incorrect.</span></div></div></div></blockquote><div><div><div><br></div><=
div><div>Yes, good point - thanks.</div><div><br></div><div>Here, then, =
is new wording for the suggested paragraph (to be added to the H.265 RTP =
payload format specification, just after the existing 'M bit' =
text):</div></div><div><br></div><div>----------</div><div>Unfortunately =
the contents of a NAL unit, alone, does not tell a RTP sender =
implementation whether or not the NAL unit ends an access unit. =
&nbsp;Instead, the implementation can obtain this information =
separately, from the encoder. &nbsp;If, however, this information is =
not&nbsp;available&nbsp;directly from the encoder (e.g., because the =
implementation is sending data that consists solely of a sequence of =
pre-encoded NAL units), then it must instead inspect subsequent NAL =
units, to determine whether or not the NAL unit ends an access unit. =
&nbsp;The following rule can be used:</div><div><div>&nbsp; &nbsp; A NAL =
unit ends an access unit if it is a VCL NAL unit, and&nbsp;the =
next-occurring VCL NAL unit has the high-order bit of the first byte =
after its NALU header equal to =
1.</div><div>----------</div></div><div><br></div></div><div =
apple-content-edited=3D"true"><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; border-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
border-spacing: 0px; ">Ross Finlayson<br>Live Networks, Inc.<br><a =
href=3D"http://www.live555.com/">http://www.live555.com/</a></span></span>=
</div><div><br></div></div></div><br></body></html>=

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From: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com>
To: Ross Finlayson <finlayson@live555.com>, "avt@ietf.org" <avt@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: [AVTCORE] Clarifying the H.265 RTP payload format specification's text on when to set the RTP "M" bit
Thread-Index: AQHOlK64MyApD0+0bEyc75pQI3FkapmM0zAQ
Date: Fri, 9 Aug 2013 13:01:02 +0000
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References: <C1F36850-2B72-4A98-97B7-8847C9C90CB0@live555.com> <3879D71E758A7E4AA99A35DD8D41D3D91D4C7AA7@xmb-rcd-x14.cisco.com> <A0D9B971-CDE0-4AF7-9A70-C280514AEF10@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D4C47@nasanexd02f.na.qualcomm.com> <5D781561-69CD-4E0D-9377-B4B26036E691@live555.com>
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Subject: Re: [AVTCORE] Clarifying the H.265 RTP payload format	specification's text on when to set the RTP "M" bit
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Though I don't think it is not really necessary, I think we can either simp=
ly add a note after the semantics of the M bit saying that the first slice =
header bit indicates that the slice is the first slice of a picture, or men=
tion this as part of the introduction to HEVC.

The addition is not really necessary because the first slice header bit is =
more expected to be used by video decoders for detection of the start of a =
new picture without relying on timing (because they might not be available =
to the decoder etc), while for the receivers at RTP level, they can always =
use the M bit and/or the RTP timestamp.

The semantics of the M bit is clearly specified, thus sender implementation=
s should follow the semantics. Those implementations that do not follow wha=
t is specified, e.g. the M bit semantics, when encapsulating NAL units into=
 packets, are bad implementations. It seems to me that the reasoning for yo=
ur suggestion is that you expect in the future a lot of H.265 RTP payload f=
ormat implementations would not follow the M bit. Nobody knows the future, =
but I hope this would not happen, as it is so easy to follow the semantics,=
 and I don't see a reason why implementers would not follow what is being s=
pecified. As I explained twice already, not as in RFC 6184, in the context =
of the H.265 payload draft, there is no reliability issue with the M bit to=
 be used for detection the end of an AU (after de-jittering of course - sam=
e for all approaches.

BR, YK

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Ross =
Finlayson
Sent: Thursday, August 08, 2013 8:15 PM
To: avt@ietf.org; payload@ietf.org
Subject: Re: [AVTCORE] Clarifying the H.265 RTP payload format specificatio=
n's text on when to set the RTP "M" bit

(I changed the "Subject:" line to make it clear that we're talking about th=
e (future) H.265 RTP payload format specification here; not the (existing) =
H.264 RTP payload format specification.)



First of all, implementers are expected to have basic knowledge of the vide=
o coding specification itself.

Yes.  However, in this case we seem to disagree on what constitutes 'basic =
knowledge'.  Determining whether or not a NAL unit ends an 'access unit' (i=
n the situation - common for many implementations - where this is not avail=
able directly from the encoder) is far from 'basic knowledge' of H.265.  Un=
less you can point to a specific section in the H.265 specification that de=
scribes this (in which case we could add a reference to that section), it s=
eems prudent to add a little text to our RTP payload format specification t=
hat clarifies this.


Of course, bad packetization/sender implementations [of the RTP 'M' bit] ca=
n make that unreliable - but bad packetization/sender implementations can m=
ake anything unreliable - thus they don't count.

I'm not sure what you're saying here.  I hope you're not saying that we sho=
uldn't really care about whether or not implementations set the 'M' bit cor=
rectly, because in that case I definitely disagree; that's the whole point =
of this email thread.



 Lastly, the suggested text is too vague and even incorrect.

Like Mohamed, you're inadvertently making my point for me :-)


For example, in the following copied part: what is "the current NAL unit"?

By this I mean the NAL unit that's being packed in the RTP packet for which=
 we are considering whether or not to set the "M' bit (or the final such NA=
L unit, if there's more than one).



Also, it is possible for non-VCL NAL unit to be between two VCL NAL units i=
n an AU, the [suggested] language would suggest that the first of the two V=
CL NAL unit ends the AU - this is definitely incorrect.

Yes, good point - thanks.

Here, then, is new wording for the suggested paragraph (to be added to the =
H.265 RTP payload format specification, just after the existing 'M bit' tex=
t):

----------
Unfortunately the contents of a NAL unit, alone, does not tell a RTP sender=
 implementation whether or not the NAL unit ends an access unit.  Instead, =
the implementation can obtain this information separately, from the encoder=
.  If, however, this information is not available directly from the encoder=
 (e.g., because the implementation is sending data that consists solely of =
a sequence of pre-encoded NAL units), then it must instead inspect subseque=
nt NAL units, to determine whether or not the NAL unit ends an access unit.=
  The following rule can be used:
    A NAL unit ends an access unit if it is a VCL NAL unit, and the next-oc=
curring VCL NAL unit has the high-order bit of the first byte after its NAL=
U header equal to 1.
----------

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/



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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Though I don&#8217;t thin=
k it is not really necessary, I think we can either simply add a note after=
 the semantics of the M bit saying that the first slice header
 bit indicates that the slice is the first slice of a picture, or mention t=
his as part of the introduction to HEVC.<o:p></o:p></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"><o:p>&nbsp;</o:p></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">The addition is not reall=
y necessary because the first slice header bit is more expected to be used =
by video decoders for detection of the start of a new picture
 without relying on timing (because they might not be available to the deco=
der etc), while for the receivers at RTP level, they can always use the M b=
it and/or the RTP timestamp.
<o:p></o:p></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"><o:p>&nbsp;</o:p></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">The semantics of the M bi=
t is clearly specified, thus sender implementations should follow the seman=
tics. Those implementations that do not follow what is specified,
 e.g. the M bit semantics, when encapsulating NAL units into packets, are b=
ad implementations. It seems to me that the reasoning for your suggestion i=
s that you expect in the future a lot of H.265 RTP payload format implement=
ations would not follow the M bit.
 Nobody knows the future, but I hope this would not happen, as it is so eas=
y to follow the semantics, and I don&#8217;t see a reason why implementers =
would not follow what is being specified. As I explained twice already, not=
 as in RFC 6184, in the context of the
 H.265 payload draft, there is no reliability issue with the M bit to be us=
ed for detection the end of an AU (after de-jittering of course &#8211; sam=
e for all approaches.<o:p></o:p></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"><o:p>&nbsp;</o:p></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">BR, YK<o:p></o:p></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"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span =
lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&qu=
ot;sans-serif&quot;"> avt-bounces@ietf.org [mailto:avt-bounces@ietf.org]
<b>On Behalf Of </b>Ross Finlayson<br>
<b>Sent:</b> Thursday, August 08, 2013 8:15 PM<br>
<b>To:</b> avt@ietf.org; payload@ietf.org<br>
<b>Subject:</b> Re: [AVTCORE] Clarifying the H.265 RTP payload format speci=
fication's text on when to set the RTP &quot;M&quot; bit<o:p></o:p></span><=
/p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">(I changed the &quot;Subject:&quot; line to make it =
clear that we're talking about the (future) H.265 RTP payload format specif=
ication here; not the (existing) H.264 RTP payload format specification.)<o=
:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">First of all, implementer=
s are expected to have basic knowledge of the video coding specification it=
self.</span><o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Yes. &nbsp;However, in this case we seem to disagree=
 on what constitutes 'basic knowledge'. &nbsp;Determining whether or not a =
NAL unit ends an 'access unit' (in the situation - common for many implemen=
tations - where this is not available directly
 from the encoder) is far from 'basic knowledge' of H.265. &nbsp;Unless you=
 can point to a specific section in the H.265 specification that describes =
this (in which case we could add a reference to that section), it seems pru=
dent to add a little text to our RTP
 payload format specification that clarifies this.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Of course, bad packetizat=
ion/sender implementations [of the RTP 'M' bit] can make that unreliable &#=
8211; but bad packetization/sender implementations can make anything
 unreliable &#8211; thus they don&#8217;t count.</span><o:p></o:p></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">I'm not sure what you're saying here. &nbsp;I hope y=
ou're not saying that we shouldn't really care about whether or not impleme=
ntations set the 'M' bit correctly, because in that case I definitely disag=
ree; that's the whole point of this email
 thread.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;Lastly, the suggest=
ed text is too vague and even incorrect.</span><o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">Like Mohamed, you're inadvertently making my point f=
or me :-)<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">For example, in the follo=
wing copied part: what is &#8220;the current NAL unit&#8221;?</span><o:p></=
o:p></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">By this I mean the NAL unit that's being packed in t=
he RTP packet for which we are considering whether or not to set the &quot;=
M' bit (or the final such NAL unit, if there's more than one). &nbsp;<o:p><=
/o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Also, it is possible for =
non-VCL NAL unit to be between two VCL NAL units in an AU, the [suggested] =
language would suggest that the first of the two VCL NAL
 unit ends the AU &#8211; this is definitely incorrect.</span><o:p></o:p></=
p>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal">Yes, good point - thanks.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Here, then, is new wording for the suggested paragra=
ph (to be added to the H.265 RTP payload format specification, just after t=
he existing 'M bit' text):<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">----------<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Unfortunately the contents of a NAL unit, alone, doe=
s not tell a RTP sender implementation whether or not the NAL unit ends an =
access unit. &nbsp;Instead, the implementation can obtain this information =
separately, from the encoder. &nbsp;If, however,
 this information is not&nbsp;available&nbsp;directly from the encoder (e.g=
., because the implementation is sending data that consists solely of a seq=
uence of pre-encoded NAL units), then it must instead inspect subsequent NA=
L units, to determine whether or not the NAL
 unit ends an access unit. &nbsp;The following rule can be used:<o:p></o:p>=
</p>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp; &nbsp; A NAL unit ends an access unit if it i=
s a VCL NAL unit, and&nbsp;the next-occurring VCL NAL unit has the high-ord=
er bit of the first byte after its NALU header equal to 1.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">----------<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span class=3D"apple-style-span">Ross Finlayson</spa=
n><br>
<span class=3D"apple-style-span">Live Networks, Inc.</span><br>
<span class=3D"apple-style-span"><a href=3D"http://www.live555.com/">http:/=
/www.live555.com/</a></span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

--_000_8BA7D4CEACFFE04BA2D902BF11719A83384D5FB3nasanexd02fnaqu_--

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From: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com>
To: Ross Finlayson <finlayson@live555.com>, "avt@ietf.org" <avt@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: [AVTCORE] Clarifying the H.265 RTP payload	format specification's text on when to set the RTP "M" bit
Thread-Index: AQHOlQCLVlfaHsfVP0SU09HfIMXjqJmNBWbA
Date: Fri, 9 Aug 2013 15:47:44 +0000
Message-ID: <8BA7D4CEACFFE04BA2D902BF11719A83384D6184@nasanexd02f.na.qualcomm.com>
References: <C1F36850-2B72-4A98-97B7-8847C9C90CB0@live555.com> <3879D71E758A7E4AA99A35DD8D41D3D91D4C7AA7@xmb-rcd-x14.cisco.com> <A0D9B971-CDE0-4AF7-9A70-C280514AEF10@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D4C47@nasanexd02f.na.qualcomm.com> <5D781561-69CD-4E0D-9377-B4B26036E691@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D5FB3@nasanexd02f.na.qualcomm.com>
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Subject: Re: [AVTCORE] Clarifying the H.265 RTP payload	format	specification's text on when to set the RTP "M" bit
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Typo: "Though I don't think it is not really necessary" -> "Though I don't =
think it is really necessary" or "Though I think it is not really necessary=
".

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Wang,=
 Ye-Kui
Sent: Friday, August 09, 2013 6:01 AM
To: Ross Finlayson; avt@ietf.org; payload@ietf.org
Subject: Re: [AVTCORE] Clarifying the H.265 RTP payload format specificatio=
n's text on when to set the RTP "M" bit

Though I don't think it is not really necessary, I think we can either simp=
ly add a note after the semantics of the M bit saying that the first slice =
header bit indicates that the slice is the first slice of a picture, or men=
tion this as part of the introduction to HEVC.

The addition is not really necessary because the first slice header bit is =
more expected to be used by video decoders for detection of the start of a =
new picture without relying on timing (because they might not be available =
to the decoder etc), while for the receivers at RTP level, they can always =
use the M bit and/or the RTP timestamp.

The semantics of the M bit is clearly specified, thus sender implementation=
s should follow the semantics. Those implementations that do not follow wha=
t is specified, e.g. the M bit semantics, when encapsulating NAL units into=
 packets, are bad implementations. It seems to me that the reasoning for yo=
ur suggestion is that you expect in the future a lot of H.265 RTP payload f=
ormat implementations would not follow the M bit. Nobody knows the future, =
but I hope this would not happen, as it is so easy to follow the semantics,=
 and I don't see a reason why implementers would not follow what is being s=
pecified. As I explained twice already, not as in RFC 6184, in the context =
of the H.265 payload draft, there is no reliability issue with the M bit to=
 be used for detection the end of an AU (after de-jittering of course - sam=
e for all approaches.

BR, YK

From: avt-bounces@ietf.org<mailto:avt-bounces@ietf.org> [mailto:avt-bounces=
@ietf.org] On Behalf Of Ross Finlayson
Sent: Thursday, August 08, 2013 8:15 PM
To: avt@ietf.org<mailto:avt@ietf.org>; payload@ietf.org<mailto:payload@ietf=
.org>
Subject: Re: [AVTCORE] Clarifying the H.265 RTP payload format specificatio=
n's text on when to set the RTP "M" bit

(I changed the "Subject:" line to make it clear that we're talking about th=
e (future) H.265 RTP payload format specification here; not the (existing) =
H.264 RTP payload format specification.)


First of all, implementers are expected to have basic knowledge of the vide=
o coding specification itself.

Yes.  However, in this case we seem to disagree on what constitutes 'basic =
knowledge'.  Determining whether or not a NAL unit ends an 'access unit' (i=
n the situation - common for many implementations - where this is not avail=
able directly from the encoder) is far from 'basic knowledge' of H.265.  Un=
less you can point to a specific section in the H.265 specification that de=
scribes this (in which case we could add a reference to that section), it s=
eems prudent to add a little text to our RTP payload format specification t=
hat clarifies this.


Of course, bad packetization/sender implementations [of the RTP 'M' bit] ca=
n make that unreliable - but bad packetization/sender implementations can m=
ake anything unreliable - thus they don't count.

I'm not sure what you're saying here.  I hope you're not saying that we sho=
uldn't really care about whether or not implementations set the 'M' bit cor=
rectly, because in that case I definitely disagree; that's the whole point =
of this email thread.


 Lastly, the suggested text is too vague and even incorrect.

Like Mohamed, you're inadvertently making my point for me :-)


For example, in the following copied part: what is "the current NAL unit"?

By this I mean the NAL unit that's being packed in the RTP packet for which=
 we are considering whether or not to set the "M' bit (or the final such NA=
L unit, if there's more than one).


Also, it is possible for non-VCL NAL unit to be between two VCL NAL units i=
n an AU, the [suggested] language would suggest that the first of the two V=
CL NAL unit ends the AU - this is definitely incorrect.

Yes, good point - thanks.

Here, then, is new wording for the suggested paragraph (to be added to the =
H.265 RTP payload format specification, just after the existing 'M bit' tex=
t):

----------
Unfortunately the contents of a NAL unit, alone, does not tell a RTP sender=
 implementation whether or not the NAL unit ends an access unit.  Instead, =
the implementation can obtain this information separately, from the encoder=
.  If, however, this information is not available directly from the encoder=
 (e.g., because the implementation is sending data that consists solely of =
a sequence of pre-encoded NAL units), then it must instead inspect subseque=
nt NAL units, to determine whether or not the NAL unit ends an access unit.=
  The following rule can be used:
    A NAL unit ends an access unit if it is a VCL NAL unit, and the next-oc=
curring VCL NAL unit has the high-order bit of the first byte after its NAL=
U header equal to 1.
----------

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/



--_000_8BA7D4CEACFFE04BA2D902BF11719A83384D6184nasanexd02fnaqu_
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<body lang=3D"EN-CA" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Typo: &#8220;</span><span=
 style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif=
&quot;;color:#1F497D">Though I don&#8217;t think it is not really necessary=
</span><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quo=
t;sans-serif&quot;;color:#1F497D">&#8221;
 -&gt; &#8220;</span><span style=3D"font-size:11.0pt;font-family:&quot;Cali=
bri&quot;,&quot;sans-serif&quot;;color:#1F497D">Though I don&#8217;t think =
it is really necessary</span><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&#8221; or &#8220;<=
/span><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:#1F497D">Though
 I think it is not really necessary</span><span style=3D"font-size:11.0pt;f=
ont-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&#8221=
;.<o:p></o:p></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"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span =
lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&qu=
ot;sans-serif&quot;"> avt-bounces@ietf.org [mailto:avt-bounces@ietf.org]
<b>On Behalf Of </b>Wang, Ye-Kui<br>
<b>Sent:</b> Friday, August 09, 2013 6:01 AM<br>
<b>To:</b> Ross Finlayson; avt@ietf.org; payload@ietf.org<br>
<b>Subject:</b> Re: [AVTCORE] Clarifying the H.265 RTP payload format speci=
fication's text on when to set the RTP &quot;M&quot; bit<o:p></o:p></span><=
/p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Though I don&#8217;t thin=
k it is not really necessary, I think we can either simply add a note after=
 the semantics of the M bit saying that the first slice header
 bit indicates that the slice is the first slice of a picture, or mention t=
his as part of the introduction to HEVC.<o:p></o:p></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"><o:p>&nbsp;</o:p></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">The addition is not reall=
y necessary because the first slice header bit is more expected to be used =
by video decoders for detection of the start of a new picture
 without relying on timing (because they might not be available to the deco=
der etc), while for the receivers at RTP level, they can always use the M b=
it and/or the RTP timestamp.
<o:p></o:p></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"><o:p>&nbsp;</o:p></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">The semantics of the M bi=
t is clearly specified, thus sender implementations should follow the seman=
tics. Those implementations that do not follow what is specified,
 e.g. the M bit semantics, when encapsulating NAL units into packets, are b=
ad implementations. It seems to me that the reasoning for your suggestion i=
s that you expect in the future a lot of H.265 RTP payload format implement=
ations would not follow the M bit.
 Nobody knows the future, but I hope this would not happen, as it is so eas=
y to follow the semantics, and I don&#8217;t see a reason why implementers =
would not follow what is being specified. As I explained twice already, not=
 as in RFC 6184, in the context of the
 H.265 payload draft, there is no reliability issue with the M bit to be us=
ed for detection the end of an AU (after de-jittering of course &#8211; sam=
e for all approaches.<o:p></o:p></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"><o:p>&nbsp;</o:p></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">BR, YK<o:p></o:p></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"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span =
lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&qu=
ot;sans-serif&quot;">
<a href=3D"mailto:avt-bounces@ietf.org">avt-bounces@ietf.org</a> [<a href=
=3D"mailto:avt-bounces@ietf.org">mailto:avt-bounces@ietf.org</a>]
<b>On Behalf Of </b>Ross Finlayson<br>
<b>Sent:</b> Thursday, August 08, 2013 8:15 PM<br>
<b>To:</b> <a href=3D"mailto:avt@ietf.org">avt@ietf.org</a>; <a href=3D"mai=
lto:payload@ietf.org">
payload@ietf.org</a><br>
<b>Subject:</b> Re: [AVTCORE] Clarifying the H.265 RTP payload format speci=
fication's text on when to set the RTP &quot;M&quot; bit<o:p></o:p></span><=
/p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">(I changed the &quot;Subject:&quot; line to make it =
clear that we're talking about the (future) H.265 RTP payload format specif=
ication here; not the (existing) H.264 RTP payload format specification.)<o=
:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">First of all, implementer=
s are expected to have basic knowledge of the video coding specification it=
self.</span><o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Yes. &nbsp;However, in this case we seem to disagree=
 on what constitutes 'basic knowledge'. &nbsp;Determining whether or not a =
NAL unit ends an 'access unit' (in the situation - common for many implemen=
tations - where this is not available directly
 from the encoder) is far from 'basic knowledge' of H.265. &nbsp;Unless you=
 can point to a specific section in the H.265 specification that describes =
this (in which case we could add a reference to that section), it seems pru=
dent to add a little text to our RTP
 payload format specification that clarifies this.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Of course, bad packetizat=
ion/sender implementations [of the RTP 'M' bit] can make that unreliable &#=
8211; but bad packetization/sender implementations can make anything
 unreliable &#8211; thus they don&#8217;t count.</span><o:p></o:p></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">I'm not sure what you're saying here. &nbsp;I hope y=
ou're not saying that we shouldn't really care about whether or not impleme=
ntations set the 'M' bit correctly, because in that case I definitely disag=
ree; that's the whole point of this email
 thread.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">&nbsp;<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;Lastly, the suggest=
ed text is too vague and even incorrect.</span><o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">Like Mohamed, you're inadvertently making my point f=
or me :-)<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">For example, in the follo=
wing copied part: what is &#8220;the current NAL unit&#8221;?</span><o:p></=
o:p></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">By this I mean the NAL unit that's being packed in t=
he RTP packet for which we are considering whether or not to set the &quot;=
M' bit (or the final such NAL unit, if there's more than one). &nbsp;<o:p><=
/o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Also, it is possible for =
non-VCL NAL unit to be between two VCL NAL units in an AU, the [suggested] =
language would suggest that the first of the two VCL NAL
 unit ends the AU &#8211; this is definitely incorrect.</span><o:p></o:p></=
p>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal">Yes, good point - thanks.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Here, then, is new wording for the suggested paragra=
ph (to be added to the H.265 RTP payload format specification, just after t=
he existing 'M bit' text):<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">----------<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Unfortunately the contents of a NAL unit, alone, doe=
s not tell a RTP sender implementation whether or not the NAL unit ends an =
access unit. &nbsp;Instead, the implementation can obtain this information =
separately, from the encoder. &nbsp;If, however,
 this information is not&nbsp;available&nbsp;directly from the encoder (e.g=
., because the implementation is sending data that consists solely of a seq=
uence of pre-encoded NAL units), then it must instead inspect subsequent NA=
L units, to determine whether or not the NAL
 unit ends an access unit. &nbsp;The following rule can be used:<o:p></o:p>=
</p>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp; &nbsp; A NAL unit ends an access unit if it i=
s a VCL NAL unit, and&nbsp;the next-occurring VCL NAL unit has the high-ord=
er bit of the first byte after its NALU header equal to 1.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">----------<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span class=3D"apple-style-span">Ross Finlayson</spa=
n><br>
<span class=3D"apple-style-span">Live Networks, Inc.</span><br>
<span class=3D"apple-style-span"><a href=3D"http://www.live555.com/">http:/=
/www.live555.com/</a></span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</body>
</html>

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Subject: Re: [AVTCORE] Clarifying the H.265 RTP payload format specification's text on when to set the RTP "M" bit
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> Though I don=92t think it is [] really necessary, I think we can =
either simply add a note after the semantics of the M bit saying that =
the first slice header bit indicates that the slice is the first slice =
of a picture, or mention this as part of the introduction to HEVC.

Because the new text is intended to help implementers figure out when to =
set the 'M' bit, it should be added just after the existing 'M' bit =
text.  (But why not just use the paragraph that I suggested?)


> The addition is not really necessary because the first slice header =
bit is more expected to be used by video decoders for detection of the =
start of a new picture without relying on timing (because they might not =
be available to the decoder etc), while for the receivers at RTP level, =
they can always use the M bit and/or the RTP timestamp.

Yes, but that's irrelevant here, because we're talking about how to =
clarify the text for implementers of RTP *senders*, not RTP receivers.  =
(If the current proposed RTP payload format specification were to be =
used only by the implementers of H.265/RTP receivers, then I would have =
no problem with it.  But that's not the case.)=20


> it is so easy to follow the semantics, and I don=92t see a reason why =
implementers would not follow what is being specified.

The reason is (as I've explained before) that for many RTP sender =
implementers who aren't very familiar with the H.265 codec =
specification, it will not be immediately obvious whether a NAL unit =
ends an access unit.  That's why I'm proposing adding a (very small) bit =
of extra text to clarify this.

I think the core of the disagreement here is that there are two =
different views on what a RTP payload format specification should be:

One view is that a RTP payload format specification should be the =
shortest, most concise possible document that describes how to =
send/receive RTP packets for a particular codec, for someone how is =
already very familiar with the codec specification.  =46rom the =
perspective of a codec designer, this point of view makes perfect sense: =
The codec specification is the 'core document'; the RTP payload format =
specification is logically just an 'appendix'.  Therefore, from this =
point of view, there seems little sense in adding extra, redundant text =
to the RTP payload format document, because it's information that is =
already available in the codec specification.

However, there are two problems with this point of view.  The first =
problem is that it makes the specification 'fragile'.  Someone who =
happens to misread the codec specification is more likely to implement =
the RTP payload format incorrectly.

The second problem with this point of view is a practical one.  Many (if =
not most) people who implement RTP payload formats are not codec =
experts.  They just want to know how to transmit data that they've =
received from an encoder, or (on the receiving end) want to know how to =
feed data that they've received from a RTP stream into a decoder.  It's =
not reasonable to expect them to have intimate knowledge of the codec =
specification (especially since they may want to implement the RTP =
payload formats for several codecs - not just one).  Nor will they =
always have time to read through the codec specification in detail to =
find information that could just as easily have been made available in =
the RTP payload format specification.

An alternative view of what a RTP payload format should be (and this is =
the view that is, I think, is shared by most members of the AVT(CORE) =
working group) is that a RTP payload format specification should have =
sufficient detail to allow someone to implement the payload format =
(sending and/or receiving), even if they have only a limited knowledge =
of the codec specification.  Even if this means adding text to the RTP =
payload format specification that might - from the point of view of the =
codec designer - seem somewhat redundant.

I hope this helps you understand why adding just a small bit of =
redundant text (such as the paragraph I suggested) to the RTP payload =
format specification (after the existing 'M' bit text) will help =
implementers, and makes it more likely that they will implement the RTP =
payload format correctly.


Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


--Apple-Mail=_0F3358AC-C016-4465-9173-887790A380C5
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"><base href=3D"x-msg://971/"></head><body =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div><blockquote =
type=3D"cite"><div lang=3D"EN-CA" link=3D"blue" vlink=3D"purple" =
style=3D"font-family: Helvetica; font-size: medium; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; =
"><div class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); ">Though I don=92t think =
it is [] really necessary, I think we can either simply add a note after =
the semantics of the M bit saying that the first slice header bit =
indicates that the slice is the first slice of a picture, or mention =
this as part of the introduction to =
HEVC.</span></div></div></div></blockquote><div><br></div>Because the =
new text is intended to help implementers figure out when to set the 'M' =
bit, it should be added just after the existing 'M' bit text. &nbsp;(But =
why not just use the paragraph that I =
suggested?)</div><div><br></div><div><br><blockquote type=3D"cite"><div =
lang=3D"EN-CA" link=3D"blue" vlink=3D"purple" style=3D"font-family: =
Helvetica; font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div =
class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p></o:p></span></div></div></div></blockquote><blockquote =
type=3D"cite"><div lang=3D"EN-CA" link=3D"blue" vlink=3D"purple" =
style=3D"font-family: Helvetica; font-size: medium; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; =
"><div class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"color: rgb(31, 73, 125); =
font-family: Calibri, sans-serif; font-size: 11pt; ">The addition is not =
really necessary because the first slice header bit is more expected to =
be used by video decoders for detection of the start of a new picture =
without relying on timing (because they might not be available to the =
decoder etc), while for the receivers at RTP level, they can always use =
the M bit and/or the RTP =
timestamp.</span></div></div></div></blockquote><div><br></div>Yes, but =
that's irrelevant here, because we're talking about how to clarify the =
text for implementers of RTP *senders*, not RTP receivers. &nbsp;(If the =
current proposed RTP payload format specification were to be used only =
by the implementers of H.265/RTP receivers, then I would have no problem =
with it. &nbsp;But that's not the =
case.)&nbsp;</div><div><br></div><div><br></div><div><blockquote =
type=3D"cite"><div lang=3D"EN-CA" link=3D"blue" vlink=3D"purple" =
style=3D"font-family: Helvetica; font-size: medium; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; =
"><div class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"color: rgb(31, 73, 125); =
font-family: Calibri, sans-serif; font-size: 11pt; ">it is so easy to =
follow the semantics, and I don=92t see a reason why implementers would =
not follow what is being =
specified.</span></div></div></div></blockquote><div><br></div>The =
reason is (as I've explained before) that for many RTP sender =
implementers who aren't very familiar with the H.265 codec =
specification, it will not be immediately obvious whether a NAL unit =
ends an access unit. &nbsp;That's why I'm proposing adding a (very =
small) bit of extra text to clarify this.</div><div><br></div><div>I =
think the core of the disagreement here is that there are two different =
views on what a RTP payload format specification should =
be:</div><div><br></div><div>One view is that a RTP payload format =
specification should be the shortest, most concise possible document =
that describes how to send/receive RTP packets for a particular codec, =
for someone how is already very familiar with the codec specification. =
&nbsp;=46rom the perspective of a codec designer, this point of view =
makes perfect sense: The codec specification is the 'core document'; the =
RTP payload format specification is logically just an 'appendix'. =
&nbsp;Therefore, from this point of view, there seems little sense in =
adding extra, redundant text to the RTP payload format document, because =
it's information that is already available in the codec =
specification.</div><div><br></div><div>However, there are two problems =
with this point of view. &nbsp;The first problem is that it makes the =
specification 'fragile'. &nbsp;Someone who happens to misread the codec =
specification is more likely to implement the RTP payload format =
incorrectly.</div><div><br></div><div>The second problem with this point =
of view is a practical one. &nbsp;Many (if not most) people who =
implement RTP payload formats are not codec experts. &nbsp;They =
just&nbsp;want to know how to transmit data that they've received from =
an encoder, or (on the receiving end) want to know how to feed data that =
they've received from a RTP stream into a decoder. &nbsp;It's not =
reasonable to expect them to have intimate knowledge of the codec =
specification (especially since they may want to implement the RTP =
payload formats for several codecs - not just one). &nbsp;Nor will they =
always have time to read through the codec specification in detail to =
find information that could just as easily have been made available in =
the RTP payload format specification.</div><div><br></div><div>An =
alternative view of what a RTP payload format should be (and this is the =
view that is, I think, is shared by most members of the AVT(CORE) =
working group) is that a RTP payload format specification should have =
sufficient detail to allow someone to implement the payload format =
(sending and/or receiving), even if they have only a limited knowledge =
of the codec specification. &nbsp;Even if this means adding text to the =
RTP payload format specification that might - from the point of view of =
the codec designer - seem somewhat redundant.</div><div><br></div><div>I =
hope this helps you understand why adding just a small bit of redundant =
text (such as the paragraph I suggested) to the RTP payload format =
specification (after the existing 'M' bit text) will help implementers, =
and makes it more likely that they will implement the RTP payload format =
correctly.</div><div><br></div><br><div apple-content-edited=3D"true">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; ">Ross =
Finlayson<br>Live Networks, Inc.<br><a =
href=3D"http://www.live555.com/">http://www.live555.com/</a></span></span>=

</div>
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From: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com>
To: Ross Finlayson <finlayson@live555.com>, "avt@ietf.org" <avt@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: [AVTCORE] Clarifying the H.264 RTP payload format specification's text on when to set the RTP "M" bit
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Date: Thu, 8 Aug 2013 16:40:15 +0000
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References: <C1F36850-2B72-4A98-97B7-8847C9C90CB0@live555.com> <3879D71E758A7E4AA99A35DD8D41D3D91D4C7AA7@xmb-rcd-x14.cisco.com> <A0D9B971-CDE0-4AF7-9A70-C280514AEF10@live555.com>
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Subject: Re: [AVTCORE] Clarifying the H.264 RTP payload format	specification's text on when to set the RTP "M" bit
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I share Mo's opinion here that there is no need to add additional text to t=
he H.264 or the H.265 draft in this aspect, rather the semantics of the M b=
it is sufficient.

First of all, implementers are expected to have basic knowledge of the vide=
o coding specification itself. If we try to explain all the coding-level de=
tails, there can be a lot.

Secondly, as I mentioned in the avtcore session in Berlin, the M bit for H.=
264 payload format is not reliable because RFC 6184 supports multi-time agg=
regation packets (MTAPs). This is not an issue any more in the H.265 payloa=
d as MTAPs are not supported anymore, thus the M bit can be a reliable indi=
cation. Of course, bad packetization/sender implementations can make that u=
nreliable - but bad packetization/sender implementations can make anything =
unreliable - thus they don't count.

Lastly, the suggested text is too vague and even incorrect. For example, in=
 the following copied part: what is "the current NAL unit"? Also, it is pos=
sible for non-VCL NAL unit to be between two VCL NAL units in an AU, the fo=
llowing language would suggest that the first of the two VCL NAL unit ends =
the AU - this is definitely incorrect.
    The current NAL unit ends an access unit if it is a VCL NAL unit, and i=
f either:
            - the next NAL unit is not a VCL NAL unit, or
            - the next NAL unit is a VCL NAL unit, and the high-order bit o=
f the first byte after its NALU header is 1.

BR, YK

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Ross =
Finlayson
Sent: Sunday, August 04, 2013 2:52 PM
To: avt@ietf.org
Subject: Re: [AVTCORE] Clarifying the H.264 RTP payload format specificatio=
n's text on when to set the RTP "M" bit

If you are willing to detect the last NALU of a frame by waiting for the fi=
rst VCL NALU of the next frame, that is very simple in both H.264 and H.265=
. The first bit after the VCL NALU header is set if it's the first VCL NALU=
 of a frame. This works in H.265 because the slice header explicitly has su=
ch a flag.

Thanks.  This is exactly the kind of information that would be useful for i=
mplementers.



Note that RTP receivers are technically forbidden to rely on the marker bit=
. But many implementations detect whether it is reliable, then use it as an=
 optimization to minimize latency if reliable.

Yes, it's unfortunate that there may be implementations out there that do n=
ot set the RTP 'M' bit correctly.  But to reduce the likelihood of this hap=
pening, we should try to make our RTP payload format specifications as clea=
r as is reasonable about when to set the 'M' bit.  (It is common - in non-i=
nteractive environments - for RTP transmitting applications to obtain their=
 encoded data as a byte stream.)



 I don't think the draft needs any more text beyond what it currently says.

On the contrary - I think you've explained that some additional text would =
be both useful and reasonable.  (Once again, because we're not MPEG, we sho=
uld not feel the need to fully-specify only the behavior that's required by=
 RTP *receivers*.)

So I suggest adding the following paragraph to the H.265 RTP payload format=
 specification, just after the existing 'M bit' text:

----------
Unfortunately the contents of a NAL unit, alone, does not tell a RTP sender=
 implementation whether or not the NAL unit ends an access unit.  Instead, =
the implementation can obtain this information separately, from the encoder=
.  If, however, this information is not directly available from the encoder=
 (e.g., because the implementation is sending data that consists solely of =
a sequence of pre-encoded NAL units), then it must instead inspect the next=
 NAL unit, to determine whether or not the current NAL unit ends an access =
unit.  The following rule can be used:
    The current NAL unit ends an access unit if it is a VCL NAL unit, and i=
f either:
          - the next NAL unit is not a VCL NAL unit, or
          - the next NAL unit is a VCL NAL unit, and the high-order bit of =
the first byte after its NALU header is 1.
----------

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I share Mo&#8217;s opinio=
n here that there is no need to add additional text to the H.264 or the H.2=
65 draft in this aspect, rather the semantics of the M bit is
 sufficient. <o:p></o:p></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"><o:p>&nbsp;</o:p></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">First of all, implementer=
s are expected to have basic knowledge of the video coding specification it=
self. If we try to explain all the coding-level details,
 there can be a lot.<o:p></o:p></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"><o:p>&nbsp;</o:p></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">Secondly, as I mentioned =
in the avtcore session in Berlin, the M bit for H.264 payload format is not=
 reliable because RFC 6184 supports multi-time aggregation
 packets (MTAPs). This is not an issue any more in the H.265 payload as MTA=
Ps are not supported anymore, thus the M bit can be a reliable indication. =
Of course, bad packetization/sender implementations can make that unreliabl=
e &#8211; but bad packetization/sender
 implementations can make anything unreliable &#8211; thus they don&#8217;t=
 count.<o:p></o:p></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"><o:p>&nbsp;</o:p></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">Lastly, the suggested tex=
t is too vague and even incorrect. For example, in the following copied par=
t: what is &#8220;the current NAL unit&#8221;? Also, it is possible
 for non-VCL NAL unit to be between two VCL NAL units in an AU, the followi=
ng language would suggest that the first of the two VCL NAL unit ends the A=
U &#8211; this is definitely incorrect.<o:p></o:p></span></p>
<p class=3D"MsoNormal">&nbsp; &nbsp; The current NAL unit ends an access un=
it if it is a VCL NAL unit, and if either:<o:p></o:p></p>
<p class=3D"MsoNormal"><span class=3D"apple-tab-span">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>- the next NAL unit is=
 not a VCL NAL unit, or<o:p></o:p></p>
<p class=3D"MsoNormal"><span class=3D"apple-tab-span">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>- the next NAL unit is=
 a VCL NAL unit, and the high-order bit of the first byte after its NALU he=
ader is 1.<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></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">BR, YK<o:p></o:p></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"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span =
lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&qu=
ot;sans-serif&quot;"> avt-bounces@ietf.org [mailto:avt-bounces@ietf.org]
<b>On Behalf Of </b>Ross Finlayson<br>
<b>Sent:</b> Sunday, August 04, 2013 2:52 PM<br>
<b>To:</b> avt@ietf.org<br>
<b>Subject:</b> Re: [AVTCORE] Clarifying the H.264 RTP payload format speci=
fication's text on when to set the RTP &quot;M&quot; bit<o:p></o:p></span><=
/p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">If you are willing to de=
tect the last NALU of a frame by waiting for the first VCL NALU of the next=
 frame, that is very simple in both H.264 and H.265. The first
 bit after the VCL NALU header is set if it&#8217;s the first VCL NALU of a=
 frame. This works in H.265 because the slice header explicitly has such a =
flag.</span><span lang=3D"EN-US"><o:p></o:p></span></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">Thanks. &nbsp;This is exactly the kind of informatio=
n that would be useful for implementers.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Note that RTP receivers =
are technically forbidden to rely on the marker bit. But many implementatio=
ns detect whether it is reliable, then use it as an optimization
 to minimize latency if reliable.</span><span lang=3D"EN-US"><o:p></o:p></s=
pan></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">Yes, it's unfortunate that there may be implementati=
ons out there that do not set the RTP 'M' bit correctly. &nbsp;But to reduc=
e the likelihood of this happening, we should try to make our RTP payload f=
ormat specifications as clear as is reasonable
 about when to set the 'M' bit. &nbsp;(It is common - in non-interactive en=
vironments - for RTP transmitting applications to obtain their encoded data=
 as a byte stream.)<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;I don&#8217;t thin=
k the draft needs any more text beyond what it currently says.</span><span =
lang=3D"EN-US"><o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">On the contrary - I think you've explained that some=
 additional text would be both useful and reasonable. &nbsp;(Once again, be=
cause we're not MPEG, we should not feel the need to fully-specify only the=
 behavior that's required by RTP *receivers*.)<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal">So I suggest adding the following paragraph to the H=
.265 RTP payload format specification, just after the existing 'M bit' text=
:<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">----------<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Unfortunately the contents of a NAL unit, alone, doe=
s not tell a RTP sender implementation whether or not the NAL unit ends an =
access unit. &nbsp;Instead, the implementation can obtain this information =
separately, from the encoder. &nbsp;If, however,
 this information is not directly available from the encoder (e.g., because=
 the implementation is sending data that consists solely of a sequence of p=
re-encoded NAL units), then it must instead inspect the next NAL unit, to d=
etermine whether or not the current
 NAL unit ends an access unit. &nbsp;The following rule can be used:<o:p></=
o:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp; &nbsp; The current NAL unit ends an access un=
it if it is a VCL NAL unit, and if either:<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span class=3D"apple-tab-span">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>- the next NAL unit is not a VCL N=
AL unit, or<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span class=3D"apple-tab-span">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>- the next NAL unit is a VCL NAL u=
nit, and the high-order bit of the first byte after its NALU header is 1.<o=
:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">----------<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span class=3D"apple-style-span"><span style=3D"font=
-size:13.5pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif&quot;;color=
:black">Ross Finlayson</span></span><span style=3D"font-size:13.5pt;font-fa=
mily:&quot;Helvetica&quot;,&quot;sans-serif&quot;;color:black"><br>
<span class=3D"apple-style-span">Live Networks, Inc.</span><br>
<span class=3D"apple-style-span"><a href=3D"http://www.live555.com/">http:/=
/www.live555.com/</a></span></span>
<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
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From: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com>
To: Ross Finlayson <finlayson@live555.com>, "avt@ietf.org" <avt@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: [payload] [AVTCORE] Clarifying the H.265 RTP payload format specification's text on when to set the RTP "M" bit
Thread-Index: AQHOlK64MyApD0+0bEyc75pQI3FkapmM0zAQgADx7oD//7IbsA==
Date: Fri, 9 Aug 2013 22:49:34 +0000
Message-ID: <8BA7D4CEACFFE04BA2D902BF11719A83384D690F@nasanexd02f.na.qualcomm.com>
References: <C1F36850-2B72-4A98-97B7-8847C9C90CB0@live555.com> <3879D71E758A7E4AA99A35DD8D41D3D91D4C7AA7@xmb-rcd-x14.cisco.com> <A0D9B971-CDE0-4AF7-9A70-C280514AEF10@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D4C47@nasanexd02f.na.qualcomm.com> <5D781561-69CD-4E0D-9377-B4B26036E691@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D5FB3@nasanexd02f.na.qualcomm.com> <69F25DB2-6926-4ACA-935D-90376A2A7BFA@live555.com>
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Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload format	specification's text on when to set the RTP "M" bit
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Now I understand your intention better - did not see that the note was inte=
nded for sender implementers.

For this purpose, indeed (as Mo pointed out earlier?) that first of all, th=
e sender needs make sure the RTP timestamp for each RTP packet is correct. =
As long as RTP timestamp is correct, it is also easy to know when the M bit=
 should be set. Or are you saying that there is no problem with RTP timesta=
mp setting but there is a problem for the M bit setting? If yes, please cla=
rify why.

I have no problem to putting some information in the line as you suggested,=
 if the suggested text is good. However, the suggested wording below still =
has a few problems (or shortcomings).

----------
Unfortunately the contents of a NAL unit, alone, does not tell a RTP sender=
 implementation whether or not the NAL unit ends an access unit.  Instead, =
the implementation can obtain this information separately, from the encoder=
.  If, however, this information is not available directly from the encoder=
 (e.g., because the implementation is sending data that consists solely of =
a sequence of pre-encoded NAL units), then it must instead inspect subseque=
nt NAL units, to determine whether or not the NAL unit ends an access unit.=
  The following rule can be used:
    A NAL unit ends an access unit if it is a VCL NAL unit, and the next-oc=
curring VCL NAL unit has the high-order bit of the first byte after its NAL=
U header equal to 1.
----------

The problems (or shortcomings) are:

-          I don't think it is good to say that the fact that a NAL unit it=
self does not indicate whether it is the last one of an AU is unfortunate.

-          Saying that the information can be obtained from the encoder is =
vague to me. I guess you meant the bitstream instead, as the encoder may be=
 absent e.g. when dealing pre-encoded video.

-          Saying that inspecting subsequent NAL units is a must is not rea=
lly correct, as I believe timing information can be used too - as mentioned=
 above, you got to make sure the RTP timestamp value is correctly set anywa=
y.

-          Last but not least, an AU may also end with a non-VCL NAL unit, =
e.g. a filler data NAL unit, an end of sequence NAL unit, an end of bitstre=
am NAL unit, a suffix SEI NAL unit, etc.

BR, YK

From: payload-bounces@ietf.org [mailto:payload-bounces@ietf.org] On Behalf =
Of Ross Finlayson
Sent: Friday, August 09, 2013 1:11 PM
To: avt@ietf.org; payload@ietf.org
Subject: Re: [payload] [AVTCORE] Clarifying the H.265 RTP payload format sp=
ecification's text on when to set the RTP "M" bit

Though I don't think it is [] really necessary, I think we can either simpl=
y add a note after the semantics of the M bit saying that the first slice h=
eader bit indicates that the slice is the first slice of a picture, or ment=
ion this as part of the introduction to HEVC.

Because the new text is intended to help implementers figure out when to se=
t the 'M' bit, it should be added just after the existing 'M' bit text.  (B=
ut why not just use the paragraph that I suggested?)



The addition is not really necessary because the first slice header bit is =
more expected to be used by video decoders for detection of the start of a =
new picture without relying on timing (because they might not be available =
to the decoder etc), while for the receivers at RTP level, they can always =
use the M bit and/or the RTP timestamp.

Yes, but that's irrelevant here, because we're talking about how to clarify=
 the text for implementers of RTP *senders*, not RTP receivers.  (If the cu=
rrent proposed RTP payload format specification were to be used only by the=
 implementers of H.265/RTP receivers, then I would have no problem with it.=
  But that's not the case.)


it is so easy to follow the semantics, and I don't see a reason why impleme=
nters would not follow what is being specified.

The reason is (as I've explained before) that for many RTP sender implement=
ers who aren't very familiar with the H.265 codec specification, it will no=
t be immediately obvious whether a NAL unit ends an access unit.  That's wh=
y I'm proposing adding a (very small) bit of extra text to clarify this.

I think the core of the disagreement here is that there are two different v=
iews on what a RTP payload format specification should be:

One view is that a RTP payload format specification should be the shortest,=
 most concise possible document that describes how to send/receive RTP pack=
ets for a particular codec, for someone how is already very familiar with t=
he codec specification.  From the perspective of a codec designer, this poi=
nt of view makes perfect sense: The codec specification is the 'core docume=
nt'; the RTP payload format specification is logically just an 'appendix'. =
 Therefore, from this point of view, there seems little sense in adding ext=
ra, redundant text to the RTP payload format document, because it's informa=
tion that is already available in the codec specification.

However, there are two problems with this point of view.  The first problem=
 is that it makes the specification 'fragile'.  Someone who happens to misr=
ead the codec specification is more likely to implement the RTP payload for=
mat incorrectly.

The second problem with this point of view is a practical one.  Many (if no=
t most) people who implement RTP payload formats are not codec experts.  Th=
ey just want to know how to transmit data that they've received from an enc=
oder, or (on the receiving end) want to know how to feed data that they've =
received from a RTP stream into a decoder.  It's not reasonable to expect t=
hem to have intimate knowledge of the codec specification (especially since=
 they may want to implement the RTP payload formats for several codecs - no=
t just one).  Nor will they always have time to read through the codec spec=
ification in detail to find information that could just as easily have been=
 made available in the RTP payload format specification.

An alternative view of what a RTP payload format should be (and this is the=
 view that is, I think, is shared by most members of the AVT(CORE) working =
group) is that a RTP payload format specification should have sufficient de=
tail to allow someone to implement the payload format (sending and/or recei=
ving), even if they have only a limited knowledge of the codec specificatio=
n.  Even if this means adding text to the RTP payload format specification =
that might - from the point of view of the codec designer - seem somewhat r=
edundant.

I hope this helps you understand why adding just a small bit of redundant t=
ext (such as the paragraph I suggested) to the RTP payload format specifica=
tion (after the existing 'M' bit text) will help implementers, and makes it=
 more likely that they will implement the RTP payload format correctly.


Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Now I understand your int=
ention better &#8211; did not see that the note was intended for sender imp=
lementers.&nbsp;
<o:p></o:p></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"><o:p>&nbsp;</o:p></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">For this purpose, indeed =
(as Mo pointed out earlier?) that first of all, the sender needs make sure =
the RTP timestamp for each RTP packet is correct. As long
 as RTP timestamp is correct, it is also easy to know when the M bit should=
 be set. Or are you saying that there is no problem with RTP timestamp sett=
ing but there is a problem for the M bit setting? If yes, please clarify wh=
y.<o:p></o:p></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"><o:p>&nbsp;</o:p></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">I have no problem to putt=
ing some information in the line as you suggested, if the suggested text is=
 good. However, the suggested wording below still has a
 few problems (or shortcomings).<o:p></o:p></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"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal">----------<o:p></o:p></p>
<p class=3D"MsoNormal">Unfortunately the contents of a NAL unit, alone, doe=
s not tell a RTP sender implementation whether or not the NAL unit ends an =
access unit. &nbsp;Instead, the implementation can obtain this information =
separately, from the encoder. &nbsp;If, however,
 this information is not&nbsp;available&nbsp;directly from the encoder (e.g=
., because the implementation is sending data that consists solely of a seq=
uence of pre-encoded NAL units), then it must instead inspect subsequent NA=
L units, to determine whether or not the NAL
 unit ends an access unit. &nbsp;The following rule can be used:<o:p></o:p>=
</p>
<p class=3D"MsoNormal">&nbsp; &nbsp; A NAL unit ends an access unit if it i=
s a VCL NAL unit, and&nbsp;the next-occurring VCL NAL unit has the high-ord=
er bit of the first byte after its NALU header equal to 1.<o:p></o:p></p>
<p class=3D"MsoNormal">----------<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></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">The problems (or shortcom=
ings) are:<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo2"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-=
list:Ignore">-<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I don&#8217;t thi=
nk it is good to say that the fact that a NAL unit itself does not indicate=
 whether it is the last one of an AU is unfortunate.
<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo2"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-=
list:Ignore">-<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Saying that the i=
nformation can be obtained from the encoder is vague to me. I guess you mea=
nt the bitstream instead, as the encoder may be absent
 e.g. when dealing pre-encoded video.<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo2"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-=
list:Ignore">-<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Saying that inspe=
cting subsequent NAL units is a must is not really correct, as I believe ti=
ming information can be used too &#8211; as mentioned above,
 you got to make sure the RTP timestamp value is correctly set anyway.<o:p>=
</o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo2"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-=
list:Ignore">-<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Last but not leas=
t, an AU may also end with a non-VCL NAL unit, e.g. a filler data NAL unit,=
 an end of sequence NAL unit, an end of bitstream NAL
 unit, a suffix SEI NAL unit, etc.<o:p></o:p></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"><o:p>&nbsp;</o:p></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">BR, YK<o:p></o:p></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"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span =
lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&qu=
ot;sans-serif&quot;"> payload-bounces@ietf.org [mailto:payload-bounces@ietf=
.org]
<b>On Behalf Of </b>Ross Finlayson<br>
<b>Sent:</b> Friday, August 09, 2013 1:11 PM<br>
<b>To:</b> avt@ietf.org; payload@ietf.org<br>
<b>Subject:</b> Re: [payload] [AVTCORE] Clarifying the H.265 RTP payload fo=
rmat specification's text on when to set the RTP &quot;M&quot; bit<o:p></o:=
p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Though I don&#8217;t thin=
k it is [] really necessary, I think we can either simply add a note after =
the semantics of the M bit saying that the first slice header
 bit indicates that the slice is the first slice of a picture, or mention t=
his as part of the introduction to HEVC.</span><o:p></o:p></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">Because the new text is intended to help implementer=
s figure out when to set the 'M' bit, it should be added just after the exi=
sting 'M' bit text. &nbsp;(But why not just use the paragraph that I sugges=
ted?)<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">The addition is not reall=
y necessary because the first slice header bit is more expected to be used =
by video decoders for detection of the start of a new picture
 without relying on timing (because they might not be available to the deco=
der etc), while for the receivers at RTP level, they can always use the M b=
it and/or the RTP timestamp.</span><o:p></o:p></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">Yes, but that's irrelevant here, because we're talki=
ng about how to clarify the text for implementers of RTP *senders*, not RTP=
 receivers. &nbsp;(If the current proposed RTP payload format specification=
 were to be used only by the implementers
 of H.265/RTP receivers, then I would have no problem with it. &nbsp;But th=
at's not the case.)&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">it is so easy to follow t=
he semantics, and I don&#8217;t see a reason why implementers would not fol=
low what is being specified.</span><o:p></o:p></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">The reason is (as I've explained before) that for ma=
ny RTP sender implementers who aren't very familiar with the H.265 codec sp=
ecification, it will not be immediately obvious whether a NAL unit ends an =
access unit. &nbsp;That's why I'm proposing
 adding a (very small) bit of extra text to clarify this.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">I think the core of the disagreement here is that th=
ere are two different views on what a RTP payload format specification shou=
ld be:<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">One view is that a RTP payload format specification =
should be the shortest, most concise possible document that describes how t=
o send/receive RTP packets for a particular codec, for someone how is alrea=
dy very familiar with the codec specification.
 &nbsp;From the perspective of a codec designer, this point of view makes p=
erfect sense: The codec specification is the 'core document'; the RTP paylo=
ad format specification is logically just an 'appendix'. &nbsp;Therefore, f=
rom this point of view, there seems little
 sense in adding extra, redundant text to the RTP payload format document, =
because it's information that is already available in the codec specificati=
on.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">However, there are two problems with this point of v=
iew. &nbsp;The first problem is that it makes the specification 'fragile'. =
&nbsp;Someone who happens to misread the codec specification is more likely=
 to implement the RTP payload format incorrectly.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">The second problem with this point of view is a prac=
tical one. &nbsp;Many (if not most) people who implement RTP payload format=
s are not codec experts. &nbsp;They just&nbsp;want to know how to transmit =
data that they've received from an encoder, or (on
 the receiving end) want to know how to feed data that they've received fro=
m a RTP stream into a decoder. &nbsp;It's not reasonable to expect them to =
have intimate knowledge of the codec specification (especially since they m=
ay want to implement the RTP payload
 formats for several codecs - not just one). &nbsp;Nor will they always hav=
e time to read through the codec specification in detail to find informatio=
n that could just as easily have been made available in the RTP payload for=
mat specification.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">An alternative view of what a RTP payload format sho=
uld be (and this is the view that is, I think, is shared by most members of=
 the AVT(CORE) working group) is that a RTP payload format specification sh=
ould have sufficient detail to allow
 someone to implement the payload format (sending and/or receiving), even i=
f they have only a limited knowledge of the codec specification. &nbsp;Even=
 if this means adding text to the RTP payload format specification that mig=
ht - from the point of view of the codec
 designer - seem somewhat redundant.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">I hope this helps you understand why adding just a s=
mall bit of redundant text (such as the paragraph I suggested) to the RTP p=
ayload format specification (after the existing 'M' bit text) will help imp=
lementers, and makes it more likely
 that they will implement the RTP payload format correctly.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal"><span class=3D"apple-style-span"><span style=3D"font=
-size:13.5pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif&quot;;color=
:black">Ross Finlayson</span></span><span style=3D"font-size:13.5pt;font-fa=
mily:&quot;Helvetica&quot;,&quot;sans-serif&quot;;color:black"><br>
<span class=3D"apple-style-span">Live Networks, Inc.</span><br>
<span class=3D"apple-style-span"><a href=3D"http://www.live555.com/">http:/=
/www.live555.com/</a></span></span>
<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

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Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload format specification's text on when to set the RTP "M" bit
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> Now I understand your intention better =96 did not see that the note =
was intended for sender implementers.=20

OK, thanks.


>  For this purpose, indeed (as Mo pointed out earlier?) that first of =
all, the sender needs make sure the RTP timestamp for each RTP packet is =
correct. As long as RTP timestamp is correct, it is also easy to know =
when the M bit should be set. Or are you saying that there is no problem =
with RTP timestamp setting but there is a problem for the M bit setting? =
If yes, please clarify why.

Figuring out the RTP timestamp isn't (nearly as much of) a problem, =
because this can be done by knowing the stream's frame rate.  But in =
this email thread, I'm focusing only on the RTP 'M' bit, because for it =
(unlike the timestamp) there's less of an incentive for a sender =
implementation to 'get it right'.  I'm worried that if the RTP payload =
format specification doesn't make it crystal clear when a NAL unit ends =
an 'access unit', then an implementer might be tempted to just not =
bother setting the bit at all.  I don't want to see that happen.  Hence =
this email thread.


> I have no problem to putting some information in the line as you =
suggested, if the suggested text is good. However, the suggested wording =
below still has a few problems (or shortcomings).
> =20
> ----------
> Unfortunately the contents of a NAL unit, alone, does not tell a RTP =
sender implementation whether or not the NAL unit ends an access unit.  =
Instead, the implementation can obtain this information separately, from =
the encoder.  If, however, this information is not available directly =
from the encoder (e.g., because the implementation is sending data that =
consists solely of a sequence of pre-encoded NAL units), then it must =
instead inspect subsequent NAL units, to determine whether or not the =
NAL unit ends an access unit.  The following rule can be used:
>     A NAL unit ends an access unit if it is a VCL NAL unit, and the =
next-occurring VCL NAL unit has the high-order bit of the first byte =
after its NALU header equal to 1.
> ----------
> =20
> The problems (or shortcomings) are:
> -          I don=92t think it is good to say that the fact that a NAL =
unit itself does not indicate whether it is the last one of an AU is =
unfortunate.

Why not?  It *is* "unfortunate".  I'm not intending to 'insult' H.265 =
here :-)


> -          Saying that the information can be obtained from the =
encoder is vague to me. I guess you meant the bitstream instead, as the =
encoder may be absent e.g. when dealing pre-encoded video.

I think you may have been confused by my use of the word "can" in the =
2nd sentence of my proposed paragraph.  Here is a rewording of the 2nd =
(& 3rd) sentences; I hope this will make things clearer:

	"Instead, the implementation may be able to obtain this =
information separately, from the encoder.  If, however, the =
implementation cannot obtain this information directly from the encoder =
(e.g., because the implementation is sending data that consists solely =
of a sequence of pre-encoded NAL units), then it must instead inspect =
subsequent NAL units, to determine whether or not the NAL unit ends an =
access unit."


> -          Last but not least, an AU may also end with a non-VCL NAL =
unit, e.g. a filler data NAL unit, an end of sequence NAL unit, an end =
of bitstream NAL unit, a suffix SEI NAL unit, etc.

Arggh!  Now, I hope, you see the problem :-)  Can you then help me =
figure out a more accurate wording for the rule?  In particular:
	1/ Suppose we have a VCL NAL unit, followed immediately by =
another VCL NAL unit with the first slice header bit set.  In this case, =
there's no doubt: The first VCL NAL unit definitely ends an 'access =
unit' - correct?
	2/ Suppose we have a VCL NAL unit, followed by a sequence of one =
or more non-VCL NAL units, followed immediately by a VCL NAL unit with =
the first slice header bit set.  In this case, which (if any) of the =
non-VCL NAL units ends an 'access unit'?  Can we figure out a rule =
(ideally, an easy rule, based solely on the "nal_unit_type"s) for this?


Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


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	charset=windows-1252

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"><base href=3D"x-msg://971/"></head><body =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div><blockquote =
type=3D"cite"><div lang=3D"EN-CA" link=3D"blue" vlink=3D"purple" =
style=3D"font-family: Helvetica; font-size: medium; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; =
"><div class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); ">Now I understand your =
intention better =96 did not see that the note was intended for sender =
implementers.&nbsp;</span></div></div></div></blockquote><div><br></div>OK=
, thanks.</div><div><br></div><div><br><blockquote type=3D"cite"><div =
lang=3D"EN-CA" link=3D"blue" vlink=3D"purple" style=3D"font-family: =
Helvetica; font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div =
class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p></o:p></span></div><div style=3D"margin: 0in 0in 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); ">&nbsp;</span><span style=3D"color: rgb(31, 73, 125); =
font-family: Calibri, sans-serif; font-size: 11pt; ">For this purpose, =
indeed (as Mo pointed out earlier?) that first of all, the sender needs =
make sure the RTP timestamp for each RTP packet is correct. As long as =
RTP timestamp is correct, it is also easy to know when the M bit should =
be set. Or are you saying that there is no problem with RTP timestamp =
setting but there is a problem for the M bit setting? If yes, please =
clarify =
why.</span></div></div></div></blockquote><div><br></div>Figuring out =
the RTP timestamp isn't (nearly as much of) a problem, because this can =
be done by knowing the stream's frame rate. &nbsp;But in this email =
thread, I'm focusing only on the RTP 'M' bit, because for it (unlike the =
timestamp) there's less of an incentive for a sender implementation to =
'get it right'. &nbsp;I'm worried that if the RTP payload format =
specification doesn't make it crystal clear when a NAL unit ends an =
'access unit', then an implementer might be tempted to just not bother =
setting the bit at all. &nbsp;I don't want to see that happen. =
&nbsp;Hence this email thread.</div><div><br><br><blockquote =
type=3D"cite"><div lang=3D"EN-CA" link=3D"blue" vlink=3D"purple" =
style=3D"font-family: Helvetica; font-size: medium; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; =
"><div class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"color: rgb(31, 73, 125); =
font-family: Calibri, sans-serif; font-size: 11pt; ">I have no problem =
to putting some information in the line as you suggested, if the =
suggested text is good. However, the suggested wording below still has a =
few problems (or shortcomings).</span></div><div style=3D"margin: 0in =
0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; =
color: rgb(31, 73, 125); ">&nbsp;</span></div><div style=3D"margin: 0in =
0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
">----------<o:p></o:p></div><div style=3D"margin: 0in 0in 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; ">Unfortunately =
the contents of a NAL unit, alone, does not tell a RTP sender =
implementation whether or not the NAL unit ends an access unit. =
&nbsp;Instead, the implementation can obtain this information =
separately, from the encoder. &nbsp;If, however, this information is =
not&nbsp;available&nbsp;directly from the encoder (e.g., because the =
implementation is sending data that consists solely of a sequence of =
pre-encoded NAL units), then it must instead inspect subsequent NAL =
units, to determine whether or not the NAL unit ends an access unit. =
&nbsp;The following rule can be used:<o:p></o:p></div><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; ">&nbsp; &nbsp; A NAL unit ends an access unit if it =
is a VCL NAL unit, and&nbsp;the next-occurring VCL NAL unit has the =
high-order bit of the first byte after its NALU header equal to =
1.<o:p></o:p></div><div style=3D"margin: 0in 0in 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">----------<o:p></o:p></div><div style=3D"margin: 0in 0in 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); ">&nbsp;</span></div><div style=3D"margin: 0in 0in =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; =
color: rgb(31, 73, 125); ">The problems (or shortcomings) =
are:<o:p></o:p></span></div><div style=3D"margin: 0in 0in 0.0001pt =
0.5in; font-size: 12pt; font-family: 'Times New Roman', serif; =
text-indent: -0.25in; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); "><span>-<span =
style=3D"font-style: normal; font-variant: normal; font-weight: normal; =
font-size: 7pt; line-height: normal; font-family: 'Times New Roman'; =
">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span =
class=3D"Apple-converted-space">&nbsp;</span></span></span></span><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); ">I don=92t think it is good to say that the fact that =
a NAL unit itself does not indicate whether it is the last one of an AU =
is unfortunate.</span></div></div></div></blockquote><div><br></div>Why =
not? &nbsp;It *is* "unfortunate". &nbsp;I'm not intending to 'insult' =
H.265 here :-)</div><div><br></div><div><br><blockquote type=3D"cite"><div=
 lang=3D"EN-CA" link=3D"blue" vlink=3D"purple" style=3D"font-family: =
Helvetica; font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div =
class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt 0.5in; font-size: 12pt; font-family: =
'Times New Roman', serif; text-indent: -0.25in; "><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); "><o:p></o:p></span></div><div style=3D"margin: 0in =
0in 0.0001pt 0.5in; font-size: 12pt; font-family: 'Times New Roman', =
serif; text-indent: -0.25in; "><span style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><span>-<span style=3D"font-style: normal; font-variant: normal; =
font-weight: normal; font-size: 7pt; line-height: normal; font-family: =
'Times New Roman'; =
">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span =
class=3D"Apple-converted-space">&nbsp;</span></span></span></span><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); ">Saying that the information can be obtained from the =
encoder is vague to me. I guess you meant the bitstream instead, as the =
encoder may be absent e.g. when dealing pre-encoded =
video.</span></div></div></div></blockquote><div><br></div>I think you =
may have been confused by my use of the word "can" in the 2nd sentence =
of my proposed paragraph. &nbsp;Here is a rewording of the 2nd (&amp; =
3rd) sentences; I hope this will make things =
clearer:</div><div><br></div><div><span style=3D"font-family: 'Times New =
Roman', serif; font-size: 16px; "><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>"Instead, the implementation may =
be able to obtain this information separately, from the encoder. =
&nbsp;If, however, the implementation cannot obtain this information =
directly from the encoder (e.g., because the implementation is sending =
data that consists solely of a sequence of pre-encoded NAL units), then =
it must instead inspect subsequent NAL units, to determine whether or =
not the NAL unit ends an access unit."</span></div><div><font =
face=3D"Times New Roman, serif"><span style=3D"font-size: =
16px;"><br></span></font><br><blockquote type=3D"cite"><div lang=3D"EN-CA"=
 link=3D"blue" vlink=3D"purple" style=3D"font-family: Helvetica; =
font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div =
class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt 0.5in; font-size: 12pt; font-family: =
'Times New Roman', serif; text-indent: -0.25in; "><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); "><o:p></o:p></span></div><div style=3D"margin: 0in =
0in 0.0001pt 0.5in; font-size: 12pt; font-family: 'Times New Roman', =
serif; text-indent: -0.25in; "><span style=3D"text-indent: -0.25in; =
font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, =
125); ">-<span style=3D"font-size: 7pt; font-family: 'Times New Roman'; =
">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</span></spa=
n><span style=3D"text-indent: -0.25in; font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); ">Last but not least, an =
AU may also end with a non-VCL NAL unit, e.g. a filler data NAL unit, an =
end of sequence NAL unit, an end of bitstream NAL unit, a suffix SEI NAL =
unit, =
etc.</span></div></div></div></blockquote><div><br></div></div>Arggh! =
&nbsp;Now, I hope, you see the problem :-) &nbsp;Can you then help me =
figure out a more accurate wording for the rule? &nbsp;In =
particular:<div><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>1/ Suppose we have a VCL NAL unit, followed immediately by =
another VCL NAL unit with the first slice header bit set. &nbsp;In this =
case, there's no doubt: The first VCL NAL unit definitely ends an =
'access unit' - correct?</div><div><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>2/ Suppose we have a VCL NAL =
unit, followed by a sequence of one or more non-VCL NAL units, followed =
immediately by a&nbsp;VCL NAL unit with the first slice header bit set. =
&nbsp;In this case, which (if any) of the non-VCL NAL units ends an =
'access unit'? &nbsp;Can we figure out a rule (ideally, an easy rule, =
based solely on the "nal_unit_type"s) for this?</div><br><br><div =
apple-content-edited=3D"true">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; ">Ross =
Finlayson<br>Live Networks, Inc.<br><a =
href=3D"http://www.live555.com/">http://www.live555.com/</a></span></span>=

</div>
<br></body></html>=

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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Ross Finlayson <finlayson@live555.com>, "avt@ietf.org" <avt@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: [AVTCORE] [payload] Clarifying the H.265 RTP payload format specification's text on when to set the RTP "M" bit
Thread-Index: AQHOlWQpaJZ07kYRdUCcKb/5v960lpmNqdXA
Date: Sat, 10 Aug 2013 03:15:02 +0000
Message-ID: <3879D71E758A7E4AA99A35DD8D41D3D91D4CC65C@xmb-rcd-x14.cisco.com>
References: <C1F36850-2B72-4A98-97B7-8847C9C90CB0@live555.com> <3879D71E758A7E4AA99A35DD8D41D3D91D4C7AA7@xmb-rcd-x14.cisco.com> <A0D9B971-CDE0-4AF7-9A70-C280514AEF10@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D4C47@nasanexd02f.na.qualcomm.com> <5D781561-69CD-4E0D-9377-B4B26036E691@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D5FB3@nasanexd02f.na.qualcomm.com> <69F25DB2-6926-4ACA-935D-90376A2A7BFA@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D690F@nasanexd02f.na.qualcomm.com> <A6FDCB43-F221-4BE0-BEDA-1F1E5394DDEC@live555.com>
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Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload format	specification's text on when to set the RTP "M" bit
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Hi Ross,

If we focus on the RTP timestamp rather than the marker bit, I think we wil=
l solve your problem. The timestamp MUST be set reliably since it is critic=
al for many RTP functions. The marker bit SHOULD be set reliably to allow r=
eceivers to detect end of frame without waiting for the next frame/timestam=
p, but this is not a MUST because it is just a latency optimization rather =
than a critical function.

If you are unable to set the RTP timestamp reliably, that is a much bigger =
issue than the marker bit. That would be an entirely different discussion, =
so I assume that is not the case.

If you are able to set the RTP timestamp reliably, then you must already ha=
ve some sort of frame boundary identification. In that case, you can set th=
e marker bit based on the timestamp changing, assuming you are willing to w=
ait for the next frame/timestamp before sending the last packet of a frame.

But you should realize that, in doing this, you have defeated the marker bi=
t optimization. You have just shifted the latency from the receiver to the =
sender, but the latency remains the same. In both cases you incur the extra=
 latency between the last packet of a frame and the first packet of the nex=
t frame.

If the sender incurs this latency to make the marker bit reliable, the rece=
iver will hopefully detect that it is reliable and not incur further latenc=
y. However, a simple receiver may incur further latency if it only looks fo=
r timestamp changes and has no logic to detect marker reliability to optimi=
ze out this latency. Also, if the receiver jitter buffer is currently longe=
r than the extra latency added by the sender, then the sender added the ext=
ra latency unnecessarily.

If the sender never sets the marker bit to avoid adding latency, the receiv=
er will incur the latency once it sees the marker bit is unreliable (or unc=
onditionally for simple receivers that only look for timestamp changes).

So there is not much benefit in making the marker bit reliable in this case=
, and there is even the risk that it will add unnecessary latency.

Regards,
Mo

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Ross =
Finlayson
Sent: Friday, August 09, 2013 8:48 PM
To: avt@ietf.org; payload@ietf.org
Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload format sp=
ecification's text on when to set the RTP "M" bit

Now I understand your intention better - did not see that the note was inte=
nded for sender implementers.

OK, thanks.



 For this purpose, indeed (as Mo pointed out earlier?) that first of all, t=
he sender needs make sure the RTP timestamp for each RTP packet is correct.=
 As long as RTP timestamp is correct, it is also easy to know when the M bi=
t should be set. Or are you saying that there is no problem with RTP timest=
amp setting but there is a problem for the M bit setting? If yes, please cl=
arify why.

Figuring out the RTP timestamp isn't (nearly as much of) a problem, because=
 this can be done by knowing the stream's frame rate.  But in this email th=
read, I'm focusing only on the RTP 'M' bit, because for it (unlike the time=
stamp) there's less of an incentive for a sender implementation to 'get it =
right'.  I'm worried that if the RTP payload format specification doesn't m=
ake it crystal clear when a NAL unit ends an 'access unit', then an impleme=
nter might be tempted to just not bother setting the bit at all.  I don't w=
ant to see that happen.  Hence this email thread.



I have no problem to putting some information in the line as you suggested,=
 if the suggested text is good. However, the suggested wording below still =
has a few problems (or shortcomings).

----------
Unfortunately the contents of a NAL unit, alone, does not tell a RTP sender=
 implementation whether or not the NAL unit ends an access unit.  Instead, =
the implementation can obtain this information separately, from the encoder=
.  If, however, this information is not available directly from the encoder=
 (e.g., because the implementation is sending data that consists solely of =
a sequence of pre-encoded NAL units), then it must instead inspect subseque=
nt NAL units, to determine whether or not the NAL unit ends an access unit.=
  The following rule can be used:
    A NAL unit ends an access unit if it is a VCL NAL unit, and the next-oc=
curring VCL NAL unit has the high-order bit of the first byte after its NAL=
U header equal to 1.
----------

The problems (or shortcomings) are:
-          I don't think it is good to say that the fact that a NAL unit it=
self does not indicate whether it is the last one of an AU is unfortunate.

Why not?  It *is* "unfortunate".  I'm not intending to 'insult' H.265 here =
:-)



-          Saying that the information can be obtained from the encoder is =
vague to me. I guess you meant the bitstream instead, as the encoder may be=
 absent e.g. when dealing pre-encoded video.

I think you may have been confused by my use of the word "can" in the 2nd s=
entence of my proposed paragraph.  Here is a rewording of the 2nd (& 3rd) s=
entences; I hope this will make things clearer:

            "Instead, the implementation may be able to obtain this informa=
tion separately, from the encoder.  If, however, the implementation cannot =
obtain this information directly from the encoder (e.g., because the implem=
entation is sending data that consists solely of a sequence of pre-encoded =
NAL units), then it must instead inspect subsequent NAL units, to determine=
 whether or not the NAL unit ends an access unit."



-          Last but not least, an AU may also end with a non-VCL NAL unit, =
e.g. a filler data NAL unit, an end of sequence NAL unit, an end of bitstre=
am NAL unit, a suffix SEI NAL unit, etc.

Arggh!  Now, I hope, you see the problem :-)  Can you then help me figure o=
ut a more accurate wording for the rule?  In particular:
            1/ Suppose we have a VCL NAL unit, followed immediately by anot=
her VCL NAL unit with the first slice header bit set.  In this case, there'=
s no doubt: The first VCL NAL unit definitely ends an 'access unit' - corre=
ct?
            2/ Suppose we have a VCL NAL unit, followed by a sequence of on=
e or more non-VCL NAL units, followed immediately by a VCL NAL unit with th=
e first slice header bit set.  In this case, which (if any) of the non-VCL =
NAL units ends an 'access unit'?  Can we figure out a rule (ideally, an eas=
y rule, based solely on the "nal_unit_type"s) for this?

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Hi Ross,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">If we focus on the RTP timestamp rather=
 than the marker bit, I think we will solve your problem. The timestamp MUS=
T be set reliably since it is critical for many RTP functions.
 The marker bit SHOULD be set reliably to allow receivers to detect end of =
frame without waiting for the next frame/timestamp, but this is not a MUST =
because it is just a latency optimization rather than a critical function.<=
o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">If you are unable to set the RTP timest=
amp reliably, that is a much bigger issue than the marker bit. That would b=
e an entirely different discussion, so I assume that is
 not the case.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">If you are able to set the RTP timestam=
p reliably, then you must already have some sort of frame boundary identifi=
cation. In that case, you can set the marker bit based on
 the timestamp changing, assuming you are willing to wait for the next fram=
e/timestamp before sending the last packet of a frame.<o:p></o:p></span></p=
>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">But you should realize that, in doing t=
his, you have defeated the marker bit optimization. You have just shifted t=
he latency from the receiver to the sender, but the latency
 remains the same. In both cases you incur the extra latency between the la=
st packet of a frame and the first packet of the next frame.<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">If the sender incurs this latency to ma=
ke the marker bit reliable, the receiver will hopefully detect that it is r=
eliable and not incur further latency. However, a simple
 receiver may incur further latency if it only looks for timestamp changes =
and has no logic to detect marker reliability to optimize out this latency.=
 Also, if the receiver jitter buffer is currently longer than the extra lat=
ency added by the sender, then the
 sender added the extra latency unnecessarily.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">If the sender never sets the marker bit=
 to avoid adding latency, the receiver will incur the latency once it sees =
the marker bit is unreliable (or unconditionally for simple
 receivers that only look for timestamp changes).<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">So there is not much benefit in making =
the marker bit reliable in this case, and there is even the risk that it wi=
ll add unnecessary latency.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Mo<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></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;"> avt-boun=
ces@ietf.org [mailto:avt-bounces@ietf.org]
<b>On Behalf Of </b>Ross Finlayson<br>
<b>Sent:</b> Friday, August 09, 2013 8:48 PM<br>
<b>To:</b> avt@ietf.org; payload@ietf.org<br>
<b>Subject:</b> Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload fo=
rmat specification's text on when to set the RTP &quot;M&quot; bit<o:p></o:=
p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-CA" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Now I unde=
rstand your intention better &#8211; did not see that the note was intended=
 for sender implementers.&nbsp;</span><span lang=3D"EN-CA"><o:p></o:p></spa=
n></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">OK, thanks.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-CA" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;For =
this purpose, indeed (as Mo pointed out earlier?) that first of all, the se=
nder needs make sure the RTP timestamp for each RTP packet is correct.
 As long as RTP timestamp is correct, it is also easy to know when the M bi=
t should be set. Or are you saying that there is no problem with RTP timest=
amp setting but there is a problem for the M bit setting? If yes, please cl=
arify why.</span><span lang=3D"EN-CA"><o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">Figuring out the RTP timestamp isn't (nearly as much=
 of) a problem, because this can be done by knowing the stream's frame rate=
. &nbsp;But in this email thread, I'm focusing only on the RTP 'M' bit, bec=
ause for it (unlike the timestamp) there's
 less of an incentive for a sender implementation to 'get it right'. &nbsp;=
I'm worried that if the RTP payload format specification doesn't make it cr=
ystal clear when a NAL unit ends an 'access unit', then an implementer migh=
t be tempted to just not bother setting
 the bit at all. &nbsp;I don't want to see that happen. &nbsp;Hence this em=
ail thread.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-CA" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I have no =
problem to putting some information in the line as you suggested, if the su=
ggested text is good. However, the suggested wording below
 still has a few problems (or shortcomings).</span><span lang=3D"EN-CA"><o:=
p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-CA" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><span lang=3D"EN-CA"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-CA">----------<o:p></o:p></span></p=
>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-CA">Unfortunately the contents of a=
 NAL unit, alone, does not tell a RTP sender implementation whether or not =
the NAL unit ends an access unit. &nbsp;Instead, the implementation can obt=
ain this information separately, from the
 encoder. &nbsp;If, however, this information is not&nbsp;available&nbsp;di=
rectly from the encoder (e.g., because the implementation is sending data t=
hat consists solely of a sequence of pre-encoded NAL units), then it must i=
nstead inspect subsequent NAL units, to determine
 whether or not the NAL unit ends an access unit. &nbsp;The following rule =
can be used:<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-CA">&nbsp; &nbsp; A NAL unit ends a=
n access unit if it is a VCL NAL unit, and&nbsp;the next-occurring VCL NAL =
unit has the high-order bit of the first byte after its NALU header equal t=
o 1.<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-CA">----------<o:p></o:p></span></p=
>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-CA" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><span lang=3D"EN-CA"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-CA" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">The proble=
ms (or shortcomings) are:</span><span lang=3D"EN-CA"><o:p></o:p></span></p>
</div>
<div style=3D"margin-left:.5in">
<p class=3D"MsoNormal" style=3D"text-indent:-.25in"><span lang=3D"EN-CA" st=
yle=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&qu=
ot;;color:#1F497D">-</span><span lang=3D"EN-CA" style=3D"font-size:7.0pt;co=
lor:#1F497D">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span cl=
ass=3D"apple-converted-space">&nbsp;</span></span><span lang=3D"EN-CA" styl=
e=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot=
;;color:#1F497D">I
 don&#8217;t think it is good to say that the fact that a NAL unit itself d=
oes not indicate whether it is the last one of an AU is unfortunate.</span>=
<span lang=3D"EN-CA"><o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">Why not? &nbsp;It *is* &quot;unfortunate&quot;. &nbs=
p;I'm not intending to 'insult' H.265 here :-)<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div style=3D"margin-left:.5in">
<p class=3D"MsoNormal" style=3D"text-indent:-.25in"><span lang=3D"EN-CA" st=
yle=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&qu=
ot;;color:#1F497D">-</span><span lang=3D"EN-CA" style=3D"font-size:7.0pt;co=
lor:#1F497D">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span cl=
ass=3D"apple-converted-space">&nbsp;</span></span><span lang=3D"EN-CA" styl=
e=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot=
;;color:#1F497D">Saying
 that the information can be obtained from the encoder is vague to me. I gu=
ess you meant the bitstream instead, as the encoder may be absent e.g. when=
 dealing pre-encoded video.</span><span lang=3D"EN-CA"><o:p></o:p></span></=
p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">I think you may have been confused by my use of the =
word &quot;can&quot; in the 2nd sentence of my proposed paragraph. &nbsp;He=
re is a rewording of the 2nd (&amp; 3rd) sentences; I hope this will make t=
hings clearer:<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span class=3D"apple-tab-span">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>&quot;Instead, the imp=
lementation may be able to obtain this information separately, from the enc=
oder. &nbsp;If, however, the implementation cannot obtain this information =
directly from the encoder
 (e.g., because the implementation is sending data that consists solely of =
a sequence of pre-encoded NAL units), then it must instead inspect subseque=
nt NAL units, to determine whether or not the NAL unit ends an access unit.=
&quot;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
<div>
<div style=3D"margin-left:.5in">
<p class=3D"MsoNormal" style=3D"text-indent:-.25in"><span lang=3D"EN-CA" st=
yle=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&qu=
ot;;color:#1F497D">-</span><span lang=3D"EN-CA" style=3D"font-size:7.0pt;co=
lor:#1F497D">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</=
span><span lang=3D"EN-CA" style=3D"font-size:11.0pt;font-family:&quot;Calib=
ri&quot;,&quot;sans-serif&quot;;color:#1F497D">Last
 but not least, an AU may also end with a non-VCL NAL unit, e.g. a filler d=
ata NAL unit, an end of sequence NAL unit, an end of bitstream NAL unit, a =
suffix SEI NAL unit, etc.</span><span lang=3D"EN-CA"><o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
<p class=3D"MsoNormal">Arggh! &nbsp;Now, I hope, you see the problem :-) &n=
bsp;Can you then help me figure out a more accurate wording for the rule? &=
nbsp;In particular:<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><span class=3D"apple-tab-span">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>1/ Suppose we have a V=
CL NAL unit, followed immediately by another VCL NAL unit with the first sl=
ice header bit set. &nbsp;In this case, there's no doubt: The first VCL NAL=
 unit definitely ends
 an 'access unit' - correct?<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span class=3D"apple-tab-span">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>2/ Suppose we have a V=
CL NAL unit, followed by a sequence of one or more non-VCL NAL units, follo=
wed immediately by a&nbsp;VCL NAL unit with the first slice header bit set.=
 &nbsp;In this case, which
 (if any) of the non-VCL NAL units ends an 'access unit'? &nbsp;Can we figu=
re out a rule (ideally, an easy rule, based solely on the &quot;nal_unit_ty=
pe&quot;s) for this?<o:p></o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal"><span class=3D"apple-style-span"><span style=3D"font=
-size:13.5pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif&quot;;color=
:black">Ross Finlayson</span></span><span style=3D"font-size:13.5pt;font-fa=
mily:&quot;Helvetica&quot;,&quot;sans-serif&quot;;color:black"><br>
<span class=3D"apple-style-span">Live Networks, Inc.</span><br>
<span class=3D"apple-style-span"><a href=3D"http://www.live555.com/">http:/=
/www.live555.com/</a></span></span>
<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
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Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload format	 specification's text on when to set the RTP "M" bit
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> If you are able to set the RTP timestamp reliably, then you must =
already have some sort of frame boundary identification.

Yes, we have a reliable way of determining when *VCL* NAL units cross a =
frame boundary: Look at the first slice header bit.  The one remaining =
problem (that I identified at the end of my last message) is: How to =
classify any *non-VCL* NAL units that appear between one VCL NAL unit =
and another VCL unit that has the first slice header bit.  Which of =
these belong to the last frame, and which belong to the next frame.

But perhaps this doesn't really matter?  Perhaps it's OK (from the point =
of view of receivers that check and try to optimize for the 'M' bit) if =
we always set the 'M' bit for the last VCL NAL unit of a frame, even if =
some subsequent non-VCL NAL units (appearing before the first VCL NAL =
unit of the next frame) also belong logically to the current frame??


> But you should realize that, in doing this, you have defeated the =
marker bit optimization. You have just shifted the latency from the =
receiver to the sender

No, not necessarily.  If the encoded video comes from a file (i.e., =
pre-recorded), then the latency involved in requiring the sender to =
examine the 'next' frame will usually be negligible.  But in any case, =
the situation we're talking about here (senders working from a sequence =
of pre-encoded NAL units) is one that usually occurs only in =
non-interactive environments, for which latency isn't crucial.


> So there is not much benefit in making the marker bit reliable in this =
case, and there is even the risk that it will add unnecessary latency.

That's certainly another possibility, I suppose: Recommend that if a =
sending implementation cannot determine whether or not a NAL unit ends =
an 'access unit' (frame) without having to scan subsequent NAL units, =
then it should not set the 'M' bit at all.  I don't like this, though =
(especially because it adds receiver latency without necessarily =
reducing sender latency (note above)).  But it's a possibility to =
consider...

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


--Apple-Mail=_41DC87E7-EDDD-43ED-AB0D-08FD287188A7
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=us-ascii

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"><base href=3D"x-msg://971/"></head><body =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div><blockquote =
type=3D"cite"><div lang=3D"EN-US" link=3D"blue" vlink=3D"purple" =
style=3D"font-family: Helvetica; font-size: medium; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; =
"><div class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; ">If you are able to set the RTP timestamp =
reliably, then you must already have some sort of frame boundary =
identification.</span></div></div></div></blockquote><div><br></div>Yes, =
we have a reliable way of determining when *VCL* NAL units cross a frame =
boundary: Look at the&nbsp;first slice header bit. &nbsp;The one =
remaining problem (that I identified at the end of my last message) is: =
How to classify any *non-VCL* NAL units that appear between one VCL NAL =
unit and another VCL unit that has the first slice header bit. =
&nbsp;Which of these belong to the last frame, and which belong to the =
next frame.</div><div><br></div><div>But perhaps this doesn't really =
matter? &nbsp;Perhaps it's OK (from the point of view of receivers that =
check and try to optimize for the 'M' bit) if we always set the 'M' bit =
for the last VCL NAL unit of a frame, even if some subsequent non-VCL =
NAL units (appearing before the first VCL NAL unit of the next frame) =
also belong logically to the current =
frame??</div><div><br></div><div><br><blockquote type=3D"cite"><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple" style=3D"font-family: =
Helvetica; font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div =
class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; ">But you should realize that, in doing this, you =
have defeated the marker bit optimization. You have just shifted the =
latency from the receiver to the =
sender</span></div></div></div></blockquote><div><br></div>No, not =
necessarily. &nbsp;If the encoded video comes from a file (i.e., =
pre-recorded), then the latency involved in requiring the sender to =
examine the 'next' frame will usually be negligible. &nbsp;But in any =
case, the situation we're talking about here (senders working from a =
sequence of pre-encoded NAL units) is one that usually occurs only in =
non-interactive environments, for which latency isn't =
crucial.</div><div><br></div><div><br></div><div><blockquote =
type=3D"cite"><div lang=3D"EN-US" link=3D"blue" vlink=3D"purple" =
style=3D"font-family: Helvetica; font-size: medium; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; =
"><div class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-family: Calibri, sans-serif; =
font-size: 11pt; ">So there is not much benefit in making the marker bit =
reliable in this case, and there is even the risk that it will add =
unnecessary =
latency.</span></div></div></div></blockquote><div><br></div></div>That's =
certainly another possibility, I suppose: Recommend that if a sending =
implementation cannot determine whether or not a NAL unit ends an =
'access unit' (frame) without having to scan subsequent NAL units, then =
it should not set the 'M' bit at all. &nbsp;I don't like this, though =
(especially because it adds receiver latency without necessarily =
reducing sender latency (note above)). &nbsp;But it's a possibility to =
consider...<br><br><div apple-content-edited=3D"true">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; ">Ross =
Finlayson<br>Live Networks, Inc.<br><a =
href=3D"http://www.live555.com/">http://www.live555.com/</a></span></span>=

</div>
<br></body></html>=

--Apple-Mail=_41DC87E7-EDDD-43ED-AB0D-08FD287188A7--

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From: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com>
To: Ross Finlayson <finlayson@live555.com>, "avt@ietf.org" <avt@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: [AVTCORE] [payload] Clarifying the H.265 RTP payload format specification's text on when to set the RTP "M" bit
Thread-Index: AQHOlYXebWo12KX/i0m4/b5pwipzfZmOdc6A
Date: Sat, 10 Aug 2013 13:54:45 +0000
Message-ID: <8BA7D4CEACFFE04BA2D902BF11719A83384D6FD4@nasanexd02f.na.qualcomm.com>
References: <C1F36850-2B72-4A98-97B7-8847C9C90CB0@live555.com> <3879D71E758A7E4AA99A35DD8D41D3D91D4C7AA7@xmb-rcd-x14.cisco.com> <A0D9B971-CDE0-4AF7-9A70-C280514AEF10@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D4C47@nasanexd02f.na.qualcomm.com> <5D781561-69CD-4E0D-9377-B4B26036E691@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D5FB3@nasanexd02f.na.qualcomm.com> <69F25DB2-6926-4ACA-935D-90376A2A7BFA@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D690F@nasanexd02f.na.qualcomm.com> <A6FDCB43-F221-4BE0-BEDA-1F1E5394DDEC@live555.com> <3879D71E758A7E4AA99A35DD8D41D3D91D4CC65C@xmb-rcd-x14.cisco.com> <F5C111A5-64A6-48AB-BBA8-8812EE49F7ED@live555.com>
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Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload format		specification's text on when to set the RTP "M" bit
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Hi Ross,

You just need one more little step to get everything right here, instead of=
 thinking of compromising the implementation :)

That is, read subclause 7.4.2.4.4 of the HEVC spec. The most relevant parag=
raphs are copied below for convenience (the first paragraph itself contains=
 the answer to your question, but the second paragraph may also provide hel=
pful information):

The first of any of the following NAL units after the last VCL NAL unit of =
a coded picture specifies the start of a new access unit:
-     access unit delimiter NAL unit (when present),
-     VPS NAL unit (when present),
-     SPS NAL unit (when present),
-     PPS NAL unit (when present),
-     Prefix SEI NAL unit (when present),
-     NAL units with nal_unit_type in the range of RSV_NVCL41..RSV_NVCL44 (=
when present),
-     NAL units with nal_unit_type in the range of UNSPEC48..UNSPEC55 (when=
 present),
-     first VCL NAL unit of a coded picture (always present).

The order of the coded pictures and non-VCL NAL units within an access unit=
 shall obey the following constraints:
-     When an access unit delimiter NAL unit is present, it shall be the fi=
rst NAL unit. There shall be at most one access unit delimiter NAL unit in =
any access unit.
-     When any prefix SEI NAL units are present, they shall not follow the =
last VCL NAL unit of the access unit.
-     NAL units having nal_unit_type equal to FD_NUT or SUFFIX_SEI_NUT, or =
in the range of RSV_NVCL45..RSV_NVCL47 or UNSPEC56..UNSPEC63 shall not prec=
ede the first VCL NAL unit of the coded picture.
-     When an end of sequence NAL unit is present, it shall be the last NAL=
 unit in the access unit other than an end of bitstream NAL unit (when pres=
ent).
-     When an end of bitstream NAL unit is present, it shall be the last NA=
L unit in the access unit.

NOTE - VPS NAL units, SPS NAL units, PPS NAL units, prefix SEI NAL units, o=
r NAL units with nal_unit_type in the range of RSV_NVCL41..RSV_NVCL44 or UN=
SPEC48..UNSPEC55, may be present in an access unit, but cannot follow the l=
ast VCL NAL unit of the coded picture within the access unit, as this condi=
tion would specify the start of a new access unit.

BR, YK

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Ross =
Finlayson
Sent: Friday, August 09, 2013 9:50 PM
To: avt@ietf.org; payload@ietf.org
Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload format sp=
ecification's text on when to set the RTP "M" bit

If you are able to set the RTP timestamp reliably, then you must already ha=
ve some sort of frame boundary identification.

Yes, we have a reliable way of determining when *VCL* NAL units cross a fra=
me boundary: Look at the first slice header bit.  The one remaining problem=
 (that I identified at the end of my last message) is: How to classify any =
*non-VCL* NAL units that appear between one VCL NAL unit and another VCL un=
it that has the first slice header bit.  Which of these belong to the last =
frame, and which belong to the next frame.

But perhaps this doesn't really matter?  Perhaps it's OK (from the point of=
 view of receivers that check and try to optimize for the 'M' bit) if we al=
ways set the 'M' bit for the last VCL NAL unit of a frame, even if some sub=
sequent non-VCL NAL units (appearing before the first VCL NAL unit of the n=
ext frame) also belong logically to the current frame??



But you should realize that, in doing this, you have defeated the marker bi=
t optimization. You have just shifted the latency from the receiver to the =
sender

No, not necessarily.  If the encoded video comes from a file (i.e., pre-rec=
orded), then the latency involved in requiring the sender to examine the 'n=
ext' frame will usually be negligible.  But in any case, the situation we'r=
e talking about here (senders working from a sequence of pre-encoded NAL un=
its) is one that usually occurs only in non-interactive environments, for w=
hich latency isn't crucial.


So there is not much benefit in making the marker bit reliable in this case=
, and there is even the risk that it will add unnecessary latency.

That's certainly another possibility, I suppose: Recommend that if a sendin=
g implementation cannot determine whether or not a NAL unit ends an 'access=
 unit' (frame) without having to scan subsequent NAL units, then it should =
not set the 'M' bit at all.  I don't like this, though (especially because =
it adds receiver latency without necessarily reducing sender latency (note =
above)).  But it's a possibility to consider...
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi Ross,<o:p></o:p></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"><o:p>&nbsp;</o:p></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">You just need one more li=
ttle step to get everything right here, instead of thinking of compromising=
 the implementation
</span><span style=3D"font-size:11.0pt;font-family:Wingdings;color:#1F497D"=
>J</span><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D"><o:p></o:p></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"><o:p>&nbsp;</o:p></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">That is, read subclause 7=
.4.2.4.4 of the HEVC spec. The most relevant paragraphs are copied below fo=
r convenience (the first paragraph itself contains the answer
 to your question, but the second paragraph may also provide helpful inform=
ation):<o:p></o:p></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"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal">The first of any of the following NAL units after th=
e last VCL NAL unit of a coded picture specifies the start of a new access =
unit:<o:p></o:p></p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; access unit delimite=
r NAL unit (when present),<o:p></o:p></p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; VPS NAL unit (when p=
resent),<o:p></o:p></p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; SPS NAL unit (when p=
resent),<o:p></o:p></p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; PPS NAL unit (when p=
resent),<o:p></o:p></p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; Prefix SEI NAL unit =
(when present),<o:p></o:p></p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; NAL units with nal_u=
nit_type in the range of RSV_NVCL41..RSV_NVCL44 (when present),<o:p></o:p><=
/p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; NAL units with nal_u=
nit_type in the range of UNSPEC48..UNSPEC55 (when present),<o:p></o:p></p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; first VCL NAL unit o=
f a coded picture (always present).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The order of the coded pictures and non-VCL NAL unit=
s within an access unit shall obey the following constraints:<o:p></o:p></p=
>
<p class=3D"MsoNormal" style=3D"margin-left:20.0pt;text-indent:-20.0pt">&#8=
211;&nbsp;&nbsp;&nbsp;&nbsp; When an access unit delimiter NAL unit is pres=
ent, it shall be the first NAL unit. There shall be at most one access unit=
 delimiter NAL unit in any access unit.<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:20.0pt;text-indent:-20.0pt">&#8=
211;&nbsp;&nbsp;&nbsp;&nbsp; When any prefix SEI NAL units are present, the=
y shall not follow the last VCL NAL unit of the access unit.<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:20.0pt;text-indent:-20.0pt">&#8=
211;&nbsp;&nbsp;&nbsp;&nbsp; NAL units having nal_unit_type equal to FD_NUT=
 or SUFFIX_SEI_NUT, or in the range of RSV_NVCL45..RSV_NVCL47 or UNSPEC56..=
UNSPEC63 shall not precede the first VCL NAL unit of the coded
 picture.<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:20.0pt;text-indent:-20.0pt">&#8=
211;&nbsp;&nbsp;&nbsp;&nbsp; When an end of sequence NAL unit is present, i=
t shall be the last NAL unit in the access unit other than an end of bitstr=
eam NAL unit (when present).<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:20.0pt;text-indent:-20.0pt">&#8=
211;&nbsp;&nbsp;&nbsp;&nbsp; When an end of bitstream NAL unit is present, =
it shall be the last NAL unit in the access unit.<o:p></o:p></p>
<p class=3D"Note1"><span lang=3D"EN-GB">NOTE&nbsp;&#8211;&nbsp;VPS NAL unit=
s, SPS NAL units, PPS NAL units, prefix SEI NAL units, or NAL units with na=
l_unit_type in the range of RSV_NVCL41..RSV_NVCL44 or UNSPEC48..UNSPEC55, m=
ay be present in an access unit, but cannot follow the
 last VCL NAL unit of the coded picture within the access unit, as this con=
dition would specify the start of a new access unit.<o:p></o:p></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"><o:p>&nbsp;</o:p></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">BR, YK<o:p></o:p></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"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span =
lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&qu=
ot;sans-serif&quot;"> avt-bounces@ietf.org [mailto:avt-bounces@ietf.org]
<b>On Behalf Of </b>Ross Finlayson<br>
<b>Sent:</b> Friday, August 09, 2013 9:50 PM<br>
<b>To:</b> avt@ietf.org; payload@ietf.org<br>
<b>Subject:</b> Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload fo=
rmat specification's text on when to set the RTP &quot;M&quot; bit<o:p></o:=
p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">If you are able to set t=
he RTP timestamp reliably, then you must already have some sort of frame bo=
undary identification.</span><span lang=3D"EN-US"><o:p></o:p></span></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">Yes, we have a reliable way of determining when *VCL=
* NAL units cross a frame boundary: Look at the&nbsp;first slice header bit=
. &nbsp;The one remaining problem (that I identified at the end of my last =
message) is: How to classify any *non-VCL* NAL
 units that appear between one VCL NAL unit and another VCL unit that has t=
he first slice header bit. &nbsp;Which of these belong to the last frame, a=
nd which belong to the next frame.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">But perhaps this doesn't really matter? &nbsp;Perhap=
s it's OK (from the point of view of receivers that check and try to optimi=
ze for the 'M' bit) if we always set the 'M' bit for the last VCL NAL unit =
of a frame, even if some subsequent non-VCL
 NAL units (appearing before the first VCL NAL unit of the next frame) also=
 belong logically to the current frame??<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">But you should realize t=
hat, in doing this, you have defeated the marker bit optimization. You have=
 just shifted the latency from the receiver to the sender</span><span lang=
=3D"EN-US"><o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">No, not necessarily. &nbsp;If the encoded video come=
s from a file (i.e., pre-recorded), then the latency involved in requiring =
the sender to examine the 'next' frame will usually be negligible. &nbsp;Bu=
t in any case, the situation we're talking about
 here (senders working from a sequence of pre-encoded NAL units) is one tha=
t usually occurs only in non-interactive environments, for which latency is=
n't crucial.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">So there is not much ben=
efit in making the marker bit reliable in this case, and there is even the =
risk that it will add unnecessary latency.</span><span lang=3D"EN-US"><o:p>=
</o:p></span></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">That's certainly anot=
her possibility, I suppose: Recommend that if a sending implementation cann=
ot determine whether or not a NAL unit ends an 'access unit' (frame) withou=
t having to scan subsequent NAL units,
 then it should not set the 'M' bit at all. &nbsp;I don't like this, though=
 (especially because it adds receiver latency without necessarily reducing =
sender latency (note above)). &nbsp;But it's a possibility to consider...<o=
:p></o:p></p>
<div>
<p class=3D"MsoNormal"><span class=3D"apple-style-span"><span style=3D"font=
-size:13.5pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif&quot;;color=
:black">Ross Finlayson</span></span><span style=3D"font-size:13.5pt;font-fa=
mily:&quot;Helvetica&quot;,&quot;sans-serif&quot;;color:black"><br>
<span class=3D"apple-style-span">Live Networks, Inc.</span><br>
<span class=3D"apple-style-span"><a href=3D"http://www.live555.com/">http:/=
/www.live555.com/</a></span></span>
<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

--_000_8BA7D4CEACFFE04BA2D902BF11719A83384D6FD4nasanexd02fnaqu_--

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From: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com>
To: Ross Finlayson <finlayson@live555.com>, "avt@ietf.org" <avt@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Thread-Topic: [AVTCORE] [payload] Clarifying the H.265 RTP payload	format specification's text on when to set the RTP "M" bit
Thread-Index: AQHOldHxrENwOy7dhUOuP7HJFLBE0pmOeOQQ
Date: Sat, 10 Aug 2013 14:16:46 +0000
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Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload	format		specification's text on when to set the RTP "M" bit
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Regardless of the following, I still think that the timestamp approach shou=
ld be the way to go. A good system should have a clear mapping of each vide=
o data unit (NAL unit in the context of H.265 and H.264) to an output times=
tamp, be it real-time encoding or pre-encoded content. For real-time encodi=
ng, the encoder knows what data units are generated for each input raw pict=
ure and thus it clearly knows the timestamp for each data unit. A pre-encod=
ed content should have been stored in some kind of video storage format tha=
t has timestamp associated with each data unit, e.g. in a file container ac=
cording to ISO base media file format, MPEG-2 systems, or whatever other fi=
le storage format. Trying to generate timestamps from a raw coded video bit=
stream without timing information for each coded picture based on a guess o=
f frame rate is really just a hack to me, as frame rate can be adaptive, pa=
rticularly at the beginning of a session in low-delay conversational applic=
ations or at locations of scene change or of need of full intra refresh or =
random accessibility, wherein your rate control algorithm can easily requir=
e skipping the encoding of some frames.

BR, YK

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Wang,=
 Ye-Kui
Sent: Saturday, August 10, 2013 6:55 AM
To: Ross Finlayson; avt@ietf.org; payload@ietf.org
Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload format sp=
ecification's text on when to set the RTP "M" bit

Hi Ross,

You just need one more little step to get everything right here, instead of=
 thinking of compromising the implementation :)

That is, read subclause 7.4.2.4.4 of the HEVC spec. The most relevant parag=
raphs are copied below for convenience (the first paragraph itself contains=
 the answer to your question, but the second paragraph may also provide hel=
pful information):

The first of any of the following NAL units after the last VCL NAL unit of =
a coded picture specifies the start of a new access unit:
-     access unit delimiter NAL unit (when present),
-     VPS NAL unit (when present),
-     SPS NAL unit (when present),
-     PPS NAL unit (when present),
-     Prefix SEI NAL unit (when present),
-     NAL units with nal_unit_type in the range of RSV_NVCL41..RSV_NVCL44 (=
when present),
-     NAL units with nal_unit_type in the range of UNSPEC48..UNSPEC55 (when=
 present),
-     first VCL NAL unit of a coded picture (always present).

The order of the coded pictures and non-VCL NAL units within an access unit=
 shall obey the following constraints:
-     When an access unit delimiter NAL unit is present, it shall be the fi=
rst NAL unit. There shall be at most one access unit delimiter NAL unit in =
any access unit.
-     When any prefix SEI NAL units are present, they shall not follow the =
last VCL NAL unit of the access unit.
-     NAL units having nal_unit_type equal to FD_NUT or SUFFIX_SEI_NUT, or =
in the range of RSV_NVCL45..RSV_NVCL47 or UNSPEC56..UNSPEC63 shall not prec=
ede the first VCL NAL unit of the coded picture.
-     When an end of sequence NAL unit is present, it shall be the last NAL=
 unit in the access unit other than an end of bitstream NAL unit (when pres=
ent).
-     When an end of bitstream NAL unit is present, it shall be the last NA=
L unit in the access unit.

NOTE - VPS NAL units, SPS NAL units, PPS NAL units, prefix SEI NAL units, o=
r NAL units with nal_unit_type in the range of RSV_NVCL41..RSV_NVCL44 or UN=
SPEC48..UNSPEC55, may be present in an access unit, but cannot follow the l=
ast VCL NAL unit of the coded picture within the access unit, as this condi=
tion would specify the start of a new access unit.

BR, YK

From: avt-bounces@ietf.org<mailto:avt-bounces@ietf.org> [mailto:avt-bounces=
@ietf.org] On Behalf Of Ross Finlayson
Sent: Friday, August 09, 2013 9:50 PM
To: avt@ietf.org<mailto:avt@ietf.org>; payload@ietf.org<mailto:payload@ietf=
.org>
Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload format sp=
ecification's text on when to set the RTP "M" bit

If you are able to set the RTP timestamp reliably, then you must already ha=
ve some sort of frame boundary identification.

Yes, we have a reliable way of determining when *VCL* NAL units cross a fra=
me boundary: Look at the first slice header bit.  The one remaining problem=
 (that I identified at the end of my last message) is: How to classify any =
*non-VCL* NAL units that appear between one VCL NAL unit and another VCL un=
it that has the first slice header bit.  Which of these belong to the last =
frame, and which belong to the next frame.

But perhaps this doesn't really matter?  Perhaps it's OK (from the point of=
 view of receivers that check and try to optimize for the 'M' bit) if we al=
ways set the 'M' bit for the last VCL NAL unit of a frame, even if some sub=
sequent non-VCL NAL units (appearing before the first VCL NAL unit of the n=
ext frame) also belong logically to the current frame??


But you should realize that, in doing this, you have defeated the marker bi=
t optimization. You have just shifted the latency from the receiver to the =
sender

No, not necessarily.  If the encoded video comes from a file (i.e., pre-rec=
orded), then the latency involved in requiring the sender to examine the 'n=
ext' frame will usually be negligible.  But in any case, the situation we'r=
e talking about here (senders working from a sequence of pre-encoded NAL un=
its) is one that usually occurs only in non-interactive environments, for w=
hich latency isn't crucial.


So there is not much benefit in making the marker bit reliable in this case=
, and there is even the risk that it will add unnecessary latency.

That's certainly another possibility, I suppose: Recommend that if a sendin=
g implementation cannot determine whether or not a NAL unit ends an 'access=
 unit' (frame) without having to scan subsequent NAL units, then it should =
not set the 'M' bit at all.  I don't like this, though (especially because =
it adds receiver latency without necessarily reducing sender latency (note =
above)).  But it's a possibility to consider...
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


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</head>
<body lang=3D"EN-CA" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regardless of the followi=
ng, I still think that the timestamp approach should be the way to go. A go=
od system should have a clear mapping of each video data
 unit (NAL unit in the context of H.265 and H.264) to an output timestamp, =
be it real-time encoding or pre-encoded content. For real-time encoding, th=
e encoder knows what data units are generated for each input raw picture an=
d thus it clearly knows the timestamp
 for each data unit. A pre-encoded content should have been stored in some =
kind of video storage format that has timestamp associated with each data u=
nit, e.g. in a file container according to ISO base media file format, MPEG=
-2 systems, or whatever other file
 storage format. Trying to generate timestamps from a raw coded video bitst=
ream without timing information for each coded picture based on a guess of =
frame rate is really just a hack to me, as frame rate can be adaptive, part=
icularly at the beginning of a session
 in low-delay conversational applications or at locations of scene change o=
r of need of full intra refresh or random accessibility, wherein your rate =
control algorithm can easily require skipping the encoding of some frames.
<o:p></o:p></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"><o:p>&nbsp;</o:p></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">BR, YK<o:p></o:p></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"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span =
lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&qu=
ot;sans-serif&quot;"> avt-bounces@ietf.org [mailto:avt-bounces@ietf.org]
<b>On Behalf Of </b>Wang, Ye-Kui<br>
<b>Sent:</b> Saturday, August 10, 2013 6:55 AM<br>
<b>To:</b> Ross Finlayson; avt@ietf.org; payload@ietf.org<br>
<b>Subject:</b> Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload fo=
rmat specification's text on when to set the RTP &quot;M&quot; bit<o:p></o:=
p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi Ross,<o:p></o:p></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"><o:p>&nbsp;</o:p></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">You just need one more li=
ttle step to get everything right here, instead of thinking of compromising=
 the implementation
</span><span style=3D"font-size:11.0pt;font-family:Wingdings;color:#1F497D"=
>J</span><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D"><o:p></o:p></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"><o:p>&nbsp;</o:p></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">That is, read subclause 7=
.4.2.4.4 of the HEVC spec. The most relevant paragraphs are copied below fo=
r convenience (the first paragraph itself contains the answer
 to your question, but the second paragraph may also provide helpful inform=
ation):<o:p></o:p></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"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal">The first of any of the following NAL units after th=
e last VCL NAL unit of a coded picture specifies the start of a new access =
unit:<o:p></o:p></p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; access unit delimite=
r NAL unit (when present),<o:p></o:p></p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; VPS NAL unit (when p=
resent),<o:p></o:p></p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; SPS NAL unit (when p=
resent),<o:p></o:p></p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; PPS NAL unit (when p=
resent),<o:p></o:p></p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; Prefix SEI NAL unit =
(when present),<o:p></o:p></p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; NAL units with nal_u=
nit_type in the range of RSV_NVCL41..RSV_NVCL44 (when present),<o:p></o:p><=
/p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; NAL units with nal_u=
nit_type in the range of UNSPEC48..UNSPEC55 (when present),<o:p></o:p></p>
<p class=3D"MsoNormal">&#8211;&nbsp;&nbsp;&nbsp;&nbsp; first VCL NAL unit o=
f a coded picture (always present).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The order of the coded pictures and non-VCL NAL unit=
s within an access unit shall obey the following constraints:<o:p></o:p></p=
>
<p class=3D"MsoNormal" style=3D"margin-left:20.0pt;text-indent:-20.0pt">&#8=
211;&nbsp;&nbsp;&nbsp;&nbsp; When an access unit delimiter NAL unit is pres=
ent, it shall be the first NAL unit. There shall be at most one access unit=
 delimiter NAL unit in any access unit.<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:20.0pt;text-indent:-20.0pt">&#8=
211;&nbsp;&nbsp;&nbsp;&nbsp; When any prefix SEI NAL units are present, the=
y shall not follow the last VCL NAL unit of the access unit.<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:20.0pt;text-indent:-20.0pt">&#8=
211;&nbsp;&nbsp;&nbsp;&nbsp; NAL units having nal_unit_type equal to FD_NUT=
 or SUFFIX_SEI_NUT, or in the range of RSV_NVCL45..RSV_NVCL47 or UNSPEC56..=
UNSPEC63 shall not precede the first VCL NAL unit of the coded
 picture.<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:20.0pt;text-indent:-20.0pt">&#8=
211;&nbsp;&nbsp;&nbsp;&nbsp; When an end of sequence NAL unit is present, i=
t shall be the last NAL unit in the access unit other than an end of bitstr=
eam NAL unit (when present).<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:20.0pt;text-indent:-20.0pt">&#8=
211;&nbsp;&nbsp;&nbsp;&nbsp; When an end of bitstream NAL unit is present, =
it shall be the last NAL unit in the access unit.<o:p></o:p></p>
<p class=3D"Note1"><span lang=3D"EN-GB">NOTE&nbsp;&#8211;&nbsp;VPS NAL unit=
s, SPS NAL units, PPS NAL units, prefix SEI NAL units, or NAL units with na=
l_unit_type in the range of RSV_NVCL41..RSV_NVCL44 or UNSPEC48..UNSPEC55, m=
ay be present in an access unit, but cannot follow the
 last VCL NAL unit of the coded picture within the access unit, as this con=
dition would specify the start of a new access unit.<o:p></o:p></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"><o:p>&nbsp;</o:p></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">BR, YK<o:p></o:p></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"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span =
lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&qu=
ot;sans-serif&quot;">
<a href=3D"mailto:avt-bounces@ietf.org">avt-bounces@ietf.org</a> [<a href=
=3D"mailto:avt-bounces@ietf.org">mailto:avt-bounces@ietf.org</a>]
<b>On Behalf Of </b>Ross Finlayson<br>
<b>Sent:</b> Friday, August 09, 2013 9:50 PM<br>
<b>To:</b> <a href=3D"mailto:avt@ietf.org">avt@ietf.org</a>; <a href=3D"mai=
lto:payload@ietf.org">
payload@ietf.org</a><br>
<b>Subject:</b> Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload fo=
rmat specification's text on when to set the RTP &quot;M&quot; bit<o:p></o:=
p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">If you are able to set t=
he RTP timestamp reliably, then you must already have some sort of frame bo=
undary identification.</span><span lang=3D"EN-US"><o:p></o:p></span></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">Yes, we have a reliable way of determining when *VCL=
* NAL units cross a frame boundary: Look at the&nbsp;first slice header bit=
. &nbsp;The one remaining problem (that I identified at the end of my last =
message) is: How to classify any *non-VCL* NAL
 units that appear between one VCL NAL unit and another VCL unit that has t=
he first slice header bit. &nbsp;Which of these belong to the last frame, a=
nd which belong to the next frame.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">But perhaps this doesn't really matter? &nbsp;Perhap=
s it's OK (from the point of view of receivers that check and try to optimi=
ze for the 'M' bit) if we always set the 'M' bit for the last VCL NAL unit =
of a frame, even if some subsequent non-VCL
 NAL units (appearing before the first VCL NAL unit of the next frame) also=
 belong logically to the current frame??<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">But you should realize t=
hat, in doing this, you have defeated the marker bit optimization. You have=
 just shifted the latency from the receiver to the sender</span><span lang=
=3D"EN-US"><o:p></o:p></span></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">No, not necessarily. &nbsp;If the encoded video come=
s from a file (i.e., pre-recorded), then the latency involved in requiring =
the sender to examine the 'next' frame will usually be negligible. &nbsp;Bu=
t in any case, the situation we're talking about
 here (senders working from a sequence of pre-encoded NAL units) is one tha=
t usually occurs only in non-interactive environments, for which latency is=
n't crucial.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">So there is not much ben=
efit in making the marker bit reliable in this case, and there is even the =
risk that it will add unnecessary latency.</span><span lang=3D"EN-US"><o:p>=
</o:p></span></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">That's certainly anot=
her possibility, I suppose: Recommend that if a sending implementation cann=
ot determine whether or not a NAL unit ends an 'access unit' (frame) withou=
t having to scan subsequent NAL units,
 then it should not set the 'M' bit at all. &nbsp;I don't like this, though=
 (especially because it adds receiver latency without necessarily reducing =
sender latency (note above)). &nbsp;But it's a possibility to consider...<o=
:p></o:p></p>
<div>
<p class=3D"MsoNormal"><span class=3D"apple-style-span"><span style=3D"font=
-size:13.5pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif&quot;;color=
:black">Ross Finlayson</span></span><span style=3D"font-size:13.5pt;font-fa=
mily:&quot;Helvetica&quot;,&quot;sans-serif&quot;;color:black"><br>
<span class=3D"apple-style-span">Live Networks, Inc.</span><br>
<span class=3D"apple-style-span"><a href=3D"http://www.live555.com/">http:/=
/www.live555.com/</a></span></span>
<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</body>
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From: "Roni Even" <ron.even.tlv@gmail.com>
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Date: Mon, 12 Aug 2013 17:08:07 +0300
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Subject: [AVTCORE] AVTCORE IETF87 meeting notes
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Hi, 

I posted the AVTcore meeting notes.  See
http://www.ietf.org/proceedings/87/minutes/minutes-87-avtcore

Please review.

Roni Even

AVTcore co-chair

 


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>Hi, =
<o:p></o:p></p><p class=3DMsoNormal>I posted the AVTcore meeting notes. =
&nbsp;See <a =
href=3D"http://www.ietf.org/proceedings/87/minutes/minutes-87-avtcore">ht=
tp://www.ietf.org/proceedings/87/minutes/minutes-87-avtcore</a><o:p></o:p=
></p><p class=3DMsoNormal>Please review.<o:p></o:p></p><p =
class=3DMsoNormal>Roni Even<o:p></o:p></p><p class=3DMsoNormal>AVTcore =
co-chair<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></body></html>
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From finlayson@live555.com  Mon Aug 12 12:25:32 2013
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Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload format specification's text on when to set the RTP "M" bit
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--Apple-Mail=_CE71244D-CC1C-464B-9F6F-FB73421345B7
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> You just need one more little step to get everything right here, =
instead of thinking of compromising the implementation J
> =20
> That is, read subclause 7.4.2.4.4 of the HEVC spec. The most relevant =
paragraphs are copied below for convenience (the first paragraph itself =
contains the answer to your question, but the second paragraph may also =
provide helpful information)

Thanks for the info; that was useful.

Here, then is the updated text of a paragraph - to be added after the =
existing 'M bit' text - that clarifies when/how to set it:
----------
Unfortunately the contents of a NAL unit, alone, do not tell a RTP =
sender implementation whether or not the NAL unit ends an access unit.  =
Instead, the implementation may be able to obtain this information =
separately, from the encoder.  If, however, the implementation cannot =
obtain this information directly from the encoder (e.g., because the =
implementation is sending data that consists solely of a sequence of =
pre-encoded NAL units), then it must instead inspect subsequent NAL =
units, to determine whether or not the NAL unit ends an access unit.  If =
the NAL units have already been timestamped, then these timestamps can =
be used to determine the access unit boundaries.  Otherwise, the access =
unit boundaries (and times) must be determined by inspecting the NAL =
units' contents - specifically, the "nal_unit_type" and (for VCL NAL =
units) the "first_slice_segment_in_pic_flag".  The following rule can be =
used:
	 A NAL unit (X) ends an access unit if the next-occurring VCL =
NAL unit (Y) has the high-order bit of the first byte after its NAL unit =
header equal to 1, and all NAL units, if any, between (X) and (Y) have a =
"nal_unit_type" in one of the following ranges: [32,35], 39, [41,44], or =
[48,55].
----------

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


--Apple-Mail=_CE71244D-CC1C-464B-9F6F-FB73421345B7
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"><base href=3D"x-msg://971/"></head><body =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div><blockquote =
type=3D"cite"><div lang=3D"EN-CA" link=3D"blue" vlink=3D"purple" =
style=3D"font-family: Helvetica; font-size: medium; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; =
"><div class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); ">You just need one more =
little step to get everything right here, instead of thinking of =
compromising the implementation<span =
class=3D"Apple-converted-space">&nbsp;</span></span><span =
style=3D"font-size: 11pt; font-family: Wingdings; color: rgb(31, 73, =
125); ">J</span><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); "><o:p></o:p></span></div><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); ">&nbsp;</span></div><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); ">That is, read subclause =
7.4.2.4.4 of the HEVC spec. The most relevant paragraphs are copied =
below for convenience (the first paragraph itself contains the answer to =
your question, but the second paragraph may also provide helpful =
information)</span></div></div></div></blockquote><div><br></div>Thanks =
for the info; that was useful.</div><div><br></div><div>Here, then is =
the updated text of a paragraph - to be added after the existing 'M bit' =
text - that clarifies when/how to set =
it:</div><div><div>----------</div><div>Unfortunately the contents of a =
NAL unit, alone, do not tell a RTP sender implementation whether or not =
the NAL unit ends an access unit. &nbsp;Instead, the implementation may =
be able to obtain this information separately, from the encoder. =
&nbsp;If, however, the implementation cannot obtain this information =
directly from the encoder (e.g., because the implementation is sending =
data that consists solely of a sequence of pre-encoded NAL units), then =
it must instead inspect subsequent NAL units, to determine whether or =
not the NAL unit ends an access unit. &nbsp;If the NAL units have =
already been timestamped, then these timestamps can be used to determine =
the access unit boundaries. &nbsp;Otherwise, the access unit boundaries =
(and times) must be determined by inspecting the NAL units' contents - =
specifically, the "nal_unit_type" and (for VCL NAL units) the =
"first_slice_segment_in_pic_flag". &nbsp;The following rule can be =
used:</div><div><span class=3D"Apple-tab-span" style=3D"white-space: =
pre; ">	</span>&nbsp;A NAL unit (X) ends an access unit if the =
next-occurring VCL NAL unit (Y) has the high-order bit of the first byte =
after its NAL unit header equal to 1, and all NAL units, if any, between =
(X) and (Y) have a "nal_unit_type" in one of the following ranges: =
[32,35], 39, [41,44], or =
[48,55].</div><div>----------</div><div><br></div><div></div></div><div =
apple-content-edited=3D"true"><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: 2; =
text-align: -webkit-auto; text-indent: 0px; text-transform: none; =
white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; ">Ross =
Finlayson<br>Live Networks, Inc.<br><a =
href=3D"http://www.live555.com/">http://www.live555.com/</a></span>
</div>
<br></body></html>=

--Apple-Mail=_CE71244D-CC1C-464B-9F6F-FB73421345B7--

From magnus.westerlund@ericsson.com  Wed Aug 14 00:47:27 2013
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Subject: [AVTCORE] This chair is back working
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WG,

Some of you might have noticed my absence the last couple of weeks both
at the meeting and on the list. The reason has been my now 7-week
daughter. I just wanted to notify you that I am now back working regular
hours. I will do my best to perform my outstanding tasks and other
duties. If you believe I have tasks to complete for you, don't hesitate
to remind me of them.

Cheers

Magnus Westerlund
AVTCORE Chair


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


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Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-clksrc-05.txt
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WG,

I have reviewed the -05 version and found some minor issues that I hope
the authors will be able to address in the update in progress anyway to
resolve the SDP directorate review. The Offer/answer section looks
acceptable even if I yet again needed to read it twice to make certain
that all cases are covered. The default statements in 6.1 and 6.1.1 I
have a tendency to overlook when I read 6.1.2 and 6.1.3.

1. Section 3:

   source level  : Source level information applies to a RTP media
      stream Source-Specific Media Attributes in the Session Description
      Protocol (SDP) [RFC5576] defines how source-level information is
      included into an SDP session description.


I think this definition looks a bit strange I would propose two
modifications to it to make it be clearer:

NEW:

   source level  : Source level information applies to a specific RTP
                                                         ^^^^^^^^
      media stream.  Source-Specific Media Attributes in the Session
                  ^^
      Description Protocol (SDP) [RFC5576] defines how source-level
      information is included into an SDP session description.


2. Section 4.8 and 5.4:

The ABNF actually is underspecified by lacking definitions. I run it in
BAP (http://tools.ietf.org/tools/bap/abnf.cgi) and got the following
response.

; integer UNDEFINED
; SP UNDEFINED
mediaclk-master = "a=ssrc:" integer SP clk-master-id
clk-master-id = "mediaclk:master-id=" master-id
timestamp-mediaclk-sl = "a:ssrc" integer SP timestamp-mediaclk
timestamp-mediaclk = "a=mediaclk:" mediaclock
mediaclock = sender / refclk / streamid / mediaclock-ext
sender = "sender" sender-ext
; token UNDEFINED
sender-ext = token
; DIGIT UNDEFINED
refclk = "direct" [ "=" 1*DIGIT ] [ rate ] [ direct-ext ]
rate = [ SP "rate=" integer "/" integer ]
direct-ext = token
streamid = ( "master-id=" master-id
streamid = ) / ( "IEEE1722=" avb-stream-id
streamid = ) / streamid-ext
master-id = EUI48
avb-stream-id = EUI64
; HEXDIG UNDEFINED
EUI48 = 5( 2HEXDIG ":" ) 2HEXDIG
EUI64 = 7( 2HEXDIG ":" ) 2HEXDIG
streamid-ext = token
; byte-string UNDEFINED
mediaclock-ext = token [ SP byte-string ]
; mediaclk-master defined but not used
; timestamp-mediaclk-sl defined but not used

As you see this results in a number of undefined labels. This needs to
be resolved. They belong to two categories:

A) The ones defined in RFC 5234. These could be easily resolved. The
most explicit way is to do the following and avoids forking definitions:

SP     = <See RFC 5234>
DIGIT  = <See RFC 5234>
HEXDIG = <See RFC 5234>

B) The second categories are more uncertain and which of there are three
labels:
integer, token and byte-string.

I guess token and byte-string are the definitions from SDP RFC 4566.
Thus they could be solved by:

token = <See RFC 4566>
byte-string = <See RFC 4566>

The integer may also be the definition from RFC 4566, however as the
only usage are in the a=ssrc attribute, I would recommend that one
actually uses the formal grammer from RFC 5576, i.e.:

   ssrc-attr = "ssrc:" ssrc-id SP attribute
   ; The base definition of "attribute" is in RFC 4566.
   ; (It is the content of "a=" lines.)

   ssrc-id = integer ; 0 .. 2**32 - 1

Thus I would suggest that one changes integer to "ssrc-id" and add
ssrc-id = <See RFC 5576>.

I think this will resolve all the issues, but please run the resulting
ABNF through BAP to check that you only get unused indications from the
top level constructs intended and no other indications.

You might have to make changes to this depending on how you are
resolving the SDP directorate review.

The above also applies to the ABNF in Section 5.4.


3. Section 4.8:

Then I am also not a fan of open definitions like the port one:
port = 1*DIGIT

This really means at least 1 Digit up to an infinite number of them. I
do prefer to set some upper limit on it. So far I don't know of a
protocol that uses more than 16-bit port numbers, although that could
change in the future. Considering what is deployed today I think the
following is much more appropriate:

port = 1*5DIGIT ; 0-65535

Or if you want to be more future safe, then
port = 1*10DIGIT
This allows for 32-bit integers



4. Section 5.2:
   The signalling optionally indicates a media clock offset value.  The
   offset indicates the RTP timestamp value at the epoch (time of
   origin) of the reference clock.  If no offset is signalled, the
   offset can be inferred at the receiver by examining RTCP sender
   reports which contain NTP and RTP timestamps which combined define a
   mapping.

This text actually raises two questions with me.

First, I am not certain that I understand the offset. So the indicated
value is an RTP timestamp, i.e. the integer representation of the 32-bit
value of the RTP timestamp. So the epoch time is dependent on the clock
system, but the current epoch for an NTP reference clock would be 1 jan
at 0:00 year 1900. Thus you need to take all the leap seconds etc into
account to determine how many time has actually passed since then and
then calculate how many wrapping of the timestamp this means. This seems
error prone, at least without being more explicit of how to do it for
the various reference clocks.

Secondly, what are the relative precedence between the SDP signaled
value and any values received in RTCP SR? Is it an initial date point
that then are replaced by any RTCP SR information? If not, does it
blocks the usage of the RTCP SR to update the relation between the "NTP"
field and the RTP TS value? This needs to be made more explicit.

5. Section 5.2 and Figure 7:

   A rate modifier may be specified.  The modifier is expressed as the
   ratio of two integers and modifies the rate specified or implied by
   the media description by this ratio.  If omitted, the rate is assumed
   to be the exact rate specified or implied by the media format.  For
   example, without a rate specification, the media clock for an 8 kHz
   G.711 audio stream will advance exactly 8000 units for each second
   advance in the reference clock from which it is derived.

and looking at this example in Figure 7:

          m=audio 5004 RTP/AVP 96
          a=rtpmap:96 L24/44100/2
          a=sendonly
          a=ts-refclk:ptp=IEEE1588-2008:39-A7-94-FF-FE-07-CB-D0:0
          a=mediaclk:direct=963214424 rate=1000/1001

I realize that I don't understand what the rate modifier means. Without
the rate modifier there would be 44100 RTP timestamp ticks per second of
reference clock. Does the rate modifier multiply with the nominal rate
so that this example means 441000*1000/1001 RTP TS ticks per second of
reference clock?

Some clarification may be required here.

6. Section 5.3:

   In a given system, master clock identifiers must be unique.  Such
   identifiers MAY be manually configured, however 17 octet string
   identifiers SHOULD be generated according to the "short-term
   persistent RTCP CNAME" algorithm as described in RFC6222 [RFC6222].

As draft-ietf-avtcore-6222bis are approved and will soon be published, I
think you should update this reference.
Please also check that the update of RFC 6222 does not cause any issue
for this.

7. Section 8.3:
When creating registries it is quite common to make it clear what
information  is the minimal required in a registration request.
Looking at these registries I would assume any registration would be
required to provide:

- Value
- Long name
- Reference
- Contact for registration

Maybe you should make this explicit.


8. Section 8.5.2:

   The source-level SDP attribute "master-id" defined by this document
   is registered with the IANA registry of SDP Parameters as follows:

   SDP Attribute ("att-field (source level)"):

     Attribute name:     master-id

     Long form:          Media clock source

     Type of name:       att-field

     Type of attribute:  source level

     Subject to charset: No

     Purpose:            See section 5 of this document

     Reference:          This document

     Values:             EUI48


I wonder if this part is needed at all. What I can read to in the
specification all source level usage with the master ID will have it
encapsulated in the mediaclk attribute, i.e. one will specify
a=ssrc:12345678 mediaclk:master-id=<foobar>

rather than

a=ssrc:12345678 master-id=<foobar>

Thus I wonder why there is a registation of master-id?

9. Section 9:

I wonder if not the classificiation of normative references are a bit
wrong. From my reading of this doc, RFC 3261 is not a normative
reference as the only statement related to SIP is the statement that SIP
supports renogitation and that may be used, but really are just an example.

Then I think the clock system/transports which has identifiers in this
specification are actual normative references, so RFC 5905, IEEE 1588-X
etc are in the wrong category.

Cheers

Magnus Westerlund

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From magnus.westerlund@ericsson.com  Wed Aug 14 06:17:40 2013
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Cc: Daesung Kwon <ds_kwon@ensec.re.kr>, 'Woo-Hwan Kim' <whkim5@ensec.re.kr>, "draft-ietf-avtcore-srtp-aes-gcm@tools.ietf.org" <draft-ietf-avtcore-srtp-aes-gcm@tools.ietf.org>, "avt@ietf.org" <avt@ietf.org>, Je Hong Park <jhpark@ensec.re.kr>
Subject: Re: [AVTCORE] Comments on the draft-ietf-avtcore-srtp-aes-gcm-06
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Kevin,

I looked over Woo-Hwan's comments and clearly 1 and 2 are resolved by
-07. Thus I wonder over issue nr 3. In -07 I do find that C_MAX for CCM
is 2^24-16, that appears to be the value he thinks should be 2^28-16 or
was the issue that is was written as 2^24+16?

Can you please confirm that -07 is correct as currently written using
2^24-16

Thanks

Magnus


On 2013-07-11 14:46, Igoe, Kevin M. wrote:

> *From:*woohwankim@gmail.com [mailto:woohwankim@gmail.com] *On Behalf Of
> *Woo-Hwan Kim
> *Sent:* Thursday, June 27, 2013 1:23 AM
> *To:* avt@ietf.org; draft-ietf-avtcore-srtp-aes-gcm@tools.ietf.org
> *Cc:* Je Hong Park; Daesung Kwon; Magnus Westerlund
> *Subject:* [AVTCORE] Comments on the draft-ietf-avtcore-srtp-aes-gcm-06
> 

> 
> 3. Section 6, Section 11.1
>  (2^24)-16 octets => (2^28)-16 octets
> 
>  
> 
> Regards, Woo-Hwan Kim
> 


-- 

Magnus Westerlund

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Hi,

I have checked the -07 version against these comments and think most are
resolved. I will comment where I believe there are still an issue.


On 2013-04-18 17:09, Magnus Westerlund wrote:
> Authors and WG,
> 
> In my review of the draft in preparing the Document Shepherd's writeup I
> found some issues that I believe needs to be fixed.
> 
> 1) Section 14.1:
> Security description [RFC 4568] defines SRTP "crypto suites"; a
>    crypto suite corresponds to a particular AEAD algorithm in SRTP.
> 
> As the registry is located on the SDP page it can be difficult to find
> and it also not named as written I propose to change this to:
> 
> Security description [RFC 4568] defines the "SRTP Crypto Suite
> Registrations" registry on the "Session Description Protocol (SDP)
> Security Descriptions" page currently located at
> "http://www.iana.org/assignments/sdp-security-descriptions/sdp-security-descriptions.xml";
> a
>    crypto suite corresponds to a particular AEAD algorithm in SRTP.


Section 15.1 now states with:

   Session description [RFC4568] defines SRTP "crypto suites".  A crypto
   suite corresponds to a particular AEAD algorithm in SRTP.  In order
   to allow SDP to signal the use of the algorithms defined in this
   document, IANA will register the following crypto suites into the
   subregistry for SRTP crypto suites under Session Description Protocol
   (SDP) Parameters:


I think this needs to be slightly corrected in two places:
 Security descriptions [RFC4568] defines SRTP "crypto suites".  A crypto
 ^^^^^^^^
   suite corresponds to a particular AEAD algorithm in SRTP.  In order
   to allow SDP to signal the use of the algorithms defined in this
   document, IANA will register the following crypto suites into the
   subregistry for SRTP crypto suites under "Session Description
                                            ^
   Protocol (SDP) Security Descriptions" Parameters:
                  ^^^^^^^^^^^^^^^^^^^^^^

This corrects both the common name of the RFC and the correct name of
the page for the sub-registries.




> 
> 
> 3) Section 14.2:
> 
> DTLS-SRTP [RFC5764] defines a DTLS-SRTP "SRTP Protection Profile";
> 
> This registry is actually named "DTLS-SRTP Protection Profiles" on
> IANA's page.

Lets wait and see what IANA says about this. There is a discrepancy
between IANA's page and what is written in the RFC.


> 
> 
> 4) IANA registry:
> 
>    On the SRTP policy Type/Value list (derived from Table 6.10.1.a of
>    [RFC3830]) we request the following addition:
> 
>       Type | Meaning                         | Possible values
>       ----------------------------------------------------------------
>        TBD | AEAD authentication tag length  | 8, 12, or 16 (in octets)
> 
> 
> This registry is called :
> 
> MIKEY Security Protocol Parameters at
> http://www.iana.org/assignments/mikey-payloads/mikey-payloads.xml
> 
> Can you please confirm that it is this registry you want to add to?

Can you please confirm that this is the right registry and in that case
use the explicit name: "MIKEY Security Protocol Parameters".


> 
> 
> 5) IANA registry:
> 
> 14.4. AEAD registry
> 
>    We request that IANA make the following additions to the AEAD
>    registry:
> 
>                  AEAD_AES_128_CCM_12     = TBD
>                  AEAD_AES_256_CCM_12     = TBD
> 
> 
> You are not using the exact correct names for the registry:
> http://www.iana.org/assignments/aead-parameters/aead-parameters.xml
> 
> Would suggest that you should write this as:
> 
>  We request that IANA make the following additions to the IANA
> Authenticated Encryption with Associated Data (AEAD) Parameters page's
> registry for "AEAD Algorithms":

This appears to not been addressed either. I would strongly suggest that
you fix it.

cheers

Magnus Westerlund

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Cc: avt@ietf.org
Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-aria-srtp-03.txt
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Authors and WG,

I have reviewed this new version. Thanks for addressing my issue. During
this review I only noticed a small number of minor issues.

Note, I have not verified in any way the test vectors. Nor am I certain
that I can spot any fact error regarding the crypto-algorithms. What I
have done is reviewed the draft consistency and correctness in their
actions to IANA and towards SRTP.


1. Section 5.3:


                              +--------------------------------------+
                              | Encryption | Encryption | AEAD Auth. |
                              | Algorithm  | Key Length | Tag Length |
                              +======================================+
    SRTP_ARIA_128_CTR_HMAC_80 |  ARIA-CTR  | 16 octets  |  80 bits   |
    SRTP_ARIA_128_CTR_HMAC_32 |  ARIA-CTR  | 16 octets  |  32 bits   |
    SRTP_ARIA_192_CTR_HMAC_80 |  ARIA-CTR  | 24 octets  |  80 bits   |
    SRTP_ARIA_192_CTR_HMAC_32 |  ARIA-CTR  | 24 octets  |  32 bits   |
    SRTP_ARIA_256_CTR_HMAC_80 |  ARIA-CTR  | 32 octets  |  80 bits   |
    SRTP_ARIA_256_CTR_HMAC_32 |  ARIA-CTR  | 32 octets  |  32 bits   |



Kim, et al.             Expires December 29, 2013              [Page 18]

Internet-Draft           ARIA Algorithm for SRTP               June 2013


    SRTP_AEAD_ARIA_128_GCM    |  ARIA-GCM  | 16 octets  | 128 bits   |
    SRTP_AEAD_ARIA_128_CCM    |  ARIA-CCM  | 16 octets  | 128 bits   |
    SRTP_AEAD_ARIA_128_GCM_12 |  ARIA-GCM  | 16 octets  |  96 bits   |
    SRTP_AEAD_ARIA_128_CCM_12 |  ARIA-CCM  | 16 octets  |  96 bits   |
    SRTP_AEAD_ARIA_128_GCM_8  |  ARIA-GCM  | 16 octets  |  64 bits   |
    SRTP_AEAD_ARIA_128_CCM_8  |  ARIA-CCM  | 16 octets  |  64 bits   |
    SRTP_AEAD_ARIA_256_GCM    |  ARIA-GCM  | 32 octets  | 128 bits   |
    SRTP_AEAD_ARIA_256_CCM    |  ARIA-CCM  | 32 octets  | 128 bits   |
    SRTP_AEAD_ARIA_256_GCM_12 |  ARIA-GCM  | 32 octets  |  96 bits   |
    SRTP_AEAD_ARIA_256_CCM_12 |  ARIA-CCM  | 32 octets  |  96 bits   |
    SRTP_AEAD_ARIA_256_GCM_8  |  ARIA-GCM  | 32 octets  |  64 bits   |
    SRTP_AEAD_ARIA_256_CCM_8  |  ARIA-CCM  | 32 octets  |  64 bits   |
                              +======================================+

           Figure 1: Mapping MIKEY parameters to AEAD algorithm

Shouldn't you split this into two tables as the last column and legend
are wrong for the first ARIA CTR + SHA-1 HMAC suits which are not AEAD
suites? That way you can use the correct labels on the last column for
auth tag lengths and in the legend.


2. Section A.2. and A.3

In section A.1 I do understand the structure for the test vectors. They
contain both just the Encrypted part and the full RTP header + payload +
ROC that authentication is calculated over, and then the resulting output.

However in A.2 and A.3 where there are AEAD algorithms I am bit
surprised over the split in Encrypted RTP payload and Authentication
tag. Due to AEAD shouldn't the relevant unit to test as input be the
full SRTP packet which contains both the encrypted and the data that is
just authenticated including the AEAD output in the payload location?

You might need a bit more explanation what the test vectors really are
so that one can correctly use them to verify ones implementation.



Next Steps:

Please address the above issues.

I personally think this is ready for going forward to WG last call when
the above is addressed. However, in that last call we will need to get a
review from someone that has sufficient crypto knowledge to be reduce
the risk for any such error making it through. Thus I intended to make
the WG last call's completion dependent on getting such a review.

Cheers

Magnus




On 2013-06-27 07:43, 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 Core Maintenance Working Group of the IETF.
> 
> 	Title           : The ARIA Algorithm and Its Use with the Secure Real-time Transport Protocol(SRTP)
> 	Author(s)       : Woo-Hwan Kim
>                           Jungkeun Lee
>                           Dong-Chan Kim
>                           Je-Hong Park
>                           Daesung Kwon
> 	Filename        : draft-ietf-avtcore-aria-srtp-03.txt
> 	Pages           : 32
> 	Date            : 2013-06-26
> 
> Abstract:
>    This document describes the use of the ARIA block cipher algorithm
>    within the Secure Real-time Transport Protocol (SRTP) for providing
>    confidentiality for the Real-time Transport Protocol (RTP) traffic
>    and for the control traffic for RTP, the Real-time Transport Control
>    Protocol (RTCP).  It details three modes of operation (CTR, CCM, GCM)
>    and a SRTP Key Derivation Function for ARIA.
> 
> 
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-avtcore-aria-srtp
> 
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-avtcore-aria-srtp-03
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-aria-srtp-03
> 
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
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> 


-- 

Magnus Westerlund

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To: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>, "draft-ietf-avtcore-idms@tools.ietf.org" <draft-ietf-avtcore-idms@tools.ietf.org>
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Cc: "avt@ietf.org" <avt@ietf.org>
Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-idms-12.txt
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Hi,

Looking at this latest version I have a comment.

section 14.4 defines the SPST registry.

Why isn't is adding the ETSI defined values from the start. If they are
known and have well known reference I don't see any issues with putting
in the values into the registry at its creation.

Value           Name                         Reference
   -----           ----                         ---------
   1               Synchronization Client       section 7 of [RFCXXXX]
   2               <Appropriate Name>           [TS183063]
   3               <Appropriate Name>           [TS183063]
   4               <Appropriate Name>           [TS183063]


Cheers

Magnus


On 2013-07-29 10:48, Brandenburg, R. (Ray) van wrote:
> Hi,
> 
> This version of the draft includes the final comments received during
> IETF LC and from Martin Thomson as Appsdir reviewer.
> 
> Best regards,
> 
> Ray
> 
> On 29 jul. 2013, at 10:45, "internet-drafts@ietf.org"
> <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
>> Core Maintenance Working Group of the IETF.
>> 
>> Title           : Inter-destination Media Synchronization using the
>> RTP Control Protocol (RTCP) Author(s)       : Ray van Brandenburg 
>> Hans Stokking Oskar van Deventer Fernando Boronat Mario Montagud 
>> Kevin Gross Filename        : draft-ietf-avtcore-idms-12.txt Pages
>> : 22 Date            : 2013-07-29
>> 
>> Abstract: This document defines a new RTP Control Protocol (RTCP)
>> Packet Type and an RTCP Extended Report (XR) Block Type to be used
>> for achieving Inter-Destination Media Synchronization (IDMS).  IDMS
>> is the process of synchronizing playout across multiple
>> geographically distributed media receivers.  Using the RTCP XR IDMS
>> Report Block defined in this document, media playout information
>> from participants in a synchronization group can be collected.
>> Based on the collected information, an RTCP IDMS Settings Packet
>> can then be sent to distribute a common target playout point to
>> which all the distributed receivers, sharing a media experience,
>> can synchronize.
>> 
>> Typical use cases in which IDMS is useful are social TV, shared 
>> service control (i.e. applications where two or more
>> geographically separated users are watching a media stream
>> together), distance learning, networked video walls, networked
>> loudspeakers, etc.
>> 
>> 
>> The IETF datatracker status page for this draft is: 
>> https://datatracker.ietf.org/doc/draft-ietf-avtcore-idms
>> 
>> There's also a htmlized version available at: 
>> http://tools.ietf.org/html/draft-ietf-avtcore-idms-12
>> 
>> A diff from the previous version is available at: 
>> http://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-idms-12
>> 
>> 
>> Please note that it may take a couple of minutes from the time of
>> submission until the htmlized version and diff are available at
>> tools.ietf.org.
>> 
>> Internet-Drafts are also available by anonymous FTP at: 
>> ftp://ftp.ietf.org/internet-drafts/
>> 
>> _______________________________________________ Audio/Video
>> Transport Core Maintenance avt@ietf.org 
>> https://www.ietf.org/mailman/listinfo/avt
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-- 

Magnus Westerlund

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Subject: [AVTCORE] Working Group Last Call on draft-ietf-avtcore-leap-second-03
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WG,

This starts the WG last call on RTP and Leap Seconds with the intended
status of Proposed Standard. Please provide any comments by 1st of
September.

https://datatracker.ietf.org/doc/draft-ietf-avtcore-leap-second

Cheers

Magnus Westerlund

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From ray.vanbrandenburg@tno.nl  Thu Aug 15 02:22:53 2013
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From: "Brandenburg, R. (Ray) van" <ray.vanbrandenburg@tno.nl>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "draft-ietf-avtcore-idms@tools.ietf.org" <draft-ietf-avtcore-idms@tools.ietf.org>
Thread-Topic: [AVTCORE] I-D Action: draft-ietf-avtcore-idms-12.txt
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Date: Thu, 15 Aug 2013 09:22:38 +0000
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Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-idms-12.txt
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Hi Magnus,

Good point! For some reason, I never really considered that to be an option=
. I guess you're right and there's no reason not to.=20

If I don't see any disagreements from others, I will add it to the next rev=
ision.=20

Thanks,

Ray

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: donderdag 15 augustus 2013 9:35
> To: Brandenburg, R. (Ray) van; draft-ietf-avtcore-idms@tools.ietf.org
> Cc: avt@ietf.org
> Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-idms-12.txt
>=20
> Hi,
>=20
> Looking at this latest version I have a comment.
>=20
> section 14.4 defines the SPST registry.
>=20
> Why isn't is adding the ETSI defined values from the start. If they are k=
nown
> and have well known reference I don't see any issues with putting in the
> values into the registry at its creation.
>=20
> Value           Name                         Reference
>    -----           ----                         ---------
>    1               Synchronization Client       section 7 of [RFCXXXX]
>    2               <Appropriate Name>           [TS183063]
>    3               <Appropriate Name>           [TS183063]
>    4               <Appropriate Name>           [TS183063]
>=20
>=20
> Cheers
>=20
> Magnus
>=20
>=20
> On 2013-07-29 10:48, Brandenburg, R. (Ray) van wrote:
> > Hi,
> >
> > This version of the draft includes the final comments received during
> > IETF LC and from Martin Thomson as Appsdir reviewer.
> >
> > Best regards,
> >
> > Ray
> >
> > On 29 jul. 2013, at 10:45, "internet-drafts@ietf.org"
> > <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
> >> Core Maintenance Working Group of the IETF.
> >>
> >> Title           : Inter-destination Media Synchronization using the
> >> RTP Control Protocol (RTCP) Author(s)       : Ray van Brandenburg
> >> Hans Stokking Oskar van Deventer Fernando Boronat Mario Montagud
> >> Kevin Gross Filename        : draft-ietf-avtcore-idms-12.txt Pages
> >> : 22 Date            : 2013-07-29
> >>
> >> Abstract: This document defines a new RTP Control Protocol (RTCP)
> >> Packet Type and an RTCP Extended Report (XR) Block Type to be used
> >> for achieving Inter-Destination Media Synchronization (IDMS).  IDMS
> >> is the process of synchronizing playout across multiple
> >> geographically distributed media receivers.  Using the RTCP XR IDMS
> >> Report Block defined in this document, media playout information from
> >> participants in a synchronization group can be collected.
> >> Based on the collected information, an RTCP IDMS Settings Packet can
> >> then be sent to distribute a common target playout point to which all
> >> the distributed receivers, sharing a media experience, can
> >> synchronize.
> >>
> >> Typical use cases in which IDMS is useful are social TV, shared
> >> service control (i.e. applications where two or more geographically
> >> separated users are watching a media stream together), distance
> >> learning, networked video walls, networked loudspeakers, etc.
> >>
> >>
> >> The IETF datatracker status page for this draft is:
> >> https://datatracker.ietf.org/doc/draft-ietf-avtcore-idms
> >>
> >> There's also a htmlized version available at:
> >> http://tools.ietf.org/html/draft-ietf-avtcore-idms-12
> >>
> >> A diff from the previous version is available at:
> >> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-idms-12
> >>
> >>
> >> Please note that it may take a couple of minutes from the time of
> >> submission until the htmlized version and diff are available at
> >> tools.ietf.org.
> >>
> >> Internet-Drafts are also available by anonymous FTP at:
> >> ftp://ftp.ietf.org/internet-drafts/
> >>
> >> _______________________________________________
> Audio/Video Transport
> >> Core Maintenance avt@ietf.org
> >> https://www.ietf.org/mailman/listinfo/avt
> > This e-mail and its contents are subject to the DISCLAIMER at
> > http://www.tno.nl/emaildisclaimer
> >
> > _______________________________________________ Audio/Video
> Transport
> > Core Maintenance avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
> >
>=20
>=20
> --
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------


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From: Dan Wing <dwing@cisco.com>
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Subject: [AVTCORE] review of draft-ietf-avt-srtp-not-mandatory-13
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I reviewed draft-ietf-avt-srtp-not-mandatory-13.  It is well-written and =
complete.

-d


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Overall a strong document, and I know it was difficult to write.

My comments:

In the Introduction, the end of
   Because of the heterogeneity of
   RTP applications and use cases, however, a single security solution
   cannot be mandated.=20
feels like it could include a citation back to =
draft-ietf-avt-srtp-not-mandatory.


In section 3.1, Depending on when AES-GCM is published compared to when =
draft-ietf-avtcore-rtp-security-options is published, updating the =
following paragraph might be prudent (so it no longer says "ongoing =
work", as it will be published work at that point),
   AES-GCM:  There is also ongoing work to define AES-GCM (Galois
      Counter Mode) and AES-CCM (Counter with CBC) authentication for
      AES-128 and AES-256.  This authentication is included in the
      cipher text which becomes expanded with the length of the
      authentication tag instead of using the SRTP authentication tag.
      This is defined in [I-D.ietf-avtcore-srtp-aes-gcm].


In section 3.1.1,=20
s/cipher suits/cipher suites/


Section 3.1.4, suggest changing
OLD:
   This can enable interworking between DTLS-SRTP and for
   example security descriptions or other keying systems where either
   part can set the key.
NEW:
   This can enable interworking between DTLS-SRTP and=20
   other keying systems where either party can set the key (e.g.,=20
   interworking with security descriptions).
(original says "either part", should say "either party").


Section 3.4 for consistency with the section title and the rest of the =
section's text, I suggest:
OLD:
   There does not appear to be significant usage of RTP over DTLS.
NEW:
   There does not appear to be significant usage of DTLS for RTP.


Section 3.6 says:
   Mechanisms have been defined that encrypt only the payload of the RTP
   packets, and leave the RTP headers and RTCP in the clear. =20
which immediately made me think, "ah, like SRTP."  But then the rest of =
the section goes on to describe non-SRTP object encryption (DRM).  Can =
that first sentence somehow be made clearer that this section is not =
describing SRTP encryption, but object encryption?  Afterall, SRTP also =
leaves RTP headers in the clear.  Also, the title of the section is =
"Payload-Only Security Mechanisms", but the second paragraph discusses =
the need for transport security (which is not 'payload security').  =
Perhaps the section could be renamed to "Object Security" or "Digital =
Rights Management" or some other term, so that the payload security and =
transport security can both be discussed without exceeding the scope of =
the section title.


Section 4.1.1,
s/might provided/might provide/


Section 4.1.3, Changing from "I", "you", and "your" to "host" and "peer" =
would keep this section more consistent with the rest of the document.


Section 4.1.3, I had some technical heartburn over these two paragraphs, =
which together make it seem like DTLS-SRTP is weaker than Security =
Descriptions, and the text claims this weakness is because DTLS-SRTP =
sends the certificate hash over signaling whereas Security Descriptions =
sends the session key over signaling:
      For example, consider a point-to-point communication system that
      uses DTLS-SRTP using self-signed certificates for key management,
      and SIP for signalling.  In such a system the end-points for the
      DTLS-SRTP handshake have securely-established keys that are not
      visible to the signalling nodes.  However, as the certificates
      used by DTLS are not bound to any PKI they can't be verified.
      Instead, hashes of the certificate are sent over the signalling
      path.  If the signalling can be trusted not to collaborate on
      performing a man-in-the-middle attack by modifying the hashes,
      then the end-points can verify that they have established keys
      with the peer they are doing signalling with.

      Systems where the key-exchange is done using the signalling
      systems, such as Security Descriptions [RFC4568] or MIKEY embedded
      in SDP [RFC4567], enable a direct binding between signalling and
      key-exchange.  Independent of DTLS-SRTP or MIKEY in SDP the actual
      security depends on the trust one can place in the signalling
      system to correctly associate the peer's identity with the key-
      exchange.

Of course, DTLS-SRTP does not need to rely solely on trust for identity, =
which is described elsewhere in this same document.  RFC4474 or a later =
section of this same document should be cited in that first paragraph =
(or in the Using Identities section, whichever the authors prefer), as =
RFC4474 provides a way to ensure the DTLS-SRTP certificates are bound to =
the identity provider's public key.  Yes, RFC4474 has some deficincies =
and perhaps those should be mentioned briefly (doesn't survive going =
through an SBC, for example).  I would add a citation and description of =
RFC4474 to that first paragraph discussing DTLS-SRTP.

The second paragraph I quoted of Section 4.1.3, I am confused that its =
first sentence discusses Security Descriptions and MIKEY, but the =
paragraphs second sentence starts with "Independent of DTLS-SRTP or =
MIKEY in SDP" -- is this an editing mistake and that second sentence =
supposed to read "Independent of **security descriptions** or MIKEY" ??  =
If it is not an editing mistake, I admit I cannot parse that second =
paragraph and it would benefit from a rewrite to make its point more =
clear.


Section 4.1.5 on privacy mentions relays.  It might be useful to add a =
citation to TURN as an example of a relay.


Section 5.1:
OLD:
   At the basic level DTLS has a number of good security properties.
   For example, to enable a man in the middle someone in the signalling
   path needs to perform an active action and modify the signalling
   message.
NEW:
   At the basic level DTLS has a number of good security properties.
   For example, to enable a man in the middle someone in the signalling
   path needs to perform an active action and modify both the signalling
.....................................................^^^^
   message and the DTLS handshake.
...........^^^^^^^^^^^^^^^^^^^^^^^

Also, what is meant by "At the basic level"?  That wording implies there =
an identified problem/weakness at some other level (advanced level).


Section 5.2,=20
   Unless one uses a Identity provider and the proposed identity
   solution [I-D.ietf-rtcweb-security-arch], the web application still
   has the possibility to insert a MITM.=20

There are other ways to detect and prevent insertion of a MitM other =
than an Identity provider -- such as verifying the certificate hash out =
of band (verifying it matches what is printed on a business card, or =
verifying the hash is the same as the previously-used hash with that =
same user as described by ZRTP, as two examples).  Could that be =
re-worded so that Identity provider is just one way to avoid insertion =
of MITM?


Section 5.2,
      Prevent tracking between sessions:  RTP CNAMEs and DTLS-SRTP
      certificates provide information that could possibly be re-used
      between session instances.  Thus to prevent tracking, the same
      information can't be re-used between different communication
      sessions.
Yes, but another solution (allowing the re-use of DTLS-SRTP =
certificates, thus saving CPU resources to create new public/private key =
pairs) is to have better TLS handshake, such as is being pursued for TLS =
1.2 so the certificates are exchanged later in the TLS exchange (after =
the session is encrypted).  I would suggest:
OLD:
      Prevent tracking between sessions:  RTP CNAMEs and DTLS-SRTP
      certificates provide information that could possibly be re-used
      between session instances.  Thus to prevent tracking, the same
      information can't be re-used between different communication
      sessions.
NEW:
      Prevent tracking between sessions:  static RTP CNAMEs and =
DTLS-SRTP
      certificates provide information that is re-used
      between session instances.  Thus to prevent tracking, the =
information
      is not re-used between sessions or the information is not sent in =
the
      clear between sessions.


Section 5.3, I suggest:
OLD:
   If these goals are to be meet with the specified
   solution there needs to exist trust in that neither the
   implementation of the client nor the platform the application runs
   can be accessed or modified by the attacker.
NEW:
   To meet these goals with the specified solution the client=20
   implementation and the application platform are trusted to=20
   protect against access and modification by an attacker.


-d





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Date: Fri, 16 Aug 2013 14:39:19 +0200
From: Christian Oien <coien@cisco.com>
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Subject: [AVTCORE] Observation in RFC 3711
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--------------040406010709080503050104
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Hello WG,

After reading RFC 3711, and making an implementation in the context of RFC
4568 for the default ciphers, and then, afterwards, looking at a couple of
existing implementations, I have come to the conclusion of having found
a mismatch between the specification and the existing implementations,
including libsrtp.

RFC 3711 section "4.3 Key Derivation" describes a formal operator DIV,
with a result having the same number of bits as its first operand.
For SRTP the first operand is the SRTP packet index, which is a 48-bit
quantity, meaning that the <label> will have 48 bits to its right XOR'ing
with master_salt.

In subsection "4.3.2 SRTCP Key Derivation" the text says "Replace the
SRTP index by the 32-bit quantity" which means that the DIV operator will
yield a 32-bit quantity.  Following the specification of key_id for SRTCP
the <label> will have 32 bits to its right when XOR'int with master_salt.

The implementations, including libsrtp, invokes this XOR with the
<label> at the same position as for SRTP.  According to the specification
this should be done 16 bits to the right of this, when invoking for SRTCP.

It seems to me that the most pragmatic thing to do is to update the
specification, not the implementations.  But any new implementer
will fail to exchange any SRTCP with software using e.g. libsrtp.
This prevents any i.e. PLI of being communicated.  The consensus of not
following the specification, has the same implication for [ DTLS ] as for
security-description key-derivation input, namely that all SRTCP session
material is non-compliant and thus implementers reading the specification
instead of a reference-implementation, will fail.

In support of making an errarta for update of RFC 3711 is that
the specifications way of doing this has, according to itself, has a
cryptographic weakness, related to the rationale of having the <label>.
I am not sure about the implications of this weakness, but it seems like
support for *not* following specification as its current text strictly
instructs via its DIV operator and the 32-bit quantity :  Re-keying
(non-zero key_derivation_rate as opposed to RFC 4568 ciphers) with
SRTP packet index 196608 gives identical encryption key as for SRTCP,
in the case that e.g. none of the latter has yet been sent (index zero).
The implication of any key-stream re-use was the reason of having <label>.

  - Christian S Oien


--------------040406010709080503050104
Content-Type: text/html; charset="ISO-8859-1"
Content-Transfer-Encoding: 7bit

<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    Hello WG,
    <br>
    <br>
    After reading RFC 3711, and making an implementation in the context
    of RFC
    <br>
    4568 for the default ciphers, and then, afterwards, looking at a
    couple of
    <br>
    existing implementations, I have come to the conclusion of having
    found
    <br>
    a mismatch between the specification and the existing
    implementations,
    <br>
    including libsrtp.
    <br>
    <br>
    RFC 3711 section "4.3 Key Derivation" describes a formal operator
    DIV,
    <br>
    with a result having the same number of bits as its first operand.
    <br>
    For SRTP the first operand is the SRTP packet index, which is a
    48-bit
    <br>
    quantity, meaning that the &lt;label&gt; will have 48 bits to its
    right XOR'ing
    <br>
    with master_salt.
    <br>
    <br>
    In subsection "4.3.2 SRTCP Key Derivation" the text says "Replace
    the
    <br>
    SRTP index by the 32-bit quantity" which means that the DIV operator
    will
    <br>
    yield a 32-bit quantity.&nbsp; Following the specification of key_id for
    SRTCP
    <br>
    the &lt;label&gt; will have 32 bits to its right when XOR'int with
    master_salt.
    <br>
    <br>
    The implementations, including libsrtp, invokes this XOR with the
    <br>
    &lt;label&gt; at the same position as for SRTP.&nbsp; According to the
    specification
    <br>
    this should be done 16 bits to the right of this, when invoking for
    SRTCP.
    <br>
    <br>
    It seems to me that the most pragmatic thing to do is to update the
    <br>
    specification, not the implementations.&nbsp; But any new implementer
    <br>
    will fail to exchange any SRTCP with software using e.g. libsrtp.
    <br>
    This prevents any i.e. PLI of being communicated.&nbsp; The consensus of
    not
    <br>
    following the specification, has the same implication for [ DTLS ]
    as for
    <br>
    security-description key-derivation input, namely that all SRTCP
    session
    <br>
    material is non-compliant and thus implementers reading the
    specification
    <br>
    instead of a reference-implementation, will fail.
    <br>
    <br>
    In support of making an errarta for update of RFC 3711 is that
    <br>
    the specifications way of doing this has, according to itself, has a
    <br>
    cryptographic weakness, related to the rationale of having the
    &lt;label&gt;.
    <br>
    I am not sure about the implications of this weakness, but it seems
    like
    <br>
    support for <b class="moz-txt-star"><span class="moz-txt-tag">*</span>not<span
        class="moz-txt-tag">*</span></b> following specification as its
    current text strictly
    <br>
    instructs via its DIV operator and the 32-bit quantity :&nbsp; Re-keying
    <br>
    (non-zero key_derivation_rate as opposed to RFC 4568 ciphers) with
    <br>
    SRTP packet index 196608 gives identical encryption key as for
    SRTCP,
    <br>
    in the case that e.g. none of the latter has yet been sent (index
    zero).
    <br>
    The implication of any key-stream re-use was the reason of having
    &lt;label&gt;.
    <br>
    <br>
    &nbsp;- Christian S Oien<br>
    <br>
  </body>
</html>

--------------040406010709080503050104--

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Subject: [AVTCORE] Milestones changed for avtcore WG
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Changed milestone "Submit SRTP Cryptographic Transforms for AES
Galois/Counter Mode (AES-GCM) and AES Counter with Cipher Block
Chaining-Message Authentication Code (AES-CCM) and corresponding
key-management profiles for Security Descriptions, MIKEY and DTLS-SRTP
for publication as proposed standard", set due date to September 2013
from September 2012.

Changed milestone "Submit an update of RFC 6222 - "Guidelines for
Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)"
including a privacy preserving algorithm for publication as proposed
standard", resolved as "Done".

Changed milestone "Submit an Overview of RTP Security Solutions as
Informational", set due date to October 2013 from October 2012.

Changed milestone "Submit RTP Clock Source Signaling as Proposed
Standard", set due date to October 2013 from November 2012.

Changed milestone "RTP and Leap Seconds as proposed standard", set due
date to October 2013 from November 2012.

Changed milestone "Submit Multiple Media Types in an RTP Session for
publication as proposed standard", set due date to November 2013 from
May 2013.

Changed milestone "Submit RTP Congestion Control: Circuit Breakers for
Unicast Sessions for publication as proposed standard", set due date
to November 2013 from May 2013.

Changed milestone "Submit SRTP Cryptographic Transform(s) on the ARIA
algorithm and corresponding key-management profiles for Security
Descriptions, MIKEY and DTLS-SRTP for publication as proposed
standard", set due date to December 2013 from December 2012.

URL: http://datatracker.ietf.org/wg/avtcore/charter/

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Subject: [AVTCORE] Milestones changed for avtcore WG
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Changed milestone "Submit SRTP Cryptographic Transforms for AES
Galois/Counter Mode (AES-GCM) and AES Counter with Cipher Block
Chaining-Message Authentication Code (AES-CCM) and corresponding
key-management profiles for Security Descriptions, MIKEY and DTLS-SRTP
for publication as proposed standard", added
draft-ietf-avtcore-srtp-aes-gcm to milestone.

Changed milestone "Submit an Overview of RTP Security Solutions as
Informational", added draft-ietf-avtcore-rtp-security-options to
milestone.

Changed milestone "Submit RTP Clock Source Signaling as Proposed
Standard", added draft-ietf-avtcore-clksrc to milestone.

Changed milestone "RTP and Leap Seconds as proposed standard", added
draft-ietf-avtcore-leap-second to milestone.

Changed milestone "Submit Multiple Media Types in an RTP Session for
publication as proposed standard", added
draft-ietf-avtcore-multi-media-rtp-session to milestone.

Changed milestone "Submit RTP Congestion Control: Circuit Breakers for
Unicast Sessions for publication as proposed standard", added
draft-ietf-avtcore-rtp-circuit-breakers to milestone.

Changed milestone "Submit in band keying mechanism for SRTP draft for
Proposed Standard", set due date to December 2013 from October 2012,
added draft-ietf-avtcore-srtp-ekt to milestone.

Changed milestone "Submit SRTP Cryptographic Transform(s) on the ARIA
algorithm and corresponding key-management profiles for Security
Descriptions, MIKEY and DTLS-SRTP for publication as proposed
standard", added draft-ietf-avtcore-aria-srtp to milestone.

Changed milestone "Submit Guidelines for using the Multiplexing
Features of RTP for Informational", added
draft-ietf-avtcore-multiplex-guidelines to milestone.

Changed milestone "Submit Update to RTP Topologies (RFC 5117) for
Informational", added draft-ietf-avtcore-rtp-topologies-update to
milestone.

URL: http://datatracker.ietf.org/wg/avtcore/charter/

From magnus.westerlund@ericsson.com  Wed Aug 21 00:40:39 2013
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Subject: Re: [AVTCORE] Milestones changed for avtcore WG
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WG,

The chairs had a discussion and decided to update the WG milestones for
quite a lot of the milestones.

You can expect some more changes in the near future. We have two things
that needs a bit of work.

First we are considering to remove this milestone:
Dec 2012 Submit Real-Time Transport Control Protocol (RTCP) in Overlay
Multicast for Proposed Standard

The reason is that it appears to be no energy for working on this in the
WG. So if someone are interested in this, you better step up and be
willing to do some work. We will send an individual message on this also.

The second thing is that we plan to create two new WG milestones for the
following drafts:
draft-ietf-avtcore-rtp-multi-stream-01
draft-ietf-avtcore-rtp-multi-stream-optimisation-00

These draft has come from the following milestone:
Submit Multiple Media Types in an RTP Session for
publication as proposed standard

However, as they are independently useful we thought it appropriate to
ensure that they have their own milestone. Milestone drafting needs to
happen and then we need approval from our AD for this.

Cheers

Magnus Westerlund
AVTCORE WG chair

On 2013-08-21 09:31, IETF Secretariat wrote:
> Changed milestone "Submit SRTP Cryptographic Transforms for AES
> Galois/Counter Mode (AES-GCM) and AES Counter with Cipher Block
> Chaining-Message Authentication Code (AES-CCM) and corresponding
> key-management profiles for Security Descriptions, MIKEY and DTLS-SRTP
> for publication as proposed standard", added
> draft-ietf-avtcore-srtp-aes-gcm to milestone.
> 
> Changed milestone "Submit an Overview of RTP Security Solutions as
> Informational", added draft-ietf-avtcore-rtp-security-options to
> milestone.
> 
> Changed milestone "Submit RTP Clock Source Signaling as Proposed
> Standard", added draft-ietf-avtcore-clksrc to milestone.
> 
> Changed milestone "RTP and Leap Seconds as proposed standard", added
> draft-ietf-avtcore-leap-second to milestone.
> 
> Changed milestone "Submit Multiple Media Types in an RTP Session for
> publication as proposed standard", added
> draft-ietf-avtcore-multi-media-rtp-session to milestone.
> 
> Changed milestone "Submit RTP Congestion Control: Circuit Breakers for
> Unicast Sessions for publication as proposed standard", added
> draft-ietf-avtcore-rtp-circuit-breakers to milestone.
> 
> Changed milestone "Submit in band keying mechanism for SRTP draft for
> Proposed Standard", set due date to December 2013 from October 2012,
> added draft-ietf-avtcore-srtp-ekt to milestone.
> 
> Changed milestone "Submit SRTP Cryptographic Transform(s) on the ARIA
> algorithm and corresponding key-management profiles for Security
> Descriptions, MIKEY and DTLS-SRTP for publication as proposed
> standard", added draft-ietf-avtcore-aria-srtp to milestone.
> 
> Changed milestone "Submit Guidelines for using the Multiplexing
> Features of RTP for Informational", added
> draft-ietf-avtcore-multiplex-guidelines to milestone.
> 
> Changed milestone "Submit Update to RTP Topologies (RFC 5117) for
> Informational", added draft-ietf-avtcore-rtp-topologies-update to
> milestone.
> 
> URL: http://datatracker.ietf.org/wg/avtcore/charter/
> _______________________________________________
> 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
----------------------------------------------------------------------


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From: Woo-Hwan Kim <whkim5@ensec.re.kr>
To: avt@ietf.org, Magnus Westerlund <magnus.westerlund@ericsson.com>
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Cc: draft-ietf-avtcore-aria-srtp@tools.ietf.org
Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-aria-srtp-03.txt
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--047d7bb04bc2523bd204e496bec5
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1. I agree to your comment and I'll divide the table into two tables, one
for 'Mapping MIKEY parameters to ARIA-CTR with HMAC algorithm' and
the other for 'Mapping MIKEY parameters to AEAD algorithm'.

2. As your comments, the encrypted payload should contain the tag in AEAD.
I'll revise the draft sooner.

Thank you for your effort and time.

Sincerely,
Woo-Hwan Kim

----- Original Message -----
From: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
To: <draft-ietf-avtcore-aria-srtp@tools.ietf.org>
Cc: <avt@ietf.org>
Sent: Thursday, August 15, 2013 4:11 PM
Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-aria-srtp-03.txt


> Authors and WG,
>
> I have reviewed this new version. Thanks for addressing my issue. During
> this review I only noticed a small number of minor issues.
>
> Note, I have not verified in any way the test vectors. Nor am I certain
> that I can spot any fact error regarding the crypto-algorithms. What I
> have done is reviewed the draft consistency and correctness in their
> actions to IANA and towards SRTP.
>
>
> 1. Section 5.3:
>
>
>                              +--------------------------------------+
>                              | Encryption | Encryption | AEAD Auth. |
>                              | Algorithm  | Key Length | Tag Length |
>                              +======================================+
>    SRTP_ARIA_128_CTR_HMAC_80 |  ARIA-CTR  | 16 octets  |  80 bits   |
>    SRTP_ARIA_128_CTR_HMAC_32 |  ARIA-CTR  | 16 octets  |  32 bits   |
>    SRTP_ARIA_192_CTR_HMAC_80 |  ARIA-CTR  | 24 octets  |  80 bits   |
>    SRTP_ARIA_192_CTR_HMAC_32 |  ARIA-CTR  | 24 octets  |  32 bits   |
>    SRTP_ARIA_256_CTR_HMAC_80 |  ARIA-CTR  | 32 octets  |  80 bits   |
>    SRTP_ARIA_256_CTR_HMAC_32 |  ARIA-CTR  | 32 octets  |  32 bits   |
>
>
>
> Kim, et al.             Expires December 29, 2013              [Page 18]
>
> Internet-Draft           ARIA Algorithm for SRTP               June 2013
>
>
>    SRTP_AEAD_ARIA_128_GCM    |  ARIA-GCM  | 16 octets  | 128 bits   |
>    SRTP_AEAD_ARIA_128_CCM    |  ARIA-CCM  | 16 octets  | 128 bits   |
>    SRTP_AEAD_ARIA_128_GCM_12 |  ARIA-GCM  | 16 octets  |  96 bits   |
>    SRTP_AEAD_ARIA_128_CCM_12 |  ARIA-CCM  | 16 octets  |  96 bits   |
>    SRTP_AEAD_ARIA_128_GCM_8  |  ARIA-GCM  | 16 octets  |  64 bits   |
>    SRTP_AEAD_ARIA_128_CCM_8  |  ARIA-CCM  | 16 octets  |  64 bits   |
>    SRTP_AEAD_ARIA_256_GCM    |  ARIA-GCM  | 32 octets  | 128 bits   |
>    SRTP_AEAD_ARIA_256_CCM    |  ARIA-CCM  | 32 octets  | 128 bits   |
>    SRTP_AEAD_ARIA_256_GCM_12 |  ARIA-GCM  | 32 octets  |  96 bits   |
>    SRTP_AEAD_ARIA_256_CCM_12 |  ARIA-CCM  | 32 octets  |  96 bits   |
>    SRTP_AEAD_ARIA_256_GCM_8  |  ARIA-GCM  | 32 octets  |  64 bits   |
>    SRTP_AEAD_ARIA_256_CCM_8  |  ARIA-CCM  | 32 octets  |  64 bits   |
>                              +======================================+
>
>           Figure 1: Mapping MIKEY parameters to AEAD algorithm
>
> Shouldn't you split this into two tables as the last column and legend
> are wrong for the first ARIA CTR + SHA-1 HMAC suits which are not AEAD
> suites? That way you can use the correct labels on the last column for
> auth tag lengths and in the legend.
>
>
> 2. Section A.2. and A.3
>
> In section A.1 I do understand the structure for the test vectors. They
> contain both just the Encrypted part and the full RTP header + payload +
> ROC that authentication is calculated over, and then the resulting output.
>
> However in A.2 and A.3 where there are AEAD algorithms I am bit
> surprised over the split in Encrypted RTP payload and Authentication
> tag. Due to AEAD shouldn't the relevant unit to test as input be the
> full SRTP packet which contains both the encrypted and the data that is
> just authenticated including the AEAD output in the payload location?
>
> You might need a bit more explanation what the test vectors really are
> so that one can correctly use them to verify ones implementation.
>
>
>
> Next Steps:
>
> Please address the above issues.
>
> I personally think this is ready for going forward to WG last call when
> the above is addressed. However, in that last call we will need to get a
> review from someone that has sufficient crypto knowledge to be reduce
> the risk for any such error making it through. Thus I intended to make
> the WG last call's completion dependent on getting such a review.
>
> Cheers
>
> Magnus
>
>
>
>
> On 2013-06-27 07:43, 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 Core Maintenance
Working Group of the IETF.
>>
>> Title           : The ARIA Algorithm and Its Use with the Secure
Real-time Transport Protocol(SRTP)
>> Author(s)       : Woo-Hwan Kim
>>                           Jungkeun Lee
>>                           Dong-Chan Kim
>>                           Je-Hong Park
>>                           Daesung Kwon
>> Filename        : draft-ietf-avtcore-aria-srtp-03.txt
>> Pages           : 32
>> Date            : 2013-06-26
>>
>> Abstract:
>>    This document describes the use of the ARIA block cipher algorithm
>>    within the Secure Real-time Transport Protocol (SRTP) for providing
>>    confidentiality for the Real-time Transport Protocol (RTP) traffic
>>    and for the control traffic for RTP, the Real-time Transport Control
>>    Protocol (RTCP).  It details three modes of operation (CTR, CCM, GCM)
>>    and a SRTP Key Derivation Function for ARIA.
>>
>>
>> The IETF datatracker status page for this draft is:
>> https://datatracker.ietf.org/doc/draft-ietf-avtcore-aria-srtp
>>
>> There's also a htmlized version available at:
>> http://tools.ietf.org/html/draft-ietf-avtcore-aria-srtp-03
>>
>> A diff from the previous version is available at:
>> http://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-aria-srtp-03
>>
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
>> _______________________________________________
>> Audio/Video Transport Core Maintenance
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>>
>

--047d7bb04bc2523bd204e496bec5
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>1. I agree to your comment and I&#39;ll divide the ta=
ble into two tables, one for=20
&#39;Mapping MIKEY parameters to ARIA-CTR with HMAC algorithm&#39; and=A0</=
div><div>the other for=20
&#39;Mapping MIKEY parameters to AEAD algorithm&#39;.<br><br>2. As your com=
ments, the=20
encrypted payload should contain the tag in AEAD.<br>I&#39;ll revise the dr=
aft sooner.<br></div><span style=3D"font-family:arial,sans-serif;font-size:=
14px"><div><span style=3D"font-family:arial,sans-serif;font-size:14px"><br>
</span></div><div><span style=3D"font-family:arial,sans-serif;font-size:14p=
x">Thank you for your effort and time.</span></div><div><span style=3D"font=
-family:arial,sans-serif;font-size:14px"><br></span></div><div><span style=
=3D"font-family:arial,sans-serif;font-size:14px">Sincerely,</span></div>
<div><span style=3D"font-family:arial,sans-serif;font-size:14px">Woo-Hwan K=
im</span></div><div><span style=3D"font-family:arial,sans-serif;font-size:1=
4px"><br></span></div>----- Original Message -----</span><br style=3D"font-=
family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">From: &quot;Mag=
nus Westerlund&quot; &lt;</span><a href=3D"mailto:magnus.westerlund@ericsso=
n.com" style=3D"font-family:arial,sans-serif;font-size:14px">magnus.westerl=
und@ericsson.com</a><span style=3D"font-family:arial,sans-serif;font-size:1=
4px">&gt;</span><br style=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">To: &lt;</span>=
<a href=3D"mailto:draft-ietf-avtcore-aria-srtp@tools.ietf.org" style=3D"fon=
t-family:arial,sans-serif;font-size:14px">draft-ietf-avtcore-aria-srtp@tool=
s.ietf.org</a><span style=3D"font-family:arial,sans-serif;font-size:14px">&=
gt;</span><br style=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">Cc: &lt;</span>=
<a href=3D"mailto:avt@ietf.org" style=3D"font-family:arial,sans-serif;font-=
size:14px">avt@ietf.org</a><span style=3D"font-family:arial,sans-serif;font=
-size:14px">&gt;</span><br style=3D"font-family:arial,sans-serif;font-size:=
14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">Sent: Thursday,=
 August 15, 2013 4:11 PM</span><br style=3D"font-family:arial,sans-serif;fo=
nt-size:14px"><span style=3D"font-family:arial,sans-serif;font-size:14px">S=
ubject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-aria-srtp-</span><span=
 style=3D"font-family:arial,sans-serif;font-size:14px">03.txt</span><br sty=
le=3D"font-family:arial,sans-serif;font-size:14px">
<br style=3D"font-family:arial,sans-serif;font-size:14px"><br style=3D"font=
-family:arial,sans-serif;font-size:14px"><span style=3D"font-family:arial,s=
ans-serif;font-size:14px">&gt; Authors and WG,</span><br style=3D"font-fami=
ly:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt; I have reviewed this new versio=
n. Thanks for addressing my issue. During</span><br style=3D"font-family:ar=
ial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; this revie=
w I only noticed a small number of minor issues.</span><br style=3D"font-fa=
mily:arial,sans-serif;font-size:14px"><span style=3D"font-family:arial,sans=
-serif;font-size:14px">&gt;</span><br style=3D"font-family:arial,sans-serif=
;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; Note, I ha=
ve not verified in any way the test vectors. Nor am I certain</span><br sty=
le=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-fami=
ly:arial,sans-serif;font-size:14px">&gt; that I can spot any fact error reg=
arding the crypto-algorithms. What I</span><br style=3D"font-family:arial,s=
ans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; have done =
is reviewed the draft consistency and correctness in their</span><br style=
=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-family=
:arial,sans-serif;font-size:14px">&gt; actions to IANA and towards SRTP.</s=
pan><br style=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt;</span><br style=3D"font-family:=
arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; 1. Section=
 5.3:</span><br style=3D"font-family:arial,sans-serif;font-size:14px"><span=
 style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br style=
=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 =A0 =A0 =A0 =A0 =A0 =A0+-----------------------------</span><span styl=
e=3D"font-family:arial,sans-serif;font-size:14px">---------+</span><br styl=
e=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Encryption | Encryptio=
n | AEAD Auth. |</span><br style=3D"font-family:arial,sans-serif;font-size:=
14px"><span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Algorithm =A0| Key=
 Length | Tag Length |</span><br style=3D"font-family:arial,sans-serif;font=
-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0+=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D</span><s=
pan style=3D"font-family:arial,sans-serif;font-size:14px">=3D=3D=3D=3D=3D=
=3D=3D=3D=3D+</span><br style=3D"font-family:arial,sans-serif;font-size:14p=
x">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRT=
P_ARIA_128_CTR_HMAC_80 | =A0ARIA-CTR =A0| 16 octets =A0| =A080 bits =A0 |</=
span><br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=
=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRTP_ARIA_128_=
CTR_HMAC_32 | =A0ARIA-CTR =A0| 16 octets =A0| =A032 bits =A0 |</span><br st=
yle=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRT=
P_ARIA_192_CTR_HMAC_80 | =A0ARIA-CTR =A0| 24 octets =A0| =A080 bits =A0 |</=
span><br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=
=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRTP_ARIA_192_=
CTR_HMAC_32 | =A0ARIA-CTR =A0| 24 octets =A0| =A032 bits =A0 |</span><br st=
yle=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRT=
P_ARIA_256_CTR_HMAC_80 | =A0ARIA-CTR =A0| 32 octets =A0| =A080 bits =A0 |</=
span><br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=
=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRTP_ARIA_256_=
CTR_HMAC_32 | =A0ARIA-CTR =A0| 32 octets =A0| =A032 bits =A0 |</span><br st=
yle=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt;</span><br style=3D"font-family:=
arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt; Kim, et al. =A0 =A0 =A0 =A0 =A0=
 =A0 Expires December 29, 2013 =A0 =A0 =A0 =A0 =A0 =A0 =A0[Page 18]</span><=
br style=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt; Internet-Draft =A0 =A0 =A0 =A0 =
=A0 ARIA Algorithm for SRTP =A0 =A0 =A0 =A0 =A0 =A0 =A0 June 2013</span><br=
 style=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt;</span><br style=3D"font-family:=
arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRT=
P_AEAD_ARIA_128_GCM =A0 =A0| =A0ARIA-GCM =A0| 16 octets =A0| 128 bits =A0 |=
</span><br style=3D"font-family:arial,sans-serif;font-size:14px"><span styl=
e=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRTP_AEAD_ARI=
A_128_CCM =A0 =A0| =A0ARIA-CCM =A0| 16 octets =A0| 128 bits =A0 |</span><br=
 style=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRT=
P_AEAD_ARIA_128_GCM_12 | =A0ARIA-GCM =A0| 16 octets =A0| =A096 bits =A0 |</=
span><br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=
=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRTP_AEAD_ARIA=
_128_CCM_12 | =A0ARIA-CCM =A0| 16 octets =A0| =A096 bits =A0 |</span><br st=
yle=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRT=
P_AEAD_ARIA_128_GCM_8 =A0| =A0ARIA-GCM =A0| 16 octets =A0| =A064 bits =A0 |=
</span><br style=3D"font-family:arial,sans-serif;font-size:14px"><span styl=
e=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRTP_AEAD_ARI=
A_128_CCM_8 =A0| =A0ARIA-CCM =A0| 16 octets =A0| =A064 bits =A0 |</span><br=
 style=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRT=
P_AEAD_ARIA_256_GCM =A0 =A0| =A0ARIA-GCM =A0| 32 octets =A0| 128 bits =A0 |=
</span><br style=3D"font-family:arial,sans-serif;font-size:14px"><span styl=
e=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRTP_AEAD_ARI=
A_256_CCM =A0 =A0| =A0ARIA-CCM =A0| 32 octets =A0| 128 bits =A0 |</span><br=
 style=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRT=
P_AEAD_ARIA_256_GCM_12 | =A0ARIA-GCM =A0| 32 octets =A0| =A096 bits =A0 |</=
span><br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=
=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRTP_AEAD_ARIA=
_256_CCM_12 | =A0ARIA-CCM =A0| 32 octets =A0| =A096 bits =A0 |</span><br st=
yle=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRT=
P_AEAD_ARIA_256_GCM_8 =A0| =A0ARIA-GCM =A0| 32 octets =A0| =A064 bits =A0 |=
</span><br style=3D"font-family:arial,sans-serif;font-size:14px"><span styl=
e=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0SRTP_AEAD_ARI=
A_256_CCM_8 =A0| =A0ARIA-CCM =A0| 32 octets =A0| =A064 bits =A0 |</span><br=
 style=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0+=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D</span><s=
pan style=3D"font-family:arial,sans-serif;font-size:14px">=3D=3D=3D=3D=3D=
=3D=3D=3D=3D+</span><br style=3D"font-family:arial,sans-serif;font-size:14p=
x">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt; =A0 =A0 =A0 =A0 =A0 Figure 1: M=
apping MIKEY parameters to AEAD algorithm</span><br style=3D"font-family:ar=
ial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt; Shouldn&#39;t you split this in=
to two tables as the last column and legend</span><br style=3D"font-family:=
arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; are wrong =
for the first ARIA CTR + SHA-1 HMAC suits which are not AEAD</span><br styl=
e=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-famil=
y:arial,sans-serif;font-size:14px">&gt; suites? That way you can use the co=
rrect labels on the last column for</span><br style=3D"font-family:arial,sa=
ns-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; auth tag l=
engths and in the legend.</span><br style=3D"font-family:arial,sans-serif;f=
ont-size:14px"><span style=3D"font-family:arial,sans-serif;font-size:14px">=
&gt;</span><br style=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt; 2. Section A.2. and A.3</span><=
br style=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt; In section A.1 I do understand =
the structure for the test vectors. They</span><br style=3D"font-family:ari=
al,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; contain bo=
th just the Encrypted part and the full RTP header + payload +</span><br st=
yle=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-fam=
ily:arial,sans-serif;font-size:14px">&gt; ROC that authentication is calcul=
ated over, and then the resulting output.</span><br style=3D"font-family:ar=
ial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt; However in A.2 and A.3 where th=
ere are AEAD algorithms I am bit</span><br style=3D"font-family:arial,sans-=
serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; surprised =
over the split in Encrypted RTP payload and Authentication</span><br style=
=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-family=
:arial,sans-serif;font-size:14px">&gt; tag. Due to AEAD shouldn&#39;t the r=
elevant unit to test as input be the</span><br style=3D"font-family:arial,s=
ans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; full SRTP =
packet which contains both the encrypted and the data that is</span><br sty=
le=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-fami=
ly:arial,sans-serif;font-size:14px">&gt; just authenticated including the A=
EAD output in the payload location?</span><br style=3D"font-family:arial,sa=
ns-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt; You might need a bit more expla=
nation what the test vectors really are</span><br style=3D"font-family:aria=
l,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; so that on=
e can correctly use them to verify ones implementation.</span><br style=3D"=
font-family:arial,sans-serif;font-size:14px"><span style=3D"font-family:ari=
al,sans-serif;font-size:14px">&gt;</span><br style=3D"font-family:arial,san=
s-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt;</span><br style=3D"font-family:=
arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; Next Steps=
:</span><br style=3D"font-family:arial,sans-serif;font-size:14px"><span sty=
le=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br style=3D"=
font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; Please add=
ress the above issues.</span><br style=3D"font-family:arial,sans-serif;font=
-size:14px"><span style=3D"font-family:arial,sans-serif;font-size:14px">&gt=
;</span><br style=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; I personal=
ly think this is ready for going forward to WG last call when</span><br sty=
le=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-fami=
ly:arial,sans-serif;font-size:14px">&gt; the above is addressed. However, i=
n that last call we will need to get a</span><br style=3D"font-family:arial=
,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; review fro=
m someone that has sufficient crypto knowledge to be reduce</span><br style=
=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-family=
:arial,sans-serif;font-size:14px">&gt; the risk for any such error making i=
t through. Thus I intended to make</span><br style=3D"font-family:arial,san=
s-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; the WG las=
t call&#39;s completion dependent on getting such a review.</span><br style=
=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-family=
:arial,sans-serif;font-size:14px">&gt;</span><br style=3D"font-family:arial=
,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; Cheers</sp=
an><br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D=
"font-family:arial,sans-serif;font-size:14px">&gt;</span><br style=3D"font-=
family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; Magnus</sp=
an><br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D=
"font-family:arial,sans-serif;font-size:14px">&gt;</span><br style=3D"font-=
family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt;</span><br style=3D"font-family:=
arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt; On 2013-06-27 07:43,=A0</span><=
a href=3D"mailto:internet-drafts@ietf.org" style=3D"font-family:arial,sans-=
serif;font-size:14px">internet-drafts@ietf.org</a><span style=3D"font-famil=
y:arial,sans-serif;font-size:14px">=A0wrote:</span><br style=3D"font-family=
:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt;</span>=
<br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:14px">&gt;&gt; A New Internet-Draft is=
 available from the on-line Internet-Drafts directories.</span><br style=3D=
"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt; =A0Thi=
s draft is a work item of the Audio/Video Transport Core Maintenance Workin=
g Group of the IETF.</span><br style=3D"font-family:arial,sans-serif;font-s=
ize:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt;</span>=
<br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:14px">&gt;&gt; Title =A0 =A0 =A0 =A0 =
=A0 : The ARIA Algorithm and Its Use with the Secure Real-time Transport Pr=
otocol(SRTP)</span><br style=3D"font-family:arial,sans-serif;font-size:14px=
">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt; Author=
(s) =A0 =A0 =A0 : Woo-Hwan Kim</span><br style=3D"font-family:arial,sans-se=
rif;font-size:14px"><span style=3D"font-family:arial,sans-serif;font-size:1=
4px">&gt;&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Jungkeun =
Lee</span><br style=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt; =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Dong-Chan Kim</span><br sty=
le=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-fami=
ly:arial,sans-serif;font-size:14px">&gt;&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 Je-Hong Park</span><br style=3D"font-family:arial,s=
ans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt; =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Daesung Kwon</span><br styl=
e=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-famil=
y:arial,sans-serif;font-size:14px">&gt;&gt; Filename =A0 =A0 =A0 =A0: draft=
-ietf-avtcore-aria-srtp-</span><span style=3D"font-family:arial,sans-serif;=
font-size:14px">03.txt</span><br style=3D"font-family:arial,sans-serif;font=
-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt; Pages =
=A0 =A0 =A0 =A0 =A0 : 32</span><br style=3D"font-family:arial,sans-serif;fo=
nt-size:14px"><span style=3D"font-family:arial,sans-serif;font-size:14px">&=
gt;&gt; Date =A0 =A0 =A0 =A0 =A0 =A0: 2013-06-26</span><br style=3D"font-fa=
mily:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt;</span>=
<br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:14px">&gt;&gt; Abstract:</span><br sty=
le=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt; =A0 =
=A0This document describes the use of the ARIA block cipher algorithm</span=
><br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"f=
ont-family:arial,sans-serif;font-size:14px">&gt;&gt; =A0 =A0within the Secu=
re Real-time Transport Protocol (SRTP) for providing</span><br style=3D"fon=
t-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt; =A0 =
=A0confidentiality for the Real-time Transport Protocol (RTP) traffic</span=
><br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"f=
ont-family:arial,sans-serif;font-size:14px">&gt;&gt; =A0 =A0and for the con=
trol traffic for RTP, the Real-time Transport Control</span><br style=3D"fo=
nt-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt; =A0 =
=A0Protocol (RTCP). =A0It details three modes of operation (CTR, CCM, GCM)<=
/span><br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=
=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt; =A0 =A0and a SRTP=
 Key Derivation Function for ARIA.</span><br style=3D"font-family:arial,san=
s-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt;</span>=
<br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:14px">&gt;&gt;</span><br style=3D"font=
-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt; The IE=
TF datatracker status page for this draft is:</span><br style=3D"font-famil=
y:arial,sans-serif;font-size:14px"><span style=3D"font-family:arial,sans-se=
rif;font-size:14px">&gt;&gt;=A0</span><a href=3D"https://datatracker.ietf.o=
rg/doc/draft-ietf-avtcore-aria-srtp" target=3D"_blank" style=3D"font-family=
:arial,sans-serif;font-size:14px">https://datatracker.ietf.org/doc/draft-ie=
tf-avtcore-aria-srtp</a><br style=3D"font-family:arial,sans-serif;font-size=
:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt;</span>=
<br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:14px">&gt;&gt; There&#39;s also a html=
ized version available at:</span><br style=3D"font-family:arial,sans-serif;=
font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt;=A0</sp=
an><a href=3D"http://tools.ietf.org/html/draft-ietf-avtcore-aria-srtp-03" t=
arget=3D"_blank" style=3D"font-family:arial,sans-serif;font-size:14px">http=
://tools.ietf.org/html/draft-ietf-avtcore-aria-srtp-03</a><br style=3D"font=
-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt;</span>=
<br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:14px">&gt;&gt; A diff from the previou=
s version is available at:</span><br style=3D"font-family:arial,sans-serif;=
font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt;=A0</sp=
an><a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-aria-sr=
tp-03" target=3D"_blank" style=3D"font-family:arial,sans-serif;font-size:14=
px">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-aria-srtp-03</a><=
br style=3D"font-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt;</span>=
<br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:14px">&gt;&gt;</span><br style=3D"font=
-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt; Intern=
et-Drafts are also available by anonymous FTP at:</span><br style=3D"font-f=
amily:arial,sans-serif;font-size:14px"><span style=3D"font-family:arial,san=
s-serif;font-size:14px">&gt;&gt;=A0</span><a href=3D"ftp://ftp.ietf.org/int=
ernet-drafts/" target=3D"_blank" style=3D"font-family:arial,sans-serif;font=
-size:14px">ftp://ftp.ietf.org/internet-drafts/</a><br style=3D"font-family=
:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt;</span>=
<br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:14px">&gt;&gt; _______________________=
_______</span><span style=3D"font-family:arial,sans-serif;font-size:14px">_=
________________</span><br style=3D"font-family:arial,sans-serif;font-size:=
14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt; Audio/=
Video Transport Core Maintenance</span><br style=3D"font-family:arial,sans-=
serif;font-size:14px"><span style=3D"font-family:arial,sans-serif;font-size=
:14px">&gt;&gt;=A0</span><a href=3D"mailto:avt@ietf.org" style=3D"font-fami=
ly:arial,sans-serif;font-size:14px">avt@ietf.org</a><br style=3D"font-famil=
y:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt;=A0</sp=
an><a href=3D"https://www.ietf.org/mailman/listinfo/avt" target=3D"_blank" =
style=3D"font-family:arial,sans-serif;font-size:14px">https://www.ietf.org/=
mailman/listinfo/avt</a><br style=3D"font-family:arial,sans-serif;font-size=
:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;&gt;</span>=
<br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:14px">&gt;</span><br></div>

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Subject: [AVTCORE] I-D Action: draft-ietf-avtcore-aria-srtp-04.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 Core Maintenance Wo=
rking Group of the IETF.

	Title           : The ARIA Algorithm and Its Use with the Secure Real-time=
 Transport Protocol(SRTP)
	Author(s)       : Woo-Hwan Kim
                          Jungkeun Lee
                          Dong-Chan Kim
                          Je-Hong Park
                          Daesung Kwon
	Filename        : draft-ietf-avtcore-aria-srtp-04.txt
	Pages           : 32
	Date            : 2013-08-23

Abstract:
   This document describes the use of the ARIA block cipher algorithm
   within the Secure Real-time Transport Protocol (SRTP) for providing
   confidentiality for the Real-time Transport Protocol (RTP) traffic
   and for the control traffic for RTP, the Real-time Transport Control
   Protocol (RTCP).  It details three modes of operation (CTR, CCM, GCM)
   and a SRTP Key Derivation Function for ARIA.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-avtcore-aria-srtp

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-avtcore-aria-srtp-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-aria-srtp-04


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

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


From magnus.westerlund@ericsson.com  Mon Aug 26 02:34:15 2013
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Subject: [AVTCORE] WG last call on draft-ietf-avtcore-aria-srtp-04
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WG,

This starts the WG last call on draft-ietf-avtcore-aria-srtp-04 to be
published as proposed standard. Please provide any feedback or review
comments by September 16. Even if you have just read it and found no
issue, please state so.

Draft:
https://datatracker.ietf.org/doc/draft-ietf-avtcore-aria-srtp

In addition any comments about current or planned implementation of this
draft are appreciated.

As a WG chair I really want some persons that really understand crypto
algorithms and/or the key-management mechanisms used to review this and
will solicit reviews outside of the WG also.

Cheers

Magnus Westerlund

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


From csp@csperkins.org  Mon Aug 26 05:31:00 2013
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Subject: Re: [AVTCORE] WG last call on draft-ietf-avtcore-aria-srtp-04
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On 26 Aug 2013, at 10:34, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:
> WG,
>=20
> This starts the WG last call on draft-ietf-avtcore-aria-srtp-04 to be
> published as proposed standard. Please provide any feedback or review
> comments by September 16. Even if you have just read it and found no
> issue, please state so.

=46rom a quick look, this looks okay to me. Some nits:

The acronym RTCP is expanded incorrectly in the abstract and =
Introduction (should be "RTP Control Protocol" to match RFC3550).=20

Appendix A: what is the byte order for the test vectors? (especially for =
fields like the seqnum and SSRC, which are commonly represented by ints =
in code)

Cheers,
Colin


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




From csp@csperkins.org  Mon Aug 26 05:57:25 2013
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From: Colin Perkins <csp@csperkins.org>
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Cc: IETF AVTCore WG <avt@ietf.org>
Subject: Re: [AVTCORE] Working Group Last Call on draft-ietf-avtcore-leap-second-03
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On 15 Aug 2013, at 08:46, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:
> This starts the WG last call on RTP and Leap Seconds with the intended
> status of Proposed Standard. Please provide any comments by 1st of
> September.
>=20
> https://datatracker.ietf.org/doc/draft-ietf-avtcore-leap-second


Section 5 says that: "No special treatment with respect to RTP =
timestamps is required in the presence of a negative leap second". This =
might be clearer written "No special treatment is needed to avoid =
ambiguity with respect to RTP timestamps in the presence of a negative =
leap second". I think some special treatment is needed, as noted in =
Section 5.2, to avoid jumps in the playout time.=20

Section 5 also says NTP includes leap seconds, and that implementations =
working to a leap-second-bearing reference need "a working =
communications channel to receive notification of leap-second =
scheduling". Is having an NTP-synchronised clock a sufficient channel to =
receive the necessary notification, or is something more needed?=20

Typos:
- Section 3, 1st paragraph: "synchronized with with the rotation"
- Section 6, 2nd paragraph: "leap-seocnd-bearing"


Cheers,
Colin



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




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Subject: Re: [AVTCORE] Working Group Last Call on draft-ietf-avtcore-leap-second-03
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--f46d0444e85530c7ac04e4dc2c90
Content-Type: text/plain; charset=ISO-8859-1

Thanks for the review. See responses below.

Kevin Gross

On Mon, Aug 26, 2013 at 6:57 AM, Colin Perkins <csp@csperkins.org> wrote:

> On 15 Aug 2013, at 08:46, Magnus Westerlund <
> magnus.westerlund@ericsson.com> wrote:
> > This starts the WG last call on RTP and Leap Seconds with the intended
> > status of Proposed Standard. Please provide any comments by 1st of
> > September.
> >
> > https://datatracker.ietf.org/doc/draft-ietf-avtcore-leap-second
>
>
> Section 5 says that: "No special treatment with respect to RTP timestamps
> is required in the presence of a negative leap second". This might be
> clearer written "No special treatment is needed to avoid ambiguity with
> respect to RTP timestamps in the presence of a negative leap second". I
> think some special treatment is needed, as noted in Section 5.2, to avoid
> jumps in the playout time.
>

That looks like a good improvement.

>
> Section 5 also says NTP includes leap seconds, and that implementations
> working to a leap-second-bearing reference need "a working communications
> channel to receive notification of leap-second scheduling". Is having an
> NTP-synchronised clock a sufficient channel to receive the necessary
> notification, or is something more needed?
>

NTP includes a mechanism for leap-second notification. That's a good start
but what we're trying to capture here is the entire communications chain
from IERS to the slaved clocks. In addition to these protocol mechanisms
for delivering leap-second notifications, master clocks must have access to
an up-to-date leap second schedule through some other communications
channel (e.g. FTP leap-seconds.list from a NIST server).

I can revise to emphasize that time protocol notifications are not
sufficient but I don't think we want to try and describe the whole
communications channel for this information. Or we can leave it as it is.
The current text is terse but correct.

>
> Typos:
> - Section 3, 1st paragraph: "synchronized with with the rotation"
> - Section 6, 2nd paragraph: "leap-seocnd-bearing"
>

Thanks. I will fix these.

>
>
> Cheers,
> Colin
>
>
>
> --
> Colin Perkins
> http://csperkins.org/
>
>
>
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>

--f46d0444e85530c7ac04e4dc2c90
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Thanks for the review. See responses below.<br><div class=
=3D"gmail_extra"><br clear=3D"all"><div><div dir=3D"ltr">Kevin Gross<br><di=
v><br></div></div></div><div class=3D"gmail_quote">On Mon, Aug 26, 2013 at =
6:57 AM, Colin Perkins <span dir=3D"ltr">&lt;<a href=3D"mailto:csp@csperkin=
s.org" target=3D"_blank">csp@csperkins.org</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 class=3D"im">On 15 Aug 2013, at 08:46, =
Magnus Westerlund &lt;<a href=3D"mailto:magnus.westerlund@ericsson.com">mag=
nus.westerlund@ericsson.com</a>&gt; wrote:<br>

&gt; This starts the WG last call on RTP and Leap Seconds with the intended=
<br>
&gt; status of Proposed Standard. Please provide any comments by 1st of<br>
&gt; September.<br>
&gt;<br>
&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-avtcore-leap-se=
cond" target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-avtcore=
-leap-second</a><br>
<br>
<br>
</div>Section 5 says that: &quot;No special treatment with respect to RTP t=
imestamps is required in the presence of a negative leap second&quot;. This=
 might be clearer written &quot;No special treatment is needed to avoid amb=
iguity with respect to RTP timestamps in the presence of a negative leap se=
cond&quot;. I think some special treatment is needed, as noted in Section 5=
.2, to avoid jumps in the playout time.<br>
</blockquote><div><br></div><div>That looks like a good improvement.=A0</di=
v><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex">
<br>
Section 5 also says NTP includes leap seconds, and that implementations wor=
king to a leap-second-bearing reference need &quot;a working communications=
 channel to receive notification of leap-second scheduling&quot;. Is having=
 an NTP-synchronised clock a sufficient channel to receive the necessary no=
tification, or is something more needed?<br>
</blockquote><div><br></div><div>NTP includes a mechanism for leap-second n=
otification. That&#39;s a good start but what we&#39;re trying to capture h=
ere is the entire communications chain from IERS to the slaved clocks. In a=
ddition to these protocol mechanisms for delivering leap-second notificatio=
ns, master clocks must have access to an up-to-date leap second schedule th=
rough some other communications channel (e.g. FTP leap-seconds.list from a =
NIST server).</div>
<div><br></div><div>I can revise to emphasize that time protocol notificati=
ons are not sufficient but I don&#39;t think we want to try and describe th=
e whole communications channel for this information. Or we can leave it as =
it is. The current text is terse but correct.</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
Typos:<br>
- Section 3, 1st paragraph: &quot;synchronized with with the rotation&quot;=
<br>
- Section 6, 2nd paragraph: &quot;leap-seocnd-bearing&quot;<br></blockquote=
><div><br></div><div>Thanks. I will fix these.=A0</div><blockquote class=3D=
"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding=
-left:1ex">

<br>
<br>
Cheers,<br>
Colin<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
<br>
<br>
--<br>
Colin Perkins<br>
<a href=3D"http://csperkins.org/" target=3D"_blank">http://csperkins.org/</=
a><br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
<br>
_______________________________________________<br>
Audio/Video Transport Core Maintenance<br>
<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt" target=3D"_blank">htt=
ps://www.ietf.org/mailman/listinfo/avt</a><br>
</div></div></blockquote></div><br></div></div>

--f46d0444e85530c7ac04e4dc2c90--

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Subject: [AVTCORE] I-D Action: draft-ietf-avtcore-srtp-aes-gcm-08.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 Core Maintenance Wo=
rking Group of the IETF.

	Title           : AES-GCM and AES-CCM Authenticated Encryption in Secure R=
TP (SRTP)
	Author(s)       : David A. McGrew
                          Kevin M. Igoe
	Filename        : draft-ietf-avtcore-srtp-aes-gcm-08.txt
	Pages           : 33
	Date            : 2013-08-26

Abstract:
   This document defines how AES-GCM and AES-CCM Authenticated
   Encryption with Associated Data algorithms can be used to provide
   confidentiality and data authentication in the SRTP protocol.



The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-avtcore-srtp-aes-gcm

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-avtcore-srtp-aes-gcm-08

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-srtp-aes-gcm-08


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

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


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On 26 Aug 2013, at 17:23, Kevin Gross <kevin.gross@avanw.com> wrote:
> Thanks for the review. See responses below.
...
> On Mon, Aug 26, 2013 at 6:57 AM, Colin Perkins <csp@csperkins.org> =
wrote:
> On 15 Aug 2013, at 08:46, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:
> > This starts the WG last call on RTP and Leap Seconds with the =
intended
> > status of Proposed Standard. Please provide any comments by 1st of
> > September.
> >
> > https://datatracker.ietf.org/doc/draft-ietf-avtcore-leap-second
...
> Section 5 also says NTP includes leap seconds, and that =
implementations working to a leap-second-bearing reference need "a =
working communications channel to receive notification of leap-second =
scheduling". Is having an NTP-synchronised clock a sufficient channel to =
receive the necessary notification, or is something more needed?
>=20
> NTP includes a mechanism for leap-second notification. That's a good =
start but what we're trying to capture here is the entire communications =
chain from IERS to the slaved clocks. In addition to these protocol =
mechanisms for delivering leap-second notifications, master clocks must =
have access to an up-to-date leap second schedule through some other =
communications channel (e.g. FTP leap-seconds.list from a NIST server).
>=20
> I can revise to emphasize that time protocol notifications are not =
sufficient but I don't think we want to try and describe the whole =
communications channel for this information. Or we can leave it as it =
is. The current text is terse but correct.

I agree that you shouldn't try to describe the whole process of how to =
do this, but I do think it needs a little more than is in the current =
draft. In particular, I can imagine someone reading the draft and =
thinking that because they have NTP, they know everything they need to =
know about leap seconds, and can just use the NTP timestamps. =46rom my =
point of view, paraphrasing what you wrote about into the draft would be =
enough to clarify this.

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




--Apple-Mail=_905A0609-9824-4EE7-8F2C-230FCDA8D8DC
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Diso-8859-1"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><div><div>On 26 Aug 2013, at 17:23, Kevin Gross &lt;<a =
href=3D"mailto:kevin.gross@avanw.com">kevin.gross@avanw.com</a>&gt; =
wrote:</div><blockquote type=3D"cite"><div dir=3D"ltr">Thanks for the =
review. See responses below.</div></blockquote>...<br><blockquote =
type=3D"cite"><div dir=3D"ltr"><div class=3D"gmail_extra"><div><div =
dir=3D"ltr">On Mon, Aug 26, 2013 at 6:57 AM, Colin Perkins <span =
dir=3D"ltr">&lt;<a href=3D"mailto:csp@csperkins.org" =
target=3D"_blank">csp@csperkins.org</a>&gt;</span> =
wrote:</div></div><div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin: 0px 0px 0px 0.8ex; =
border-left-width: 1px; border-left-color: rgb(204, 204, 204); =
border-left-style: solid; padding-left: 1ex; position: static; z-index: =
auto; "><div class=3D"im">On 15 Aug 2013, at 08:46, Magnus Westerlund =
&lt;<a =
href=3D"mailto:magnus.westerlund@ericsson.com">magnus.westerlund@ericsson.=
com</a>&gt; wrote:<br>

&gt; This starts the WG last call on RTP and Leap Seconds with the =
intended<br>
&gt; status of Proposed Standard. Please provide any comments by 1st =
of<br>
&gt; September.<br>
&gt;<br>
&gt; <a =
href=3D"https://datatracker.ietf.org/doc/draft-ietf-avtcore-leap-second" =
target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-avtcore-leap=
-second</a><br></div></blockquote></div></div></div></blockquote>...<br><b=
lockquote type=3D"cite"><div dir=3D"ltr"><div class=3D"gmail_extra"><div =
class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin: =
0px 0px 0px 0.8ex; border-left-width: 1px; border-left-color: rgb(204, =
204, 204); border-left-style: solid; padding-left: 1ex; position: =
relative; z-index: 0; "><div class=3D"im">Section 5 also says NTP =
includes leap seconds, and that implementations working to a =
leap-second-bearing reference need "a working communications channel to =
receive notification of leap-second scheduling". Is having an =
NTP-synchronised clock a sufficient channel to receive the necessary =
notification, or is something more =
needed?</div></blockquote><div><br></div><div>NTP includes a mechanism =
for leap-second notification. That's a good start but what we're trying =
to capture here is the entire communications chain from IERS to the =
slaved clocks. In addition to these protocol mechanisms for delivering =
leap-second notifications, master clocks must have access to an =
up-to-date leap second schedule through some other communications =
channel (e.g. FTP leap-seconds.list from a NIST server).</div>
<div><br></div><div>I can revise to emphasize that time protocol =
notifications are not sufficient but I don't think we want to try and =
describe the whole communications channel for this information. Or we =
can leave it as it is. The current text is terse but =
correct.</div></div></div></div></blockquote><br></div><div =
apple-content-edited=3D"true"><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; border-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"font-size: 9px; "><div>I agree that =
you shouldn't try to describe the whole process of how to do this, but I =
do think it needs a little more than is in the current draft. In =
particular, I can imagine someone reading the draft and thinking that =
because they have NTP, they know everything they need to know about leap =
seconds, and can just use the NTP timestamps. =46rom my point of view, =
paraphrasing what you wrote about into the draft would be enough to =
clarify =
this.</div><div><br></div><div>--&nbsp;</div><div></div><div>Colin =
Perkins</div><div><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a></div><div><br></d=
iv></span></span><br class=3D"Apple-interchange-newline">

</div>
<br></body></html>=

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References: <C1F36850-2B72-4A98-97B7-8847C9C90CB0@live555.com> <3879D71E758A7E4AA99A35DD8D41D3D91D4C7AA7@xmb-rcd-x14.cisco.com> <A0D9B971-CDE0-4AF7-9A70-C280514AEF10@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D4C47@nasanexd02f.na.qualcomm.com> <5D781561-69CD-4E0D-9377-B4B26036E691@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D5FB3@nasanexd02f.na.qualcomm.com> <69F25DB2-6926-4ACA-935D-90376A2A7BFA@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D690F@nasanexd02f.na.qualcomm.com> <A6FDCB43-F221-4BE0-BEDA-1F1E5394DDEC@live555.com> <3879D71E758A7E4AA99A35DD8D41D3D91D4CC65C@xmb-rcd-x14.cisco.com> <F5C111A5-64A6-48AB-BBA8-8812EE49F7ED@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D6FD4@nasanexd02f.na.qualcomm.com> <B995CF20-66A4-47B9-B141-B650290F8DC7@live555.com>
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Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload format	specification's text on when to set the RTP "M" bit
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Hi Ross,

Sorry for a slow response. I started to edit this draft today, and have inc=
luded the following note after the semantics of the M bit:

Informative note: The content of a NAL unit does not tell whether or not th=
e NAL unit is the last NAL unit, in decoding order, of an access unit.  An =
RTP sender implementation may obtain this information from the video encode=
r.  If, however, the implementation cannot obtain this information directly=
 from the encoder, e.g., when the stream was pre-encoded, and also there is=
 no timestamp allocated for each NAL unit, then the sender implementation c=
an inspect subsequent NAL units in decoding order to determine whether or n=
ot the NAL unit is the last NAL unit of an access unit as follows.  A NAL u=
nit naluX is the last NAL unit of an access unit if the next VCL NAL unit n=
aluY in decoding order has the high-order bit of the first byte after its N=
AL unit header equal to 1, and all NAL units between naluX and naluY, when =
present, have nal_unit_type in the range of 32 to 35, inclusive, equal to 3=
9, in the range of 41 to 44, inclusive, or in the range of 48 to 55, inclus=
ive.

Would you be fine with this wording?

BR, YK


From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Ross =
Finlayson
Sent: Monday, August 12, 2013 12:25 PM
To: avt@ietf.org; payload@ietf.org
Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload format sp=
ecification's text on when to set the RTP "M" bit

You just need one more little step to get everything right here, instead of=
 thinking of compromising the implementation :)

That is, read subclause 7.4.2.4.4 of the HEVC spec. The most relevant parag=
raphs are copied below for convenience (the first paragraph itself contains=
 the answer to your question, but the second paragraph may also provide hel=
pful information)

Thanks for the info; that was useful.

Here, then is the updated text of a paragraph - to be added after the exist=
ing 'M bit' text - that clarifies when/how to set it:
----------
Unfortunately the contents of a NAL unit, alone, do not tell a RTP sender i=
mplementation whether or not the NAL unit ends an access unit.  Instead, th=
e implementation may be able to obtain this information separately, from th=
e encoder.  If, however, the implementation cannot obtain this information =
directly from the encoder (e.g., because the implementation is sending data=
 that consists solely of a sequence of pre-encoded NAL units), then it must=
 instead inspect subsequent NAL units, to determine whether or not the NAL =
unit ends an access unit.  If the NAL units have already been timestamped, =
then these timestamps can be used to determine the access unit boundaries. =
 Otherwise, the access unit boundaries (and times) must be determined by in=
specting the NAL units' contents - specifically, the "nal_unit_type" and (f=
or VCL NAL units) the "first_slice_segment_in_pic_flag".  The following rul=
e can be used:
           A NAL unit (X) ends an access unit if the next-occurring VCL NAL=
 unit (Y) has the high-order bit of the first byte after its NAL unit heade=
r equal to 1, and all NAL units, if any, between (X) and (Y) have a "nal_un=
it_type" in one of the following ranges: [32,35], 39, [41,44], or [48,55].
----------

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi Ross,<o:p></o:p></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"><o:p>&nbsp;</o:p></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">Sorry for a slow response=
. I started to edit this draft today, and have included the following note =
after the semantics of the M bit:<o:p></o:p></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"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal" style=3D"margin-left:.9in">Informative note: The con=
tent of a NAL unit does not tell whether or not the NAL unit is the last NA=
L unit, in decoding order, of an access unit.&nbsp; An RTP sender implement=
ation may obtain this information from the
 video encoder.&nbsp; If, however, the implementation cannot obtain this in=
formation directly from the encoder, e.g., when the stream was pre-encoded,=
 and also there is no timestamp allocated for each NAL unit, then the sende=
r implementation can inspect subsequent
 NAL units in decoding order to determine whether or not the NAL unit is th=
e last NAL unit of an access unit as follows.&nbsp; A NAL unit naluX is the=
 last NAL unit of an access unit if the next VCL NAL unit naluY in decoding=
 order has the high-order bit of the
 first byte after its NAL unit header equal to 1, and all NAL units between=
 naluX and naluY, when present, have nal_unit_type in the range of 32 to 35=
, inclusive, equal to 39, in the range of 41 to 44, inclusive, or in the ra=
nge of 48 to 55, inclusive.<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></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">Would you be fine with th=
is wording?<o:p></o:p></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"><o:p>&nbsp;</o:p></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">BR, YK<o:p></o:p></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"><o:p>&nbsp;</o:p></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"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span =
lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&qu=
ot;sans-serif&quot;"> avt-bounces@ietf.org [mailto:avt-bounces@ietf.org]
<b>On Behalf Of </b>Ross Finlayson<br>
<b>Sent:</b> Monday, August 12, 2013 12:25 PM<br>
<b>To:</b> avt@ietf.org; payload@ietf.org<br>
<b>Subject:</b> Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload fo=
rmat specification's text on when to set the RTP &quot;M&quot; bit<o:p></o:=
p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">You just need one more li=
ttle step to get everything right here, instead of thinking of compromising=
 the implementation<span class=3D"apple-converted-space">&nbsp;</span></spa=
n><span style=3D"font-size:11.0pt;font-family:Wingdings;color:#1F497D">J</s=
pan><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">That is, read subclause 7=
.4.2.4.4 of the HEVC spec. The most relevant paragraphs are copied below fo=
r convenience (the first paragraph itself contains the answer
 to your question, but the second paragraph may also provide helpful inform=
ation)</span><o:p></o:p></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">Thanks for the info; that was useful.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Here, then is the updated text of a paragraph - to b=
e added after the existing 'M bit' text - that clarifies when/how to set it=
:<o:p></o:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal">----------<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Unfortunately the contents of a NAL unit, alone, do =
not tell a RTP sender implementation whether or not the NAL unit ends an ac=
cess unit. &nbsp;Instead, the implementation may be able to obtain this inf=
ormation separately, from the encoder.
 &nbsp;If, however, the implementation cannot obtain this information direc=
tly from the encoder (e.g., because the implementation is sending data that=
 consists solely of a sequence of pre-encoded NAL units), then it must inst=
ead inspect subsequent NAL units, to
 determine whether or not the NAL unit ends an access unit. &nbsp;If the NA=
L units have already been timestamped, then these timestamps can be used to=
 determine the access unit boundaries. &nbsp;Otherwise, the access unit bou=
ndaries (and times) must be determined by
 inspecting the NAL units' contents - specifically, the &quot;nal_unit_type=
&quot; and (for VCL NAL units) the &quot;first_slice_segment_in_pic_flag&qu=
ot;. &nbsp;The following rule can be used:<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span class=3D"apple-tab-span">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>&nbsp;A NAL unit (X) ends an acces=
s unit if the next-occurring VCL NAL unit (Y) has the high-order bit of the=
 first byte after its NAL unit header equal to 1, and all NAL units, if any=
, between (X)
 and (Y) have a &quot;nal_unit_type&quot; in one of the following ranges: [=
32,35], 39, [41,44], or [48,55].<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">----------<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span class=3D"apple-style-span"><span style=3D"font=
-size:13.5pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif&quot;;color=
:black">Ross Finlayson</span></span><span style=3D"font-size:13.5pt;font-fa=
mily:&quot;Helvetica&quot;,&quot;sans-serif&quot;;color:black"><br>
<span class=3D"apple-style-span">Live Networks, Inc.</span><br>
<span class=3D"apple-style-span"><a href=3D"http://www.live555.com/">http:/=
/www.live555.com/</a></span></span>
<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
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From: Ross Finlayson <finlayson@live555.com>
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Date: Mon, 26 Aug 2013 22:38:53 -0700
Message-Id: <969F6DF9-BABB-4438-BAC0-E0F36049ED63@live555.com>
References: <C1F36850-2B72-4A98-97B7-8847C9C90CB0@live555.com> <3879D71E758A7E4AA99A35DD8D41D3D91D4C7AA7@xmb-rcd-x14.cisco.com> <A0D9B971-CDE0-4AF7-9A70-C280514AEF10@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D4C47@nasanexd02f.na.qualcomm.com> <5D781561-69CD-4E0D-9377-B4B26036E691@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D5FB3@nasanexd02f.na.qualcomm.com> <69F25DB2-6926-4ACA-935D-90376A2A7BFA@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D690F@nasanexd02f.na.qualcomm.com> <A6FDCB43-F221-4BE0-BEDA-1F1E5394DDEC@live555.com> <3879D71E758A7E4AA99A35DD8D41D3D91D4CC65C@xmb-rcd-x14.cisco.com> <F5C111A5-64A6-48AB-BBA8-8812EE49F7ED@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D6FD4@nasanexd02f.na.qualcomm.com> <B995CF20-66A4-47B9-B141-B650290F8DC7@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83385120E1@nasanexd02f.na.qualcomm.com>
To: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com>
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Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload format	 specification's text on when to set the RTP "M" bit
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On Aug 26, 2013, at 9:50 PM, "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com> =
wrote:

> Hi Ross,
> =20
> Sorry for a slow response. I started to edit this draft today, and =
have included the following note after the semantics of the M bit:
> =20
> Informative note: The content of a NAL unit does not tell whether or =
not the NAL unit is the last NAL unit, in decoding order, of an access =
unit.  An RTP sender implementation may obtain this information from the =
video encoder.  If, however, the implementation cannot obtain this =
information directly from the encoder, e.g., when the stream was =
pre-encoded, and also there is no timestamp allocated for each NAL unit, =
then the sender implementation can inspect subsequent NAL units in =
decoding order to determine whether or not the NAL unit is the last NAL =
unit of an access unit as follows.  A NAL unit naluX is the last NAL =
unit of an access unit if the next VCL NAL unit naluY in decoding order =
has the high-order bit of the first byte after its NAL unit header equal =
to 1, and all NAL units between naluX and naluY, when present, have =
nal_unit_type in the range of 32 to 35, inclusive, equal to 39, in the =
range of 41 to 44, inclusive, or in the range of 48 to 55, inclusive.
> =20
> Would you be fine with this wording?

Yes, this looks good to me.


Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


--Apple-Mail=_A2B275B2-78D8-432C-9001-6B09994EF54F
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=us-ascii

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"><base href=3D"x-msg://971/"></head><body =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Aug 26, 2013, =
at 9:50 PM, "Wang, Ye-Kui" &lt;<a =
href=3D"mailto:yekuiw@qti.qualcomm.com">yekuiw@qti.qualcomm.com</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div lang=3D"EN-CA" link=3D"blue" vlink=3D"purple" =
style=3D"font-family: Helvetica; font-size: medium; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; =
"><div class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); ">Hi =
Ross,<o:p></o:p></span></div><div style=3D"margin: 0in 0in 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); ">&nbsp;</span></div><div style=3D"margin: 0in 0in =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; =
color: rgb(31, 73, 125); ">Sorry for a slow response. I started to edit =
this draft today, and have included the following note after the =
semantics of the M bit:<o:p></o:p></span></div><div style=3D"margin: 0in =
0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; =
color: rgb(31, 73, 125); ">&nbsp;</span></div><div style=3D"margin: 0in =
0in 0.0001pt 0.9in; font-size: 12pt; font-family: 'Times New Roman', =
serif; ">Informative note: The content of a NAL unit does not tell =
whether or not the NAL unit is the last NAL unit, in decoding order, of =
an access unit.&nbsp; An RTP sender implementation may obtain this =
information from the video encoder.&nbsp; If, however, the =
implementation cannot obtain this information directly from the encoder, =
e.g., when the stream was pre-encoded, and also there is no timestamp =
allocated for each NAL unit, then the sender implementation can inspect =
subsequent NAL units in decoding order to determine whether or not the =
NAL unit is the last NAL unit of an access unit as follows.&nbsp; A NAL =
unit naluX is the last NAL unit of an access unit if the next VCL NAL =
unit naluY in decoding order has the high-order bit of the first byte =
after its NAL unit header equal to 1, and all NAL units between naluX =
and naluY, when present, have nal_unit_type in the range of 32 to 35, =
inclusive, equal to 39, in the range of 41 to 44, inclusive, or in the =
range of 48 to 55, inclusive.<o:p></o:p></div><div style=3D"margin: 0in =
0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; =
color: rgb(31, 73, 125); ">&nbsp;</span></div><div style=3D"margin: 0in =
0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; =
color: rgb(31, 73, 125); ">Would you be fine with this =
wording?</span></div></div></div></blockquote><div><br></div>Yes, this =
looks good to me.</div><br><br><div apple-content-edited=3D"true">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; ">Ross =
Finlayson<br>Live Networks, Inc.<br><a =
href=3D"http://www.live555.com/">http://www.live555.com/</a></span></span>=

</div>
<br></body></html>=

--Apple-Mail=_A2B275B2-78D8-432C-9001-6B09994EF54F--

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Subject: [AVTCORE] I-D Action: draft-ietf-avtcore-leap-second-04.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 Core Maintenance Wo=
rking Group of the IETF.

	Title           : RTP and Leap Seconds
	Author(s)       : Kevin Gross
                          Ray van Brandenburg
	Filename        : draft-ietf-avtcore-leap-second-04.txt
	Pages           : 8
	Date            : 2013-08-27

Abstract:
   This document discusses issues that arise when RTP sessions span
   Coordinated Universal Time (UTC) leap seconds.  It updates RFC 3550
   to describe how RTP senders and receivers should behave in the
   presence of leap seconds.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-avtcore-leap-second

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-avtcore-leap-second-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-leap-second-04


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

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


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--047d7bb0435ca4f70104e4efc4c4
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This revision should resolve Colin's recent comments.

Kevin Gross
+1-303-447-0517
Media Network Consultant
AVA Networks - www.AVAnw.com <http://www.avanw.com/>


On Tue, Aug 27, 2013 at 9:34 AM, <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 Core Maintenance
> Working Group of the IETF.
>
>         Title           : RTP and Leap Seconds
>         Author(s)       : Kevin Gross
>                           Ray van Brandenburg
>         Filename        : draft-ietf-avtcore-leap-second-04.txt
>         Pages           : 8
>         Date            : 2013-08-27
>
> Abstract:
>    This document discusses issues that arise when RTP sessions span
>    Coordinated Universal Time (UTC) leap seconds.  It updates RFC 3550
>    to describe how RTP senders and receivers should behave in the
>    presence of leap seconds.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-avtcore-leap-second
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-avtcore-leap-second-04
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-leap-second-04
>
>
> Please note that it may take a couple of minutes from the time of
> submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>

--047d7bb0435ca4f70104e4efc4c4
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">This revision should resolve Colin&#39;s recent comments.<=
/div><div class=3D"gmail_extra"><br clear=3D"all"><div><div dir=3D"ltr">Kev=
in Gross<br><div>+1-303-447-0517</div><div>Media Network Consultant<br><div=
>AVA Networks -=A0<a href=3D"http://www.avanw.com/" target=3D"_blank">www.A=
VAnw.com</a></div>
</div></div></div>
<br><br><div class=3D"gmail_quote">On Tue, Aug 27, 2013 at 9:34 AM,  <span =
dir=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blan=
k">internet-drafts@ietf.org</a>&gt;</span> wrote:<br><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex">
<br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=A0This draft is a work item of the Audio/Video Transport Core Maintenance =
Working Group of the IETF.<br>
<br>
=A0 =A0 =A0 =A0 Title =A0 =A0 =A0 =A0 =A0 : RTP and Leap Seconds<br>
=A0 =A0 =A0 =A0 Author(s) =A0 =A0 =A0 : Kevin Gross<br>
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Ray van Brandenburg<br>
=A0 =A0 =A0 =A0 Filename =A0 =A0 =A0 =A0: draft-ietf-avtcore-leap-second-04=
.txt<br>
=A0 =A0 =A0 =A0 Pages =A0 =A0 =A0 =A0 =A0 : 8<br>
=A0 =A0 =A0 =A0 Date =A0 =A0 =A0 =A0 =A0 =A0: 2013-08-27<br>
<br>
Abstract:<br>
=A0 =A0This document discusses issues that arise when RTP sessions span<br>
=A0 =A0Coordinated Universal Time (UTC) leap seconds. =A0It updates RFC 355=
0<br>
=A0 =A0to describe how RTP senders and receivers should behave in the<br>
=A0 =A0presence of leap seconds.<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-avtcore-leap-second"=
 target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-avtcore-leap=
-second</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-avtcore-leap-second-04" ta=
rget=3D"_blank">http://tools.ietf.org/html/draft-ietf-avtcore-leap-second-0=
4</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-leap-secon=
d-04" target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtco=
re-leap-second-04</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
_______________________________________________<br>
Audio/Video Transport Core Maintenance<br>
<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt" target=3D"_blank">htt=
ps://www.ietf.org/mailman/listinfo/avt</a><br>
</blockquote></div><br></div>

--047d7bb0435ca4f70104e4efc4c4--

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The following errata report has been submitted for RFC3711,
"The Secure Real-time Transport Protocol (SRTP)".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=3711&eid=3712

--------------------------------------
Type: Technical
Reported by: Christian S Oien <coien@cisco.com>

Section: 4.3.2

Original Text
-------------
Replace the SRTP index by the 32-bit quantity: 0 || SRTCP index
(i.e., excluding the E-bit, replacing it with a fixed 0-bit), and use


Corrected Text
--------------
Replace the SRTP index by the 48-bit quantity: 0 || SRTCP index
(i.e., excluding the E-bit, replacing it with a fixed 0-bit and then
prepending a 16-bits zero padding), and use


Notes
-----
Replacing with a 32-bit quantity means that the DIV operator will
yield a 32-bit quantity.  Following the specification of key_id for SRTCP
the <label> will have 32 bits to its right when XOR'ing with master_salt.

The majority of implementations, including libsrtp, invokes this XOR with the
<label> at the same position as for SRTP.  According to the specification
this should be done 16 bits to the right of this, when invoking for SRTCP.

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. 

--------------------------------------
RFC3711 (draft-ietf-avt-srtp-09)
--------------------------------------
Title               : The Secure Real-time Transport Protocol (SRTP)
Publication Date    : March 2004
Author(s)           : M. Baugher, D. McGrew, M. Naslund, E. Carrara, K. Norrman
Category            : PROPOSED STANDARD
Source              : Audio/Video Transport
Area                : Real-time Applications and Infrastructure
Stream              : IETF
Verifying Party     : IESG

From mzanaty@cisco.com  Tue Aug 27 15:58:13 2013
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com>
Thread-Topic: [AVTCORE] [payload] Clarifying the H.265 RTP payload	format specification's text on when to set the RTP "M" bit
Thread-Index: AQHOo3jjnEEyNQElLUmyU/ypjzM8DQ==
Date: Tue, 27 Aug 2013 22:58:04 +0000
Message-ID: <75CC2B0E-28F2-452C-9CE5-060DB8DFEE6B@cisco.com>
References: <C1F36850-2B72-4A98-97B7-8847C9C90CB0@live555.com> <3879D71E758A7E4AA99A35DD8D41D3D91D4C7AA7@xmb-rcd-x14.cisco.com> <A0D9B971-CDE0-4AF7-9A70-C280514AEF10@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D4C47@nasanexd02f.na.qualcomm.com> <5D781561-69CD-4E0D-9377-B4B26036E691@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D5FB3@nasanexd02f.na.qualcomm.com> <69F25DB2-6926-4ACA-935D-90376A2A7BFA@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D690F@nasanexd02f.na.qualcomm.com> <A6FDCB43-F221-4BE0-BEDA-1F1E5394DDEC@live555.com> <3879D71E758A7E4AA99A35DD8D41D3D91D4CC65C@xmb-rcd-x14.cisco.com> <F5C111A5-64A6-48AB-BBA8-8812EE49F7ED@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83384D6FD4@nasanexd02f.na.qualcomm.com> <B995CF20-66A4-47B9-B141-B650290F8DC7@live555.com> <8BA7D4CEACFFE04BA2D902BF11719A83385120E1@nasanexd02f.na.qualcomm.com>
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Cc: Ross Finlayson <finlayson@live555.com>, "avt@ietf.org" <avt@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload	format	specification's text on when to set the RTP "M" bit
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From: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com>
To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
Thread-Topic: [AVTCORE] [payload] Clarifying the H.265 RTP payload	format specification's text on when to set the RTP "M" bit
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Date: Wed, 28 Aug 2013 01:36:20 +0000
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Cc: Ross Finlayson <finlayson@live555.com>, "avt@ietf.org" <avt@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload	format	specification's text on when to set the RTP "M" bit
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com>
Thread-Topic: [AVTCORE] [payload] Clarifying the H.265 RTP payload	format specification's text on when to set the RTP "M" bit
Thread-Index: AQHOo3jjnEEyNQElLUmyU/ypjzM8DZmqKt0A///IteA=
Date: Wed, 28 Aug 2013 04:21:36 +0000
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Cc: Ross Finlayson <finlayson@live555.com>, "avt@ietf.org" <avt@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload	format	specification's text on when to set the RTP "M" bit
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From: Woo-Hwan Kim <whkim5@ensec.re.kr>
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Subject: Re: [AVTCORE] WG last call on draft-ietf-avtcore-aria-srtp-04
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I have searched for the acronym RTCP in several RFC documents.
RTCP is an acronym for RTP Control Protocol in RFCs 3550 and 3611, but
it is an acronym for Real-time-Transport Control Protocol in RFCs 3605,
3711, 4585.
I think any expression is admissible since both of them are widely used and
there is no confusion when one or the other is used.
If necessary, I'll correct the draft.

The byte order for the test vector is the network order (called big endian)
which has been adopted by most documents.
I'll add a comment in the next revision.

Thanks for your comments.

Sincerely,
Woo-Hwan Kim

> Date: Mon, 26 Aug 2013 13:30:52 +0100
> From: Colin Perkins <csp@csperkins.org>
> From a quick look, this looks okay to me. Some nits:
>
> The acronym RTCP is expanded incorrectly in the abstract and Introduction
(should be "RTP Control Protocol" to match RFC3550).
>
> Appendix A: what is the byte order for the test vectors? (especially for
fields like the seqnum and SSRC, which are commonly represented by ints in
code)
>
> Cheers,
> Colin
>
>
> --
> Colin Perkins
> http://csperkins.org/
>

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<div dir=3D"ltr"><div><div><font face=3D"arial, sans-serif"><span style=3D"=
font-size:14px">I have searched for the acronym RTCP in several RFC documen=
ts.</span></font></div><div><font face=3D"arial, sans-serif"><span style=3D=
"font-size:14px">RTCP is an acronym for RTP Control Protocol in RFCs 3550 a=
nd 3611, but</span></font></div>
<div><font face=3D"arial, sans-serif"><span style=3D"font-size:14px">it is =
an acronym for Real-time-Transport Control Protocol in RFCs 3605, 3711, 458=
5.</span></font></div><div><font face=3D"arial, sans-serif"><span style=3D"=
font-size:14px">I think any expression is admissible since both of them are=
 widely used and there is no confusion when one or the other is used.</span=
></font></div>
<div><font face=3D"arial, sans-serif"><span style=3D"font-size:14px">If nec=
essary, I&#39;ll correct the draft.</span></font></div><div><font face=3D"a=
rial, sans-serif"><span style=3D"font-size:14px"><br></span></font></div><d=
iv><font face=3D"arial, sans-serif"><span style=3D"font-size:14px">The byte=
 order for the test vector is the network order (called big endian) which h=
as been adopted by most documents.</span></font></div>
<div><font face=3D"arial, sans-serif"><span style=3D"font-size:14px">I&#39;=
ll add a comment in the next revision.</span></font></div><div><font face=
=3D"arial, sans-serif"><span style=3D"font-size:14px"><br></span></font></d=
iv><div>
<font face=3D"arial, sans-serif"><span style=3D"font-size:14px">Thanks for =
your comments.</span></font></div><div><font face=3D"arial, sans-serif"><sp=
an style=3D"font-size:14px"><br></span></font></div><div><font face=3D"aria=
l, sans-serif"><span style=3D"font-size:14px">Sincerely,</span></font></div=
>
<div><font face=3D"arial, sans-serif"><span style=3D"font-size:14px">Woo-Hw=
an Kim</span></font></div><div style=3D"font-family:arial,sans-serif;font-s=
ize:14px"><br></div></div><span style=3D"font-family:arial,sans-serif;font-=
size:14px">&gt; Date: Mon, 26 Aug 2013 13:30:52 +0100</span><br style=3D"fo=
nt-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; From: Coli=
n Perkins &lt;</span><a href=3D"mailto:csp@csperkins.org" style=3D"font-fam=
ily:arial,sans-serif;font-size:14px">csp@csperkins.org</a><span style=3D"fo=
nt-family:arial,sans-serif;font-size:14px">&gt;</span><div>
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; From a qui=
ck look, this looks okay to me. Some nits:</span><br style=3D"font-family:a=
rial,sans-serif;font-size:14px"><span style=3D"font-family:arial,sans-serif=
;font-size:14px">&gt;</span><br style=3D"font-family:arial,sans-serif;font-=
size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; The acrony=
m RTCP is expanded incorrectly in the abstract and Introduction (should be =
&quot;RTP Control Protocol&quot; to match RFC3550).</span><br style=3D"font=
-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt; Appendix A: what is the byte or=
der for the test vectors? (especially for fields like the seqnum and SSRC, =
which are commonly represented by ints in code)</span><br style=3D"font-fam=
ily:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt; Cheers,</span><br style=3D"font=
-family:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; Colin</spa=
n><br style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"=
font-family:arial,sans-serif;font-size:14px">&gt;</span><br style=3D"font-f=
amily:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br =
style=3D"font-family:arial,sans-serif;font-size:14px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:14px">&gt; --</span><br style=3D"font-fami=
ly:arial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt; Colin Perk=
ins</span><br style=3D"font-family:arial,sans-serif;font-size:14px"><span s=
tyle=3D"font-family:arial,sans-serif;font-size:14px">&gt;=A0</span><a href=
=3D"http://csperkins.org/" target=3D"_blank" style=3D"font-family:arial,san=
s-serif;font-size:14px">http://csperkins.org/</a><br style=3D"font-family:a=
rial,sans-serif;font-size:14px">
<span style=3D"font-family:arial,sans-serif;font-size:14px">&gt;</span><br>=
</div></div>

--f46d04428e401e105404e4fa712f--

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Thanks, this looks good to me.
Colin


On 27 Aug 2013, at 16:45, Kevin Gross <kevin.gross@avanw.com> wrote:

> This revision should resolve Colin's recent comments.
>=20
> Kevin Gross
> +1-303-447-0517
> Media Network Consultant
> AVA Networks - www.AVAnw.com
>=20
>=20
> On Tue, Aug 27, 2013 at 9:34 AM, <internet-drafts@ietf.org> wrote:
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
>  This draft is a work item of the Audio/Video Transport Core =
Maintenance Working Group of the IETF.
>=20
>         Title           : RTP and Leap Seconds
>         Author(s)       : Kevin Gross
>                           Ray van Brandenburg
>         Filename        : draft-ietf-avtcore-leap-second-04.txt
>         Pages           : 8
>         Date            : 2013-08-27
>=20
> Abstract:
>    This document discusses issues that arise when RTP sessions span
>    Coordinated Universal Time (UTC) leap seconds.  It updates RFC 3550
>    to describe how RTP senders and receivers should behave in the
>    presence of leap seconds.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-avtcore-leap-second
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-avtcore-leap-second-04
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-leap-second-04
>=20
>=20
> Please note that it may take a couple of minutes from the time of =
submission
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
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>=20
> _______________________________________________
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--=20
Colin Perkins
http://csperkins.org/




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<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Thanks, this looks good to me.<div>Colin</div><div><br></div><div><br><div><div>On 27 Aug 2013, at 16:45, Kevin Gross &lt;<a href="mailto:kevin.gross@avanw.com">kevin.gross@avanw.com</a>&gt; wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div dir="ltr">This revision should resolve Colin's recent comments.</div><div class="gmail_extra"><br clear="all"><div><div dir="ltr">Kevin Gross<br><div>+1-303-447-0517</div><div>Media Network Consultant<br><div>AVA Networks -&nbsp;<a href="http://www.avanw.com/" target="_blank">www.AVAnw.com</a></div>
</div></div></div>
<br><br><div class="gmail_quote">On Tue, Aug 27, 2013 at 9:34 AM,  <span dir="ltr">&lt;<a href="mailto:internet-drafts@ietf.org" target="_blank">internet-drafts@ietf.org</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
A New Internet-Draft is available from the on-line Internet-Drafts directories.<br>
&nbsp;This draft is a work item of the Audio/Video Transport Core Maintenance Working Group of the IETF.<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; Title &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : RTP and Leap Seconds<br>
&nbsp; &nbsp; &nbsp; &nbsp; Author(s) &nbsp; &nbsp; &nbsp; : Kevin Gross<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Ray van Brandenburg<br>
&nbsp; &nbsp; &nbsp; &nbsp; Filename &nbsp; &nbsp; &nbsp; &nbsp;: draft-ietf-avtcore-leap-second-04.txt<br>
&nbsp; &nbsp; &nbsp; &nbsp; Pages &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : 8<br>
&nbsp; &nbsp; &nbsp; &nbsp; Date &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;: 2013-08-27<br>
<br>
Abstract:<br>
&nbsp; &nbsp;This document discusses issues that arise when RTP sessions span<br>
&nbsp; &nbsp;Coordinated Universal Time (UTC) leap seconds. &nbsp;It updates RFC 3550<br>
&nbsp; &nbsp;to describe how RTP senders and receivers should behave in the<br>
&nbsp; &nbsp;presence of leap seconds.<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href="https://datatracker.ietf.org/doc/draft-ietf-avtcore-leap-second" target="_blank">https://datatracker.ietf.org/doc/draft-ietf-avtcore-leap-second</a><br>
<br>
There's also a htmlized version available at:<br>
<a href="http://tools.ietf.org/html/draft-ietf-avtcore-leap-second-04" target="_blank">http://tools.ietf.org/html/draft-ietf-avtcore-leap-second-04</a><br>
<br>
A diff from the previous version is available at:<br>
<a href="http://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-leap-second-04" target="_blank">http://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-leap-second-04</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submission<br>
until the htmlized version and diff are available at <a href="http://tools.ietf.org/" target="_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href="ftp://ftp.ietf.org/internet-drafts/" target="_blank">ftp://ftp.ietf.org/internet-drafts/</a><br>
<br>
_______________________________________________<br>
Audio/Video Transport Core Maintenance<br>
<a href="mailto:avt@ietf.org">avt@ietf.org</a><br>
<a href="https://www.ietf.org/mailman/listinfo/avt" target="_blank">https://www.ietf.org/mailman/listinfo/avt</a><br>
</blockquote></div><br></div>
_______________________________________________<br>Audio/Video Transport Core Maintenance<br><a href="mailto:avt@ietf.org">avt@ietf.org</a><br>https://www.ietf.org/mailman/listinfo/avt<br></blockquote></div><br><div apple-content-edited="true">
<span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; font-size: medium; "><span class="Apple-style-span" style="font-size: 9px; "><div><br class="Apple-interchange-newline"><br class="khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div>Colin Perkins</div><div><a href="http://csperkins.org/">http://csperkins.org/</a></div><div><br></div></span></span><br class="Apple-interchange-newline">

</div>
<br></div></body></html>
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, draft-ietf-avtcore-aria-srtp@tools.ietf.org, avt@ietf.org
Subject: Re: [AVTCORE] WG last call on draft-ietf-avtcore-aria-srtp-04
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On 28 Aug 2013, at 05:29, Woo-Hwan Kim <whkim5@ensec.re.kr> wrote:
> I have searched for the acronym RTCP in several RFC documents.
> RTCP is an acronym for RTP Control Protocol in RFCs 3550 and 3611, but
> it is an acronym for Real-time-Transport Control Protocol in RFCs =
3605, 3711, 4585.
> I think any expression is admissible since both of them are widely =
used and there is no confusion when one or the other is used.
> If necessary, I'll correct the draft.

Please correct it. The expansion in RFC 3550 is definitive. We shouldn't =
perpetuate the mistakes in other RFCs.

> The byte order for the test vector is the network order (called big =
endian) which has been adopted by most documents. I'll add a comment in =
the next revision.

Thanks!
Colin



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From: "Wang, Ye-Kui" <yekuiw@qti.qualcomm.com>
To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
Thread-Topic: [AVTCORE] [payload] Clarifying the H.265 RTP payload	format specification's text on when to set the RTP "M" bit
Thread-Index: AQHOo3jm1moEaDpZa0yx3vgEUVhvCZmp1oPggACkDgCAAGUh8A==
Date: Wed, 28 Aug 2013 17:55:11 +0000
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Cc: Ross Finlayson <finlayson@live555.com>, "avt@ietf.org" <avt@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Subject: Re: [AVTCORE] [payload] Clarifying the H.265 RTP payload	format	specification's text on when to set the RTP "M" bit
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From alan.b.johnston@gmail.com  Wed Aug 28 11:32:14 2013
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From: Alan Johnston <alan.b.johnston@gmail.com>
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Subject: [AVTCORE] Review of draft-ietf-avtcore-rtp-security-options-04
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--047d7bb045080ca3ad04e506354f
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Here's my review of draft-ietf-avtcore-rtp-security-options-04.

Summary:  The document is well-written and useful.  There are a few
technical issues to resolve but is otherwise ready to progress.

- Alan -

Numbers refer to sections in the document.

3.1.1:

It is surprising no mention is made of RFC 4474 in this section (or in the
whole document) as DTLS-SRTP keying relies on an integrity protected
signaling channel provided by RFC 4474.  When mentioning RFC 4474, it needs
to be pointed out that RFC 4474 is effectively being deprecated by the STIR
work and has proved to be undeployable and also ineffective for E.164
identities.

3.1.5:

Might be worth a single sentence about what makes ZRTP unique, such as:
 "ZRTP provides best effort encryption independent of the signaling
protocol and utilizes key continuity, Short Authentication Strings, or a
PKI for authentication."

5.1:

"There also exists a solution that enables the fingerprints to be bound to
identities established by the first proxy for each user [RFC4916]."  I
don't see how SIP connected identity relates at all to this.  Is this
perhaps meant to reference RFC 4474?

5.2:

"But, unless the application and its server is intending to compromise the
communication security, they can provide a secure and integrity-protected
exchange of the certificate hashes thus preventing any man-in-the-middle
(MITM) from inserting itself in the key-exchange."  This is possible, but
not trivial, and I doubt few WebRTC signaling channels will provide this.
 This statement needs some support for it to be included.

5.2:

Might also be worth a mention that ZRTP can be used to verify the
DTLS-SRTP fingerprints used without relying on any third parties as
described in draft-johnston-rtcweb-zrtp.

Nits:

2:

"it's" -> "its"

2.1:

"end-points verifiable identity" -> "end-point's verifiable identity"

--047d7bb045080ca3ad04e506354f
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr"><span style=3D"font-family:arial,sans-serif;font-size:13px=
">Here&#39;s my review of draft-ietf-avtcore-rtp-</span><span style=3D"font=
-family:arial,sans-serif;font-size:13px">security-options-04.</span><br sty=
le=3D"font-family:arial,sans-serif;font-size:13px">
<br style=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">Summary: =A0The document is well=
-written and useful. =A0There are a few</span><br style=3D"font-family:aria=
l,sans-serif;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">technical issue=
s to resolve but is otherwise ready to progress.</span><br style=3D"font-fa=
mily:arial,sans-serif;font-size:13px"><br style=3D"font-family:arial,sans-s=
erif;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">- Alan -</span>=
<br style=3D"font-family:arial,sans-serif;font-size:13px"><br style=3D"font=
-family:arial,sans-serif;font-size:13px"><span style=3D"font-family:arial,s=
ans-serif;font-size:13px">Numbers refer to sections in the document.</span>=
<br style=3D"font-family:arial,sans-serif;font-size:13px">
<br style=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">3.1.1:</span><br style=3D"font-f=
amily:arial,sans-serif;font-size:13px"><br style=3D"font-family:arial,sans-=
serif;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">It is surprisin=
g no mention is made of RFC 4474 in this section (or in the</span><br style=
=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"font-family=
:arial,sans-serif;font-size:13px">whole document) as DTLS-SRTP keying relie=
s on an integrity protected</span><br style=3D"font-family:arial,sans-serif=
;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">signaling chann=
el provided by RFC 4474. =A0When mentioning RFC 4474, it needs</span><br st=
yle=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"font-fam=
ily:arial,sans-serif;font-size:13px">to be pointed out that RFC 4474 is eff=
ectively being deprecated by the STIR</span><br style=3D"font-family:arial,=
sans-serif;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">work and has pr=
oved to be undeployable and also ineffective for E.164</span><br style=3D"f=
ont-family:arial,sans-serif;font-size:13px"><span style=3D"font-family:aria=
l,sans-serif;font-size:13px">identities.</span><br style=3D"font-family:ari=
al,sans-serif;font-size:13px">
<br style=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">3.1.5:</span><br style=3D"font-f=
amily:arial,sans-serif;font-size:13px"><br style=3D"font-family:arial,sans-=
serif;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">Might be worth =
a single sentence about what makes ZRTP unique, such as:</span><br style=3D=
"font-family:arial,sans-serif;font-size:13px"><span style=3D"font-family:ar=
ial,sans-serif;font-size:13px">=A0&quot;ZRTP provides best effort encryptio=
n independent of the signaling</span><br style=3D"font-family:arial,sans-se=
rif;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">protocol and ut=
ilizes key continuity, Short Authentication Strings, or a</span><br style=
=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"font-family=
:arial,sans-serif;font-size:13px">PKI for authentication.&quot;</span><br s=
tyle=3D"font-family:arial,sans-serif;font-size:13px">
<br style=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">5.1:</span><br style=3D"font-fam=
ily:arial,sans-serif;font-size:13px"><br style=3D"font-family:arial,sans-se=
rif;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">&quot;There als=
o exists a solution that enables the fingerprints to be bound to</span><br =
style=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:13px">identities established by the first =
proxy for each user [RFC4916].&quot; =A0I</span><br style=3D"font-family:ar=
ial,sans-serif;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">don&#39;t see h=
ow SIP connected identity relates at all to this. =A0Is this</span><br styl=
e=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"font-famil=
y:arial,sans-serif;font-size:13px">perhaps meant to reference RFC 4474?</sp=
an><br style=3D"font-family:arial,sans-serif;font-size:13px">
<br style=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">5.2:</span><br style=3D"font-fam=
ily:arial,sans-serif;font-size:13px"><br style=3D"font-family:arial,sans-se=
rif;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">&quot;But, unle=
ss the application and its server is intending to compromise the</span><br =
style=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"font-f=
amily:arial,sans-serif;font-size:13px">communication security, they can pro=
vide a secure and integrity-protected</span><br style=3D"font-family:arial,=
sans-serif;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">exchange of the=
 certificate hashes thus preventing any man-in-the-middle</span><br style=
=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"font-family=
:arial,sans-serif;font-size:13px">(MITM) from inserting itself in the key-e=
xchange.&quot; =A0This is possible, but</span><br style=3D"font-family:aria=
l,sans-serif;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">not trivial, an=
d I doubt few WebRTC signaling channels will provide this.</span><br style=
=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"font-family=
:arial,sans-serif;font-size:13px">=A0This statement needs some support for =
it to be included.=A0</span><br style=3D"font-family:arial,sans-serif;font-=
size:13px">
<br style=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">5.2:</span><br style=3D"font-fam=
ily:arial,sans-serif;font-size:13px"><br style=3D"font-family:arial,sans-se=
rif;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">Might also be w=
orth a mention that ZRTP can be used to verify the</span><br style=3D"font-=
family:arial,sans-serif;font-size:13px"><span style=3D"font-family:arial,sa=
ns-serif;font-size:13px">DTLS-SRTP fingerprints used without relying on any=
 third parties as</span><br style=3D"font-family:arial,sans-serif;font-size=
:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">described in dr=
aft-johnston-rtcweb-zrtp.</span><br style=3D"font-family:arial,sans-serif;f=
ont-size:13px"><br style=3D"font-family:arial,sans-serif;font-size:13px"><s=
pan style=3D"font-family:arial,sans-serif;font-size:13px">Nits:</span><br s=
tyle=3D"font-family:arial,sans-serif;font-size:13px">
<br style=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">2:</span><br style=3D"font-famil=
y:arial,sans-serif;font-size:13px"><br style=3D"font-family:arial,sans-seri=
f;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">&quot;it&#39;s&=
quot; -&gt; &quot;its&quot;</span><br style=3D"font-family:arial,sans-serif=
;font-size:13px"><br style=3D"font-family:arial,sans-serif;font-size:13px">=
<span style=3D"font-family:arial,sans-serif;font-size:13px">2.1:</span><br =
style=3D"font-family:arial,sans-serif;font-size:13px">
<br style=3D"font-family:arial,sans-serif;font-size:13px"><span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">&quot;end-points verifiable iden=
tity&quot; -&gt; &quot;end-point&#39;s verifiable identity&quot;</span><br>
</div>

--047d7bb045080ca3ad04e506354f--

From jonathan@vidyo.com  Wed Aug 28 11:54:47 2013
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Date: Wed, 28 Aug 2013 14:54:32 -0400
Thread-Topic: [AVTCORE] WG last call on draft-ietf-avtcore-aria-srtp-04
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Subject: Re: [AVTCORE] WG last call on draft-ietf-avtcore-aria-srtp-04
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This document needs a section addressing encryption of RTP header extension=
s (RFC 6904), since that's now a requirement for SRTP encryption algorithms=
.

On Aug 26, 2013, at 5:34 AM, Magnus Westerlund <magnus.westerlund@ericsson.=
com> wrote:

> WG,
>=20
> This starts the WG last call on draft-ietf-avtcore-aria-srtp-04 to be
> published as proposed standard. Please provide any feedback or review
> comments by September 16. Even if you have just read it and found no
> issue, please state so.
>=20
> Draft:
> https://datatracker.ietf.org/doc/draft-ietf-avtcore-aria-srtp
>=20
> In addition any comments about current or planned implementation of this
> draft are appreciated.
>=20
> As a WG chair I really want some persons that really understand crypto
> algorithms and/or the key-management mechanisms used to review this and
> will solicit reviews outside of the WG also.
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
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> Audio/Video Transport Core Maintenance
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>=20

--
Jonathan Lennox
jonathan@vidyo.com



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Subject: [AVTCORE] I-D Action: draft-ietf-avtcore-idms-13.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 Core Maintenance Wo=
rking Group of the IETF.

	Title           : Inter-destination Media Synchronization using the RTP Co=
ntrol Protocol (RTCP)
	Author(s)       : Ray van Brandenburg
                          Hans Stokking
                          Oskar van Deventer
                          Fernando Boronat
                          Mario Montagud
                          Kevin Gross
	Filename        : draft-ietf-avtcore-idms-13.txt
	Pages           : 23
	Date            : 2013-08-29

Abstract:
   This document defines a new RTP Control Protocol (RTCP) Packet Type
   and an RTCP Extended Report (XR) Block Type to be used for achieving
   Inter-Destination Media Synchronization (IDMS).  IDMS is the process
   of synchronizing playout across multiple geographically distributed
   media receivers.  Using the RTCP XR IDMS Report Block defined in this
   document, media playout information from participants in a
   synchronization group can be collected.  Based on the collected
   information, an RTCP IDMS Settings Packet can then be sent to
   distribute a common target playout point to which all the distributed
   receivers, sharing a media experience, can synchronize.

   Typical use cases in which IDMS is useful are social TV, shared
   service control (i.e. applications where two or more geographically
   separated users are watching a media stream together), distance
   learning, networked video walls, networked loudspeakers, etc.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-avtcore-idms

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-avtcore-idms-13

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-idms-13


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

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


From csp@csperkins.org  Thu Aug 29 05:13:26 2013
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From: Colin Perkins <csp@csperkins.org>
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Cc: draft-ietf-avtcore-rtp-security-options@tools.ietf.org, "avt@ietf.org" <avt@ietf.org>
Subject: Re: [AVTCORE] comments on draft-ietf-avtcore-rtp-security-options-04
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Dan,

Thank you for the detailed review. These are all good comments, which =
we'll address in the next version.=20

Addressing the comments on Section 4.1.3 and 5.2 required some fairly =
extensive text changes. I'd be grateful if you could check these when =
the revised draft appears.=20

Cheers,
Colin



On 16 Aug 2013, at 03:18, Dan Wing <dwing@cisco.com> wrote:
> Overall a strong document, and I know it was difficult to write.
>=20
> My comments:
>=20
> In the Introduction, the end of
>   Because of the heterogeneity of
>   RTP applications and use cases, however, a single security solution
>   cannot be mandated.=20
> feels like it could include a citation back to =
draft-ietf-avt-srtp-not-mandatory.
>=20
>=20
> In section 3.1, Depending on when AES-GCM is published compared to =
when draft-ietf-avtcore-rtp-security-options is published, updating the =
following paragraph might be prudent (so it no longer says "ongoing =
work", as it will be published work at that point),
>   AES-GCM:  There is also ongoing work to define AES-GCM (Galois
>      Counter Mode) and AES-CCM (Counter with CBC) authentication for
>      AES-128 and AES-256.  This authentication is included in the
>      cipher text which becomes expanded with the length of the
>      authentication tag instead of using the SRTP authentication tag.
>      This is defined in [I-D.ietf-avtcore-srtp-aes-gcm].
>=20
>=20
> In section 3.1.1,=20
> s/cipher suits/cipher suites/
>=20
>=20
> Section 3.1.4, suggest changing
> OLD:
>   This can enable interworking between DTLS-SRTP and for
>   example security descriptions or other keying systems where either
>   part can set the key.
> NEW:
>   This can enable interworking between DTLS-SRTP and=20
>   other keying systems where either party can set the key (e.g.,=20
>   interworking with security descriptions).
> (original says "either part", should say "either party").
>=20
>=20
> Section 3.4 for consistency with the section title and the rest of the =
section's text, I suggest:
> OLD:
>   There does not appear to be significant usage of RTP over DTLS.
> NEW:
>   There does not appear to be significant usage of DTLS for RTP.
>=20
>=20
> Section 3.6 says:
>   Mechanisms have been defined that encrypt only the payload of the =
RTP
>   packets, and leave the RTP headers and RTCP in the clear. =20
> which immediately made me think, "ah, like SRTP."  But then the rest =
of the section goes on to describe non-SRTP object encryption (DRM).  =
Can that first sentence somehow be made clearer that this section is not =
describing SRTP encryption, but object encryption?  Afterall, SRTP also =
leaves RTP headers in the clear.  Also, the title of the section is =
"Payload-Only Security Mechanisms", but the second paragraph discusses =
the need for transport security (which is not 'payload security').  =
Perhaps the section could be renamed to "Object Security" or "Digital =
Rights Management" or some other term, so that the payload security and =
transport security can both be discussed without exceeding the scope of =
the section title.
>=20
>=20
> Section 4.1.1,
> s/might provided/might provide/
>=20
>=20
> Section 4.1.3, Changing from "I", "you", and "your" to "host" and =
"peer" would keep this section more consistent with the rest of the =
document.
>=20
>=20
> Section 4.1.3, I had some technical heartburn over these two =
paragraphs, which together make it seem like DTLS-SRTP is weaker than =
Security Descriptions, and the text claims this weakness is because =
DTLS-SRTP sends the certificate hash over signaling whereas Security =
Descriptions sends the session key over signaling:
>      For example, consider a point-to-point communication system that
>      uses DTLS-SRTP using self-signed certificates for key management,
>      and SIP for signalling.  In such a system the end-points for the
>      DTLS-SRTP handshake have securely-established keys that are not
>      visible to the signalling nodes.  However, as the certificates
>      used by DTLS are not bound to any PKI they can't be verified.
>      Instead, hashes of the certificate are sent over the signalling
>      path.  If the signalling can be trusted not to collaborate on
>      performing a man-in-the-middle attack by modifying the hashes,
>      then the end-points can verify that they have established keys
>      with the peer they are doing signalling with.
>=20
>      Systems where the key-exchange is done using the signalling
>      systems, such as Security Descriptions [RFC4568] or MIKEY =
embedded
>      in SDP [RFC4567], enable a direct binding between signalling and
>      key-exchange.  Independent of DTLS-SRTP or MIKEY in SDP the =
actual
>      security depends on the trust one can place in the signalling
>      system to correctly associate the peer's identity with the key-
>      exchange.
>=20
> Of course, DTLS-SRTP does not need to rely solely on trust for =
identity, which is described elsewhere in this same document.  RFC4474 =
or a later section of this same document should be cited in that first =
paragraph (or in the Using Identities section, whichever the authors =
prefer), as RFC4474 provides a way to ensure the DTLS-SRTP certificates =
are bound to the identity provider's public key.  Yes, RFC4474 has some =
deficincies and perhaps those should be mentioned briefly (doesn't =
survive going through an SBC, for example).  I would add a citation and =
description of RFC4474 to that first paragraph discussing DTLS-SRTP.
>=20
> The second paragraph I quoted of Section 4.1.3, I am confused that its =
first sentence discusses Security Descriptions and MIKEY, but the =
paragraphs second sentence starts with "Independent of DTLS-SRTP or =
MIKEY in SDP" -- is this an editing mistake and that second sentence =
supposed to read "Independent of **security descriptions** or MIKEY" ??  =
If it is not an editing mistake, I admit I cannot parse that second =
paragraph and it would benefit from a rewrite to make its point more =
clear.
>=20
>=20
> Section 4.1.5 on privacy mentions relays.  It might be useful to add a =
citation to TURN as an example of a relay.
>=20
>=20
> Section 5.1:
> OLD:
>   At the basic level DTLS has a number of good security properties.
>   For example, to enable a man in the middle someone in the signalling
>   path needs to perform an active action and modify the signalling
>   message.
> NEW:
>   At the basic level DTLS has a number of good security properties.
>   For example, to enable a man in the middle someone in the signalling
>   path needs to perform an active action and modify both the =
signalling
> .....................................................^^^^
>   message and the DTLS handshake.
> ...........^^^^^^^^^^^^^^^^^^^^^^^
>=20
> Also, what is meant by "At the basic level"?  That wording implies =
there an identified problem/weakness at some other level (advanced =
level).
>=20
>=20
> Section 5.2,=20
>   Unless one uses a Identity provider and the proposed identity
>   solution [I-D.ietf-rtcweb-security-arch], the web application still
>   has the possibility to insert a MITM.=20
>=20
> There are other ways to detect and prevent insertion of a MitM other =
than an Identity provider -- such as verifying the certificate hash out =
of band (verifying it matches what is printed on a business card, or =
verifying the hash is the same as the previously-used hash with that =
same user as described by ZRTP, as two examples).  Could that be =
re-worded so that Identity provider is just one way to avoid insertion =
of MITM?
>=20
>=20
> Section 5.2,
>      Prevent tracking between sessions:  RTP CNAMEs and DTLS-SRTP
>      certificates provide information that could possibly be re-used
>      between session instances.  Thus to prevent tracking, the same
>      information can't be re-used between different communication
>      sessions.
> Yes, but another solution (allowing the re-use of DTLS-SRTP =
certificates, thus saving CPU resources to create new public/private key =
pairs) is to have better TLS handshake, such as is being pursued for TLS =
1.2 so the certificates are exchanged later in the TLS exchange (after =
the session is encrypted).  I would suggest:
> OLD:
>      Prevent tracking between sessions:  RTP CNAMEs and DTLS-SRTP
>      certificates provide information that could possibly be re-used
>      between session instances.  Thus to prevent tracking, the same
>      information can't be re-used between different communication
>      sessions.
> NEW:
>      Prevent tracking between sessions:  static RTP CNAMEs and =
DTLS-SRTP
>      certificates provide information that is re-used
>      between session instances.  Thus to prevent tracking, the =
information
>      is not re-used between sessions or the information is not sent in =
the
>      clear between sessions.
>=20
>=20
> Section 5.3, I suggest:
> OLD:
>   If these goals are to be meet with the specified
>   solution there needs to exist trust in that neither the
>   implementation of the client nor the platform the application runs
>   can be accessed or modified by the attacker.
> NEW:
>   To meet these goals with the specified solution the client=20
>   implementation and the application platform are trusted to=20
>   protect against access and modification by an attacker.
>=20
>=20
> -d
>=20
>=20
>=20
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



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




From csp@csperkins.org  Thu Aug 29 05:58:15 2013
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Alan,

Thank you for the review. Some comments inline.

Cheers,
Colin



On 28 Aug 2013, at 19:32, Alan Johnston <alan.b.johnston@gmail.com> =
wrote:
> Here's my review of draft-ietf-avtcore-rtp-security-options-04.
>=20
> Summary:  The document is well-written and useful.  There are a few =
technical issues to resolve but is otherwise ready to progress.
>=20
> - Alan -
>=20
> Numbers refer to sections in the document.
>=20
> 3.1.1:
>=20
> It is surprising no mention is made of RFC 4474 in this section (or in =
the whole document) as DTLS-SRTP keying relies on an integrity protected =
signaling channel provided by RFC 4474.  When mentioning RFC 4474, it =
needs to be pointed out that RFC 4474 is effectively being deprecated by =
the STIR work and has proved to be undeployable and also ineffective for =
E.164 identities.

I have added a reference to RFC 4474 in Sections 4.1.3 and 4.1.4 as a =
result of Dan Wing's comments. I'm not familiar with the STIR work, or =
the other issues, can you suggest text for Section 3.1.1?

> 3.1.5:
>=20
> Might be worth a single sentence about what makes ZRTP unique, such =
as:
>  "ZRTP provides best effort encryption independent of the signaling
> protocol and utilizes key continuity, Short Authentication Strings, or =
a PKI for authentication."

Done.

> 5.1:
>=20
> "There also exists a solution that enables the fingerprints to be =
bound to identities established by the first proxy for each user =
[RFC4916]."  I don't see how SIP connected identity relates at all to =
this.  Is this perhaps meant to reference RFC 4474?

Yes, I think so.

> 5.2:
>=20
> "But, unless the application and its server is intending to compromise =
the communication security, they can provide a secure and =
integrity-protected exchange of the certificate hashes thus preventing =
any man-in-the-middle (MITM) from inserting itself in the key-exchange." =
 This is possible, but not trivial, and I doubt few WebRTC signaling =
channels will provide this. This statement needs some support for it to =
be included.=20

Dan Wing had comments on that section also, and it's been rewritten in =
the -05 version of the draft. That specific text has been removed, but =
that's not the only change, and I'd appreciate review to see if the new =
text makes sense.

> 5.2:
>=20
> Might also be worth a mention that ZRTP can be used to verify the
> DTLS-SRTP fingerprints used without relying on any third parties as
> described in draft-johnston-rtcweb-zrtp.

I added a mention of ZRTP here.

> Nits:
>=20
> 2:
>=20
> "it's" -> "its"
>=20
> 2.1:
>=20
> "end-points verifiable identity" -> "end-point's verifiable identity"

Fixed.

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




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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 Core Maintenance Wo=
rking Group of the IETF.

	Title           : Options for Securing RTP Sessions
	Author(s)       : Magnus Westerlund
                          Colin Perkins
	Filename        : draft-ietf-avtcore-rtp-security-options-05.txt
	Pages           : 32
	Date            : 2013-08-29

Abstract:
   The Real-time Transport Protocol (RTP) is used in a large number of
   different application domains and environments.  This heterogeneity
   implies that different security mechanisms are needed to provide
   services such as confidentiality, integrity and source authentication
   of RTP/RTCP packets suitable for the various environments.  The range
   of solutions makes it difficult for RTP-based application developers
   to pick the most suitable mechanism.  This document provides an
   overview of a number of security solutions for RTP, and gives
   guidance for developers on how to choose the appropriate security
   mechanism.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-avtcore-rtp-security-options

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-avtcore-rtp-security-options-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-rtp-security-options-=
05


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

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From alan.b.johnston@gmail.com  Thu Aug 29 06:48:02 2013
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From: Alan Johnston <alan.b.johnston@gmail.com>
To: Colin Perkins <csp@csperkins.org>
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Subject: Re: [AVTCORE] Review of draft-ietf-avtcore-rtp-security-options-04
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--f46d043bdfde83189104e5165a7c
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Colin,

Thanks for making the changes.  A few comments below.

- Alan -


On Thu, Aug 29, 2013 at 7:57 AM, Colin Perkins <csp@csperkins.org> wrote:

> Alan,
>
> Thank you for the review. Some comments inline.
>
> Cheers,
> Colin
>
>
>
> On 28 Aug 2013, at 19:32, Alan Johnston <alan.b.johnston@gmail.com> wrote:
> > Here's my review of draft-ietf-avtcore-rtp-security-options-04.
> >
> > Summary:  The document is well-written and useful.  There are a few
> technical issues to resolve but is otherwise ready to progress.
> >
> > - Alan -
> >
> > Numbers refer to sections in the document.
> >
> > 3.1.1:
> >
> > It is surprising no mention is made of RFC 4474 in this section (or in
> the whole document) as DTLS-SRTP keying relies on an integrity protected
> signaling channel provided by RFC 4474.  When mentioning RFC 4474, it needs
> to be pointed out that RFC 4474 is effectively being deprecated by the STIR
> work and has proved to be undeployable and also ineffective for E.164
> identities.
>
> I have added a reference to RFC 4474 in Sections 4.1.3 and 4.1.4 as a
> result of Dan Wing's comments. I'm not familiar with the STIR work, or the
> other issues, can you suggest text for Section 3.1.1?
>

Here's the proposed charter for STIR:
http://datatracker.ietf.org/wg/stir/charter/

draft-jennings-dispatch-rfc4474bis discusses the problems with RFC 4474 and
some proposed changes.

I would suggest text in this section mention that DTLS-SRTP relies on an
integrity protected signaling channel based on RFC 4474, but that there is
proposed underway in the IETF to try to make this workable.


>
> > 3.1.5:
> >
> > Might be worth a single sentence about what makes ZRTP unique, such as:
> >  "ZRTP provides best effort encryption independent of the signaling
> > protocol and utilizes key continuity, Short Authentication Strings, or a
> PKI for authentication."
>
> Done.
>
> > 5.1:
> >
> > "There also exists a solution that enables the fingerprints to be bound
> to identities established by the first proxy for each user [RFC4916]."  I
> don't see how SIP connected identity relates at all to this.  Is this
> perhaps meant to reference RFC 4474?
>
> Yes, I think so.
>
> > 5.2:
> >
> > "But, unless the application and its server is intending to compromise
> the communication security, they can provide a secure and
> integrity-protected exchange of the certificate hashes thus preventing any
> man-in-the-middle (MITM) from inserting itself in the key-exchange."  This
> is possible, but not trivial, and I doubt few WebRTC signaling channels
> will provide this. This statement needs some support for it to be included.
>
> Dan Wing had comments on that section also, and it's been rewritten in the
> -05 version of the draft. That specific text has been removed, but that's
> not the only change, and I'd appreciate review to see if the new text makes
> sense.
>

I will look it over when -05 is out.



>
> > 5.2:
> >
> > Might also be worth a mention that ZRTP can be used to verify the
> > DTLS-SRTP fingerprints used without relying on any third parties as
> > described in draft-johnston-rtcweb-zrtp.
>
> I added a mention of ZRTP here.
>
> > Nits:
> >
> > 2:
> >
> > "it's" -> "its"
> >
> > 2.1:
> >
> > "end-points verifiable identity" -> "end-point's verifiable identity"
>
> Fixed.
>
> --
> Colin Perkins
> http://csperkins.org/
>
>
>
>

--f46d043bdfde83189104e5165a7c
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Colin,<div><br></div><div>Thanks for making the changes. =
=A0A few comments below.</div><div><br></div><div>- Alan -</div><div class=
=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Thu, Aug 29, 2013 at=
 7:57 AM, Colin Perkins <span dir=3D"ltr">&lt;<a href=3D"mailto:csp@csperki=
ns.org" target=3D"_blank">csp@csperkins.org</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">Alan,<br>
<br>
Thank you for the review. Some comments inline.<br>
<br>
Cheers,<br>
Colin<br>
<br>
<br>
<br>
On 28 Aug 2013, at 19:32, Alan Johnston &lt;<a href=3D"mailto:alan.b.johnst=
on@gmail.com">alan.b.johnston@gmail.com</a>&gt; wrote:<br>
&gt; Here&#39;s my review of draft-ietf-avtcore-rtp-security-options-04.<br=
>
&gt;<br>
&gt; Summary: =A0The document is well-written and useful. =A0There are a fe=
w technical issues to resolve but is otherwise ready to progress.<br>
&gt;<br>
&gt; - Alan -<br>
&gt;<br>
&gt; Numbers refer to sections in the document.<br>
&gt;<br>
&gt; 3.1.1:<br>
&gt;<br>
&gt; It is surprising no mention is made of RFC 4474 in this section (or in=
 the whole document) as DTLS-SRTP keying relies on an integrity protected s=
ignaling channel provided by RFC 4474. =A0When mentioning RFC 4474, it need=
s to be pointed out that RFC 4474 is effectively being deprecated by the ST=
IR work and has proved to be undeployable and also ineffective for E.164 id=
entities.<br>

<br>
I have added a reference to RFC 4474 in Sections 4.1.3 and 4.1.4 as a resul=
t of Dan Wing&#39;s comments. I&#39;m not familiar with the STIR work, or t=
he other issues, can you suggest text for Section 3.1.1?<br></blockquote>
<div><br></div><div>Here&#39;s the proposed charter for STIR: <a href=3D"ht=
tp://datatracker.ietf.org/wg/stir/charter/">http://datatracker.ietf.org/wg/=
stir/charter/</a>=A0</div><div><br></div><div>draft-jennings-dispatch-rfc44=
74bis discusses the problems with RFC 4474 and some proposed changes.<br>
</div><div><br></div><div>I would suggest text in this section mention that=
 DTLS-SRTP relies on an integrity protected signaling channel based on RFC =
4474, but that there is proposed underway in the IETF to try to make this w=
orkable.</div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex">
<br>
&gt; 3.1.5:<br>
&gt;<br>
&gt; Might be worth a single sentence about what makes ZRTP unique, such as=
:<br>
&gt; =A0&quot;ZRTP provides best effort encryption independent of the signa=
ling<br>
&gt; protocol and utilizes key continuity, Short Authentication Strings, or=
 a PKI for authentication.&quot;<br>
<br>
Done.<br>
<br>
&gt; 5.1:<br>
&gt;<br>
&gt; &quot;There also exists a solution that enables the fingerprints to be=
 bound to identities established by the first proxy for each user [RFC4916]=
.&quot; =A0I don&#39;t see how SIP connected identity relates at all to thi=
s. =A0Is this perhaps meant to reference RFC 4474?<br>

<br>
Yes, I think so.<br>
<br>
&gt; 5.2:<br>
&gt;<br>
&gt; &quot;But, unless the application and its server is intending to compr=
omise the communication security, they can provide a secure and integrity-p=
rotected exchange of the certificate hashes thus preventing any man-in-the-=
middle (MITM) from inserting itself in the key-exchange.&quot; =A0This is p=
ossible, but not trivial, and I doubt few WebRTC signaling channels will pr=
ovide this. This statement needs some support for it to be included.<br>

<br>
Dan Wing had comments on that section also, and it&#39;s been rewritten in =
the -05 version of the draft. That specific text has been removed, but that=
&#39;s not the only change, and I&#39;d appreciate review to see if the new=
 text makes sense.<br>
</blockquote><div><br></div><div>I will look it over when -05 is out.</div>=
<div><br></div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,2=
04);border-left-style:solid;padding-left:1ex">

<br>
&gt; 5.2:<br>
&gt;<br>
&gt; Might also be worth a mention that ZRTP can be used to verify the<br>
&gt; DTLS-SRTP fingerprints used without relying on any third parties as<br=
>
&gt; described in draft-johnston-rtcweb-zrtp.<br>
<br>
I added a mention of ZRTP here.<br>
<br>
&gt; Nits:<br>
&gt;<br>
&gt; 2:<br>
&gt;<br>
&gt; &quot;it&#39;s&quot; -&gt; &quot;its&quot;<br>
&gt;<br>
&gt; 2.1:<br>
&gt;<br>
&gt; &quot;end-points verifiable identity&quot; -&gt; &quot;end-point&#39;s=
 verifiable identity&quot;<br>
<br>
Fixed.<br>
<span class=3D""><font color=3D"#888888"><br>
--<br>
Colin Perkins<br>
<a href=3D"http://csperkins.org/" target=3D"_blank">http://csperkins.org/</=
a><br>
<br>
<br>
<br>
</font></span></blockquote></div><br></div></div>

--f46d043bdfde83189104e5165a7c--

From magnus.westerlund@ericsson.com  Thu Aug 29 22:53:30 2013
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Cc: Colin Perkins <csp@csperkins.org>, avt@ietf.org
Subject: Re: [AVTCORE] Review of draft-ietf-avtcore-rtp-security-options-04
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Hi,

See below for proposal regarding Section 3.1.1.

On 2013-08-29 15:47, Alan Johnston wrote:
> On Thu, Aug 29, 2013 at 7:57 AM, Colin Perkins <csp@csperkins.org
> <mailto:csp@csperkins.org>> wrote:
>
>     >
>     > Numbers refer to sections in the document.
>     >
>     > 3.1.1:
>     >
>     > It is surprising no mention is made of RFC 4474 in this section
>     (or in the whole document) as DTLS-SRTP keying relies on an
>     integrity protected signaling channel provided by RFC 4474.  When
>     mentioning RFC 4474, it needs to be pointed out that RFC 4474 is
>     effectively being deprecated by the STIR work and has proved to be
>     undeployable and also ineffective for E.164 identities.
> 
>     I have added a reference to RFC 4474 in Sections 4.1.3 and 4.1.4 as
>     a result of Dan Wing's comments. I'm not familiar with the STIR
>     work, or the other issues, can you suggest text for Section 3.1.1?
> 
> 
> Here's the proposed charter for STIR:
> http://datatracker.ietf.org/wg/stir/charter/ 
> 
> draft-jennings-dispatch-rfc4474bis discusses the problems with RFC 4474
> and some proposed changes.
> 
> I would suggest text in this section mention that DTLS-SRTP relies on an
> integrity protected signaling channel based on RFC 4474, but that there
> is proposed underway in the IETF to try to make this workable.

I tried drafting some text for this. Feedback appreciated

   DTLS-SRTP key management can use the signalling protocol in four
   ways.  First, to agree on using DTLS-SRTP for media security.
   Secondly, to determine the network location (address and port) where
   each side is running a DTLS listener to let the parts perform the
   key-management handshakes that generate the keys used by SRTP.
   Thirdly, to exchange hashes of each side's certificates to bind these
   to the signalling, and ensure there is no man-in-the-middle attack.
   Finally to provide an assertable identity, e.g. [RFC4474] that can be
   used to prevent modification of the signalling and the exchange of
   certificate hashes.  That way enabling binding between the key-
   exchange and the signalling.

   This usage is well defined for SIP/SDP in [RFC5763], and in most
   cases can be adopted for use with other bi-directions signalling
   solutions.  It should be noted that there is work underway of
   revisiting the SIP Identity mechanism [RFC4474] in the STIR WG.

Cheers

Magnus Westerlund

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Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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Cc: IETF AVTCore WG <avt@ietf.org>
Subject: Re: [AVTCORE] WG last call on draft-ietf-avtcore-aria-srtp-04
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(this message was apparently not posted to the AVT mailing list.  =
Resending.)

On Aug 26, 2013, at 2:34 AM, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:

> WG,
>=20
> This starts the WG last call on draft-ietf-avtcore-aria-srtp-04 to be
> published as proposed standard. Please provide any feedback or review
> comments by September 16. Even if you have just read it and found no
> issue, please state so.
>=20
> Draft:
> https://datatracker.ietf.org/doc/draft-ietf-avtcore-aria-srtp

The document seems to just register the IANA codepoints for Security =
Descriptions, DTLS-SRTP, and MIKEY.  For that reason, it probably needs =
to be standards track (I note that ARIA itself is informational).

Nits:  Section 5.1's title should be "Security Descriptions" or =
"Security Descriptions (SDES)".  Section 5.1, change "Security =
description" to "SDP Security Descriptions" (proper capitalization and =
pluralization of RFC4568's short name).

-d


>=20
> In addition any comments about current or planned implementation of =
this
> draft are appreciated.
>=20
> As a WG chair I really want some persons that really understand crypto
> algorithms and/or the key-management mechanisms used to review this =
and
> will solicit reviews outside of the WG also.
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From internet-drafts@ietf.org  Fri Aug 30 07:19:32 2013
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Subject: [AVTCORE] I-D Action: draft-ietf-avtcore-srtp-aes-gcm-09.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 Core Maintenance Wo=
rking Group of the IETF.

	Title           : AES-GCM and AES-CCM Authenticated Encryption in Secure R=
TP (SRTP)
	Author(s)       : David A. McGrew
                          Kevin M. Igoe
	Filename        : draft-ietf-avtcore-srtp-aes-gcm-09.txt
	Pages           : 33
	Date            : 2013-08-30

Abstract:
   This document defines how AES-GCM and AES-CCM Authenticated
   Encryption with Associated Data algorithms can be used to provide
   confidentiality and data authentication in the SRTP protocol.



The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-avtcore-srtp-aes-gcm

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-avtcore-srtp-aes-gcm-09

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Subject: [AVTCORE] I-D Action: draft-ietf-avtcore-rtp-security-options-06.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 Core Maintenance Wo=
rking Group of the IETF.

	Title           : Options for Securing RTP Sessions
	Author(s)       : Magnus Westerlund
                          Colin Perkins
	Filename        : draft-ietf-avtcore-rtp-security-options-06.txt
	Pages           : 32
	Date            : 2013-08-30

Abstract:
   The Real-time Transport Protocol (RTP) is used in a large number of
   different application domains and environments.  This heterogeneity
   implies that different security mechanisms are needed to provide
   services such as confidentiality, integrity and source authentication
   of RTP/RTCP packets suitable for the various environments.  The range
   of solutions makes it difficult for RTP-based application developers
   to pick the most suitable mechanism.  This document provides an
   overview of a number of security solutions for RTP, and gives
   guidance for developers on how to choose the appropriate security
   mechanism.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-avtcore-rtp-security-options

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http://tools.ietf.org/html/draft-ietf-avtcore-rtp-security-options-06

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06


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From csp@csperkins.org  Fri Aug 30 08:48:20 2013
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Dan, Alan,

Thanks for your review. This version hopefully addresses your comments =
on the -04 draft.=20

Thanks,
Colin



On 30 Aug 2013, at 16:43, 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 Core =
Maintenance Working Group of the IETF.
>=20
> 	Title           : Options for Securing RTP Sessions
> 	Author(s)       : Magnus Westerlund
>                          Colin Perkins
> 	Filename        : draft-ietf-avtcore-rtp-security-options-06.txt
> 	Pages           : 32
> 	Date            : 2013-08-30
>=20
> Abstract:
>   The Real-time Transport Protocol (RTP) is used in a large number of
>   different application domains and environments.  This heterogeneity
>   implies that different security mechanisms are needed to provide
>   services such as confidentiality, integrity and source =
authentication
>   of RTP/RTCP packets suitable for the various environments.  The =
range
>   of solutions makes it difficult for RTP-based application developers
>   to pick the most suitable mechanism.  This document provides an
>   overview of a number of security solutions for RTP, and gives
>   guidance for developers on how to choose the appropriate security
>   mechanism.
>=20
>=20
> The IETF datatracker status page for this draft is:
> =
https://datatracker.ietf.org/doc/draft-ietf-avtcore-rtp-security-options
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-avtcore-rtp-security-options-06

...

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




From ron.even.tlv@gmail.com  Fri Aug 30 09:36:01 2013
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Subject: [AVTCORE] FW: New Version Notification for draft-wu-avtcore-dynamic-pt-usage-00.txt
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Hi,
Apologize for the cross posting. Please comment only in avtcore mailing =
list

We have submitted a draft that clarifies the usage of dynamic payload =
type numbers.
This is based on email discussion in MMUSIC=20
Thanks
Roni Even

-----Original Message-----
From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]=20
Sent: 30 August, 2013 7:29 PM
To: Qin Wu; Wenson Wu; Rachel Huang; Roni Even; Rachel Huang
Subject: New Version Notification for =
draft-wu-avtcore-dynamic-pt-usage-00.txt


A new version of I-D, draft-wu-avtcore-dynamic-pt-usage-00.txt
has been successfully submitted by Qin Wu and posted to the IETF =
repository.

Filename:	 draft-wu-avtcore-dynamic-pt-usage
Revision:	 00
Title:		 Guideline for dynamic payload type number usage policy
Creation date:	 2013-08-30
Group:		 Individual Submission
Number of pages: 5
URL:             =
http://www.ietf.org/internet-drafts/draft-wu-avtcore-dynamic-pt-usage-00.=
txt
Status:          =
http://datatracker.ietf.org/doc/draft-wu-avtcore-dynamic-pt-usage
Htmlized:        =
http://tools.ietf.org/html/draft-wu-avtcore-dynamic-pt-usage-00


Abstract:
   The RTP Profile for Audio and Video Conferences with Minimal Control
   (RTP/AVP) is the basis for many other profiles, such as the Secure
   Real-time Transport Protocol (RTP/SAVP), the Extended RTP Profile for
   Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
   AVPF), and the Extended Secure RTP Profile for RTCP-Based Feedback
   (RTP/SAVPF).  This document updates RFC 3551 and provide guidelines
   for payload type number usage policy.

                                                                         =
        =20


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

The IETF Secretariat


From dwing@cisco.com  Fri Aug 30 19:05:35 2013
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From: Dan Wing <dwing@cisco.com>
In-Reply-To: <8595A60A-EF38-400C-B662-DC8CB99CAC1D@csperkins.org>
Date: Fri, 30 Aug 2013 19:05:17 -0700
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References: <20130830154323.6690.24778.idtracker@ietfa.amsl.com> <8595A60A-EF38-400C-B662-DC8CB99CAC1D@csperkins.org>
To: Colin Perkins <csp@csperkins.org>
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Subject: Re: [AVTCORE] I-D Action: draft-ietf-avtcore-rtp-security-options-06.txt
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On Aug 30, 2013, at 8:48 AM, Colin Perkins <csp@csperkins.org> wrote:

> Dan, Alan,
>=20
> Thanks for your review. This version hopefully addresses your comments =
on the -04 draft.=20

Thanks.  -06 looks good.

-d


>=20
> Thanks,
> Colin
>=20
>=20
>=20
> On 30 Aug 2013, at 16:43, 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 Core =
Maintenance Working Group of the IETF.
>>=20
>> 	Title           : Options for Securing RTP Sessions
>> 	Author(s)       : Magnus Westerlund
>>                         Colin Perkins
>> 	Filename        : draft-ietf-avtcore-rtp-security-options-06.txt
>> 	Pages           : 32
>> 	Date            : 2013-08-30
>>=20
>> Abstract:
>>  The Real-time Transport Protocol (RTP) is used in a large number of
>>  different application domains and environments.  This heterogeneity
>>  implies that different security mechanisms are needed to provide
>>  services such as confidentiality, integrity and source =
authentication
>>  of RTP/RTCP packets suitable for the various environments.  The =
range
>>  of solutions makes it difficult for RTP-based application developers
>>  to pick the most suitable mechanism.  This document provides an
>>  overview of a number of security solutions for RTP, and gives
>>  guidance for developers on how to choose the appropriate security
>>  mechanism.
>>=20
>>=20
>> The IETF datatracker status page for this draft is:
>> =
https://datatracker.ietf.org/doc/draft-ietf-avtcore-rtp-security-options
>>=20
>> There's also a htmlized version available at:
>> http://tools.ietf.org/html/draft-ietf-avtcore-rtp-security-options-06
>=20
> ...
>=20
> --=20
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20

